irclog2html for #asterisk on 20061009

`00:00.01`	`jimbo-`	`how far is the final release of 1.4?`
`00:03.47`	`jimbo-`	`hello`
`00:04.00`	`jimbo-`	`anyone here?`
`00:05.11`	`wmandra`	`is anyone here using broadvoice????`
`00:05.45`	`file`	`I have enough pens to last me 2 years`
`00:07.00`	`bkw__`	`I read that as "I have enough penis to last me 2 years"`
`00:14.35`	`hohum`	`never heard of broadvoice`
`00:14.38`	`hohum`	`heard of broadwing`
`00:14.41`	`hohum`	`heard of broadvox`
`00:14.44`	`hohum`	`not broadvoice`
`00:15.02`	`*** part/#asterisk jimbo- (i=jhio8838@214.sub-75-193-216.myvzw.com)`
`00:16.56`	`*** join/#asterisk Cyt (n=danielcy@85.75.176.202)`
`00:17.14`	`Cyt`	`Does anybody have asterisk working with lines from INX (internationalnumber.com)`
`00:18.42`	`*** join/#asterisk mv00 (i=HiTMaN@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk)`
`00:24.23`	`hohum`	`how do I get rid of 407 Proxy Authorization Required messages?`
`00:32.49`	`*** join/#asterisk mv00 (i=HiTMaN@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk)`
`00:33.42`	`*** join/#asterisk Schulich (n=Jazba@165.154.36.183)`
`00:38.22`	`KuJaX`	`what ports do i need to open with iptables for asterisk incoming and outgoing?`
`00:39.56`	`*** join/#asterisk re-pete (n=repete@24.96.201.72)`
`00:43.12`	`*** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-624ab91f9fb703e5)`
`00:43.49`	`*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net)`
`00:44.10`	`wmandra`	`hey guys... is anyone here using Comcast for their ISP???`
`00:50.22`	`Cyt`	`Hi, please: I'm usign in sip.conf register => NUMBER:mypass@sip.intlno.com/101 . The problem is, It rings straight to the extension 101, ignoring any dialplan created in extensions.conf. What did I do wrong, please! Thank you (with other providers this works fine)`
`00:54.26`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`00:54.42`	`*** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org)`
`00:54.50`	`*** join/#asterisk pabluss (n=aquicamb@200.75.1.29)`
`00:55.13`	`pabluss`	`afternoon`
`00:56.47`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`00:59.06`	`wmandra`	`afternoon pabluss`
`00:59.11`	`shellshark`	`evening`
`00:59.45`	`pabluss`	`greetings from stgo asterisk server working :D`
`01:07.18`	`*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net)`
`01:07.31`	`rob0`	`wmandra: I'm on Comcast.`
`01:07.33`	`wmandra`	`evening all`
`01:08.01`	`wmandra`	`hey rob0: were you experiencing any issues with VOIP service today??`
`01:08.11`	`rbd`	`is it possible to have sip bind to two different ports (basically I'd like to have it so that sip trunks could come in as one of two different contexts)`
`01:08.13`	`rob0`	`Didn't have any calls.`
`01:09.11`	`pabluss`	`wmandra where do you from?`
`01:09.33`	`rob0`	`Noo Joisee`
`01:09.55`	`wmandra`	`thanks..... asterisk was unable to register to my provider all day, they said the last registration attempt was early this morning... after rebooting asterisk, router, etc still nothing, then i did a port scan and 5060 was blocked.... then all of a sudden about half an hour ago it started working again`
`01:10.08`	`wmandra`	`NJ`
`01:10.11`	`re-pete`	`~seen jart`
`01:10.25`	`jbot`	`jart is currently on #asterisk (18h 46m 17s). Has said a total of 12 messages. Is idling for 2h 45m 9s, last said: 'they're just mad their stock is 1/20th what it used to be'.`
`01:10.43`	`rob0`	`weird!! I wouldn't put it past them to start blocking SIP.`
`01:10.45`	`pabluss`	`here Santiago of Chile`
`01:11.05`	`re-pete`	`my wife is from Santiago`
`01:11.13`	`pabluss`	`yes?`
`01:11.18`	`re-pete`	`Si`
`01:11.27`	`pabluss`	`opss rene?`
`01:11.46`	`re-pete`	`I've been there once for vacation and loved it.`
`01:12.22`	`*** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)`
`01:12.31`	`a1fa`	`has anybody heard anything about sip on psp`
`01:13.40`	`wmandra`	`i actually just called broadvoice to apologize for blaming them..... i had a friend in CT do a portscan and UDP 5060 was definately blocked 45 minutes ago`
`01:13.53`	`a1fa`	`wmandra : fuck broadvoice`
`01:14.00`	`a1fa`	`its always their fault`
`01:14.42`	`wmandra`	`i want to dump broadvoice, but still can't figure out who is gonna be any better.... it seems all the provders suck`
`01:14.50`	`rob0`	`Are you sure the friend didn't have the blockage on his end?`
`01:14.50`	`wmandra`	`lol@a1fa`
`01:15.03`	`a1fa`	`broadvoice is dirt cheap`
`01:15.04`	`pabluss`	`somebody have information about howto to connect two or more asterisk server ?`
`01:15.11`	`a1fa`	`make sure you use more then 1000 minutes a month to punish them`
`01:15.25`	`tzafrir`	`pabluss, there's a page on voip-info`
`01:15.27`	`wmandra`	`yeah right, on all three of my lines`
`01:15.40`	`wmandra`	`i'm actually considering teliax right now`
`01:15.56`	`shellshark`	`shellshark.net is decent`
`01:16.00`	`wmandra`	`i'd really like to find a provider that will let me set CID for outbound`
`01:16.10`	`rob0`	`I've got Asterlink, no problems. I set CID.`
`01:16.10`	`shellshark`	`truely unlimited outbound plan for $15/mo... hard to beat it`
`01:16.26`	`shellshark`	`rob0: what's it cost?`
`01:16.35`	`litage`	`how long does Digium generally take when you want to re-register a g729 license because you've added/removed/switched a NIC?`
`01:16.36`	`rob0`	`shellshark sounds possibly biased. :)`
`01:16.47`	`wmandra`	`ya think :)`
`01:16.47`	`shellshark`	`rob0: perhaps ;)`
`01:17.07`	`pabluss`	`ok tzafir there we go...`
`01:17.12`	`pabluss`	`thanks`
`01:18.00`	`rob0`	`Asterlink extreme, 800 number, US outbound calls under 2c/Min, not sure of exact amount. $2/month fee.`
`01:19.10`	`rob0`	`I never set up my Asterlink inbound number (or did it wrong more likely) but I use IPKall and Stanaphone for inbound (free).`
`01:20.38`	`wmandra`	`i was considering didx for inbound`
`01:22.04`	`shellshark`	`wmandra: there is also virtualphoneline.com that doesnt charge you a $20 membership fee, and is owned by the same people as DIDx (Super Technologies)`
`01:22.26`	`wmandra`	`ok now i'm going nuts.... does anyone happen to know the wiki page that explained the problem when you have two incoming numbers coming from the same provider * routes them to the same extension even when you have them configured to use different ones`
`01:22.40`	`wmandra`	`yeah, i saw that`
`01:23.33`	`rob0`	`virtualphoneline.com Web site unusable without flash :(`
`01:23.40`	`shellshark`	`wmandra: err, [inbound] exten => _12345,1,Dial(SIP/100) exten => _12346,1,Dial(SIP/101)`
`01:23.49`	`shellshark`	`rob0: once you login it's usable`
`01:24.01`	`a1fa`	`anybody know anything about psp?`
`01:24.10`	`a1fa`	`regarding voip on psp`
`01:24.26`	`rob0`	`I don't have an account, was just looking.`
`01:24.31`	`wmandra`	`shellshark: thanks... let me give that a try`
`01:24.34`	`shellshark`	`wmandra: it's impossible for a call destined for 12346 to go to extension 100, in this case`
`01:24.38`	`shellshark`	`wmandra: and vice versa`
`01:25.11`	`*** join/#asterisk droops_mobile (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net)`
`01:25.11`	`shellshark`	`wmandra: most likely you have something like [inbound] exten => _X.,1,Dial(SIP/100)`
`01:25.27`	`shellshark`	`wmandra: which, of course, will match everything ;-)`
`01:25.28`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`01:26.30`	`wmandra`	`actually right now i have exten => 9735551212,1,Dial.... and exten => 9735551213,1,Dial....`
`01:26.43`	`*** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net)`
`01:28.14`	`wmandra`	`nope the _ didn't help`
`01:30.00`	`shellshark`	`and you did an extensions reload right?`
`01:30.13`	`shellshark`	`after changing and saving your extensions.conf`
`01:30.21`	`wmandra`	`yup`
`01:30.34`	`wmandra`	`all the calls appear to come in the last number`
`01:30.46`	`shellshark`	`odd`
`01:30.54`	`shellshark`	`talk to your provider about it`
`01:31.26`	`*** join/#asterisk supjigatr (n=syslod@152.53.17.26)`
`01:31.29`	`shellshark`	`you could setup a debugging AGI at one of the exten statements to see what the DNID is`
`01:33.40`	`*** part/#asterisk pabluss (n=aquicamb@200.75.1.29)`
`01:35.41`	`*** join/#asterisk jo3sm1th (i=jo3sm1th@247.sub-75-192-66.myvzw.com)`
`01:35.53`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`01:36.56`	`jo3sm1th`	`Hi`
`01:37.12`	`jo3sm1th`	`Any good conferencing about VOIP technology on IRC lately?`
`01:37.22`	`*** join/#asterisk dovid (n=dovi5988@85.159.160.196)`
`01:37.41`	`dovid`	`morning all`
`01:38.30`	`shellshark`	`evening`
`01:38.55`	`dovid`	`same`
`01:42.07`	`*** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net)`
`02:04.11`	`wmandra`	`shellshark: i think i figured out the problem on my icoming calls..... * only uses the /XXXXXXX in the register line to see if the incoming call is authorized, or to determine the extension to goto sometimes...... when you have multiple registrations to the same provider and use that provider for both incoming / outgoing calls when the call comes in * checks the inbound context for an extension matching the inbound number, then it lo`
`02:04.57`	`shellshark`	`ah`
`02:05.07`	`shellshark`	`you shouldnt have multiple register lines :P`
`02:05.20`	`shellshark`	`you only have to register if you're sending calls out to someone, you know`
`02:05.32`	`shellshark`	`you dont have to register for inbound calls, that's what you setup a peer for`
`02:07.23`	`wmandra`	`actually you have that backwards i think :P the register is waht tells the provider what IP address to route incoming calls to..... the peer entries are for outbound calls`
`02:07.59`	`shellshark`	`nope`
`02:08.29`	`shellshark`	`register is for outbound, peer is for inbound`
`02:11.07`	`wmandra`	`eh... either way i still need all three for outbound since each line can only handle 1 concurent call`
`02:11.44`	`shellshark`	`who misled you to believe that?`
`02:12.16`	`wmandra`	`what? one concurrent call or the register/peer thing?`
`02:13.32`	`shellshark`	`one concurrent call`
`02:13.36`	`shellshark`	`per register statement`
`02:14.46`	`wmandra`	`broadvoice only allows one concurent call.... if a call comes in on XXX1 and during that call i try to dial out on that line they won't let the call through`
`02:17.18`	`wmandra`	`either way... if i don't register one of my lines with broadvoice no calls come into * from that number and if i don't add a peer entry for a given line no calls from * can go out on that line`
`02:17.40`	`shellshark`	`ah, so by registering twice (once per account you have with them), you are able to dial out via the second register when someone calls you on the first number (taking up the one allocated channel of the first account)`
`02:18.10`	`wmandra`	`exactly`
`02:20.17`	`*** join/#asterisk fromvega (n=eu@200-161-218-43.dsl.telesp.net.br)`
`02:20.23`	`fromvega`	`Hello!`
`02:20.25`	`fromvega`	`<PROTECTED>`
`02:29.08`	`*** join/#asterisk snickn (i=nick@light.teleri.net)`
`02:29.30`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`02:31.36`	`*** join/#asterisk carmen (i=ix@c-24-91-185-85.hsd1.ma.comcast.net)`
`02:31.40`	`carmen`	`is there a ncurses softphone?`
`02:31.52`	`carmen`	`or maybe even more minimall... echo stuff to a LUFS to effect the call state?`
`02:34.40`	`*** join/#asterisk jake1932 (n=Administ@pool-68-236-49-109.phil.east.verizon.net)`
`02:35.14`	`rob0`	`carmen: * itself can act as a softphone.`
`02:36.29`	`jake1932`	`can someone give me any pointers to compiling cdr_shell at pbxfreeware.com? I tried 'gcc cdr_shell.c -o cdr_shell' but I get a few errors.`
`02:36.38`	`carmen`	`rob0: ah just noticed the alsa config option`
`02:36.43`	`carmen`	`thx`
`02:36.58`	`carmen`	`ekiga has serious issues. and twinkle requires qt. an afaik theres nothing else maintained`
`02:40.10`	`rob0`	`I installed * on my laptop, but it seems my sound card is too lousy. No one could hear me when I was on the phone.`
`02:40.43`	`*** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx)`
`02:41.03`	`carmen`	`maybe you need the 'mic boost' option`
`02:41.08`	`carmen`	`usually adds another 20db or somethign`
`02:42.54`	`tzafrir`	`jake1932, what system? what errors? pastebin the build log`
`02:43.46`	`jake1932`	`tzafrir: it's just an older P3 system. I'll pastebin the output. Is the build statement correct?`
`02:44.56`	`*** part/#asterisk fromvega (n=eu@200-161-218-43.dsl.telesp.net.br)`
`02:45.32`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`02:45.56`	`jake1932`	`hmm... pastebin.ca is not responding for me`
`02:46.30`	`Cyt`	`jake1932: It's really slow. I just found this one: http://paste.nintendev.com/`
`02:46.32`	`tzafrir`	`jake1932, I meant: version of astersik, OS, etc.`
`02:46.40`	`De_Mon`	`??pastebin`
`02:46.46`	`tzafrir`	`~pb`
`02:46.47`	`jbot`	`it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`02:47.03`	`De_Mon`	`channels.debian.net? heh`
`02:47.05`	`jake1932`	`http://paste.nintendev.com/pastebin.php?show=36`
`02:47.14`	`jake1932`	`tnx Cyt`
`02:47.31`	`jake1932`	`I'm running 1.2.9.1`
`02:47.53`	`jake1932`	`on debian - 2.6.16`
`02:47.53`	`*** join/#asterisk cvaldess (n=hello@75.Red-88-18-160.staticIP.rima-tde.net)`
`02:47.58`	`cvaldess`	`Hi`
`02:48.49`	`cvaldess`	`any one here testing uCasterisk for Blackfin DSP??`
`02:50.11`	`tzafrir`	`asterisk.h is not found`
`02:50.35`	`tzafrir`	`do you have asterisk-dev installed?`
`02:50.50`	`jake1932`	`nope - lemme try that`
`02:50.57`	`tzafrir`	`cvaldess, I've looked at that`
`02:51.10`	`tzafrir`	`jake1932, that is: if you installed from debs`
`02:51.18`	`jake1932`	`hmm'`
`02:51.41`	`jake1932`	`I might have originally - it's been a while`
`02:51.57`	`cvaldess`	`tzafrir> ???`
`02:52.20`	`Cyt`	`jake1932: your welcome ;)`
`02:52.20`	`*** join/#asterisk Stp1800 (n=Stp1800@atlsfl-bundle-69-167-93-164.atlsfl.adelphia.net)`
`02:52.25`	`tzafrir`	`cvaldess, basically Asterisk 1.4 is quite close to building on th BF. It's a saner starting point`
`02:53.32`	`cvaldess`	`tzafrir> will buy stamp to start testing`
`02:54.00`	`tzafrir`	`There is a BF simulator called SkyEye. I wonder how useful it is`
`02:57.12`	`cvaldess`	`tzafrir> nice tool`
`02:57.34`	`BiGuRoOt`	`hey, anybody can help with rxfax application? i�ve installed with sucess, but i not receive fax...`
`03:05.06`	`*** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net)`
`03:09.43`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`03:09.54`	`*** join/#asterisk SomethingISODD (n=DanC@h109.42.63.69.cable.ottr.cablerocket.net)`
`03:10.05`	`SomethingISODD`	`hello all question does anyone know if asterisk will support mp5`
`03:10.13`	`SomethingISODD`	`h.264`
`03:12.15`	`rbd`	`can I use static extensions and realtime extensions (stored in mysql) at the same time, and in the same context?`
`03:13.33`	`cvaldess`	`rdb> yes`
`03:13.43`	`cvaldess`	`some example at wiki`
`03:14.09`	`cvaldess`	`in same context no`
`03:14.42`	`cvaldess`	`don�t know.. never test it but samples at waki`
`03:15.33`	`rbd`	`cvaldess: okay...so in my case, I'd like to be able to handle incoming SIP calls (to context sip-incoming) via realtime extensions for all but a single extension (extension 500)...now I know that I could put this ext in the database, but if I could just have it in as static for performance reasons that would be the best`
`03:16.20`	`rbd`	`I might be able to forward it to a context with just the static ext entry, then forward that to my realtime extensions context`
`03:16.39`	`rbd`	`err forward everything that doesn't match that static extension entry`
`03:17.24`	`cvaldess`	`The way RealTime Extensions work is through a switch statement in the dialplan`
`03:18.41`	`cvaldess`	`http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions`
`03:23.53`	`rbd`	`cvaldess: yeah looking at that. thanks for the help`
`03:30.16`	`Cyt`	`I'm having a hard time with INX (internationalnumber.com). I have the dialplan done for it but asterisk seems to ignore this and ring directly on the extension. Could somebody take a look and give me some suggestion? THANKS!!!! (sip.conf: http://paste.nintendev.com/pastebin.php?show=34 extensions.conf http://paste.nintendev.com/pastebin.php?show=35)`
`03:34.39`	`SomethingISODD`	`anyone know of any Software video phones that will run on linux`
`03:45.58`	`FuriousGeorge`	`anyone using sipphone`
`03:46.00`	`FuriousGeorge`	`?`
`03:46.56`	`FuriousGeorge`	`my parents use a voip offering from att which will only work with their att modem device. its reliable as hell. ive yet to find a service that has the same consistant quality`
`03:47.29`	`FuriousGeorge`	`then again, since i have never had any great success with voip and *, i guess i cant say for sure whether its the provider or the server`
`03:47.42`	`FuriousGeorge`	`users complain especially that calling parties cant hear them`
`03:47.47`	`FuriousGeorge`	`they "break up"`
`03:48.03`	`shellshark`	`that's a codec:bandwidth issue`
`03:48.35`	`shellshark`	`you're trying to shove too large of a voice channel down too small of a data pipe`
`03:49.16`	`shellshark`	`if you use the g729 or g723 codec it performs a lot better`
`03:49.24`	`shellshark`	`but you can't do things like fax, etc across it`
`03:53.32`	`*** join/#asterisk bmg505 (n=leon@c1-205-12.rndf.isadsl.co.za)`
`03:58.06`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`04:01.09`	`*** join/#asterisk fiber0pti (n=John@c-68-35-13-238.hsd1.nm.comcast.net)`
`04:08.27`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`04:09.01`	`*** join/#asterisk fiber0pti (n=John@c-68-35-13-238.hsd1.nm.comcast.net)`
`04:11.21`	`FuriousGeorge`	`shellshark: i know this is late, but we have cable internet down here`
`04:11.26`	`FuriousGeorge`	`bandwidth isnt the problem`
`04:11.36`	`FuriousGeorge`	`whats 64 kb when you get 10M down`
`04:11.53`	`FuriousGeorge`	`and 2 MB up`
`04:12.00`	`shellshark`	`down is not the problem (hence why you hear them fine), it's up`
`04:12.09`	`shellshark`	`i've never seen a cable company give 2mbps upload`
`04:12.26`	`shellshark`	`i've got business class cable here and i'm lucky to get 768kbps`
`04:13.00`	`shellshark`	`also, the nature of cable services in general is another huge issue, as the latency is all over the board in general`
`04:13.01`	`FuriousGeorge`	`cabelvision in nj here`
`04:13.14`	`shellshark`	`and they can not guarantee latency, of course`
`04:13.46`	`shellshark`	`run a constant ping to somewhere like google or yahoo, for a whole 24 hour period, and see what your max reply was`
`04:13.55`	`shellshark`	`and your average`
`04:14.04`	`FuriousGeorge`	`thats a good idea`
`04:14.31`	`shellshark`	`if your max is insanely different from the average, you know you experience lag spikes`
`04:14.38`	`FuriousGeorge`	`but like i was saying, same isp, with a closed voip service from ATT = much more consistant quality`
`04:14.41`	`shellshark`	`which cause hiccups all the time with VoIP`
`04:14.59`	`shellshark`	`some codecs handle buffering better also`
`04:15.15`	`shellshark`	`they might have a hardware jitterbuffer in their modem also`
`04:15.20`	`FuriousGeorge`	`they use ulaw too`
`04:15.29`	`FuriousGeorge`	`that could be`
`04:15.44`	`shellshark`	`there are all kinds of factors that are at play`
`04:16.45`	`FuriousGeorge`	`so even with a business class ISP, reliable voip is not gonna happen`
`04:17.14`	`shellshark`	`also, their server could sit less hops away from your other provider, which is a huge probability, seeing how ATT feeds 95% of the cable companies in the US with upstream bandwidth`
`04:17.40`	`FuriousGeorge`	`that sounds feasable`
`04:18.05`	`shellshark`	`FuriousGeorge: business class cable, no... business class SDSL, T1, T3, OC1, OC3, etc, sure it's possible to have consistant quality`
`04:18.45`	`FuriousGeorge`	`thats what i figured. shoot, if im gonna pay $300/mo for internet i'd better get some latency guarentees`
`04:18.56`	`shellshark`	`SDSL can do that`
`04:19.08`	`shellshark`	`you'll get slower speeds, but you'll gain an SLA`
`04:19.22`	`FuriousGeorge`	`whts the cost on that?`
`04:19.25`	`shellshark`	`SLA > speed, in most cases, including this scenario ;)`
`04:19.31`	`shellshark`	`depends on the provider`
`04:19.33`	`FuriousGeorge`	`i havent seen it around here`
`04:19.48`	`shellshark`	`you're in jersey, you should be able to get FIOS from verizon`
`04:20.40`	`FuriousGeorge`	`not yet`
`04:20.45`	`shellshark`	`not sure if that is plaugued with the same problems as cable or not, but seeing that it's fiber and not copper, i'm sure they can give you more consistant service`
`04:20.50`	`FuriousGeorge`	`at least not where im at`
`04:21.12`	`FuriousGeorge`	`i was gonna take a wait and see approach with that as its so new`
`04:21.31`	`FuriousGeorge`	`but afaik thats gonna be a residential and business offering like cable`
`04:21.40`	`FuriousGeorge`	`with no SLA of that sort`
`04:22.01`	`shellshark`	`i'm sure they'll offer an SLA to business customers`
`04:22.15`	`shellshark`	`it'd be an easy SLA to honor ;)`
`04:22.16`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`04:22.50`	`FuriousGeorge`	`not the sort with guarenteed latency and uptime i mean`
`04:23.35`	`FuriousGeorge`	`found an sdsl probider but the cheapest offering is 120 bucks for 128k`
`04:23.36`	`shellshark`	`well maybe not quite, since it's not a dedicated circuit`
`04:23.55`	`shellshark`	`SDSL is a dedicated circuit ;)`
`04:24.09`	`shellshark`	`and $120/mo for a dedicated circuit is DAMN cheap`
`04:24.18`	`shellshark`	`if you don't agree, quote a DS1 ;)`
`04:24.27`	`FuriousGeorge`	`i know what ur saying`
`04:24.56`	`shellshark`	`for someone looking for reliability, a dedicated circuit is the only way to go`
`04:25.16`	`FuriousGeorge`	`most of my customers want voip, but none make enough toll calling that it makes sense to get one of these enterprise class connections`
`04:26.03`	`FuriousGeorge`	`verizon adsl is aweful, cable is slightly better, but to pay 300 bucks a month for your ISP does not recoup the premium for most of my clientel`
`04:26.12`	`shellshark`	`a DS1 is a far cry from being able to be called "enterprise" :P`
`04:26.48`	`shellshark`	`you running a wifi hotspot out of your house or something? :)`
`04:26.55`	`FuriousGeorge`	`shellshark: im just using the enterprise term to separate business class oferings of cable and adsl from t1/t3/ds1 etc`
`04:27.38`	`FuriousGeorge`	`so now when i bitch about my business class isp, you will know im not talking about a ts1 line`
`04:27.41`	`FuriousGeorge`	`*ds1`
`04:27.50`	`FuriousGeorge`	`or t1 or whatever`
`04:28.52`	`shellshark`	`yeah, you're talking about a shared connection with no SLA with higher bandwidth cap and some buzzwords thrown in ;)`
`04:29.35`	`shellshark`	`no wonder your quality is not consistant`
`04:29.37`	`shellshark`	`:P`
`04:30.01`	`FuriousGeorge`	`sure when u put it that way`
`04:30.09`	`shellshark`	`also, where are your clients in relation to the server? on the same LAN as the server itself? or also on the internet?`
`04:30.15`	`FuriousGeorge`	`same lan`
`04:30.22`	`shellshark`	`good`
`04:30.25`	`FuriousGeorge`	`and my provider is god knows where`
`04:30.29`	`FuriousGeorge`	`Mi i think`
`04:30.41`	`shellshark`	`i was going to say if they are on the internet also, then you're doubling your bandwidth`
`04:30.42`	`FuriousGeorge`	`im gonna try taking stats with different providers`
`04:30.53`	`FuriousGeorge`	`try sippone out too`
`04:30.56`	`FuriousGeorge`	`they are a big one`
`04:30.58`	`shellshark`	`so that single 64kbps channel just turned into 128kbps`
`04:31.03`	`shellshark`	`try me out :)`
`04:31.11`	`FuriousGeorge`	`who is me?`
`04:31.15`	`shellshark`	`shellshark.net`
`04:32.10`	`shellshark`	`i've got dual redundant DS3 circuits coming in at different ends of a building from two different tier 1 providers to a dual processor, dual-core 2.8ghz p4 xeon with 4gb of RAM`
`04:32.35`	`shellshark`	`(server it sitting in a secured co-location facility in downtown chicago)`
`04:33.23`	`De_Mon`	`ooO compTIA linux+ certified`
`04:33.29`	`shellshark`	`hehe`
`04:33.45`	`De_Mon`	`which providers?`
`04:33.49`	`shellshark`	`i'm no english major, and i needed some filler content :P`
`04:34.09`	`shellshark`	`Savvis and Genuity IIRC`
`04:34.22`	`JT`	`so all this bandwidth... going into a single server?`
`04:34.24`	`De_Mon`	`well you got the ajax, perl, python and ruby going for you!`
`04:34.56`	`De_Mon`	`JT it better not be 1 server!`
`04:36.41`	`shellshark`	`JT: of course not`
`04:37.06`	`JT`	`it certainly sounded like it was going to 1 server`
`04:39.04`	`*** join/#asterisk fafnir (n=notfaf@unaffiliated/fafnir)`
`04:39.15`	`shellshark`	`a single server could not handle all of that bandwidth by itself :)`
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`04:40.42`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`04:44.09`	`JT`	`sure, but who says you're using all the bandwidth? :P`
`04:44.51`	`BiGuRoOt`	`how can i do for asterisk record outgoind call on determined channel?`
`04:45.09`	`BiGuRoOt`	`outgoind = outgoing`
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`05:15.53`	`Flauto`	`anyone is using gtalk here`
`05:20.56`	`Flauto`	`it is quiet`
`05:20.59`	`Flauto`	`hello`
`05:22.29`	`stephane_`	`jour`
`05:23.21`	`Flauto`	`hi stephane`
`05:23.28`	`Flauto`	`what's up`
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`06:03.13`	`Merphy`	`hi everybody`
`06:03.26`	`Merphy`	`I am very new to asterisk`
`06:04.50`	`qdk`	`(me is taking notes. This might be important. :-P`
`06:05.48`	`Merphy`	`I am in a process to propose IP phone solution to my org. It is a commercial org which provided ip phone services. will ASTERISK suffice?`
`06:06.09`	`carmen`	`yeo`
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`07:07.02`	`dmcn`	`totally off topic: i need contact to someone with a mobile from an english mobile phone company (orange, o2, vodafone etc.) to help with a single test of an sms gateway - query me if you can help :)`
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`07:15.47`	`Makenshi`	`dmcn ya there?`
`07:16.11`	`dmcn`	`Makenshi, i am :)`
`07:16.19`	`Makenshi`	`right let's go!`
`07:17.26`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
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`07:45.51`	`Aurs`	`how do I enable queue logging?`
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`07:46.19`	`*** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-fd6abe14dc847e81)`
`07:46.19`	`hohum`	`how do I get rid of 407 Proxy Authorization Required messages?`
`07:50.28`	`*** join/#asterisk oQPa (i=name@237.Red-83-44-33.dynamicIP.rima-tde.net)`
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`07:53.33`	`tzafrir`	`an interesting problem with busy detection:`
`07:54.27`	`tzafrir`	`our telco does not provide hangup notification, and thus we use busydetect`
`07:54.58`	`tzafrir`	`However I have just noticed a way to leave such an analog zaptel trunk off-hook:`
`07:55.35`	`tzafrir`	`call an "incomplete" number. After a while the telco returns a congenstion tone and considers the call ended.`
`07:55.48`	`tzafrir`	`But Asterisk does not detect that as end of call`
`07:55.58`	`*** join/#asterisk oQPa (i=Ftv@237.Red-83-44-33.dynamicIP.rima-tde.net)`
`07:57.05`	`tzafrir`	`Any way around this?`
`07:57.07`	`lilalinux`	`when I specify multiple channels in one dial command, asterisk fails. e.g. Dial(SIP/sk & SIP/sm) "Oct 9 09:48:15 WARNING[10806]: chan_sip.c:1980 create_addr: No such host: sk"`
`07:57.43`	`tzafrir`	`do you have a sip peer named sk?`
`07:57.49`	`lilalinux`	`sk and sm and yes`
`07:57.50`	`tzafrir`	`Maybe remove that space?`
`07:58.21`	`lilalinux`	`i'll try that`
`08:01.12`	`*** join/#asterisk oQPa (n=Spain@237.Red-83-44-33.dynamicIP.rima-tde.net)`
`08:04.40`	`lilalinux`	`tzafrir: thx, that was it`
`08:06.38`	`lilalinux`	`fixed the wiki`
`08:07.16`	`hohum`	`how do I get rid of 407 Proxy Authorization Required messages?`
`08:07.35`	`*** join/#asterisk tparcina (n=tparcina@lns01-1341.dsl.iskon.hr)`
`08:08.08`	`lilalinux`	`hohum: did you specify your credentials?`
`08:08.14`	`lilalinux`	`(sorry for asking)`
`08:09.32`	`dorel__`	`is anyone trying to install hudlite on debian?`
`08:12.42`	`lilalinux`	`dorel__: wow, that's awesome, is it free?`
`08:12.49`	`tparcina`	`IAX questio: when I dial Dial(IAX2/zg2/${EXTEN:1},70,t)`
`08:13.20`	`tparcina`	`do I use username/pass that are entered in iax.conf for zg2 user?`
`08:13.22`	`tzafrir`	`no. non-free`
`08:13.36`	`tzafrir`	`apt-get install op-panel`
`08:14.22`	`hohum`	`lilalinux: I'm working with an endpoint that doesn't support auth on invite`
`08:14.35`	`hohum`	`I also have insecure=very set on all the relevant sip.conf entries`
`08:14.36`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`08:14.48`	`hohum`	`and Asterisk is still spewing 407s`
`08:15.08`	`dorel__`	`lilalinux: yeah, you're going to install it?`
`08:15.09`	`tzafrir`	`dorel__, did you see my question above? Have you encountered anything similar?`
`08:15.14`	`tparcina`	`so, is that the right way to make authorized iax calls?`
`08:15.39`	`dorel__`	`tzafrir: oh sorry i missed it, let me scroll up`
`08:16.08`	`*** join/#asterisk oej (n=oej@195.18.201.3)`
`08:17.08`	`dorel__`	`tzafrir: ive encountered something similar but not an "incomplete" number`
`08:17.33`	`*** join/#asterisk Splat (n=Splat@220-253-136-10.TAS.netspace.net.au)`
`08:20.20`	`jmls`	`Aurs: ping`
`08:20.35`	`jmls`	`aurs: look in /etc/asterisk/logger.conf`
`08:20.43`	`jmls`	`by default queue logging is on`
`08:20.47`	`*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)`
`08:20.53`	`jmls`	`but have a look for queue_log = no`
`08:21.00`	`hohum`	`fuck this is annoying`
`08:21.03`	`jmls`	`and change it to queue_log = yes`
`08:21.24`	`Aurs`	`jmls: had a old logger.conf. but found it in the sample conf`
`08:22.40`	`jmls`	`aurs: all sorted now ?`
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`08:33.34`	`*** join/#asterisk hank (n=hank@netwichtig.de)`
`08:33.36`	`hank`	`hi`
`08:35.28`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.4)`
`08:36.09`	`tengulre`	`how many E1 card allowed in a single system?`
`08:36.57`	`Cyt[away]`	`Please, does anybody knows this error? I tryied on google but could not find a good answer: -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from IP..`
`08:38.52`	`tengulre`	`anybody here??`
`08:39.54`	`tengulre`	`how many E1 cards can allowed in a single system?`
`08:40.11`	`hank`	`tengulre: you should be a bit more patient...`
`08:40.29`	`tengulre`	`hank: :)`
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`08:46.26`	`tengulre`	`hank: can you help me?`
`08:47.08`	`hank`	`tengulre: nope :( sorry... im struggling to get asterisk working and understand things as well`
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`08:48.05`	`benjk`	`tenguire, best is not to use more than one or two cards`
`08:48.15`	`benjk`	`ideally one card per system`
`08:50.13`	`tengulre`	`benjk: I have six E1 cards, I want install them in a same machine. but someone tell me that only support 256 lines in a single machine when asterisk's version is 1.2.9 before,`
`08:50.33`	`tengulre`	`I want to know version is 1.4 later.`
`08:52.36`	`jmls`	`tengulre: you are asking for trouble installing that many cards in a single machine. Apart from the disaster that will happen if the machine fails (all lines go down) the load alone will be quite high`
`08:52.58`	`jmls`	`tengulre: strongly recommend that you go for at least 3 machines with 2 cards per machine`
`08:53.18`	`*** join/#asterisk tengulre11 (n=tengulre@221.11.5.180)`
`08:53.27`	`tengulre11`	`sorry, I just offline!`
`08:53.44`	`tengulre11`	`please re-tell me !`
`08:54.04`	`jmls`	`tengulre: you are asking for trouble installing that many cards in a single machine. Apart from the disaster that will happen if the machine fails (all lines go down) the load alone will be quite high`
`08:54.09`	`jmls`	`tengulre: strongly recommend that you go for at least 3 machines with 2 cards per machine`
`08:54.52`	`qdk`	`tengulre11: the I/O will kill your machine... and 6 x E1 is "only" about 180 channels.`
`08:55.42`	`tengulre11`	`oh!`
`08:55.48`	`jmls`	`qdk: not if they are quad E1's ...`
`08:56.19`	`jmls`	`<PROTECTED>`
`08:57.04`	`tengulre11`	`jmls: is the 2 cards per machine is best?`
`08:57.12`	`tengulre11`	`s/is/does`
`08:58.17`	`qdk`	`jmls: ififififif.`
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`08:59.02`	`jmls`	`qdk: if my aunty had testicles she would be my uncle.`
`08:59.08`	`jmls`	`:)`
`08:59.31`	`jmls`	`tengulre11: yes. But how many ports does your E1 card have ? 1,2, or 4 ?`
`08:59.32`	`qdk`	`jmls: exactly.`
`08:59.40`	`hohum`	`you should NEVER have more than 4 T1s per machine`
`08:59.52`	`jmls`	`qdK: asked tengulre11 a leadiing question :)`
`09:00.29`	`hohum`	`4 T1s or 4 E1s`
`09:00.30`	`hohum`	`max`
`09:00.53`	`jmls`	`hohum: 4 ports or 4 cards ?`
`09:01.42`	`qdk`	`the power of failover/load balancing also rule single point of failure even if one machine could utilize all channels without problems.`
`09:02.13`	`jmls`	`qdK: not without dropping all your calls until the failover happens`
`09:03.04`	`jmls`	`qdk: or do you know something about keeping a call up whilst moving it to another machine ?`
`09:05.29`	`qdk`	`jmls: atm. calls are lost in failover as far as i know, i plan to build my business accepting that fact, for now anyway... Im building my structure around state sharing, but no call failover.`
`09:06.10`	`qdk`	`jmls: http://193.164.155.120/files/visios/VoIP-eksempel.jpg <- my first draft of my structure... with just about no details on it.`
`09:09.30`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
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`09:14.45`	`oliviert`	`hi guys`
`09:15.00`	`jmls`	`yikes! documentation :)`
`09:15.09`	`oliviert`	`i do have a small pb, with my digium TDM400P`
`09:15.45`	`oliviert`	`when i enter ztcfg -VV, it gives me no error, but tells that there's 0 channels configured.`
`09:16.01`	`oliviert`	`I do have 4 fxo ports on the card, what do i miss ?`
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`09:23.28`	`Aurs`	`jmls: yes, tnx`
`09:24.11`	`tparcina`	`iax - how to make authenticated calls?`
`09:25.28`	`hank`	`what means "distinctive ring"?`
`09:27.04`	`qdk`	`hank: out-of-context... a specific kind of ring.`
`09:27.30`	`tzafrir`	`several different ring tones`
`09:27.45`	`tzafrir`	`For analog FXO`
`09:27.48`	`hank`	`qdk: ok thanks`
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`09:30.13`	`hank`	`is an ntba something specific to euroisdn or even germany?`
`09:34.54`	`*** join/#asterisk kristalino (i=kristali@gateway/tor/x-fe57a7cc9fabea62)`
`09:58.24`	`florz`	`hank: Well, it's just not called NTBA, but NT1 instead ;-)`
`09:58.55`	`hank`	`florz: i see, thanks`
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`10:00.52`	`backblue`	`morning!`
`10:01.30`	`dovid`	`morning`
`10:16.04`	`*** join/#asterisk mut (n=ana@65.111.222.120)`
`10:19.23`	`*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
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`10:39.26`	`dovid`	`.`
`10:39.49`	`evilmnky`	`where is the best source for documentation on AEL2`
`10:40.11`	`dovid`	`~seen evilmnky`
`10:40.28`	`jbot`	`evilmnky is currently on #debian (1h 30m 19s) #asterisk (1h 30m 19s). Has said a total of 3 messages. Is idling for 39s, last said: 'where is the best source for documentation on AEL2'.`
`10:40.28`	`evilmnky`	`?`
`10:40.34`	`HaffiH`	`hello, everyone, is anyone here that has some expiriance with , FAXes and Asterisk, useing ZAP Digium Wildcard TE110P E1.`
`10:40.55`	`dalbaech`	`have you tried http://www.voip-info.org/wiki/view/Asterisk+AEL2 yet?`
`10:40.58`	`evilmnky`	`dovid: dont perform commands on me.. monkey`
`10:41.23`	`HaffiH`	`I just did a hardware upgrade and after that my Faxes come partial almost always :S`
`10:42.15`	`evilmnky`	`dalbaech, yeah: was looking for something a little more detailed but atleast it's a start`
`10:49.28`	`tzafrir`	`HaffiH, how do you connect the fax?`
`10:50.38`	`*** join/#asterisk kevin_m (i=spyro2@3digit.de)`
`10:51.27`	`HaffiH`	`Tzafrir: the asterisk is reciving the fax it self and emailing`
`10:51.47`	`tzafrir`	`"self" == ? rxfax?`
`10:52.46`	`*** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net)`
`10:52.51`	`flackes`	`Good morning all`
`10:53.04`	`flackes`	`any one that can give me some help with a GXP- 2000 locking problem?`
`10:53.19`	`HaffiH`	`Tza: yes`
`10:53.37`	`flackes`	`yes to me or yes to tza :P`
`10:53.59`	`HaffiH`	`Flackers: yes to you`
`10:54.04`	`flackes`	`Nice`
`10:54.36`	`flackes`	`well i have sevral gxp - 2000 and they work fine for ages.. then all of a sudden.. one or two of the phones will just report as busy all the time`
`10:54.40`	`flackes`	`untill i reboot it`
`10:54.43`	`flackes`	`then it works fine again`
`10:55.08`	`flackes`	`have the same problem with the HT - 496.. but cant even reboot that through its interface... have to power it off as it locks completly`
`10:55.16`	`HaffiH`	`Sorry flackes, I miss typed, I was saying yes to Tzafrir :p`
`10:55.21`	`flackes`	`LOL`
`10:56.18`	`hohum`	`how do I stop Asterisk from being an RTP proxy`
`10:56.46`	`hohum`	`I have nat=no canreinvite=yes on the relevant peers`
`10:56.52`	`hohum`	`and it still proxies RTP`
`10:57.37`	`*** part/#asterisk HaffiH (i=haffi@adsl5-253.simnet.is)`
`10:59.57`	`*** join/#asterisk oej (n=oej@193.213.21.232)`
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`11:10.59`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`11:11.51`	`stephane_`	`re`
`11:21.24`	`*** join/#asterisk Budairc (n=chatzill@200.215.57.174)`
`11:27.56`	`*** join/#asterisk negativecreep (n=xaeem@host210-2-170-89.isb.dancom.net.pk)`
`11:28.15`	`negativecreep`	`any folk in here who can help with a2billing?`
`11:36.02`	`RoyK`	`billing? wtf would you bill people?`
`11:36.20`	`negativecreep`	`RoyK: requirement.`
`11:39.37`	`RoyK`	`negativecreep: can't you just be nice and give it all away?`
`11:39.39`	`RoyK`	`:)`
`11:41.16`	`negativecreep`	`RoyK: if it was just me, i would love this world to be a free place.`
`11:41.27`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`11:41.50`	`RoyK`	`lol`
`11:44.17`	`*** join/#asterisk soylentgreen (n=fgast@reliant-fxp0.muc.axsp.de)`
`11:45.32`	`Aurs`	`billing... bangs head on keyboarj08jkbk.n ffsjak�bfas`
`11:46.48`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:46.54`	`hank`	`a question about cables: i have a computer with 2 hfc interfaces one in nt and one in te mode. we have 2 or 3 bri lines and 1 pri line from the provider. there is a commercial hardware pbx attached to the pri. would it be correct to connect an ntba to one of the bri lines and to the te card and the nt card to an ntba (without current) via a crossed cable and this ntba to an isdn plug of the commercial pbx?`
`11:47.14`	`hank`	`and: yes, i _am_ confused by all this. just in case it sounds like i am...`
`11:51.07`	`*** join/#asterisk gerhard7 (n=gerhard@82-171-119-13.dsl.ip.tiscali.nl)`
`11:58.32`	`negativecreep`	`lol`
`11:58.45`	`stephane_`	`re`
`11:58.47`	`negativecreep`	`RoyK: after the fast. ;)`
`11:59.14`	`RoyK`	`hehe`
`11:59.18`	`*** join/#asterisk soylentgreen (n=fgast@reliant-fxp0.muc.axsp.de)`
`11:59.24`	`RoyK`	`yeah`
`12:04.09`	`negativecreep`	`a2billing is killiing me now.`
`12:09.07`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`12:12.52`	`*** join/#asterisk adorah (n=admin@84.94.127.28.cable.012.net.il)`
`12:15.18`	`qdk`	`negativecreep: sounds seriously... remember to notify next of ken. :-P`
`12:15.56`	`negativecreep`	`heh..`
`12:16.03`	`negativecreep`	`the call is passed fine to a2b`
`12:16.06`	`negativecreep`	`but then a2b exits.`
`12:16.07`	`negativecreep`	`:(`
`12:16.16`	`negativecreep`	`i shall check after the fast is over.`
`12:16.19`	`negativecreep`	`need coffee.`
`12:16.20`	`*** part/#asterisk negativecreep (n=xaeem@host210-2-170-89.isb.dancom.net.pk)`
`12:16.25`	`*** join/#asterisk tuxd00d (n=tuxinato@128.187.128.38)`
`12:16.50`	`qdk`	`negativecreep: have you look in logs? CLI? debug echoes in a2b?`
`12:17.12`	`qdk`	`looked*`
`12:17.14`	`lilalinux`	`how do I make a pattern that accepts 2-4 digits?`
`12:17.40`	`lilalinux`	`is there a ? quantifier like in regex?`
`12:17.59`	`qdk`	`lilalinux: something like _NNNN.`
`12:18.53`	`qdk`	`lilalinux: forgot how to make the 2-4 part, you might need to have 3 seperate extensions for that.`
`12:19.06`	`lilalinux`	`yeah, that's what im doing at the moment`
`12:19.11`	`lilalinux`	`thought there would be something smarter`
`12:19.24`	`lilalinux`	`What wildcard can I use that includes + ?`
`12:19.35`	`qdk`	`Niklas-: hi there. :-)`
`12:19.43`	`lilalinux`	`for international calls`
`12:19.45`	`*** join/#asterisk vgster (n=vgster@170.252.64.1)`
`12:20.29`	`Niklas-`	`hi ;)`
`12:21.23`	`qdk`	`lilalinux: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns <- doesnt seem like its possible to make the (in perl) {2,4} regexp style.`
`12:21.44`	`qdk`	`lilalinux: i would make the 3 extensions and use macros for them.`
`12:21.49`	`qdk`	`lilalinux: or something like that.`
`12:23.27`	`lilalinux`	`k`
`12:23.40`	`lilalinux`	`and what about the wildcard for [+0-9]?`
`12:23.57`	`qdk`	`lilalinux: not sure what you mean?`
`12:24.14`	`qdk`	`lilalinux: you mean a "dialout" char?`
`12:27.37`	`lilalinux`	`yeah`
`12:27.40`	`lilalinux`	`+49 69 ...`
`12:27.52`	`lilalinux`	`country prefix`
`12:28.58`	`qdk`	`lilalinux: use 00 in stead of +?`
`12:30.03`	`qdk`	`lilalinux: btw. i meant X not N in my earlier example. sorry.`
`12:32.21`	`*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`12:35.04`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`12:41.53`	`*** join/#asterisk postel_ (n=jp@wikimedia/Postel)`
`12:42.33`	`*** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se)`
`12:48.17`	`*** join/#asterisk _deg_ (n=deg@201.15.217.96)`
`12:49.10`	`hank`	`lilalinux: frankfurter?`
`12:49.39`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`12:50.29`	`lilalinux`	`hank: richtich :)`
`12:50.55`	`hank`	`lilalinux: ei guude wie? :)`
`12:52.32`	`*** join/#asterisk FlatFoot (n=simon@80.88.192.113)`
`12:52.41`	`FlatFoot`	`afternoon all`
`12:52.48`	`FlatFoot`	`anyone in from australia ?`
`12:58.24`	`*** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com)`
`12:58.25`	`creativx`	`no`
`12:58.28`	`creativx`	`but i know a guy in australia`
`12:58.53`	`FlatFoot`	`ah just needed to answer a strange pub question`
`12:59.25`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`12:59.35`	`qdk`	`The company /me is working for is expanding to Australia... yet another useless piece of info. :-)`
`13:00.09`	`creativx`	`crikey`
`13:00.23`	`jeremy_g`	`what the heck is wrong with DNID`
`13:00.30`	`creativx`	`i fixed my dnid`
`13:00.31`	`jeremy_g`	`i set (DNID=20323)`
`13:00.51`	`jeremy_g`	`and NoOP(${DNID}) would still wont show it`
`13:01.13`	`jeremy_g`	`the above two lines are after each other in the same context`
`13:01.47`	`jeremy_g`	`set(__DNID=20323)`
`13:01.51`	`jeremy_g`	`and even this doesnt work`
`13:02.01`	`jeremy_g`	`whats wrong?? my configuration? my concept`
`13:03.30`	`creativx`	`well`
`13:03.41`	`creativx`	`good question`
`13:05.32`	`creativx`	`can you give the var another name`
`13:05.33`	`creativx`	`for funs sake`
`13:05.34`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`13:06.24`	`*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)`
`13:07.11`	`*** join/#asterisk frop (n=mrmr@adsl-212-20.37-151.net24.it)`
`13:07.13`	`frop`	`lo`
`13:07.30`	`*** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it)`
`13:10.29`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.231)`
`13:11.05`	`jeremy_g`	`creativx:i want to change the DNID variable`
`13:11.20`	`jeremy_g`	`creativx:do you know the DNID and RDNID stuff`
`13:12.00`	`creativx`	`the dnid var is moved to ${CALLERID(DNID)}`
`13:12.09`	`creativx`	`well ofcouse depending on what version you have`
`13:12.15`	`*** join/#asterisk mtaht4 (n=m@c-68-83-153-44.hsd1.nj.comcast.net)`
`13:13.47`	`jeremy_g`	`svn-r17M`
`13:13.47`	`jeremy_g`	`1.2.10`
`13:14.18`	`creativx`	`yeah try ${CALLERID(xx)}`
`13:14.31`	`creativx`	`where xx might be dnid, rdnid, num, all`
`13:14.34`	`creativx`	`i dont remember all:)`
`13:14.50`	`creativx`	`Gets or sets Caller*ID data on the channel. The allowable datatypes`
`13:14.50`	`creativx`	`are "all", "name", "num", "ANI", "DNID", "RDNIS".`
`13:15.55`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:17.04`	`*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`13:17.11`	`*** join/#asterisk Aristotles (n=Invincib@d211-31-241-93.dsl.nsw.optusnet.com.au)`
`13:17.48`	`*** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au)`
`13:18.45`	`oliviert`	`hi all, i do have problem to setup incoming call for zaptel device`
`13:18.51`	`oliviert`	`anyone can help ?`
`13:19.01`	`*** join/#asterisk ondrej (n=ondrej@ubuntu/member/ondrej)`
`13:19.18`	`ondrej`	`mm all`
`13:19.33`	`mosty`	`oliviert, be more specific, what is the problem exactly?`
`13:20.30`	`tzafrir`	`oliviert, someone might, if you told what it was`
`13:20.48`	`ondrej`	`I would like to create setup in which our support people would paste telephone number into custom application and asterisk would dial the number and get it connected to support people hardphone. Unfortunately I don't know how this is called in english which makes using google quite difficult.`
`13:20.53`	`oliviert`	`mosty: i do have 4 fxo ports, well my main pb is configuring zapata.conf and zapata-auto.conf, and then inbound route`
`13:21.53`	`oliviert`	`what i do want is that any incoming call be taken by ivr and then be on hold until a phone take it`
`13:22.14`	`oliviert`	`but at this time, asterisk does not take incoming call`
`13:22.26`	`Mercestes`	`oliviert: What shows up under zap show status?`
`13:22.53`	`mosty`	`well the ivr thing is done in the dialplan (ie extensions.conf), meanwhile have you setup /etc/zaptel.conf and /etc/asterisk/zapata.conf ?`
`13:24.03`	`oliviert`	`Mercestes: zap show status ?`
`13:24.08`	`dorel__`	`tzafrir; are you currently employed in a voip industry in the country/`
`13:24.13`	`Mercestes`	`yea......`
`13:24.17`	`*** join/#asterisk arisjr (n=arisjr@galois.wahtec.com.br)`
`13:24.45`	`arisjr`	`Hi folks`
`13:25.17`	`Mercestes`	`you type it in the cli...`
`13:25.27`	`Mercestes`	`zap means zaptel, show means....show, and status means....status.`
`13:25.31`	`tzafrir`	`dorel__, http://linmagazine.co.il/node/view/28685`
`13:25.44`	`oliviert`	`mosty: i do think that zaptel.conf is ok, i got fxsks= 1 to 4`
`13:26.01`	`oliviert`	`and zone declare as fr`
`13:26.24`	`arisjr`	`need a help. I still need ztdummy if I use a board that uses other driver and channel (in not zaptel at all)`
`13:26.30`	`arisjr`	`?`
`13:26.51`	`mosty`	`oliviert, and a context to send calls to?`
`13:26.59`	`tzafrir`	`/etc/zaptel.conf is for the kernel-level zaptel . /etc/asterisk/zapata.conf is for the (userspace) asterisk zaptel channel`
`13:27.10`	`oliviert`	`Wildcard TDM400P REV I Board 1 OK 0 0 0`
`13:27.11`	`oliviert`	`ZTDUMMY/1 1 UNCONFIGUR 0 0 0`
`13:27.14`	`mosty`	`oliviert, ignore my last comment, that goes in zapata.conf`
`13:27.22`	`tzafrir`	`arisjr, no`
`13:27.36`	`*** join/#asterisk kristalino (i=kristali@gateway/tor/x-61b592aae8a2e7fb)`
`13:27.47`	`arisjr`	`I mean for moh and conference: do I still need ztdummy if I use a board that DONT use zaptel at all?`
`13:28.12`	`Mercestes`	`That looks like one zap channel to me. But, ok, close enough for now.`
`13:28.15`	`oliviert`	`mosty: the context defined is: from-zaptel`
`13:28.18`	`tzafrir`	`arisjr, it is easy to check if you have a valid timing source: run zttest`
`13:28.22`	`Mercestes`	`oliviert: Do you see anything in the CLI when you dial your number??`
`13:28.31`	`*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)`
`13:28.35`	`drcode`	`hi all`
`13:28.36`	`drcode`	`whats up`
`13:28.44`	`oliviert`	`when dialing out is ok, as i do it under SIP/voip`
`13:28.52`	`arisjr`	`I dont have any zapata channel, I dont use zapata`
`13:28.58`	`oliviert`	`and i do need analog lines only for incoming calls`
`13:28.59`	`drcode`	`I need help with sip`
`13:29.05`	`drcode`	`I am realy newbiew`
`13:29.09`	`tzafrir`	`oliviert, set verbose 3 first`
`13:29.23`	`oliviert`	`done`
`13:29.37`	`arisjr`	`I use a board that has it own channel and driver.`
`13:29.39`	`RoyK`	`~pb`
`13:29.40`	`jbot`	`from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`13:29.41`	`Mercestes`	`oliviert: now dial your # and see what the cli says (if anything)`
`13:29.44`	`tzafrir`	`arisjr, so what board do you have?`
`13:29.46`	`drcode`	`I want to connect with sip client pc to pc`
`13:29.49`	`arisjr`	`digivoice`
`13:29.57`	`arisjr`	`Its a brazilian board`
`13:29.58`	`drcode`	`and also option to call pc to phone lines (I have regular modem)`
`13:30.06`	`*** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR)`
`13:30.31`	`nXOR`	`hello people, would someone amongst you have any expeerience with visdn`
`13:30.41`	`tzafrir`	`"pc to phone" uses some kind of VoIP protocol. Which is in, in your case?`
`13:30.43`	`oliviert`	`i do have the TDM04B`
`13:30.56`	`drcode`	`yes`
`13:31.04`	`oliviert`	`it's the TDM400P with 4 fxo modules`
`13:31.13`	`jeremy_g`	`creativx:no man, that didnt solve it for sure. ${CALLERID(dnid)} and ${DNID} are same. latter is obsolete but still works. i see in the logs 501=20232 where i try to do DNID=20232`
`13:31.21`	`nXOR`	`drcode: is that confirmational response to my quesry ?`
`13:31.27`	`jeremy_g`	`i want to change the DNID`
`13:31.33`	`nXOR`	`query*`
`13:31.36`	`arisjr`	`tzafir, this question is to me?`
`13:31.40`	`quid246`	`ArisJr: Can I send you a PM?`
`13:31.45`	`drcode`	`I am in sip`
`13:31.55`	`drcode`	`sorry for my dumb qustions`
`13:31.58`	`arisjr`	`yes`
`13:32.15`	`oliviert`	`Mercestes: dialing out is working with a sip operator, and this one is working, i do want to handle dialing in, not on sip, but analog lines`
`13:32.37`	`mosty`	`oliviert, what do you see in the asterisk console when you dial in?`
`13:32.49`	`oliviert`	`mosty: hold on`
`13:33.16`	`mosty`	`oliviert, paste it on a paste website somewhere if there's a lot of output`
`13:33.43`	`oliviert`	`mosty: nothing, then i guess, that my channels are not well configure, as asterisk do not hang up the line`
`13:33.55`	*** join/#asterisk trelane` (n=trelane@unaffiliated/trelane)
`13:34.08`	`mosty`	`Mercestes, perhaps console is easier to understand than cli`
`13:34.14`	`drcode`	`to astriks need spicle card from sip client to astriks ? or I can also use regualr modem?`
`13:34.39`	`mosty`	`oliviert, did you run ztcfg ?`
`13:35.11`	`jeremy_g`	`is it possible to change the EXTEN and DNID variable?`
`13:35.39`	`oliviert`	`mosty: yes, give me a paste url, i show you what ztcfg tell me`
`13:36.06`	`mosty`	`oliviert, there's one in the channel topic`
`13:38.30`	`arisjr`	`For moh and conference do I still need ztdummy if I use a board that DONT use zaptel at all?`
`13:39.09`	trelane`	`CDR's a wonderful think`
`13:39.11`	trelane`	`thing`
`13:39.18`	`oliviert`	`mosty: pastbin.ca is not loading`
`13:39.32`	`Mercestes`	trelane`: Salesman? Yea, cdr's make me happy..:)
`13:39.34`	`quid246`	`trelane: calling LD on the company dollar?`
`13:39.42`	`[TK]D-Fender`	`arisjr : Never needed ztdummy for MoH.`
`13:39.46`	`quid246`	`Or caling "entertianment srvices"?`
`13:39.50`	trelane`	`Mercestes, company operator... the phone's in "Do Not Disturb"`
`13:40.10`	trelane`	`quid246, company operator that can't be fucked to talk with the customers`
`13:40.12`	`Mercestes`	trelane`: twitch Seen that one before..lol. You can disable do not disturb on a polycom.
`13:40.22`	`jeremy_g`	`is it possible to change the EXTEN and DNID variable?`
`13:40.24`	`quid246`	`haha`
`13:40.29`	trelane`	`Mercestes, you can disable it on the snoms too, I leave it enabled so they can get up to take a leak`
`13:40.43`	`jeremy_g`	`<PROTECTED>`
`13:40.50`	`quid246`	`trelane: Just give them a piss bottle instead`
`13:41.30`	`arisjr`	`[TK]D-Fender, you mean, you use what board for pstn?`
`13:41.48`	`[TK]D-Fender`	`arisjr : I don't do PSTN at home. VoIP terminated through PRI on teir end.`
`13:42.13`	`[TK]D-Fender`	`arisjr : For MeetMe, yes you will require a Zaptel timing source, but not for MoH.`
`13:42.39`	`[TK]D-Fender`	`arisjr : this is app_conference which doesn't require a timer like MeetMe that you might want to consider (3rd party app)`
`13:43.13`	`jeremy_g`	`can someone pay a little heed to my question`
`13:43.31`	`arisjr`	`[TK]D-Fender, I use a file MoH source (asterisk), and when I get mor than one call on this board, music gets crappy`
`13:43.36`	`jeremy_g`	`<PROTECTED>`
`13:43.42`	`mosty`	`oliviert, google for another done`
`13:43.46`	`mosty`	`another one, even`
`13:44.11`	`[TK]D-Fender`	`arisjr : never heard of that before....`
`13:44.22`	`arisjr`	`[TK]D-Fender, and the asterisk book (future of telephny) tells that ztdummy is for moh and conference`
`13:44.37`	`[TK]D-Fender`	`jeremy_g : Have you considerd that that variable may be READ ONLY?`
`13:44.39`	`Mercestes`	`jeremy_g: I am not 100% certain you can change the DNID jeremy_g but try DNID= instead of "set(DNID)." set requires a special function and I don't believe DNID has that functionality.`
`13:44.48`	`arisjr`	`[TK]D-Fender, so I lost`
`13:44.57`	`arisjr`	`[TK]D-Fender, so I'm lost`
`13:45.01`	`[TK]D-Fender`	`arisjr : I do MoH here without it so the answer is "NO"`
`13:45.04`	`Mercestes`	`or....it could be read only...>.>`
`13:45.09`	`*** join/#asterisk Benj0 (n=BenjO@lev92-1-81-57-180-205.fbx.proxad.net)`
`13:45.12`	`*** join/#asterisk pingwin (i=pingwin@gateway/tor/x-1dd11c236b6ace85)`
`13:45.15`	`mosty`	`arisjr, try it without ztdummy and see if you get an error or not`
`13:46.36`	`arisjr`	`that's the case. I was using without it!! and I was having the same errors I was having with it. the manufactor of the board said I must use ztdummy`
`13:46.45`	`arisjr`	`I think HE is lost`
`13:47.01`	`mosty`	`arisjr, what error message are you getting?`
`13:47.18`	`arisjr`	`no error. just comunication problems`
`13:47.24`	`mosty`	`such as?`
`13:47.29`	`arisjr`	`bad sound and stuff`
`13:47.39`	`mosty`	`bad in what way?`
`13:48.25`	`arisjr`	`music all wrong, like lost packets on PSTN, which by the way, is hard to happen`
`13:49.24`	`arisjr`	`can be codec problem, converting problem, channel problem, ...`
`13:49.25`	`oliviert`	`mosty: can't find anyother on asterisk`
`13:49.46`	`oliviert`	`mosty: all paste bot are dedicated to some channels, but not asterisk`
`13:50.08`	`arisjr`	`but I need to certify that I dont need ztdummy at all if I dont use zapata`
`13:50.28`	`mosty`	`oliviert, rafb.net/paste/`
`13:51.12`	`arisjr`	`lets say, a SIP only asterisk need ztdummy to conference or moh? or for conference...`
`13:51.42`	`pingwin[work]`	`hey I'm having a little difficulty with the queue, any hel;p?`
`13:52.07`	`*** join/#asterisk uwe (n=uwe@dogbert.palnet.com)`
`13:52.10`	`mosty`	`arisjr, ztdummy is needed for timing when you don't have a hardware timer (most commonly on a zaptel board). but i can't remember which features need a timer like that`
`13:52.25`	`oliviert`	`mosty: http://rafb.net/paste/results/Fm19cM67.html`
`13:52.34`	`[TK]D-Fender`	`arisjr : I do not have an Zaptel hardware OR ztdummy. I use MoH. Its FINE. end of story.`
`13:53.14`	`pingwin[work]`	`all I'm trying to get is when you are in the queue, I want to be able to hit '*' or something to be able to exit the queue and direct to a dial, or voicemail`
`13:53.17`	`jeremy_g`	`[TK]D-Fender:yes i now reallize. actually i want to change the From: field of the dial made thru asterisk. i want my other soft phone being called to get a different callerid than current.`
`13:53.22`	`oliviert`	`mosty: zapata.conf : http://rafb.net/paste/results/7MATJ361.html`
`13:53.33`	`pingwin[work]`	`having a little difficulty finding info on it`
`13:53.58`	`[TK]D-Fender`	`pingwin[work] : Set a context in the queue definition and any single-digit exten in there will allw you to exit the queue to it.`
`13:54.11`	`pingwin[work]`	`awesome, thank you`
`13:54.21`	`jeremy_g`	`[TK]D-Fender:i am running an application on the other softphone that takes decision based on the From: field of the SIP INVITE. How can i change that in * . I am expecting to change some CHANNEL_VARIABLE=mynewextension and then Dial(mysoftphone)`
`13:54.31`	`hank`	`'grml.org' i hate that word ;)`
`13:54.38`	`[TK]D-Fender`	`jeremy_g : the set the CALLERID. Set(CALLERID(num)=12345)`
`13:54.43`	`oliviert`	`mosty: zapata-auto.conf: http://rafb.net/paste/results/4pfIZg17.html`
`13:54.50`	`[TK]D-Fender`	`jeremy_g : Set(CALLERID(name)=Schmuck)`
`13:55.08`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`13:55.08`	`*** mode/#asterisk [+o anthm] by ChanServ`
`13:57.44`	`drcode`	`what chip and good hardware recommanded(with good sound)`
`13:58.20`	`Mercestes`	`oliviert: you said you see nothing in the console when you dial your #?`
`13:58.30`	`jeremy_g`	`[TK]D-Fender:this is idiotic.`
`13:58.49`	`xheliox`	`Anyone have a recommendation for buying custom voice prompts? I need someone other than Allison...`
`13:59.02`	`jeremy_g`	`[TK]D-Fender:either you don't understand my question`
`13:59.13`	`oliviert`	`Meersestes: yep, because dialing out is not using the fxos`
`13:59.18`	`*** join/#asterisk jmls (n=asterisk@62.49.235.130)`
`13:59.40`	`oliviert`	`mosty: any idea about my config files ?`
`13:59.47`	`jeremy_g`	`how do i know which variables are read-only?`
`13:59.57`	`Mercestes`	`oliviert: Dumb question. Are you sure your # is ringing to your Zap device??`
`14:00.00`	`jeremy_g`	`is EXTEN ready only`
`14:00.18`	`Mercestes`	`jeremy_g: EXTEN is read only, but if you do a "goto" it will change the EXTEN variable.`
`14:00.44`	`Mercestes`	`jeremy_g: so if you do a goto(69me,1) then your ${EXTEN} will be 69me`
`14:01.13`	`oliviert`	`Mercestes: yes, because while i was playing with the conf files, one itme it hang up, but said to me that no line was wonfigure on this number...`
`14:01.34`	`oliviert`	`Mercestes: did you have a look at my zapata.conf ?`
`14:01.56`	`Mercestes`	`oliviert: Got it open now.`
`14:02.38`	`[TK]D-Fender`	`jeremy_g : CALLERID = "from"`
`14:02.39`	`Mercestes`	`oliviert: And in extensions.conf you have a context [from-zaptel] with an exten => yourNumber,1,Do(something)?`
`14:03.44`	`drcode`	`if I have two sip client I can connect both into astriks?`
`14:04.06`	`drcode`	`and use astriks as sip server (pc to pc call )?`
`14:04.14`	`mosty`	`drcode, yes`
`14:04.25`	`drcode`	`k`
`14:04.41`	`mosty`	`drcode, but learn to spell it properly, else you will have trouble finding docs on google. it's "asterisk"`
`14:04.55`	`Mercestes`	`drcode: You define the clients in sip.conf. host = asterisk1 on asterisk2 and host=asterisk2 on asterisk1. With usernames, passwords, type=friend.`
`14:04.59`	`drcode`	`is there document , how to setup sip server with sip client ?`
`14:05.07`	`drcode`	`k`
`14:05.09`	`Mercestes`	`drcode: Then you can do a dial(sip@<whatever you called it in sip.conf>)`
`14:05.13`	`mosty`	`drcode, there are lots, look on google`
`14:05.15`	`drcode`	`k`
`14:05.24`	`drcode`	`I will try to install`
`14:05.33`	`drcode`	`what is freepbx?`
`14:05.40`	`*** join/#asterisk Qwell_ (i=qwell@unaffiliated/qwell)`
`14:05.40`	`drcode`	`it can manage astrikes?`
`14:05.40`	`*** mode/#asterisk [+o Qwell_] by ChanServ`
`14:06.02`	`oliviert`	`Mercestes: no, i do have extensions at the moment only for sip phone`
`14:06.27`	`Mercestes`	`oliviert: that would be your problem.`
`14:06.59`	`mosty`	`oliviert: if you set verbose 10 and set debug 10 you should at least see the zap line ringing in the asterisk console when you dial in though`
`14:07.00`	`tzafrir`	`drcode, even worse, someone might even decide to sue you for mixing Asterix and UN*X: http://mobilix.org/ ;-)`
`14:07.01`	`oliviert`	`Mercestes: ok that mean that i need to configure a from-zaptel extensions`
`14:07.11`	`Mercestes`	`drcode: it's something that would require you to go to #freepbx to get support on instead of #asterisk...:)`
`14:07.20`	`drcode`	`k`
`14:07.29`	`oliviert`	`mosty: i was in asterisk CLI, how do i access the console then ?`
`14:07.50`	`Mercestes`	`mosty: I guess cli was easier than console afterall.`
`14:07.52`	`mosty`	`oliviert, same thing`
`14:07.55`	`mosty`	`heh`
`14:08.23`	`oliviert`	`mosty: i saw that i do not have /etc/zaptel.conf at all, is it a pb ?`
`14:09.45`	`oliviert`	`mosty: error, it is there, with fxsks=1-4`
`14:10.21`	`ondrej`	`[one more time]: I would like to create setup in which our support people would paste telephone number into our custom application (python whatever) and asterisk would dial this number and get it connected to support people hardphone.`
`14:10.23`	`oliviert`	`Mercestes: why does the from-zaptel need an extension ?`
`14:11.16`	`mosty`	`oliviert: use an extension that matches anything if you want all incoming zap calls treated the same`
`14:11.54`	`mosty`	`ondrej, google "asterisk click to call"`
`14:11.59`	`Mercestes`	`oliviert: Because the asterisk developers were just crazy like that.`
`14:12.20`	`oliviert`	`Mercestes: oh !!`
`14:12.25`	`jmls`	`<PROTECTED>`
`14:12.55`	`Mercestes`	`oliviert: try _X.,1,Playback(tt-weasels)`
`14:13.39`	`jeremy_g`	`Mercestes:you are the man`
`14:13.54`	`Mercestes`	`jeremy_g: I am?`
`14:14.04`	`jeremy_g`	`it worked`
`14:14.17`	`Mercestes`	`it did?`
`14:14.19`	`Mercestes`	`O.O`
`14:14.19`	`jeremy_g`	`but partially`
`14:14.22`	`jeremy_g`	`:)`
`14:14.28`	`Mercestes`	`oh good..I was starting to worry.`
`14:14.35`	`Mercestes`	`D-Fender usually pwns me indiscriminately.`
`14:14.48`	`*** join/#asterisk wulfy814 (n=lorentz@70.90.221.73)`
`14:15.45`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`14:16.22`	`Mercestes`	`jeremy_g: in fact, I believe D-Fender suggested changing the callerid(name) or callerid(number) which is not read only. That might work as well if you can read that information.`
`14:16.29`	`*** join/#asterisk toddf (i=rhsld1z6@net-66-210-111-62.theshop.net)`
`14:16.34`	trelane`	`President Bush says U.S. working to confirm North Korea's nuke test claim`
`14:16.47`	trelane`	`did he not log into the usgs site and check for the big rumble at 1430-1450 zulu?`
`14:17.22`	`toddf`	`this may be a faq, but .. if one enables transfer capabilities, how does one dial out and enter '0#' to a remote line?`
`14:17.25`	`ondrej`	`mosty: thanks, that's exactly what I needed (i didn't know what to search for...)`
`14:17.37`	`coppice`	`trelane: that doesn't prove it was a nuclear blast. some say it looks fake`
`14:17.46`	`jmls`	`trelane: Read - "we got caught with our pants down on thanksgiving and are scrambling to come up with some form of coherent response coz we had no idea that the bastards would ignore us"`
`14:17.57`	`*** join/#asterisk pifiu-laptop (n=someone@216.5.79.1)`
`14:18.49`	`toddf`	`aka you use '#' to initate transfer, how do you send '#' to the remote party?`
`14:18.56`	`toddf`	`## doesn't seem to work ;-(`
`14:19.33`	`oliviert`	`Mercestes: my pb is that asterisk does not hang up the line`
`14:19.54`	`oliviert`	`Mercestes: do i need to create that from-zaptel extensions ?`
`14:20.21`	`Mercestes`	`oliviert: easy enough. [from-zaptel] _X.,1,Answer()`
`14:20.39`	`Mercestes`	`_X.,2,Playback(Goodbye)`
`14:20.44`	`Mercestes`	`_X.,3,Hangup()`
`14:21.03`	`*** join/#asterisk boris2 (n=pjf@adsl-ull-238-218.47-151.net24.it)`
`14:21.53`	`mosty`	`oliviert, so incoming calls enter your dialplan correctly now? google for "asterisk hangup detection" for some starters on that`
`14:22.11`	`*** part/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR)`
`14:22.16`	`jmls`	`oliviert: don't forget to either restart * or "extensions reload" from the cli`
`14:22.24`	`*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`14:23.46`	`bXi`	`yo`
`14:23.56`	`bXi`	`i'm having an issue with visdn and asterisk`
`14:24.00`	`oliviert`	`jmls: i did restart asterisk`
`14:24.27`	`bXi`	`asterisk is seeing when i call the number asigned to visdn0`
`14:24.29`	`jmls`	`oliviert: ok, I was just checking`
`14:24.50`	`bXi`	`but on the other end i get something telling me that the number is unavailable`
`14:24.55`	`bXi`	`asterisk gives me this http://pastebin.ca/195405`
`14:25.19`	`*** join/#asterisk pabluss (n=aquicamb@200.75.1.29)`
`14:25.22`	`pabluss`	`morning`
`14:25.38`	`jmls`	`pabluss: afternoon ;)`
`14:27.35`	`*** part/#asterisk StyleWarz (i=stylewar@euphoria.evil-packet.org)`
`14:28.35`	`pabluss`	`jmls: where do you from?`
`14:29.17`	`*** join/#asterisk NormanASD (n=norman@206.135.58.98)`
`14:29.26`	`pabluss`	`jmls: i'm in Stgo of Chile`
`14:30.04`	`jmls`	`pabluss: UK`
`14:30.43`	`jeremy_g`	`Mercestes:if i dial an extension, where i am greeted and goto(some other extensio) then in that some other extension, is it possible to do a hangup for the first extension that i dialled.`
`14:30.54`	`jeremy_g`	`my case is a little weird`
`14:31.30`	`Mercestes`	`jeremy_g: no, because calls don't "spawn" they are only moved to new directives. exactly what are you trying to do? Give me an example.`
`14:36.53`	`*** join/#asterisk dovid (n=dovi5988@85.159.160.196)`
`14:37.26`	`oliviert`	`Mercestes: ok, now asterisk hang up the line, but it says that the # i dialed is incorrect`
`14:37.42`	`Mercestes`	`...`
`14:37.53`	`mosty`	`oliviert, change your dialplan`
`14:38.34`	`hohum`	`how do I get Asterisk to STOP acting like an RTP proxy when it bridges 2 endpoints together`
`14:39.24`	`Mercestes`	`What do you want it to do, hohum?`
`14:39.39`	`toerkeium`	`Guys, I have added "55,1,ChanSpy(Exten)" to my extensions.conf, how can I make that when I dial 55 and then after the extension number, I can hear the conversation for that extension? instead of adding ChanSpy(Exten) statically ?`
`14:39.45`	`oliviert`	`mosty: it says that the # i dialed is not in service, but my dialplan set the incoming call to go to the sip phone or to IVR`
`14:40.52`	`mosty`	`oliviert: is the voice you hear coming from asterisk or from your phone service provider?`
`14:40.57`	`jmls`	`toerkeium: *55,1,Read(SpyOn)`
`14:41.11`	`toerkeium`	`thanks jmls, gonna try that`
`14:41.16`	`jmls`	`*55,2,ChanSpy(${Spyon})`
`14:41.51`	`toerkeium`	`nice, thank you very much`
`14:42.39`	`mosty`	`hohum, see the canreinvite option in sip.conf`
`14:42.39`	`oliviert`	`mosty: from asterisk`
`14:42.39`	`*** join/#asterisk soylentgreen (n=fgast@193.238.89.34)`
`14:43.02`	`mosty`	`oliviert, then your dialplan is broken, or you haven't reloaded the diaplan since you changed it last`
`14:43.06`	`oliviert`	`mosty: in asterisk report, i got the log that it answered the line, and direct it to s`
`14:43.35`	`oliviert`	`mosty: well for every change i made i did restart asterisk, to be sure everything is reloaded`
`14:44.22`	`oliviert`	`mosty: what i want is an ivr answer the line, say a kind of welcome, and then transfer it to any phone available`
`14:44.54`	`hohum`	`mosty: well I have canreinvite=yes on the relevant peers`
`14:45.03`	`hohum`	`and it STILL insists on being an RTP proxy`
`14:45.08`	`hohum`	`insists rather`
`14:45.15`	`mosty`	`hohum, i suspect that the phones also need to support that`
`14:45.50`	`hohum`	`how does the asterisk box know that the phones won't do it?`
`14:46.06`	`hohum`	`its just information contained in the SDP is all`
`14:46.35`	`[TK]D-Fender`	`hohum : You need both ends of the call to say "canreinvite=yes"/`
`14:46.39`	`mosty`	`asterisk could just not pass reinvite's on`
`14:46.42`	`oliviert`	`mosty: my from-zaptel dialplan: http://rafb.net/paste/results/qMfevR71.html`
`14:47.25`	`*** join/#asterisk jaxzn (n=s123@S0106006097940f68.vw.shawcable.net)`
`14:47.47`	`*** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net)`
`14:47.51`	`hohum`	`both ends do though`
`14:48.02`	`hohum`	`both ends say canreinvite=yes`
`14:48.07`	`hohum`	`asterisk is being stubborne about it`
`14:48.18`	`jaxzn`	`does anyone know how one would be able to dial asterisk box, hang up, then have asterisk call back the number who just dialed with dialtone ready ?`
`14:48.30`	`mosty`	`hohum, but do the phones support it?`
`14:48.45`	`hohum`	`mosty:yes`
`14:49.14`	`dovid`	`jaxzn: yes, there are scripts out there that will do it or u can write ur own agi script`
`14:49.46`	`mosty`	`oliviert, that looks rather complex, why don't you just play the message and then dial all the phones you want to ring?`
`14:49.50`	`jaxzn`	`dovid, can you point me to some of them ?`
`14:49.50`	`*** join/#asterisk saftsack (n=saftsack@p54A7EA4C.dip.t-dialin.net)`
`14:50.01`	`dovid`	`google`
`14:50.06`	`dovid`	`for call back`
`14:50.26`	`dovid`	`here`
`14:50.26`	`dovid`	`http://www.google.com/search?sourceid=navclient&ie=UTF-8&rls=GGLR,GGLR:2006-36,GGLR:en&q=asterisk+call+back+script`
`14:50.35`	`Flauto`	`anyone uses gtalk with`
`14:50.39`	`Flauto`	`please help me`
`14:51.11`	`oliviert`	`mosty: it what i'm tring to do, i even try to set it up with freepbx, without success`
`14:51.29`	`*** join/#asterisk murf (n=steve_mu@216.166.159.235)`
`14:52.02`	`mosty`	`oliviert, set debug 10, set verbose 10 and watch the console/cli when you dial in. you should see how it steps through the dialplan and what any errors are`
`14:53.33`	`*** part/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`14:58.17`	`[TK]D-Fender`	`oliviert : Um... what are you trying to do in there? That approah is lloking pretty "off".`
`15:00.18`	`*** join/#asterisk afrosheen (n=cj@txprotoa2.august.net)`
`15:02.13`	`oliviert`	`mosty: ok here is the dialplan how it occurs: http://rafb.net/paste/results/zjkIM198.html`
`15:04.29`	`*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)`
`15:06.27`	`*** join/#asterisk sloth_ (n=sloth@pool-162-83-220-153.ny5030.east.verizon.net)`
`15:07.13`	`sloth_`	`hello room`
`15:07.25`	`[TK]D-Fender`	`oliviert : Fine , but WHAt are you trying to do?`
`15:07.45`	`pabluss`	`hi [TK]D-Fender, morning`
`15:07.57`	`Optic`	`hi`
`15:08.00`	`Optic`	`happy turkey day`
`15:08.00`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`15:08.18`	`*** join/#asterisk watchy (n=watchy@office2.gwhsi.com)`
`15:08.33`	`watchy`	`anyone able to paste me a small dialplan for international calls?`
`15:09.05`	`[TK]D-Fender`	`watchy : exten => _011.,1,Dial(Zap/g1/${EXTEN})`
`15:09.13`	`[TK]D-Fender`	`watchy : there`
`15:10.29`	`watchy`	`thanks`
`15:10.39`	`syzygyBSD`	`[TK]D-Fender: you make it look so easy`
`15:10.44`	`watchy`	`customer couldnt dial int and i never even thought about adding it to there server`
`15:10.54`	`watchy`	`i didnt realise you had to dial 011 first of all`
`15:11.01`	`afrosheen`	`syzygyBSD, it's easier than typing his nick for autocompletion :)`
`15:11.05`	`*** join/#asterisk HobNobblin (n=HobNobbl@65.204.35.98)`
`15:11.18`	`[TK]D-Fender`	`watchy : Wake up and smell the dial-tone!`
`15:11.29`	`watchy`	`hehe hows monday treating you tk`
`15:11.37`	`syzygyBSD`	`I don't know, his nick is really easy for auto completion`
`15:11.40`	`[TK]D-Fender`	`watchy : feh.`
`15:11.57`	`watchy`	`i feel the same way`
`15:12.15`	`FlatFoot`	`Optic: Happy Turkey Day ?????????`
`15:12.17`	`[TK]D-Fender`	`Its monday... at least I'm not at work.`
`15:12.28`	`HobNobblin`	`Any feelings from anyone on the GXP-2000?`
`15:12.48`	`afrosheen`	`nope`
`15:13.26`	`[TK]D-Fender`	`HobNobblin : Hunk of shit.`
`15:13.35`	`toerkeium`	`jmls: while spying.. is there any way I could change the extension number without hunging up and dialing *55 again?`
`15:13.46`	`*** join/#asterisk RoyK (n=roy@ti211310a080-9888.bb.online.no)`
`15:13.49`	`HobNobblin`	`Really? Any suggestions in that price range?`
`15:14.17`	`[TK]D-Fender`	`HobNobblin : not that low. Look at Aastra or Polycom if you want something decent. Linksys is a 3rd rank option.`
`15:14.22`	`sloth_`	`I am using WaitExten() and the 't' extension to repeat a message if no digits are pressed. What I would like to have happen is if the 't' extension is reached 3 times for the call to be hungup. What is the best way to implement this?`
`15:14.32`	`watchy`	`i woke up this morning puking`
`15:14.40`	`watchy`	`i think ive got scroat cancer`
`15:14.40`	`*** join/#asterisk pigpen (n=mark@fw.seamans.cc)`
`15:14.44`	`HobNobblin`	`Thanks`
`15:14.53`	`[TK]D-Fender`	`sloth_ : About 10 lines of dialplan logic to maintain a counter.`
`15:14.56`	`afrosheen`	`HobNobblin, how much are you willing to shell out`
`15:15.17`	`afrosheen`	`watchy, scroat cancer? like, in your balls?`
`15:15.20`	`HobNobblin`	`I was hoping to be in the $100-$150ish range`
`15:15.31`	`[TK]D-Fender`	`HobNobblin : What vkind of call volume are you going to have on this phone? need speakerphone? got/planning on getting PoE?`
`15:15.42`	`watchy`	`tk: i got a co with 8 lines, they might be adding 4 more should i go with a t1 trunk or what? they are analog now`
`15:15.59`	`hohum`	`man this is pissing me off`
`15:16.01`	`watchy`	`afrosheen: well it burns when i pee`
`15:16.04`	`afrosheen`	`HobNobblin, the new polycom 430's are around that price`
`15:16.14`	`afrosheen`	`watchy, hookers?`
`15:16.20`	`[TK]D-Fender`	`watchy : Depends on cost and what you really want. I'd say yes if you hit 8 line and a fractional PRI isn't too much more.`
`15:16.22`	`hohum`	`asterisk is so ubershitty at doing certain things`
`15:16.24`	`HobNobblin`	`no PoE. speakerphone is essential. these phones will be used on a manufacturing floor too so need to be durable`
`15:16.25`	`watchy`	`afro: indeed`
`15:16.29`	`pigpen`	`polycom 430: Note: no dnd button...my customer really disliked that.`
`15:16.39`	`[TK]D-Fender`	`HobNobblin : Basic use I guess?`
`15:16.46`	`afrosheen`	`watchy, go get a shot`
`15:17.02`	`syzygyBSD`	`hohum: like what?`
`15:17.07`	`watchy`	`wouldnt that be an insult to my sister`
`15:17.13`	`[TK]D-Fender`	`pigpen : MAKE one for them. And you an still go into DND, just not native witha single button.`
`15:17.13`	`afrosheen`	`lol`
`15:17.26`	`stephane_`	`re`
`15:17.27`	`*** join/#asterisk anthonyl (i=anthony@nat/digium/x-005b02bdd459d0d8)`
`15:17.28`	`hohum`	`syzgyBSD: I want it to STOP proxying RTP streams`
`15:17.37`	`afrosheen`	`pigpen, bah, put it in your dialplan`
`15:17.37`	`hohum`	`I have canreinvite=yes on ALL relevant peers`
`15:17.41`	`watchy`	`you like the new firmware on polycoms tk? the 3.0 or whatever`
`15:17.43`	`HobNobblin`	`I'm probably looking at getting basic but durable units for the manufacturing floor and more feature full units for office`
`15:17.47`	`pigpen`	`yeah..now they have to go through the menu...I have not looked into creating a button.`
`15:17.48`	`hohum`	`but it INSISTS on being in the RTP stream`
`15:18.04`	`syzygyBSD`	`hohum: do you have insecure=very?`
`15:18.10`	`pigpen`	`but yes..the dialplan is probably a better route, especially with the privacy manager I am running.`
`15:18.22`	`hohum`	`syz: yes, on some, do I need to turn that off?`
`15:18.22`	`afrosheen`	`pigpen, that's right, polycoms come with keycaps and you can relabel stuff`
`15:18.39`	`[TK]D-Fender`	`HobNobblin : ok, then if you're looking on a budget, get an Aastra 9112 for that "beat around" phone, and either 480i's or Polycom's for the office. www.telephonydepot.com.`
`15:18.42`	`syzygyBSD`	`no, should be on all`
`15:18.46`	`hohum`	`okay`
`15:18.47`	`hohum`	`thanks`
`15:18.49`	`hohum`	`let me try that`
`15:19.00`	`HobNobblin`	`thanks, I'll take a look at those`
`15:19.07`	`hohum`	`what about cancallforward=yes?`
`15:19.22`	`[TK]D-Fender`	`watchy : SIP 2.0.1. is faster than previous releases and add much better NAT support, as well as audio tweaking capabilities.`
`15:19.46`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`15:20.01`	`syzygyBSD`	`hohum: that shouldn't matter`
`15:20.46`	`afrosheen`	`[TK]D-Fender, are you talking about the new polycom firmware?`
`15:21.16`	`[TK]D-Fender`	`afrosheen : yes`
`15:21.37`	`tamp4x`	`anyone here have a load balanced asterisk set up?`
`15:21.44`	`*** join/#asterisk in (n=int@24-107-57-39.dhcp.stls.mo.charter.com)`
`15:22.01`	`afrosheen`	`our latest batch of 501's shipped with the new firmware, nobody told us that when we ordered them...so now we maintain 2 configs for each set`
`15:22.36`	`watchy`	`you need a config for each set?`
`15:22.47`	`watchy`	`is it that much different?`
`15:24.02`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`15:25.27`	`sloth_`	`tk: for my counter should exten => t,1,Set(i=1) be a good start?`
`15:25.29`	`*** join/#asterisk pifiu-laptop (n=someone@216.5.79.1)`
`15:26.14`	`*** join/#asterisk DasTech (n=DasTech@ppp-71-128-71-74.dsl.irvnca.pacbell.net)`
`15:26.21`	`afrosheen`	`watchy, well one ftp folder holds the older firmware, one holds the newer, so we maintain 2 sets of everything`
`15:26.33`	`afrosheen`	`watchy, and then we have 2 logins, bla bla`
`15:26.54`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`15:27.45`	`[TK]D-Fender`	`sloth_ : Nope. Logic flaw. that will get set on EVERY timout.....`
`15:28.31`	`sloth_`	`Ah, where is the best place for this logic, or is this logic not logical at all?`
`15:28.42`	`*** join/#asterisk SomethingISODD (n=Dan@h109.42.63.69.cable.ottr.cablerocket.net)`
`15:28.51`	`SomethingISODD`	`hello how do i change a run tone from usa to euro??`
`15:29.24`	`[TK]D-Fender`	`sloth_ : Use your imagination and think step-by-step how the system yshould think as the call progresses. This is programming 101`
`15:29.27`	`watchy`	`afro: oh yea, i didnt even think about that`
`15:29.28`	`uwe`	`hello, im trying to configure asterisk with zaptel tdm400p connected to regular phone line, so far everything is fine except that when i try to call out it rings once/twice and then i get this tu-ta-ti sound (couldnt describe it better ;) ) i suppose this is related to the zone i define in zaptel.conf, am i correct?`
`15:30.06`	`watchy`	`im kinda outta it today, i woke up sick and i still dont feel good`
`15:31.11`	`*** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net)`
`15:31.23`	`SomethingISODD`	`anyone??`
`15:31.33`	`SomethingISODD`	i have changed it in zapatel and zaptel.conf`
`15:31.36`	`SomethingISODD`	conf`s
`15:31.53`	`[TK]D-Fender`	`SomethingISODD : And what are you listening to it on?`
`15:32.08`	`SomethingISODD`	`shitty GS`
`15:32.39`	`[TK]D-Fender`	`SomethingISODD : then tahts whree you need to tell it.`
`15:32.39`	`SomethingISODD`	`ya but i thought Asterisk is the thing that generates the ring tone`
`15:32.51`	`[TK]D-Fender`	`SomethingISODD : Because sip only tells a phone the call is "ringing", or "busy" or whatever and its the GS that decides what tones to play.`
`15:33.10`	`[TK]D-Fender`	`SomethingISODD : No, SIP sends a STATE, not the audio you think is associated with that state.`
`15:33.44`	`SomethingISODD`	`oh ok.`
`15:33.46`	`SomethingISODD`	`thanks`
`15:34.15`	`uwe`	`um, or can someone tell me what the name of the voice tu-ta-ti is or a reference where i can find out? keywords?`
`15:34.17`	`SomethingISODD`	`if i called in on a did with the ring tone change, would that tell me if it actually changed?`
`15:34.35`	`[TK]D-Fender`	`SomethingISODD : And if you can't set it to the zone you want, add that to the list of "Even more reasons to not touch GS even WITH a 10' pole"`
`15:34.56`	`SomethingISODD`	`[TK]D-Fender, oh i know it is`
`15:35.02`	`[TK]D-Fender`	`SomethingISODD : Hearing is believing.`
`15:35.02`	`*** part/#asterisk HobNobblin (n=HobNobbl@65.204.35.98)`
`15:35.03`	`SomethingISODD`	`GS is shit..`
`15:35.06`	`wunderkin`	`uwe: ta-ta sound? heheheheh you mean the tri-tone/SIT when a number is disconnected?`
`15:35.37`	`*** join/#asterisk Ebola (n=Ebola@host81-132-187-57.range81-132.btcentralplus.com)`
`15:36.18`	`wunderkin`	`uwe: maybe add a w or a few before you dial the number`
`15:36.35`	`jmls`	`toerkeium: show application extenspy`
`15:37.13`	`uwe`	`hehee.. tahnks a lot wunderkin`
`15:37.18`	`uwe`	`*thanks`
`15:37.22`	`jmls`	`toerkeium: the best way is to set the SPYGROUP variable then you can cycle through all matching extensions`
`15:38.11`	`uwe`	`i think ive got everything working now :D`
`15:41.45`	`oliviert`	`where are stored the on hold music on the file system ?`
`15:41.56`	`DasTech`	`man the fbsd portis way behine`
`15:42.01`	`DasTech`	`behind`
`15:42.07`	`DasTech`	`its still 1.2.9.1`
`15:42.09`	`DasTech`	`grrr`
`15:42.21`	`DasTech`	`who is doingthe port ?`
`15:43.49`	`afrosheen`	`oliviert, probably /var/lib/asterisk/moh or close to that`
`15:45.38`	`oliviert`	`afrosheen: thanks`
`15:47.03`	`anthonyl`	`bonjour`
`15:47.10`	`*** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net)`
`15:47.21`	`*** join/#asterisk vgster (n=vgster@170.252.64.1)`
`15:51.02`	`*** join/#asterisk mut (n=ana@65.111.222.120)`
`15:53.03`	`DasTech`	`anyone here built 1.4 on bsd yet ?`
`15:54.43`	`*** join/#asterisk jimbo- (n=jhio8838@cpe-74-65-239-78.nyc.res.rr.com)`
`15:54.53`	`jimbo-`	`question regarding asterisk 1.4`
`15:55.19`	`jimbo-`	`configure & make won't build res_odbc`
`15:55.31`	`jimbo-`	`i have unixODBC-devel installed`
`15:55.34`	`jimbo-`	`fedora core 4`
`15:55.42`	`jimbo-`	`is there anything I'm missing?`
`15:55.46`	`*** join/#asterisk jake1932 (n=Administ@pool-68-236-49-109.phil.east.verizon.net)`
`15:56.53`	`jake1932`	`anthm: i found cdr_shell on your site. any pointers in how to integrate it with asterisk?`
`15:59.15`	`tamp4x`	`anyone here dso a load balanced solution with asterisk?`
`15:59.25`	`tamp4x`	`for dynamically registering users`
`16:00.01`	`*** join/#asterisk slobberknocker (n=ckwall@63.149.122.93)`
`16:00.23`	`*** join/#asterisk kristalino (i=kristali@gateway/tor/x-de786dfa8cc0b4a1)`
`16:00.24`	`*** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net)`
`16:01.04`	`*** join/#asterisk luke-jr_work (n=luke-jr@adsl-70-128-250-253.dsl.ksc2mo.swbell.net)`
`16:01.11`	`DasTech`	`get a f5 or build a load balancer`
`16:01.21`	`anthm`	`you just compile it and load it`
`16:01.24`	`tamp4x`	`f5?`
`16:01.33`	`anthm`	`if it still works =0`
`16:01.57`	`DasTech`	`its a loadbalance`
`16:02.29`	`jake1932`	`anthm: i was going to modify it a bit to do http stuff with libcurl. didn't seem to compile (at least on 1.2.9.1). Maybe i was not doing it right`
`16:04.24`	`anthm`	`go to the src of asterisk`
`16:04.31`	`anthm`	`cp ./contrib/scripts/astxs /bin`
`16:04.42`	`anthm`	`chmod u+rx /bin/astxs`
`16:05.00`	`anthm`	`astxs /path/to/cdr_shell.c`
`16:05.18`	`anthm`	`if you dont have a /usr/src/asterisk you need to add`
`16:05.25`	`jake1932`	`i do`
`16:05.34`	`anthm`	`ok that's all then`
`16:05.42`	`luke-jr_work`	`Trying to debug a tinny-like sound on calls... do Sangoma cards just suck?`
`16:05.42`	`jake1932`	`very nice. tnx!`
`16:06.23`	`DasTech`	`no sangommas rock`
`16:06.24`	`*** join/#asterisk folsson (n=folsson@h100n2fls33o985.telia.com)`
`16:06.28`	`luke-jr_work`	`(IP-only calls work fine)`
`16:06.33`	`DasTech`	`we use them left and right`
`16:07.06`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`16:07.33`	`jmls`	`we use them centre ..`
`16:08.15`	`*** part/#asterisk jimbo- (n=jhio8838@cpe-74-65-239-78.nyc.res.rr.com)`
`16:08.26`	`luke-jr_work`	`DasTech, could I have it misconfigured such then? :\`
`16:09.09`	`DasTech`	`could be. when I have a fre min I will get you the confis form our working units`
`16:09.17`	`DasTech`	`ich card do you have ?`
`16:09.25`	`DasTech`	`so I get you the right ones`
`16:09.40`	`DasTech`	`currently updating 54 servers`
`16:09.50`	`DasTech`	`12 of wich are fbsd/asterisk`
`16:09.55`	`luke-jr_work`	`A101`
`16:10.05`	`DasTech`	`ok`
`16:10.25`	`luke-jr_work`	`I suspect the card mainly because bridged analog-to-analog faxes fail`
`16:11.02`	`[TK]D-Fender`	`luke-jr_ : You must be doing something very wrong, or have some phenominally bad luck.`
`16:11.10`	`luke-jr_work`	`and like I said, SIP-to-analog (or vice versa) also have a tin-like sound on the SIP end around the SIP-speaker's voice`
`16:11.30`	`luke-jr_work`	`hm`
`16:11.55`	`luke-jr_work`	`or the guy who put this system together originally was getting rid of a bad card o.o`
`16:11.56`	`[TK]D-Fender`	`luke-jr_ : You need to make sure you have your cards firmware up to date, and the latest Wanpipe. They HAVE had some very know problems with faxing, but even Digium disavows any responsibility for faxing on their cards.`
`16:12.07`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`16:12.08`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:12.08`	`*** mode/#asterisk [+o mog] by ChanServ`
`16:12.08`	`jake1932`	`anthm: no errors, but the .so is not in the asterisk mods folder.`
`16:12.54`	`DasTech`	`you have to put it there`
`16:13.02`	`*** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-9a13ccb52447277d)`
`16:13.02`	`DasTech`	`jake`
`16:13.15`	`DasTech`	`unless you mod themakefile to move it for you`
`16:13.15`	`jake1932`	`<PROTECTED>`
`16:13.16`	`*** join/#asterisk adorah (n=admin@84.94.127.28.cable.012.net.il)`
`16:13.30`	`jake1932`	`(i just did a full search)`
`16:13.31`	`DasTech`	`then it did not make`
`16:13.56`	`jake1932`	`gcc-3.4 -I/usr/src/asterisk -I/usr/src/asterisk/include -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O0 -DZAPTEL_OPTIMIZATIONS -DAST_JB -fomit-frame-pointer -fPIC -c /usr/src/ic/cdr_shell.c -o /usr/src/ic/cdr_shell.o`
`16:14.16`	`jake1932`	`wouldn't it have given an error if it didn't compile?`
`16:14.55`	`*** join/#asterisk andresmujica (n=andresmu@201.244.245.86)`
`16:15.32`	`DasTech`	`brb fone work`
`16:15.50`	`anthm`	`if you want it to install`
`16:15.55`	`anthm`	`astxs -install /path/to/cdr_shell.c`
`16:16.05`	`anthm`	`if your box is already up`
`16:16.10`	`anthm`	`astxs -autoload -install /path/to/cdr_shell.c`
`16:16.21`	`andresmujica`	`anyone knows something about r2dtmf ???`
`16:16.25`	`luke-jr_work`	`so where does Sangoma distribute their firmware? :\`
`16:16.25`	`jake1932`	`yep - tried that too (found it on your site)`
`16:16.51`	`jake1932`	`shouldn't it create a .so?`
`16:17.25`	`DasTech`	`anthm what is this site ?`
`16:17.25`	`anthm`	`if you use autoload and ast is not running the install wont work`
`16:17.28`	`anthm`	`try just install`
`16:17.37`	`jake1932`	`ok`
`16:17.53`	`*** join/#asterisk postel (n=jp@wikimedia/Postel)`
`16:18.06`	`[TK]D-Fender`	`luke-jr_ : its all on their FTP`
`16:18.25`	`anthm`	`i think he's talking about http://www.pbxfreeware.org`
`16:18.43`	`jake1932`	`indeed`
`16:19.22`	`jake1932`	`asterisk:/usr/src/ic# astxs -install /usr/src/cdr_shell.c`
`16:19.23`	`jake1932`	`gcc-3.4 -I/usr/src/asterisk -I/usr/src/asterisk/include -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O0 -DZAPTEL_OPTIMIZATIONS -DAST_JB -fomit-frame-pointer -fPIC -c /usr/src/cdr_shell.c -o /usr/src/cdr_shell.o`
`16:19.40`	`jake1932`	`that's all i get`
`16:20.39`	`*** join/#asterisk toxap (n=toxap@213.227.193.75)`
`16:21.46`	`anthm`	`gcc -shared -Xlinker -x -o /usr/src/cdr_shell.so /usr/src/cdr_shell.o`
`16:21.49`	`anthm`	`should be there too`
`16:21.51`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:22.10`	`anthm`	`maybe the astxs is broke ?`
`16:23.18`	`anthm`	`http://www.freeswitch.org/asterisk_stuff/astxs`
`16:23.27`	`anthm`	`try that one it works for me`
`16:23.32`	`jake1932`	`it's looking for cdr_shell.o`
`16:23.36`	`anthm`	`if that doesnt work the makefile is probably broke`
`16:23.41`	`jake1932`	`ok`
`16:24.00`	`anthm`	`or you can just run that command i just mentioned by hand`
`16:24.08`	`toxap`	`hi, please give me link for free g729 codec, i want to check some tests at asterisk h323`
`16:24.20`	`jmls`	`anthm: hope you don't mind, I updated #5161 to work with the latest svn trunk`
`16:24.37`	`anthm`	`no that's ok`
`16:24.48`	`anthm`	`that one's getting elderly`
`16:25.00`	`anthm`	`i think it's 13 mos old now`
`16:25.17`	`jmls`	`anthm: I was wanting to use it to allow me to hook into the manager events so that I could then use jabber to send them on`
`16:25.33`	`anthm`	`I wrote an entire new application from scratch in the time that patch has festered in there`
`16:25.47`	`jmls`	`anthm: which one is that`
`16:25.47`	`anthm`	`yep that is one of the reasons i made it to begin with`
`16:25.49`	`toddf`	`am I the only one who notices that when enabling transfers on outbound calls, one cannot send the digit '#' to an outbound call ?`
`16:26.00`	`anthm`	`so you could have apps get events internally`
`16:26.06`	`jake1932`	`anthm: doesn't seem to compile to .o`
`16:26.10`	`toxap`	`hi, please give me link for free g729 codec, i want to check some tests at asterisk h323`
`16:26.24`	`anthm`	`jmls http://www.freeswitch.org`
`16:27.22`	`jmls`	`oh, that small app ;)`
`16:27.30`	`jake1932`	`gcc: /usr/src/ic/cdr_shell.o: No such file or directory`
`16:28.25`	`anthm`	`based on your input i would assume gcc -shared -Xlinker -x -o /usr/src/cdr_shell.so /usr/src/cdr_shell.o`
`16:28.29`	`anthm`	`would complete the task`
`16:29.04`	`anthm`	`did you try the other verison of astxs i pasted you ?`
`16:29.14`	`jake1932`	`yep`
`16:29.22`	`jake1932`	`it's actually the one i have`
`16:29.29`	`*** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`16:29.31`	`jake1932`	`(but i wget'd it anyways)`
`16:30.24`	`*** join/#asterisk slobberknocker (n=ckwall@63.149.122.93)`
`16:31.16`	`*** join/#asterisk _alex_mx_ (n=alex@200.78.229.18)`
`16:31.37`	`docelmo`	`Anyone in here from orlando know PHP and Web Design?`
`16:32.57`	`anthm`	`try http://www.freeswitch.org/asterisk_stuff/cheat_sh`
`16:33.00`	`anthm`	`dl that`
`16:33.05`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:33.05`	`*** mode/#asterisk [+o mog] by ChanServ`
`16:33.08`	`anthm`	`sh cheat_sh /usr/src/cdr_shell`
`16:33.27`	`anthm`	`that might give you a so in /usr/src`
`16:33.45`	`jake1932`	`:o)`
`16:33.58`	`jake1932`	`that did it! tnx`
`16:34.07`	`anthm`	`np`
`16:35.34`	`*** join/#asterisk fiber0pti (n=John@207.114.199.107)`
`16:35.57`	`toxap`	`hi, please give me link for free g729 codec, i want to check some tests at asterisk h323`
`16:36.21`	`jake1932`	`<PROTECTED>`
`16:36.29`	`anthm`	`i bet the more times you ask that question in succession the less likely you will ever get it ;)`
`16:37.10`	`docelmo`	`hehe`
`16:37.15`	`toxap`	`jake1932, I know, but i not have $ in internet`
`16:37.26`	`docelmo`	`Then you dont have g729`
`16:37.27`	`docelmo`	`:)`
`16:37.48`	`jake1932`	`we can work on a barter system`
`16:38.06`	`key2`	`docelmo: still had no time to reencode the video ?`
`16:38.07`	`docelmo`	`possible.. but what does he have that anyone may want?`
`16:38.08`	`brad_mssw`	`toxap: the free for educational use version is linked on voip-info.org`
`16:38.22`	`docelmo`	`I just moved.. Lemme see if it looks decient`
`16:38.57`	`*** part/#asterisk DasTech (n=DasTech@ppp-71-128-71-74.dsl.irvnca.pacbell.net)`
`16:40.01`	`docelmo`	`no the video looks good the audio blows..`
`16:41.38`	`*** join/#asterisk arisjr (n=arisjr@galois.wahtec.com.br)`
`16:44.21`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`16:46.10`	`luke-jr_work`	`[TK]D-Fender, not docs on what to do with them`
`16:47.54`	`luke-jr_work`	`or rather, on how to get the serial number`
`16:49.45`	`*** join/#asterisk Mugatu_ (n=mugatu@unaffiliated/Mugatu)`
`16:49.59`	`arisjr`	`hi folks... xlite and asteriesk, all with gsm. I am havin some noise on conversations. can someone help me?`
`16:51.28`	`Qwell[]`	`Don't most conversations consist of some amount of noise?`
`16:51.48`	`Qwell[]`	`I mean, it would be pretty silly to call somebody, and have both ends muted the entire time`
`16:52.27`	`DrkShdw`	`depends on your definition of noise. by 'noise' I assume you mean 'sound' by his 'noise' i assume he means 'noise' :P`
`16:53.21`	`sloth_`	`TK, thanks you set me on the right path (regarding the counter)`
`16:53.29`	`Qwell[]`	`DrkShdw: yes, indeed. :)`
`16:53.50`	`Qwell[]`	`DrkShdw: the point really, is what type of noise. There are multiple types of "bad" noise, all with different causes/solutions`
`16:54.18`	`arisjr`	`its like a bad tunned radio...`
`16:54.32`	`arisjr`	`using gsm on both ends`
`16:54.42`	`arisjr`	`of course`
`16:57.12`	`De_Mon`	`of course?`
`16:57.22`	`arisjr`	`hmmm bad english`
`16:57.56`	`De_Mon`	`well its just kinda odd to say 'of course' when what you said isn't at all obvious.`
`16:59.49`	`arisjr`	`other thing is with some calls, the communication is like a choper (helicopter)... and one side hears, and the other dont.`
`17:00.06`	`*** join/#asterisk doolph (n=doo@200.46.148.58)`
`17:00.18`	`luke-jr_work`	`bad bandwidth management?`
`17:00.31`	`arisjr`	`is with pstn.`
`17:00.46`	`arisjr`	`and gsm on the end`
`17:00.53`	`arisjr`	`hmmm`
`17:01.43`	`arisjr`	`I am with asterisk on DMZ. I can put it on internal net to minimize latency on the firewall.`
`17:01.49`	`[TK]D-Fender`	`sloth_ : Set your counter before you start looping. Then increment it on "t" or "i" and check if you exceed your maximum. If you don't jsut jsut BELOW the point where you initiaize it to play your menu again.`
`17:03.11`	`arisjr`	`luke-jr_work, this choper noise... Is it normally a bandwitdh problem?`
`17:09.28`	`luke-jr_work`	`arisjr, wtf? gsm doesn't run over PSTN`
`17:09.55`	`*** join/#asterisk ScurvyDawg (n=scurvyda@S0106000d883f28a0.gv.shawcable.net)`
`17:10.30`	`*** join/#asterisk rgsteele (n=chatzill@nat-pool.agora-net.com)`
`17:10.41`	`ScurvyDawg`	`I am a newb to Asterisk could you reccomend some documentation for me to start?`
`17:11.02`	`luke-jr_work`	`voip-info.org`
`17:11.11`	`Qwell[]`	`~docs`
`17:11.14`	`jbot`	`somebody said docs was 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`
`17:11.14`	`ScurvyDawg`	`thank you`
`17:11.15`	`Qwell[]`	`~book`
`17:11.16`	`jbot`	`methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`17:11.17`	`arisjr`	`luke-jr_work, yeap... client -> gsm -> asterisk -> slin, gsm codec -> PSTN`
`17:11.29`	`rgsteele`	`Hey guys...Gotta question. I just set up my outgoing calls to get routed through a channel bank provided by the phone company. However, after about 30 seconds on an outgoing call, it begins to break up so badly that I have to hang up. Any ideas?`
`17:11.31`	`ScurvyDawg`	`Ohhh a book perfect`
`17:11.54`	`arisjr`	`luke-jr_work, even`
`17:11.59`	`rgsteele`	`Btw, running asterisk 1.2.9.1`
`17:13.05`	`arisjr`	`luke-jr_work, digium boards work on slinear, isn't it?`
`17:14.15`	`sloth_`	`TK: Thanks I go it.`
`17:15.01`	`ScurvyDawg`	`awesome response for docs thanks all :)`
`17:16.06`	`luke-jr_work`	`arisjr, needless to say, client->asterisk deals with bandwidth`
`17:16.55`	`arisjr`	`ok... I get your point. but focusing on the problem, this helicopter noise is always bandwodth problem?`
`17:17.25`	`luke-jr_work`	`arisjr, no idea, my experience is mostly via IP`
`17:17.28`	`doolph`	`my sip.conf registry command is not working any clue?`
`17:18.36`	`luke-jr_work`	`arisjr, cutting in and out suggests bandwidth to me, and only one end hearing the other suggests NAT`
`17:20.14`	`rgsteele`	`Ok, well how about a more simple question: Is there any jitter control via any of the asterisk configuration files?`
`17:20.28`	`quid246`	`Rgsteele: IAX has jitter buffer.`
`17:20.44`	`doolph`	`anyone?`
`17:21.18`	`rgsteele`	`quid246: I've enabled the jitterbuffer in the iax.conf`
`17:21.19`	`[TK]D-Fender`	`sloth_ : Care to show your approach?`
`17:21.33`	`rgsteele`	`But, I still get horrible break-up after about 20 or 30 seconds.`
`17:21.57`	`nvzn`	`does the asterisk 1.4 still require ztdummy or does it use POSIX timers?`
`17:22.01`	`quid246`	`rgsteele: Local or to a provider?`
`17:22.34`	`[TK]D-Fender`	`nvzn : Doesn't technically require eitehr depending on what you want to do.`
`17:22.59`	`rgsteele`	`quid246: To a provider. The outgoing calls get passed to the Asterisk box, which has two TDM400p cards in it. The TDM400P's have analog lines going from them to a channel bank provided by the phone company.`
`17:23.14`	`rgsteele`	`The channel bank terminates the T1 to the phone company.`
`17:23.17`	`doolph`	`what's new with asterisk 1.4?`
`17:23.23`	`nvzn`	`[TK]D-Fender: i was under the impression that the timer was needed for musiconhold and echo stuff`
`17:23.35`	`quid246`	`RG: Do you get breakup on internal calls IAX-to-IAX?`
`17:23.42`	`[TK]D-Fender`	`nvzn : Not for MoH, and what do you mean "echo stuff"?`
`17:24.03`	`nvzn`	`[TK]D-Fender: echo cancellation`
`17:24.45`	`[TK]D-Fender`	`nvzn : EC on what interface?`
`17:25.34`	`nvzn`	`[TK]D-Fender: im not using one`
`17:25.52`	`rgsteele`	`quid246: Double-checking`
`17:25.56`	`[TK]D-Fender`	`nvzn : Then what are you talking about?`
`17:25.57`	`rgsteele`	`Stand by...`
`17:26.00`	`nvzn`	`[TK]D-Fender: so what exactly do these timers do`
`17:26.13`	`nvzn`	`[TK]D-Fender: ztdummy for instance`
`17:26.25`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`17:26.59`	`[TK]D-Fender`	`nvzn : Theya re usef for IAX2 trunking timing, and MeetMe conference timing in cases where you don't have a hardware source.`
`17:27.13`	`*** join/#asterisk andrebarbosa (n=andrebar@83.240.148.220)`
`17:27.19`	`nvzn`	`[TK]D-Fender: so i need it then, since i have an IAX2 trunk`
`17:27.28`	`rgsteele`	`quid246: I can confirm that it does not break up IAX-to-IAX. Calls between internal office phones go from one phone, to the asterisk box, then back to whichever other internal office phone.`
`17:27.33`	`[TK]D-Fender`	`nvzn : Are you indeed using it in "trunk" mode?`
`17:27.42`	`nvzn`	`[TK]D-Fender: yes`
`17:28.21`	`rgsteele`	`quid246: Thinking this might warrant a call to the phone company.`
`17:28.31`	`[TK]D-Fender`	`nvzn : then that is a case where you will need a timing source.`
`17:28.53`	`nvzn`	`[TK]D-Fender: so will 1.4 require ztdummy or does it use POSIX timers?`
`17:29.08`	`andrebarbosa`	`i have a doubt on meetme app`
`17:29.24`	`andrebarbosa`	`if i have on meetme.conf an entry: 1000,1111,1111`
`17:29.50`	`andrebarbosa`	`and if I enter on a dynamic conference vusing meetme(\|Md)`
`17:29.58`	`andrebarbosa`	`and then choose 1000 conference`
`17:30.04`	`andrebarbosa`	`it will not ask for the pin number`
`17:30.07`	`[TK]D-Fender`	`nvzn : ztdummy USES POSIX timers unless you are using a kernel that doesn't support them in which case it uses a UHCI USB interface for that.`
`17:30.15`	`andrebarbosa`	`it should work like this?`
`17:31.15`	`nvzn`	`[TK]D-Fender: does the latter case apply to a freebsd kernel?`
`17:31.27`	`[TK]D-Fender`	`andrebarbosa : youa re supposed to passing it the PIN when you CALL it.`
`17:31.37`	`[TK]D-Fender`	`nvzn : Not sure as far as BSD is concerned`
`17:31.41`	`doolph`	`anyone know where's g729 key stored?`
`17:32.26`	`*** join/#asterisk DocHolliday (i=RogerRab@gateway/tor/x-0bddec5ad5969c89)`
`17:32.36`	`andrebarbosa`	`[TK]D-Fender, the problem is that meetme app does no ask for pin number when entering using meetme(\|Md) (dynamicly)`
`17:33.01`	`[TK]D-Fender`	`andrebarbosa : MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe conference`
`17:33.08`	`*** part/#asterisk andresmujica (n=andresmu@201.244.245.86)`
`17:33.21`	`andrebarbosa`	`ya`
`17:33.21`	`[TK]D-Fender`	`andrebarbosa : read the BIG PRINT and pass it a damn PIN to us.`
`17:33.21`	`[TK]D-Fender`	`use*`
`17:33.45`	`[TK]D-Fender`	`andrebarbosa : if its "dynamic then where the hell is it expect to get a PIn from if you don't PASS it?`
`17:34.25`	`andrebarbosa`	`ok, so`
`17:34.48`	`SplasPood`	`andrebarbosa: you'd wanna use Authenticate() before to handle prompting for the pin...`
`17:34.51`	`*** join/#asterisk backblue (n=igor@82.102.1.42)`
`17:34.59`	`*** join/#asterisk hank (n=hank@netwichtig.de)`
`17:35.05`	`backblue`	`hi, anyone here, played with gsm gateways and bri's?`
`17:35.23`	`andrebarbosa`	`if i got an user on 1000 room (each enter using Meetme(1000) ) and it HAD to enter the pin`
`17:35.41`	`andrebarbosa`	`and then if someone goes using Meetm(\|dM) and enter 1000`
`17:35.46`	`andrebarbosa`	`it will be ask for the pin`
`17:36.08`	`andrebarbosa`	`but if noone is in the 1000 room`
`17:36.20`	`andrebarbosa`	`the MeetMe(\|dM) will not ask the pin`
`17:36.39`	`andrebarbosa`	`is there a way to reserve a conference? even if there is noone inside of it?`
`17:36.56`	`*** part/#asterisk pabluss (n=aquicamb@200.75.1.29)`
`17:38.56`	`*** join/#asterisk saftsack (n=saftsack@p54A7EA4C.dip.t-dialin.net)`
`17:39.24`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`17:41.01`	`backblue`	`andrebarbosa: portugues?`
`17:41.49`	`*** join/#asterisk LostFrog (n=reallyno@wsip-68-225-90-115.dc.dc.cox.net)`
`17:42.07`	`andrebarbosa`	`backblue, yes :)`
`17:43.01`	`backblue`	`andrebarbosa: quem es?`
`17:43.03`	`backblue`	`ashes?`
`17:43.13`	`andrebarbosa`	`? nao`
`17:43.18`	`backblue`	`hum ok`
`17:43.40`	`backblue`	`o ashes � que tava a fazer uma cena dessas de conferencias`
`17:43.46`	`backblue`	`ja fez acho`
`17:44.06`	`andrebarbosa`	`ha algum # de asterisk portugues?`
`17:44.06`	`backblue`	`andrebarbosa: conhe�o-te? :)`
`17:44.20`	`backblue`	`andrebarbosa: na ptnet, aqui nao.`
`17:44.22`	`andrebarbosa`	`acho que nao.. nao venho muito ao irc`
`17:44.31`	`andrebarbosa`	`ah`
`17:44.34`	`*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
`17:44.34`	`andrebarbosa`	`#asterisk tb?`
`17:44.35`	`rgsteele`	`quid246: Just plugged an analog phone into the channel bank and got no breakup`
`17:44.36`	`diclophis-work`	`hello all`
`17:44.45`	`backblue`	`andrebarbosa: de onde �s? nop, canal voip.`
`17:44.53`	`andrebarbosa`	`ok`
`17:45.25`	`backblue`	`andrebarbosa: alguma empresa? ou mais um curioso? :)`
`17:46.32`	`andrebarbosa`	`empresa`
`17:46.34`	`andrebarbosa`	`critical software`
`17:46.43`	`backblue`	`andrebarbosa: ha giro! :) sou da 3gntw.`
`17:46.47`	`sloth_`	`sure thing: exten => s,1,Set(TOCOUNT=0)`
`17:46.47`	`sloth_`	`exten => s,2,Set(TOCOUNT=$[${TOCOUNT} + 1])`
`17:46.47`	`sloth_`	`exten => s,n,Set(TIMEOUT(digit)=3)`
`17:46.48`	`sloth_`	`exten => s,n,Set(TIMEOUT(response)=5)`
`17:46.48`	`sloth_`	`exten => s,n,Background(webpda/webpda-menu)`
`17:46.48`	`sloth_`	`exten => s,n,Noop(This message has been heard ${TOCOUNT} times)`
`17:46.49`	`sloth_`	`exten => s,n,WaitExten(2)`
`17:46.51`	`sloth_`	`exten => t,1,GotoIf($[ ${TOCOUNT} = 3 ]?t,2:s,2)`
`17:46.53`	`sloth_`	`exten => t,2,Voicemail(u500@webpda)`
`17:47.02`	`backblue`	`andrebarbosa: conhe�es?`
`17:47.05`	`[TK]D-Fender`	`sloth_ : PASTEBIN`
`17:47.06`	`sloth_`	`should have used pastbin (sorry)`
`17:47.47`	`[TK]D-Fender`	`sloth_ : Excellent work. Spot-on.`
`17:47.55`	`sloth_`	`Thanks man`
`17:47.55`	`andrebarbosa`	`nao conheco..`
`17:48.00`	`andrebarbosa`	`lisboa?`
`17:48.10`	`[TK]D-Fender`	`sloth_ : I would suggest you create a SEPERATE counter for invalid attempts too.`
`17:48.13`	`diclophis-work`	`why would a SIP channel not report ringing?`
`17:48.22`	`backblue`	`andrebarbosa: nop, famalic�o.`
`17:48.31`	`backblue`	`somos o provider voip .pt mais barato e com mais qualidade`
`17:48.32`	`[TK]D-Fender`	`sloth_ : Anti-Slacker countermeasures`
`17:48.46`	`backblue`	`mas n � o nosso mercado, mas normalmente � por o que o pessoal de voip nos conhe�e`
`17:48.58`	`backblue`	`n � o mercado principal i mean`
`17:49.00`	`[TK]D-Fender`	`sloth_ : I use "iCount" and "tCount" personally.`
`17:49.19`	`andrebarbosa`	`eu aqui trabalho numa caixa que integra varios serviços, e o asterisk é um deles`
`17:49.21`	`backblue`	`o nosso major goal, � deployment de centrais ip.`
`17:49.31`	`andrebarbosa`	`temos alguns acordos com isp voip`
`17:49.33`	`andrebarbosa`	`tipo a netcall`
`17:49.37`	`sloth_`	`nice, good idea`
`17:49.44`	`backblue`	`� tipo o ipbrick da iportalmais`
`17:49.47`	`backblue`	`tb trabalhei l�`
`17:49.58`	`andrebarbosa`	`ya parecido com o ipbrick`
`17:50.00`	`backblue`	`andrebarbosa: nos somos melhores que a netcal`
`17:50.06`	`backblue`	`n � deitar abaixo o trabalho deles`
`17:50.09`	`backblue`	`nada disso`
`17:50.15`	`LostFrog`	`~pb`
`17:50.20`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`17:50.20`	`backblue`	`somos mais competitivos, � o que queria dizer`
`17:50.33`	`andrebarbosa`	`os gajos abusam um bocadito nas chamadas para gsm :\`
`17:50.41`	`backblue`	`nos temos os pre�os mais baratos`
`17:50.46`	`backblue`	`de .pt`
`17:50.51`	`backblue`	`e vao baixar mais`
`17:51.11`	`backblue`	`mas nos agora vamos lan�ar dois tarif�rios, bqr e lcr`
`17:51.16`	`backblue`	`o cliente escolhe o que quiser`
`17:51.24`	`backblue`	`mas vo�es integram com centrais antigas?`
`17:51.36`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`17:51.41`	`aydiosmio`	`viagra?`
`17:51.53`	`LostFrog`	`Is there anyone around that could help me with a blind transfer problem?`
`17:51.57`	`LostFrog`	`It looks like this: http://pastebin.ca/195528`
`17:52.07`	`sloth_`	`Is anyone using Ruby/Rails with Asterisk?`
`17:52.12`	`LostFrog`	`I have a problem with uni-directional audio.`
`17:52.20`	`backblue`	`LostFrog: mat issues?`
`17:52.21`	`backblue`	`nat`
`17:52.27`	`LostFrog`	`Nope.. all VPNed.`
`17:52.37`	`LostFrog`	`And direct calls work fine.`
`17:52.46`	`backblue`	`no nat, so you can see your udp rtp trafic?`
`17:53.15`	`LostFrog`	`yep.`
`17:54.23`	`backblue`	`firewalls?`
`17:54.57`	`DocHolliday`	`i just got yelled at for not making a password the person's name.. gives up on life`
`17:55.22`	`Pj_`	`just put 0000`
`17:55.24`	`Pj_`	`they all love it`
`17:55.29`	`wunderkin`	`THE PASSWORD SHOULD BE 'password', nubb!`
`17:55.40`	`Pj_`	`nah password is too long`
`17:55.53`	`wunderkin`	`admin?`
`17:55.54`	`Pj_`	`people type passw... and they forget the end by the time they get there`
`17:55.56`	`DocHolliday`	`i think 1234 is more appropriate`
`17:56.09`	`*** join/#asterisk VijayG (i=vijay@202.131.145.234)`
`17:56.11`	`Pj_`	`admin is not bad, but only for admin accounts otherwise it ruins the fun`
`17:56.30`	`Pj_`	`DocHolliday: your users will tell you that 0000 is faster to type`
`17:56.34`	`DocHolliday`	`yeah seriously, i got called a "hack"..`
`17:56.35`	`Pj_`	`especially if you got commercials`
`17:56.55`	`DocHolliday`	`ouch :(`
`17:57.01`	`VijayG`	`hello everyone`
`17:57.05`	`DocHolliday`	`if that ever happens i'm getting out of IT`
`17:57.16`	`Pj_`	`(start packing)`
`17:57.23`	`VijayG`	`i need to configure G729 codec in my asterisk server`
`17:57.26`	`kristalino`	`do i need a kernel module for the linksys spa3000 ?`
`17:57.33`	`VijayG`	`how can i do that`
`17:57.38`	`VijayG`	`can anyone guide me for that`
`17:58.16`	`LostFrog`	`Buy it, VijayG. It comes with instructions.`
`17:58.18`	`pingwin[work]`	`is there a way to make a single playback function call play multiple files in order to construct a sentence?`
`17:59.11`	`LostFrog`	`pingwin[work]: show application playback`
`17:59.42`	`pingwin[work]`	`rockin, thanks`
`17:59.59`	`VijayG`	`Hello <LostFrog>, i need to test it, i think there is something called as educational or free version for g729 as well`
`18:00.27`	`LostFrog`	`But that would be illegal in most countries, and I couldn't possibly help you.`
`18:00.34`	`LostFrog`	`You may want to google it. ;)`
`18:00.45`	`VijayG`	`in india its legal`
`18:00.48`	`VijayG`	`i am based in india`
`18:00.52`	`doolph`	`Sep 26 23:28:56 DEBUG[6183] db.c: Unable to find key '4755845097' in family`
`18:00.54`	`doolph`	`any idea?`
`18:00.55`	`VijayG`	`let me just try to find out`
`18:01.19`	`LostFrog`	`Well.. when I used it in my India office, I figured it out by googling.`
`18:01.28`	`LostFrog`	`But here in the US it is illegal.`
`18:01.28`	`LostFrog`	`so..`
`18:02.00`	`*** join/#asterisk hilacha (n=joel@200-171-99-250.dsl.telesp.net.br)`
`18:02.25`	`VijayG`	`let me try...`
`18:03.05`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`18:03.28`	`*** join/#asterisk _deg_ (n=deg@200.163.193.247)`
`18:04.01`	`VijayG`	`Can you give me some hints for the same please`
`18:04.10`	`VijayG`	`i tried looking for the same on google`
`18:04.17`	`hmmhesays`	`heh`
`18:04.24`	`VijayG`	`but could not find much info about the educational version`
`18:04.55`	`LostFrog`	`Some people are just dense.`
`18:05.08`	`*** join/#asterisk Ebola (n=Ebola@host81-132-187-57.range81-132.btcentralplus.com)`
`18:05.16`	`hilacha`	`hi all. I have a asterisk server running fine to my voip provider but i have noise in the voice stream in the dialer voip audio only. The noise appears like a AM radio trying to tune. Can someone, plz, give me a way to find the root of my problem?`
`18:05.30`	`*** join/#asterisk blebleble (i=godie@caesar.godie.net)`
`18:05.52`	`blebleble`	`getting a '> dialparties.agi: extstate: 1' when dialing an extension, it goes straight to voicemail, yet all the settings are correct anyone have any ideas on how to troubleshoot this`
`18:06.46`	`*** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)`
`18:07.12`	`doolph`	`how can I debug sip registry command??`
`18:07.25`	`LostFrog`	`sip debug?`
`18:07.49`	`doolph`	`it doesnt want to get registered with the isp`
`18:09.08`	`doolph`	`I have register => 1299:xxxx@201.227.202.53:5060/1299 in sip.conf`
`18:09.17`	`doolph`	`but sip show registry doesnt show nothing`
`18:09.37`	`LostFrog`	`It's really hard to trace rtp on a running system with multiple channels going.. :(`
`18:11.00`	`*** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.109.193)`
`18:11.53`	`doolph`	`this is getting me sick with this server`
`18:14.47`	`a1fa`	`doolph : do sip show peer`
`18:18.12`	`*** join/#asterisk cian (n=cian@cian.ws)`
`18:18.23`	`LostFrog`	`Hmm.. bind transfers don't work with native transfer.`
`18:18.52`	`LostFrog`	`+l`
`18:19.19`	`*** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net)`
`18:19.34`	`*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`18:19.42`	`LostFrog`	`Actually.. vice versa.`
`18:20.32`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`18:20.56`	`*** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`18:21.20`	`brad_mssw`	`VijayG: just look here : http://www.voip-info.org/wiki-Asterisk+G.729+Licensing .. if you're still interested in the 'free' version, scan down the page to the 'opensource implementation', there's a link to the for educational use one`
`18:21.34`	`tamp4x`	`anyone here dso a load balanced solution with asterisk (dynamically regestering peers) ?`
`18:22.10`	`docelmo`	`ya me`
`18:22.11`	`docelmo`	`:)`
`18:22.22`	`docelmo`	`Im speaking at astricon on the subject`
`18:22.36`	`backblue`	`docelmo: soo, do a pdf so we can read`
`18:22.37`	`backblue`	`:)`
`18:22.47`	`docelmo`	`you will get one from the show..`
`18:22.56`	`backblue`	`i'm very interested too`
`18:23.00`	`docelmo`	`or better yet.. come listen, learn, ask questions.. :)`
`18:23.03`	`LostFrog`	`IDamn, I wish I had a training budet.`
`18:23.14`	`backblue`	`docelmo: no money. no time.`
`18:23.17`	`LostFrog`	`well.. I do.. it just happens to be $0.`
`18:23.25`	`tamp4x`	`same reason as backblue`
`18:23.56`	`tamp4x`	`i was thinking of hacking the chan_sip to do it, with what ever stupid reply the phone gives back grom the server that is communicating with it`
`18:24.05`	`tamp4x`	`grom=from`
`18:24.27`	`docelmo`	`there is a easier way.. SER -> Asterisk`
`18:24.43`	`tamp4x`	`yes`
`18:24.44`	`tamp4x`	`but`
`18:24.49`	`doolph`	`anyone had before problems with register command in sip.conf?`
`18:24.52`	`tamp4x`	`when an invite comes from a carrier`
`18:24.59`	`tamp4x`	`to asterisk`
`18:25.03`	`backblue`	`do you want to suport multiple registrations on asterisk? only that?`
`18:25.33`	`LostFrog`	`Is ther a good document out there on redundancy with asterisk and sip?`
`18:25.38`	`tamp4x`	`phone is registered to server A (via ser 300 redirect), call comes in to server B, B sends to phone, phone will reject invite`
`18:25.38`	`docelmo`	`I do IP and digest auth in SER then tell asterisk what to do..`
`18:26.04`	`docelmo`	`Phone is registered to SER not asterisk`
`18:26.16`	`docelmo`	`redirects in asterisk suck`
`18:26.30`	`backblue`	`sip + asterisk sucks`
`18:26.31`	`backblue`	`:)`
`18:26.42`	`docelmo`	`no that would be h323`
`18:26.43`	`tamp4x`	`how does asterisk send the call to the phone`
`18:26.53`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`18:26.53`	`docelmo`	`asterisk doesnt.. SER does`
`18:26.54`	`docelmo`	`:)`
`18:26.55`	`Pj_`	`with a club`
`18:27.01`	`backblue`	`well, i dont like chan_sip`
`18:27.09`	`docelmo`	`I dont think many do..`
`18:27.17`	`tamp4x`	`what if the phone is busy`
`18:27.20`	`docelmo`	`chan_sip was bastardized..`
`18:27.27`	`tamp4x`	`if you have your carrier send calls to ser`
`18:27.30`	`docelmo`	`Then SER sends it to one of the asterisk boxes for voicemail`
`18:27.48`	`tamp4x`	`o you have some section for 486`
`18:27.50`	`*** join/#asterisk freebsd_fan (n=unsure@hdkbib3.hdk.gu.se)`
`18:27.54`	`cekc`	`is there a way to debug what zaptel hardware is currently up to?`
`18:27.56`	`tamp4x`	`didnt think of that`
`18:28.16`	`docelmo`	`Your not the first ITSP I have built clustered.. :)`
`18:29.00`	`*** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`18:29.45`	`backblue`	`docelmo: so you should be able to answer my question, with realtime+cache in asterisk, when i do reload, to re-read some information that it's allready cached, i loose my nat keep alives, anyway to round that?`
`18:30.47`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:31.47`	`*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)`
`18:32.55`	`docelmo`	`nope.. Welcome to the limitations of Realtime`
`18:33.09`	`*** part/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)`
`18:33.37`	`docelmo`	`You have to wait for the phone to re-register.. I set my registration timeouts for 1 minute so they are kept as upto date as possible`
`18:33.38`	`*** join/#asterisk riksta (n=rick@89.241.86.77)`
`18:33.45`	`LostFrog`	`That's a lot of traffic.`
`18:34.45`	`backblue`	`docelmo: you do that on the phone, right?`
`18:34.50`	`docelmo`	`registrations? well yes and no.. only a few bytes.. however SER does this w/o a problem.. asterisk may bitch tho`
`18:34.53`	`docelmo`	`yep`
`18:35.01`	`docelmo`	`instead of 3600 set it to 60`
`18:35.28`	`*** join/#asterisk Ebola (n=Ebola@host81-132-187-57.range81-132.btcentralplus.com)`
`18:35.37`	`backblue`	`yes, that's what i do too`
`18:35.46`	`backblue`	`but can be bad`
`18:35.51`	`*** join/#asterisk oej (n=oej@62.92.148.159)`
`18:35.54`	`backblue`	`chansip it's bad`
`18:36.14`	`backblue`	`makes no senso to me use ser+asterisk`
`18:36.26`	`backblue`	`asterisk should suport sip well`
`18:36.48`	`*** join/#asterisk afrosheen (n=cj@txprotoa2.august.net)`
`18:36.52`	`docelmo`	`asterisk yes if one asterisk box.. if your clustering asterisk then SER is your best friend`
`18:37.12`	`docelmo`	`OLLE!`
`18:37.20`	`afrosheen`	`is anyone here familiar with g729 licensing?`
`18:37.30`	`docelmo`	`what do you wanna know?`
`18:37.31`	`oej`	`YES!`
`18:37.50`	`afrosheen`	`I'm trying to estimate how many licenses we'll need, total`
`18:37.54`	`backblue`	`docelmo: well why not use 1 asterisk box, to do the ser job? and many asterisk another boxes?`
`18:38.37`	`docelmo`	`hehe.. Asterisk can only handle 300 calls MAX if you're lucky.. SER can handle 10,000+`
`18:38.42`	`docelmo`	`you do the math`
`18:38.44`	`afrosheen`	`I'm not sure if g729 has to be licensed per device, per channel or what`
`18:38.49`	`docelmo`	`per channel`
`18:39.05`	`backblue`	`docelmo: well, that its one more problem with asterisk.`
`18:39.06`	`docelmo`	`if you plan to do A-LEG and B-LEG g729 then you dont need it.. Just use pass thru`
`18:39.06`	`backblue`	`:)`
`18:39.09`	`afrosheen`	`so if my polycoms will do g729 to the server, is that a channel or is a channel when they dial outgoing`
`18:39.10`	`denon`	`per concurrent channel`
`18:39.21`	`docelmo`	`else you will need g729 in the middle for transcoding`
`18:39.30`	`afrosheen`	`yeah I do NOT want the server transcoding`
`18:39.45`	`afrosheen`	`since the phones can do it, might as well pass the stream to the endpoint (sip trunk)`
`18:39.52`	`backblue`	`docelmo: but i dont mind to use ser in a clustered env`
`18:40.08`	`backblue`	`docelmo: but i mind asterisk does not suport well sip as ser do`
`18:40.13`	`backblue`	`even if i only use one machine.`
`18:40.35`	`docelmo`	`Asterisk is SER on steroids.. SER is a proxy nothing more.. its light weight..`
`18:40.54`	`afrosheen`	`commpartners supports ulaw and g729, so I will just need licenses for g729 per channel going to commpartners, correct?`
`18:40.55`	`backblue`	`cluster it's a very specific implementation normally, so costumization it's normal, so as using ser.`
`18:41.34`	`backblue`	`docelmo: makes no sense to me, if i want to install a pbx, i have to install ser in the same machine of asterisk, to have efficient suport for sip.`
`18:41.54`	`docelmo`	`no not really although you can`
`18:42.18`	`docelmo`	`if just a small pbx then all you need is asterisk if > 300 Calls total then you cluster`
`18:42.23`	`backblue`	`well, no realtime`
`18:42.29`	`backblue`	`no multiple registrations`
`18:42.40`	`backblue`	`no alphanumeric sip users`
`18:42.59`	`backblue`	`no , no , no , no , ...`
`18:43.05`	`backblue`	`alots of no, in the sip protocol`
`18:43.05`	`docelmo`	`No need, yes you can and yes you can`
`18:43.14`	`docelmo`	`You're just new..`
`18:43.26`	`docelmo`	`Get some hard core experience behind you then you will see`
`18:43.41`	`backblue`	`i have experience`
`18:43.45`	`backblue`	`why do you say that?`
`18:43.47`	`backblue`	`explain please.`
`18:43.49`	`docelmo`	`SIP is 100000000 times better than H323 and 1000000 times better than IAX`
`18:43.57`	`docelmo`	`Cause your acting like a newn`
`18:43.59`	`docelmo`	`newb`
`18:44.37`	`justinu\|laptop`	`in this channel, iax p0wns`
`18:44.42`	`andrebarbosa`	`well i deploy a voip solution on my company based on IAX extensions.. and was the worst thing i could do..`
`18:44.51`	`andrebarbosa`	`lots of delay on the calls`
`18:45.03`	`andrebarbosa`	`tried everything.. then i fallbak to SIP and everything works fine`
`18:45.03`	`andrebarbosa`	`:s`
`18:45.27`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`18:45.27`	`afrosheen`	`iax > sip`
`18:45.53`	`andrebarbosa`	`i could not explain the delay on the calls`
`18:46.04`	`andrebarbosa`	`> 1000ms sometimes`
`18:46.06`	`andrebarbosa`	`on LAN calls`
`18:46.21`	`afrosheen`	`wow`
`18:46.26`	`andrebarbosa`	`i got less delay on intercontinental calls`
`18:46.26`	`afrosheen`	`I have no idea how you could do that`
`18:46.27`	`andrebarbosa`	`:s`
`18:46.34`	`andrebarbosa`	`neither I`
`18:46.35`	`andrebarbosa`	`!`
`18:46.52`	`rbd`	`does extconfig.conf use res_odbc.conf for resolving the database name? (i.e. if I enter "extensions => mysql,asterisk,extensions_table", does it look for an entry called 'asterisk' in res_odbc.conf (if not, where does it find the mysql connection info?)`
`18:46.54`	`andrebarbosa`	`try turning off jitter, codecs, ..`
`18:47.01`	`backblue`	`we have a lots of iax implementations without problems`
`18:47.05`	`backblue`	`and sip too`
`18:47.15`	`backblue`	`but we are allways limited to what chan_sip can do`
`18:47.20`	`andrebarbosa`	`ya`
`18:47.27`	`docelmo`	`Put it this way you will never see me openly say use IAX it rules..`
`18:47.29`	`*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)`
`18:47.30`	`afrosheen`	`I just like iax better because of how it magically defeats firewalls`
`18:47.33`	`drcode`	`hi all`
`18:47.37`	`docelmo`	`IAX has its own place in asterisk..`
`18:47.40`	`docelmo`	`disabled..`
`18:47.44`	`Corydon-w`	`rbd: nope, that's only for odbc connections`
`18:47.44`	`justinu\|laptop`	`lol`
`18:47.52`	`drcode`	`if I have nat, can I also use sip?`
`18:47.58`	`andrebarbosa`	`ya afrosheen`
`18:48.02`	`backblue`	`docelmo: i dont agree with you, iax it's very good for trunking at least.`
`18:48.03`	`andrebarbosa`	`my option was based on that too`
`18:48.10`	`*** join/#asterisk CuCullin (n=smwuser@38.251.246.11)`
`18:48.12`	`Greek-Boy`	`yeah IAX rules for trunking`
`18:48.17`	`Greek-Boy`	`SIP good for phones and clients`
`18:48.27`	`backblue`	`yeah, you dont have nat issues with that.`
`18:48.28`	`afrosheen`	`Greek-Boy, just summed it up nicely`
`18:48.32`	`Corydon-w`	`rbd: res_config_mysql looks in res_mysql.conf`
`18:48.39`	`drcode`	`sip can be used in nat?`
`18:48.42`	`CuCullin`	`hey all - I'm having a bit of trouble finding any info on using asterisk with Comcast's Digital Voice offering. Anyone know where I can find something on this?`
`18:48.48`	`afrosheen`	`drcode, single-sided nat, yes`
`18:49.03`	`afrosheen`	`CuCullin, I think they are 2 different worlds`
`18:49.04`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`18:49.05`	`rbd`	`Corydon-w: ahh okay... that file didn't seem to ship in my asterisk distro`
`18:49.07`	`_deg_`	`is this possible to have something like this on zapata.conf?`
`18:49.07`	`docelmo`	`backblue haha.. try IAX trunking at grater than 20 channels..`
`18:49.16`	`drcode`	`well , I have try to use x-lite`
`18:49.16`	`_deg_`	`channel => 1,3,5,8`
`18:49.20`	`rbd`	`I'll grab it from the default distro...this is asterisk 1.2.7 I believe`
`18:49.20`	`CuCullin`	`afrosheen: as in... ?`
`18:49.21`	`afrosheen`	`docelmo, is there some hard coded limitation?`
`18:49.25`	`Corydon-w`	`rbd: it's in the addons package`
`18:49.28`	`_deg_`	`insted of channel => 1-8`
`18:49.32`	`drcode`	`I am inside nat , and other not , it didn't secess`
`18:49.38`	`afrosheen`	`CuCullin, digital voice is comcast's internal digital phone service...asterisk is not`
`18:49.38`	`drcode`	`any one mybe use it?`
`18:49.38`	`docelmo`	`afrosheen no but IAX turns to shit > 20`
`18:49.47`	`afrosheen`	`docelmo, define 'turns to shit'`
`18:49.51`	`backblue`	`docelmo: probably, because ztdummy`
`18:49.54`	`docelmo`	`and your CPU level goes 100%`
`18:49.54`	`rbd`	`Corydon-w: okay, so realtime extensions will work out of the box, right? I just need to grab res-mysql.conf from addons and set it up.`
`18:50.03`	`docelmo`	`backblue I tried with actual hardware.. same`
`18:50.06`	`docelmo`	`its something in IAX`
`18:50.13`	`backblue`	`and with posix timers?`
`18:50.17`	`Corydon-w`	`rbd: or you could set up mysql in unixODBC`
`18:50.20`	`andrebarbosa`	`I have a problem on realtime regarding SIP`
`18:50.31`	`andrebarbosa`	`re-directs don't work`
`18:50.32`	`docelmo`	`andrebarbosa ??`
`18:50.32`	`andrebarbosa`	`:s`
`18:50.45`	`rbd`	`Corydon-w: okay, thanks much. I assume I could use something like sqlLite via unixODBC as well`
`18:50.45`	`_deg_`	`is ther someone to help me here with zapata.conf?`
`18:50.46`	`andrebarbosa`	`on IAX notransfer works fine`
`18:51.02`	`Corydon-w`	`rbd: perhaps`
`18:51.15`	`afrosheen`	`docelmo, you weren't doing any transcoding over iax were you?`
`18:51.22`	`backblue`	`docelmo: you must agree with me, sip implementation could be very much better.`
`18:51.30`	`CuCullin`	`afrosheen: as a PBX, I should be able to use Asterisk to manage multiple phones in my homes, assigning them after the call comes in, and possibly, if comcast supports it, use asterisk to handle incoming calls through Comcasts system without the phone line, as in logging in somehow. Thats what I'm wondering about.`
`18:51.41`	`*** join/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net)`
`18:51.49`	`drcode`	`is there doc that expline about sip and nat?`
`18:51.53`	`docelmo`	`backblue Mark Spencer and I spoke about it @ VON a couple weeks ago`
`18:51.58`	`docelmo`	`yes even he knows it.`
`18:52.01`	`*** join/#asterisk Woland (n=lucifer@fox.perm.ru)`
`18:52.11`	`docelmo`	`drcode yes the RFC`
`18:52.14`	`backblue`	`i know they do, but they only want to sell cards.`
`18:52.15`	`Corydon-w`	`CuCullin: Comcast does not run their voip service over the same network as their broadband`
`18:52.16`	`Woland`	`�� ?`
`18:52.32`	`docelmo`	`Corydon76-home yes they do`
`18:52.36`	`CuCullin`	`yes they do`
`18:52.40`	`docelmo`	`it runs on the same frequency`
`18:52.41`	`Corydon-w`	`docelmo: no, they don't`
`18:52.54`	`Corydon-w`	`They run a completely separate network over the same lines`
`18:53.16`	`backblue`	`well, i have to go`
`18:53.20`	`backblue`	`cya`
`18:53.22`	`CuCullin`	`Corydon, how so? I have a single cable modem which also gives me my analog line.`
`18:53.26`	`docelmo`	`unless they are doing something I dont know of.. Cause when I was doing some diagnostics they same freq was used for the modem and the ATA portion`
`18:53.30`	`Woland`	`heil`
`18:53.31`	`Corydon-w`	`CuCullin: correct`
`18:53.46`	`Corydon-w`	`CuCullin: the cable modem is terminating two distinct networks`
`18:53.51`	`CuCullin`	`ah.`
`18:54.00`	`afrosheen`	`corydon is right`
`18:54.07`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`18:54.16`	`Woland`	`can anybody tell me how to write extension for calling to addresses like user@domain ?`
`18:54.27`	`CuCullin`	`ok, so effectively the only way for me to set up a PBX would be making use of the analog line, and not going over my network`
`18:54.36`	`Corydon-w`	`CuCullin: correct`
`18:54.39`	`afrosheen`	`CuCullin, yep`
`18:54.41`	`CuCullin`	`well thats a shame.`
`18:54.47`	`docelmo`	`Woland exten => user, 1, Dial(SIP/user@domain)`
`18:54.51`	`afrosheen`	`you're not going to bridge their system to asterisk just because they're using voip`
`18:54.53`	`CuCullin`	`no soft-phone capabilities with them either I suppose?`
`18:55.04`	`luke-jr_work`	`Corydon-w, maybe he can get access to the phone network?`
`18:55.06`	`Corydon-w`	`Also correct`
`18:55.11`	`docelmo`	`who comcast? no`
`18:55.16`	`CuCullin`	`docelmo: yes`
`18:55.21`	`Corydon-w`	`luke-jr_work: not without violating the TOS`
`18:55.36`	`luke-jr_work`	`Corydon-w, well, that applies to Vonage as well =p`
`18:55.40`	`CuCullin`	`Well thats a lie right there. I'll have to try and get some credit for that load of bs.`
`18:55.52`	`docelmo`	`Their network is almost the same as Brighthouse in flordia which I used to work for..`
`18:56.12`	`CuCullin`	`I wish I could say I'd drop comcast, but I have no cable provider alternatives, and cant beat the price...`
`18:56.14`	`docelmo`	`its 100% enclosed on the coax/fiber.. You need an ATA to get to it`
`18:56.23`	`*** join/#asterisk lero (n=rootz@200.192.160.100)`
`18:56.26`	`lero`	`hi`
`18:56.31`	`lero`	`anyone use munin with asterisk plugins?`
`18:56.57`	`toddf`	`does anybody use the transfer capabilities of asterisk? How do you dial out with the ability to transfer the call to another phone, yet send '#' to remote pbx systems?`
`18:57.02`	`Woland`	`docelmo: in this case asterisk understands extension as 'call to user in context domain', isn't it?`
`18:57.07`	`luke-jr_work`	`Can I setup a LAN using the cable provider's lines independent of their ISP network? ;p`
`18:57.22`	`luke-jr_work`	`toddf, I just don't send #`
`18:57.23`	`docelmo`	`yes.. on the 2nd user use ${EXTEN} forgot that part..`
`18:57.35`	`Woland`	`docelmo: is there a way to write "exten`
`18:57.42`	`Woland`	`docelmo: is there a way to write "exten => user@domain ..."`
`18:57.49`	`toddf`	`luke: but, some systems require it; when I call my isp to talk to anybody I must first dial 0#`
`18:57.50`	`CuCullin`	`well screw me. Thanks guys.`
`18:58.01`	`Woland`	`docelmo: or wildcard ".@domain"`
`18:58.36`	`docelmo`	`CuCullin uhh no.. :)`
`18:58.42`	`Corydon-w`	`CuCullin: woohoo`
`18:58.51`	`CuCullin`	`rhetorical guys :-P`
`18:59.01`	`docelmo`	`Woland dude you are confusing the hell outa me`
`18:59.08`	`*** join/#asterisk HaMYaI (i=HaMYaI@125.25.133.235)`
`18:59.09`	`docelmo`	`Check out app_dial on the wiki`
`18:59.11`	`CuCullin`	`I doubt my ass is all that inviting... though Comcast would seem to disagree.`
`18:59.27`	`*** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz)`
`18:59.29`	`CuCullin`	`oh well... im off to play with asterisk in other ways`
`18:59.32`	`Corydon-w`	`What, is it hairy or something?`
`18:59.33`	`Woland`	`docelmo: ok $+)`
`19:00.02`	`CuCullin`	`Corydon: just an assumption really. I don't tend to rate my ass.`
`19:00.08`	`CuCullin`	`Must be the lack of myspace usage.`
`19:00.24`	`*** part/#asterisk Woland (n=lucifer@fox.perm.ru)`
`19:00.28`	`Corydon-w`	`Most people don't... they allow others to rate it...`
`19:00.36`	`*** part/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz)`
`19:01.31`	`CuCullin`	`shrug I'll leave that to the gf. Thanks for the help though.`
`19:02.01`	`stephane_`	`re`
`19:02.14`	`afrosheen`	`ismyasshotorisitnot.com`
`19:02.41`	`[TK]D-Fender`	`toddf : the idea is not to use *'s DTMF based features for anything you don't have to. Normal SIP phone has a dedicate transfer button which has nothing to do with DTMF therby leaving the "#" key alone for normal use at all times`
`19:05.48`	`doolph`	`db.c: Unable to find key 'SIP/2029950_in' in family 'cfb'`
`19:08.08`	`*** part/#asterisk jmls (n=asterisk@62.49.235.130)`
`19:11.15`	`*** join/#asterisk blaylock (n=seth@snap.helixsystems.com)`
`19:13.35`	`*** join/#asterisk Alric (n=nbowyer@avantacom.com)`
`19:15.32`	`*** join/#asterisk funxion (n=nunya@63.214.236.169)`
`19:15.47`	`drcode`	`asteriks need mysql or other db?`
`19:15.53`	`luke-jr_work`	`no`
`19:15.56`	`funxion`	`doesnt need it`
`19:15.58`	`Alric`	`Anyone know if you can make a Sipura ATA auto answer?`
`19:16.08`	`drcode`	`k`
`19:16.09`	`drcode`	`thnx`
`19:16.30`	`funxion`	`does anyone know the asterisk variable for the destination channel of a call ?`
`19:16.48`	`funxion`	`provided the variable is called after the call ha been routed`
`19:16.51`	`luke-jr_work`	`Alric, how can you make an analog phone autoanswer??`
`19:17.02`	`funxion`	`good point luke-jr_work`
`19:17.11`	`Alric`	`It doesn't have to be an analog phone on the end of an ATA...`
`19:17.34`	`funxion`	`it is an alanlog telephony adapter`
`19:17.42`	`funxion`	`so it does have to be analog`
`19:17.50`	`a1fa`	`Alric : get a phone-pickup-monkey`
`19:18.07`	`a1fa`	`everytime your ata rings, the monkey picks it up`
`19:18.15`	`a1fa`	`bcos he is attached to the phone`
`19:18.18`	`Alric`	`Yes, its analog. No, its not a phone.`
`19:18.21`	`a1fa`	`and he gets shocked`
`19:19.28`	`funxion`	`a1fa do you know what the asterisk variable for the destination channel is?`
`19:19.56`	`Alric`	`Not sure how else to get this Viking thing to pick up. It doesn't seem to want to on its own.`
`19:20.10`	`funxion`	`does ${channel} return the inbound or outbound side?`
`19:20.57`	`funxion`	`Alric I dont thin you can force it through the ata`
`19:21.41`	`funxion`	`the analog end needs to have its own autoanswer function`
`19:23.24`	`Alric`	`That does not make me happy :)`
`19:24.00`	`[TK]D-Fender`	`Alric : its up to the Viking to pick up. Go read the manual to make sure you bought the right kind of unit`
`19:25.20`	`*** join/#asterisk ramtha (n=tk@p5088BA90.dip0.t-ipconnect.de)`
`19:25.22`	`ramtha`	`hi`
`19:25.43`	`ramtha`	`how can i set a call limt to a iax2 connection?`
`19:28.17`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`19:28.19`	`afrosheen`	`the viking needs the following code : $explore = [foreign lands] ; if (encounter_natives) goto {plunder}`
`19:28.29`	`Alric`	`Well, looks like what I have is just an amp, and expects an open channel with... audio.`
`19:28.55`	`[TK]D-Fender`	`Alric : then you got the wrong one...`
`19:31.00`	`*** part/#asterisk lero (n=rootz@200.192.160.100)`
`19:33.48`	`*** join/#asterisk cian (n=cian@cian.ws)`
`19:34.27`	`*** join/#asterisk toerkeium (i=oo@201.216.206.221)`
`19:39.08`	`*** join/#asterisk mv00 (i=HiTMaN@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk)`
`19:43.10`	`*** join/#asterisk VijayG (i=vijay@202.131.145.234)`
`19:43.12`	`VijayG`	`hello`
`19:43.32`	`VijayG`	`i am not able to make calls from sip trunks`
`19:43.40`	`VijayG`	`i am able to do so from iax trunks`
`19:43.56`	`VijayG`	`whereas the same sip.conf is working fine at other server of mine`
`19:44.07`	`VijayG`	`any idea what could be the issue with this specific server`
`19:45.07`	`*** part/#asterisk CuCullin (n=smwuser@38.251.246.11)`
`19:46.00`	`[TK]D-Fender`	`VijayG : You have not showed us your configs, so NO. pastebin.ca is your friend.`
`19:46.25`	`VijayG`	`the problem does not looks from configuration of sip.conf`
`19:46.29`	`FuriousGeorge`	`so, notably missing from the comprehensive IM support in 1.4 seems to be simple`
`19:46.33`	`VijayG`	`because the same configuration exactly same`
`19:46.34`	`FuriousGeorge`	`as in SIMPLE`
`19:46.44`	`FuriousGeorge`	`am i missing something?`
`19:46.47`	`VijayG`	`is running in other server and is just fine`
`19:47.03`	`mog`	`simple is lame FuriousGeorge`
`19:47.04`	`mog`	`^_^`
`19:47.09`	`doolph`	`where's asterisk database stored?`
`19:47.16`	`VijayG`	`in case i need to reinstall asterisk`
`19:47.19`	`mog`	`but you could use the jabber support in asterisk to gateway with simple`
`19:47.22`	`VijayG`	`do i need to first stop it`
`19:47.28`	`VijayG`	`and then install it again`
`19:47.32`	`FuriousGeorge`	`mog: just the man i wanted to ask about that`
`19:47.34`	`FuriousGeorge`	`:)`
`19:47.46`	`VijayG`	`or just install it`
`19:48.13`	`FuriousGeorge`	`mog: how could one gateway with simple though? are there some canned scripts that will switch between simple and XMPP`
`19:48.21`	`mog`	`yeah`
`19:48.22`	`mog`	`for ser`
`19:48.30`	`mog`	`ser talks to jabber jabber talks to asterisk`
`19:49.08`	`FuriousGeorge`	`mog: yeah i heard that jabber talks to asterisk now but i havent been able to find documentation. maybe you could talk to the developer for me :)`
`19:49.21`	`mog`	`there is in the docs`
`19:49.55`	`FuriousGeorge`	`mog: i made progdocs, which one is it in?`
`19:50.08`	`mog`	`its in the docs folder`
`19:51.19`	`toerkeium`	`guys, withing AMI I can know when I extension is (idle, ringing, unavailable, ringing, not found). Is there any way to know, or check when an extension is answered by an answer machine?`
`19:51.32`	`*** join/#asterisk blebleble (i=godie@caesar.godie.net)`
`19:52.18`	`blebleble`	`ok hopefully an easy / dumb question, i have a digium tdm400p, if i have 4 analog lines comming into the card how many fxs modules do i need? 1 or 4?`
`19:52.36`	`a1fa`	`4`
`19:52.37`	`[TK]D-Fender`	`blebleble : NONE`
`19:52.38`	`FuriousGeorge`	`mog: im looking in there now. you dont mean jabber and jingle.txt do you?`
`19:52.41`	`a1fa`	`jeje`
`19:52.48`	`mog`	`sure`
`19:52.58`	`[TK]D-Fender`	`blebleble : FXS modules are for PHONES, not LINE.`
`19:53.15`	`a1fa`	`he ment to say FXO`
`19:53.46`	`FuriousGeorge`	`mog: i was wondering more about (for instance) sample configurations ive seen that use dialplan app jabbersend, which i havent found documentation for`
`19:53.48`	`blebleble`	`TK: err sorry yah FXO`
`19:53.48`	`[TK]D-Fender`	`a1fa : Sorry, I never load chan_psychic.so its too unreliable.`
`19:54.00`	`*** join/#asterisk DaneM (n=DaneM@70.135.56.182)`
`19:54.01`	`blebleble`	`a1fa: thanks`
`19:54.01`	`mog`	`show application jabbersend`
`19:54.02`	`[TK]D-Fender`	`blebleble : You want 4 ports, you need 4 modules.`
`19:54.06`	`FuriousGeorge`	`mog: and there is no show application jabbersend`
`19:54.08`	`blebleble`	`thanks guys`
`19:54.12`	`DaneM`	`Hello`
`19:54.16`	`[TK]D-Fender`	`blebleble ; np`
`19:54.23`	`mog`	`yes there is`
`19:54.32`	`mog`	`you probably didnt build res_jabber`
`19:54.34`	`toerkeium`	`what about my question? is it too stupid or too complex? :P)`
`19:54.38`	`mog`	`or chan_gtalk`
`19:54.43`	`DaneM`	`I have a question about using a 4-wire rj11 cable to connect 2 lines to a X100P card. Anybody know how to do that?`
`19:54.44`	`mog`	`do a make menuselect`
`19:54.48`	`mog`	`you will see it didnt compile`
`19:55.02`	`Qwell[]`	`DaneM: the x100p is 1 fxo`
`19:55.05`	`FuriousGeorge`	`mog: thats odd cuz i selected it, lemme check`
`19:55.08`	`cekc`	`asterisk can link to jabber?`
`19:55.08`	`Qwell[]`	`so...no, you can't`
`19:55.15`	`mog`	`cekc, yes`
`19:55.16`	`[TK]D-Fender`	`DaneM : You can't.`
`19:55.17`	`mog`	`in 1.4`
`19:55.17`	`DaneM`	`I have 2 cards, but I'm having trouble getting the second card to see line2`
`19:55.19`	`cekc`	`sweet`
`19:55.32`	`DaneM`	`(no dial tone, etc.)`
`19:55.51`	`cekc`	`what does it do, notify you of calls and voicemail?`
`19:56.09`	`DaneM`	`yes, line 1 works perfectly with all my extensions. Line 2 won't even pick up.`
`19:56.22`	`DaneM`	`(incoming calls)`
`19:57.15`	`FuriousGeorge`	`mog: i see res jabber option 10 checked, im making again maybe i forgot to make install or something`
`19:57.53`	`DaneM`	`Do I have to configure something in one of the files to get the X100P to see the second line (which I believe is the outer pair of wires)?`
`19:58.09`	`pingwin[work]`	`is there a method to require the call screening to work on every phone call? dials and queues?`
`19:58.20`	`FuriousGeorge`	`mog: still not there. is there something besides the option under resources res_jabber that needs to be checked off`
`19:58.35`	`mog`	`res_jabber is all you need for that , chan_gtalk is for calling`
`19:58.39`	`mog`	`do an ls in res folder`
`19:58.40`	`FuriousGeorge`	`do i need to run autoconf again after make menuselect?`
`19:58.42`	`mog`	`did it compile?`
`19:58.51`	`FuriousGeorge`	`mog: yeah`
`19:59.19`	`mog`	`then is it installed and loaded?`
`19:59.29`	`Flauto`	`mog, i have a problem with gtalk`
`20:00.17`	`Flauto`	`how can i check if gtalk is loaded`
`20:00.38`	`mog`	`show modules`
`20:00.53`	`pingwin[work]`	`or, is there a macro for recording a temporary message?`
`20:01.25`	`DaneM`	`brb`
`20:01.36`	`FuriousGeorge`	`mog: ok when i show modules i dont see res_jabber`
`20:01.41`	`FuriousGeorge`	`but it did compile`
`20:01.56`	`*** join/#asterisk jeremy_g (i=jeremy_g@c213-100-17-43.swipnet.se)`
`20:01.57`	`mog`	`see if its in /usr/lib/asterisk/modules`
`20:02.15`	`Flauto`	`show modules`
`20:02.15`	`Flauto`	`No such command 'show modules' (type 'help' for help)`
`20:02.30`	`FuriousGeorge`	`mog: no it isnt`
`20:02.38`	`FuriousGeorge`	`do i need to run autoconf again or something?`
`20:02.41`	`mog`	`then you didnt make install FuriousGeorge`
`20:02.54`	`mog`	`Flauto, show modules is a command its in the core`
`20:03.02`	`mog`	`its always there in 1.4`
`20:03.08`	`mog`	`in trunk its core show modules`
`20:03.51`	`Flauto`	`okay`
`20:04.09`	`Flauto`	`i do have gtalk at /usr/lib/asterisk/modules`
`20:04.14`	`*** join/#asterisk cian (n=cian@cian.ws)`
`20:05.51`	`Flauto`	`No such command 'core show modules' (type 'help' for help)`
`20:06.02`	`FuriousGeorge`	`mog: i started over and i can say with certainty that while make will build res_jabber make install will not install it. i could manually copy it to usr/lib/asterisk/modules, right?`
`20:06.04`	`*** join/#asterisk oej (n=oej@62.92.148.159)`
`20:06.18`	`mog`	`yes`
`20:08.16`	`Flauto`	`mog, in my case, i see chan_gtalk.so* chan_jingle.so* res_jabber.so* all in there at /var/lib/asterisk/modules`
`20:08.24`	`Flauto`	`but when i tried to call`
`20:08.34`	`*** join/#asterisk p1p (i=p1p@64.200.16.100)`
`20:08.35`	`Flauto`	`it tells me that gtalk channel is not registered`
`20:09.36`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`20:09.44`	`FuriousGeorge`	`mog: did i mention i made sure iksemel was installed? anyway, now i manually copied it into my modules dir but its not loading the module. should i put an entry in modules.conf or something?`
`20:10.10`	`mog`	`in modules.conf it should autoload`
`20:10.31`	`Flauto`	`okay`
`20:10.33`	`Flauto`	`let me see`
`20:10.59`	`Flauto`	`yes, it is autoload=yes`
`20:14.10`	`FuriousGeorge`	`mog: i made samples again just in case. i even looked in the asterisk source under configs. there is no mention of res_jabber in modules.conf or modules.conf.sampke`
`20:14.22`	`mog`	`exactly`
`20:14.29`	`FuriousGeorge`	`i tried manually putting and entry in there, and it still doesnt work`
`20:14.29`	`mog`	`it has autoload=yes`
`20:14.38`	`mog`	`youll notice most modules dont have one`
`20:14.40`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`20:14.40`	`FuriousGeorge`	`that it does`
`20:14.44`	`FuriousGeorge`	`i did notice`
`20:14.48`	`FuriousGeorge`	`i misunderstood you`
`20:14.51`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`20:14.53`	`Flauto`	`mog, i still don't get it when i tried show modules or core show modules`
`20:15.23`	`*** join/#asterisk bobloblian (n=bob@net-252-14.northwestel.net)`
`20:15.35`	`FuriousGeorge`	`Flauto: you have to use show modules from the asterisk cli`
`20:15.38`	`*** join/#asterisk WAudette (n=WAudette@71.237.146.239)`
`20:15.58`	`*** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`20:16.34`	`Flauto`	`yes, i tried`
`20:16.52`	`bobloblian`	`I have been trying to find if there is a way to playback a message to a sip softphone while it is on-hook`
`20:16.57`	`bobloblian`	`is this possible?`
`20:17.53`	`cekc`	`do I need libpri for anything if I don't have ISDN lines?`
`20:18.24`	`DaneM`	`Does anybody know how to handle 2 incoming analog lines using 2 X100P cards?`
`20:18.45`	`FuriousGeorge`	`so, mog: ive copied the res_jabber file into usr/lib/asterisk/modules manually and its still not loading the module. any idea why?`
`20:18.49`	`DaneM`	`The trick is that line 2 requires a 4-lead rj11`
`20:19.02`	`mog`	`you have a jabber.conf?`
`20:19.15`	`*** join/#asterisk Domingues (n=domingue@200-170-201-152.core01.spo.ifx.net.br)`
`20:19.15`	`FuriousGeorge`	`mog: yes that i do have`
`20:19.19`	`Corydon-w`	`Then it's an RJ12, not an RJ11`
`20:19.21`	`cekc`	`DaneM: make a custom phone cable`
`20:19.25`	`DaneM`	`aah..thanks`
`20:19.35`	`DaneM`	`ok. Do you just cut the inner pair?`
`20:20.03`	`DaneM`	`...or does the x100p require the inner pair and not use the outer pair?`
`20:20.06`	`Flauto`	`[Oct 9 15:19:44] WARNING[18215]: channel.c:2874 ast_request: No channel type registered for 'gtalk'`
`20:20.07`	`Flauto`	`[Oct 9 15:19:44] WARNING[18215]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'gtalk' (cause 66 - Channel not implemented)`
`20:20.09`	`Corydon-w`	`Do you have a pair of crimpers?`
`20:20.17`	`DaneM`	`yup`
`20:20.20`	`mog`	`then asterisk should load it or crash`
`20:20.38`	`Corydon-w`	`Then you can wire each x100p with a separate pair`
`20:20.45`	`FuriousGeorge`	`asterisk -cvvvvvvvvvv\|grep jabber shows no mention whatsoever and i cant reload res`
`20:20.49`	`FuriousGeorge`	`_jabber`
`20:21.11`	`Corydon-w`	`WhOrange, Blue, WhBlue, Orange`
`20:21.14`	`DaneM`	`Can the x100p make use of the outer pair at all?`
`20:21.19`	`Corydon-w`	`No`
`20:21.21`	`DaneM`	`k`
`20:21.46`	`DaneM`	`...so just to summarize, I need to make the outer pair into the inner pair when connecting to the card?`
`20:21.49`	`mog`	`do a show version Corydon-w`
`20:21.52`	`mog`	`er FuriousGeorge`
`20:21.55`	`mog`	`my bad Corydon-w`
`20:21.55`	`Domingues`	`Hello all, I am using a script in perl using AGI to do automatic re-route of calls, after I start to use the script I realized that DTMF stoped to work, the script is blocking the DTMF, does anybody know how to pass throught`
`20:21.58`	`Corydon-w`	`DaneM: correct`
`20:22.04`	`DaneM`	`sweet! Thanks!`
`20:22.33`	`Corydon-w`	`DaneM: just make sure you reverse the order when swapping the outer pair to the inner pair`
`20:22.54`	`Corydon-w`	`i.e. Orange/WhOrange`
`20:22.54`	`FuriousGeorge`	`mog: Asterisk 1.4.0-beta2`
`20:23.02`	`DaneM`	`ok thanks`
`20:23.09`	`mog`	`i dont see how that is possible FuriousGeorge`
`20:23.27`	`Domingues`	`Hello all, I am using a script in perl using AGI to do automatic re-route of calls, after I start to use the script I realized that DTMF stoped to work, when i call some phone if URA the DTMF is blocked, does anybody know how to pass throught the DTMF`
`20:23.43`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`20:24.16`	`FuriousGeorge`	`mog: i kid you not`
`20:24.56`	`FuriousGeorge`	`mog: i sent you a /msg`
`20:25.10`	`mog`	`didnt recieve it`
`20:25.47`	`*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`20:26.25`	`Flauto`	`mog, i tried to use show modules and core show modules, both did not work`
`20:26.25`	`Domingues`	`does any body already have problem in DTMF using AGI?`
`20:26.43`	`FuriousGeorge`	`mog: still nothing?`
`20:27.19`	`mog`	`yes`
`20:28.11`	`FuriousGeorge`	`mog: yes as in no message received :) i basically offered to allow you to ssh in if you didnt believe me, but i obviously cant send my credentials via the room`
`20:29.41`	`toerkeium`	`guys, anyone know why if a extensions is not bein used, AMI reports 1 (in use) when I check the extension? what could be wrong ?`
`20:32.20`	`*** join/#asterisk blebleble (i=godie@caesar.godie.net)`
`20:32.46`	`blebleble`	`i just had an asterisk box use 70gb of debug logs to /var/log/asterisk/full , how can i turn that debugging off, i did sip no debug but still going`
`20:33.29`	`blebleble`	`like tons of 'DEBUG[6020] channel.c: Scheduling timer at 0 sample intervals'`
`20:34.00`	`jmls`	`blebleble: logger rotate`
`20:34.22`	`blebleble`	`jmls: it is rotating them but obviously i dont want 70gb of logs and all the debug output anyways`
`20:34.31`	`jmls`	`alos check your logger.conf settings and remove the debug options`
`20:34.55`	`jmls`	`(those two statements should be the other way round)`
`20:35.22`	`blebleble`	`great thanks, thats what i was looking for`
`20:35.23`	`bobloblian`	`can I get a message to play to a siphone as soon as it registers?`
`20:35.53`	`bobloblian`	`the playback description says not all channels support playing messages while on hook, but I can't find any more info about that`
`20:37.10`	`*** join/#asterisk backblue (n=moo@87.196.0.45)`
`20:38.11`	`stubert`	`hey, anyone know what the relaxdtmf setting does in sip.conf???`
`20:38.54`	`*** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net)`
`20:39.42`	`Domingues`	`hello all, does anybody already have problems with DTMF using AGI script`
`20:40.20`	`pingwin[work]`	`inside of a macro, how can I remove the extensions from the previous context?`
`20:41.45`	`*** join/#asterisk mv00 (n=HiTMaN@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk)`
`20:43.03`	`toerkeium`	`dont know why.. once extension 5001 calls extension 5006, when the call ends, "sip show channels" shows still the call`
`20:43.08`	`toerkeium`	`anyone any idea why?`
`20:43.51`	`pingwin[work]`	`if you dial it twice or more, will it recycle the channel or continue to the next?`
`20:44.37`	`*** join/#asterisk KranZ (n=user@sme.bestline.net)`
`20:44.38`	`toerkeium`	`if I try to dial to 5006 extension, it will give me busy`
`20:44.50`	`pingwin[work]`	`that is odd`
`20:45.54`	`toerkeium`	`do you know what should I check? I am pretty lost`
`20:46.43`	`pingwin[work]`	`i'm new to this... so I'm not too sure`
`20:46.55`	`pingwin[work]`	`is it over a sip/iax/zap channel?`
`20:47.48`	`bobloblian`	`toerkeium: I am no expert either, but I would suggest asterisk -vvvvvr to enable the cli, then sip debug so you can watch what happens`
`20:47.51`	`bobloblian`	`might get a clue that way`
`20:48.11`	`pingwin[work]`	`anyone have a tip for my macro problem?`
`20:51.24`	`aptura`	`do you have it saves on pastebin`
`20:51.27`	`syzygyBSD`	`Afternoon everyone`
`20:51.38`	`*** part/#asterisk Domingues (n=domingue@200-170-201-152.core01.spo.ifx.net.br)`
`20:51.54`	`jmls`	`pingwin[work]: if you are in a macro, it has it's own "extensions". I don't understand the question`
`20:52.03`	`[TK]D-Fender`	`pingwin[work] : What extensions from the previous macro interfere with anything? Pastbin it for us`
`20:53.37`	`pingwin[work]`	`well i'm calling a macro, and in it are part of instructions, ie dial 1 or 2 for different things. but instead of going to the extension within the macro, it goes to the extension from the context I called the macro from`
`20:53.53`	`pingwin[work]`	`if that doesn't clarify, I'll pastebin`
`20:55.35`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`20:55.56`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.231)`
`20:59.14`	`*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`20:59.42`	`Mercestes`	`Hey. What would cause a sip phone to immediately answer a call (before ring back) and provide no voicepath? The phone is not forwarded.`
`21:00.08`	`[TK]D-Fender`	`pingwin[work] : Yes, please pastebin.`
`21:01.54`	`toddf`	D-Fender: I don't have a normal sip phone, I have a sipura ata and a cisco ata186 with standard pots phones ;-( or I also want to tell my pbx to transfer the call when I'm taking a call on my mobile, thus '#' must go to the pbx too; I was hoping there was some sortof `escape' for # in that case
`21:03.17`	`[TK]D-Fender`	`toddf : You should not use "#" transfers with Sipura ATA's. They support signalling things like that using hook-flash. Read the manual.`
`21:03.33`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`21:04.04`	`toddf`	`D-Fender: aaah, I see.. (beginning to see the light) .. do not let asterisk interpret # as special, let a separate transfer signaling thing happen...`
`21:04.32`	`[TK]D-Fender`	`toddf : Exactly.`
`21:05.36`	`toddf`	`know if there's any hope for the cisco ata186?? (but I will read the manual on that one also)`
`21:05.39`	`[TK]D-Fender`	`Sipura/Linksys ATA's are pretty smart and cover virtually every kind of feature you can imagine possible through a POTS interface in a non-diruptive manner. As SIP does not have a "flash" button, that is the gateway to all these extra features.`
`21:05.53`	`[TK]D-Fender`	`toddf : No idea on the 186's inner workings, sorry`
`21:06.18`	`toddf`	`ic`
`21:06.42`	`DaneM`	`OK...I've made a custom cable to hook line 2 up to the second X100P. I have reversed the two outer wires and put them on the inside. Unfortunately, line 2 isn't picking up reliably. Any ideas?`
`21:06.51`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.154)`
`21:07.04`	`toddf`	`my wife doesn't like the # thingey so she can't check cc balances and such, so I hope I can fix the ata186 to do this too .. the sipura is so much easier to carry with me than the ata186 ;-(`
`21:10.35`	`DaneM`	`Do you have to set up anything special in the configuration files to get line 2 working (on X100P card 2)?`
`21:11.19`	`*** join/#asterisk sloth (n=sloth@cpe-69-203-212-182.nyc.res.rr.com)`
`21:14.00`	`[TK]D-Fender`	`DaneM : There is no "line 2" on an X100P. Get it?`
`21:23.00`	`*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net)`
`21:23.30`	`*** part/#asterisk toddf (i=rhsld1z6@net-66-210-111-62.theshop.net)`
`21:23.47`	`DaneM`	`I understand that they can only use one line per card. Earlier somebody told me that I could access line 2 incoming calls by creating a custom wire and plugging it into a second card`
`21:24.33`	`DaneM`	`do I need to make any changes to configuration files in order to get this to answer?`
`21:26.13`	`KranZ`	`DaneM: for one, you need 2 cards and 2 pairs of wires for 2 lines`
`21:26.24`	`DaneM`	`got that`
`21:26.33`	`KranZ`	`each card gets 1 pair (2 wires)`
`21:26.39`	`*** join/#asterisk beyond (n=beyond@200.192.160.100)`
`21:26.43`	`DaneM`	`ok. Got that.`
`21:26.47`	`KranZ`	`you'll also need to subscribe to a 2nd line from your phone company`
`21:26.52`	`DaneM`	`got that`
`21:27.19`	`KranZ`	`you can also buy a splitter which plugs into your wall outlet and has two ports (1 for each line)`
`21:27.25`	`DaneM`	`got that :-)`
`21:27.32`	`KranZ`	`then use 2 cords (1 for each card)`
`21:27.46`	`DaneM`	`ok. I have one plugged into each card.`
`21:28.03`	`KranZ`	`have you plugged a phone in each cord to verify that it's working?`
`21:28.20`	`DaneM`	`the wire going into card 2 has the outer pair and the inner pair switched, and the (former) outer pair crossed.`
`21:28.31`	`KranZ`	`to test each line (before trying to plug them into the cards)`
`21:28.38`	`DaneM`	`hmmm...no, I haven't. I'll go do that.`
`21:28.41`	`KranZ`	`you shouldn't need to mess with the wiring if you have a splitter`
`21:28.47`	`KranZ`	`just use a standard phone line`
`21:28.50`	`KranZ`	`2 wire`
`21:28.57`	`KranZ`	`single line`
`21:29.02`	`*** join/#asterisk pingwin[work] (i=pingwin@gateway/tor/x-56b5695f0cb380b9)`
`21:29.35`	`DaneM`	`OK. Then the splitter I have is just for connecting two phones into one 2-line jack. That must why I was told to make a custom wire.`
`21:29.39`	`*** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie)`
`21:30.08`	`KranZ`	`the splitter replaces the need for a custom wire`
`21:30.15`	`pingwin[work]`	`when calling a queue item, I want to have multiple sound clips play, anyone know how this can be achieved?`
`21:30.20`	`KranZ`	`makes it look like 2 separate jacks, each a different line`
`21:30.43`	`DaneM`	`OK. I'll look into getting one. one min while I check to see if the custom job works.`
`21:30.48`	`pingwin[work]`	`jmls: [TK]D-Fender thanks trying to help before with the macro problem. Used read instead of waitexten and that took care of it. I think it's a problem with macro in general`
`21:31.07`	`KranZ`	`on each jack on the splitter, they are both wired like single line jacks, ie: inside 2 wires are live`
`21:31.09`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`21:31.13`	`[TK]D-Fender`	`pingwin[work] : you aren't supposed to make IVR's in Macro's at all...`
`21:31.47`	`pingwin[work]`	`what's an IVR?`
`21:31.49`	`pingwin[work]`	`sorry`
`21:32.06`	`hohum`	`Interactive Voice Response`
`21:32.11`	`hohum`	`IE`
`21:32.20`	`hohum`	`"hit 3 if your phone is blue"`
`21:32.42`	`pingwin[work]`	`ahhh, then how are you supposed to?`
`21:32.52`	`hohum`	`pingwin: AGI scripts`
`21:32.52`	`KranZ`	`diff contexts`
`21:33.10`	`pingwin[work]`	`because a goto doesn't have the ability to return where it was called from on it's own. You'd have to set variables prior to goto`
`21:33.12`	`[TK]D-Fender`	`pingwin[work] : Us a context like you're supposed to, and not a macro`
`21:33.34`	`KranZ`	`didn't they add gosub and return?`
`21:33.39`	`pingwin[work]`	`it was impossible to do with just contexts`
`21:33.56`	`KranZ`	`i have a few ivr's which work fine in contexts`
`21:34.00`	`KranZ`	`never messed with agi scripts`
`21:34.01`	`[TK]D-Fender`	`pingwin[work] : Impossible? I highly doubt that.`
`21:34.06`	`hohum`	`if you have a problem that doesn't fit within an asterisk dial plan, the you should be using AGI scripts`
`21:34.07`	`pingwin[work]`	`macro's are perfectly capible doing the job. you can include => ${MACRO_CONTEXT} if that's an issue`
`21:34.48`	`[TK]D-Fender`	`pingwin[work] : You try that..... you can't unse VARIABLES for an Include....`
`21:34.55`	`DaneM`	`KranZ: Looks like my wire is bad. I'll go pick up a splitter. Thanks for the help!`
`21:34.56`	`pingwin[work]`	`regardless, my problem is taken care of for me.`
`21:34.57`	`pingwin[work]`	`it works for me.`
`21:35.12`	`pingwin[work]`	`I know AGI scripts would have worked, but i didn't want to go through learning the integration methods and all that prior just looking for quick and fast`
`21:35.33`	`cekc`	`what is the command to see the kernel version I am running?`
`21:35.35`	`hohum`	`quick and fast is dangerous, right is the way`
`21:35.57`	`pingwin[work]`	`[TK]D-Fender: well it would if macro's operated in that way. but just like the variable include, macro's don't operate on the internally defined extensions`
`21:36.05`	`pingwin[work]`	`uname -a`
`21:36.59`	`pingwin[work]`	`but my problem now is I have a queue, and I want the announce to be a succession of sounds. Like "Caller is "&${RECORDED_FILE}`
`21:37.28`	`wmandra`	`afternoon all.... has anyone else had trouble with SIP and comcast over the past few days???`
`21:37.30`	`pingwin[work]`	`I've already tried making the announce overload as being an & delimited string of file names. and it created the announce override string properly`
`21:37.34`	`sbingner`	`are there any known issues w/ talking from 1.2 * to 1.4 *? -- the authentication seems to be rejected for some reason`
`21:37.59`	`pingwin[work]`	`but it tries to open it as a static file. not breaking it up as if it were passed directly into a Playback or background function`
`21:39.05`	`*** part/#asterisk LostFrog (n=reallyno@wsip-68-225-90-115.dc.dc.cox.net)`
`21:39.51`	`pigpen`	`Got reject for frame 27, retransmitting frame 27 now, updating n_r! < what is this on my PRI?`
`21:41.31`	`*** join/#asterisk mbison42 (n=meverts@drake.neopolitan.com)`
`21:41.41`	`fiber0pti`	`Anyone want to try out a new operator panel and give me feedback?`
`21:43.13`	`jmls`	`always willing to look at new stuff`
`21:43.50`	`pigpen`	`sure...`
`21:44.57`	`*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`21:46.12`	`mbison42`	`probably a frequent question, but anyone have an opinion on a good VOIP provider for business use, 4-8 lines? teliax isn;t working out...`
`21:46.32`	`ShadowHntr`	`mbison42: i've been looking into VoicePulse.`
`21:46.41`	`ShadowHntr`	`they're really friendly with asterisk.`
`21:46.45`	`fiber0pti`	`jmls: Great. Head on over to i9technologies.com/isymphony download the server and client and let me know what ya think`
`21:46.54`	`fiber0pti`	`jmls: If you need installation help, feel free to ask :)`
`21:47.09`	`wmandra`	`has anyone here experienced any problems over the last fw days with comcast and SIP??`
`21:47.12`	`fiber0pti`	`pigpen: More than welcome too! let me know if you need help`
`21:47.47`	`bobloblian`	`playback\|noanswer <== says "Not all channels support playing messages while still on hook." what channels do support this?`
`21:49.34`	`bobloblian`	`or better yet, where can I find some relevant information?`
`21:51.33`	`*** join/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca)`
`21:51.47`	`fjean5`	`hi guys`
`21:52.43`	`*** join/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai)`
`21:52.57`	`xai`	`can asterisk set TOS bits in sip and IAX?`
`21:53.46`	`fjean5`	`xai: check here: http://www.voip-info.org/wiki-Asterisk+sip+tos`
`21:57.34`	`fjean5`	`anyone knows a company that do asterisk platform leases`
`21:58.11`	`fiber0pti`	`jmls: did you get the download?`
`21:58.21`	`fiber0pti`	`pigpen: did you get the download, too?`
`21:58.37`	`*** join/#asterisk Dr-Linux\|work (n=Linux@202.59.73.131)`
`22:00.31`	`*** join/#asterisk qoop (n=qoop@d36-115-191.home1.cgocable.net)`
`22:01.27`	`*** part/#asterisk qoop (n=qoop@d36-115-191.home1.cgocable.net)`
`22:03.10`	`xai`	`Would it be better to use port-forwarded IAX instead of routing it through the VPN between offices?`
`22:03.25`	`*** part/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca)`
`22:04.06`	`*** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net)`
`22:05.16`	`doolph`	`hello, anyone have idea why my register command in sip.conf is not working?`
`22:06.44`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`22:08.45`	`[TK]D-Fender`	`doolph : pastebin your sip.conf (mask only passwords please)`
`22:08.58`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`22:10.08`	`*** join/#asterisk vlt (n=daniel@dslb-088-073-227-023.pools.arcor-ip.net)`
`22:13.06`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`22:13.06`	`*** mode/#asterisk [+o mog] by ChanServ`
`22:21.53`	`FuriousGeorge`	`mog: im at a total loss here with this res_jabber module that isnt being installed. did anything come to you, cuz ive tried everything i can think of`
`22:27.08`	`xai`	`I have an office that routes SIP packets via VPN (ipsec) from the main (asterisk) office, directly to their phones. I'm thinking of putting small * appliances there and just use IAX routing/NAT. I think that would be much faster than using ipsec.`
`22:27.18`	`mog`	`no FuriousGeorge you could just jabber me`
`22:27.35`	`xai`	`every time you send phone traffice, it gets delay due to encryption/decryption .`
`22:29.34`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`22:31.52`	`FuriousGeorge`	`mog: one sec`
`22:33.11`	`FuriousGeorge`	`mog check your jabber client`
`22:33.34`	`mog`	`nothing`
`22:33.43`	`mog`	`send a message to mogorman@astjab.org or mogorman@digium.com`
`22:33.47`	`*** join/#asterisk findlay (n=justin@67.137.24.115)`
`22:36.46`	`*** join/#asterisk coldoutside (n=root@38.99.66.231)`
`22:37.20`	`*** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net)`
`22:37.36`	`coldoutside`	`I'm having voice problems with my Asterisk/IAX setup and I need some help troubleshooting. Can anyone give some pointers, or direct me to a good resource to target my problem?`
`22:38.26`	`kratzers`	`voip-info.org is a good place to look`
`22:38.30`	`remmo`	`iax debug`
`22:39.55`	`Crad`	`man trixbox is pissing me off :-x`
`22:40.23`	`coldoutside`	`I've looked through voip-info.org, but haven't solved my problem yet`
`22:40.25`	`sbingner`	`Crad: that's what it's designed to do`
`22:40.33`	`Crad`	`the extras are working good, the core stuff I need to get sip trunking working dont seem to be working at all`
`22:40.39`	`Crad`	`:-\`
`22:40.44`	`coldoutside`	`or even really been able to identify it. this has been going on for several months now.`
`22:41.15`	`*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)`
`22:42.51`	`findlay`	`what is it called when it sounds like the audio is interspersed with rapid spaces of silence, kind of like a pulsating sound?`
`22:43.16`	`findlay`	`or even better, where can I read about sound quality troubleshooting?`
`22:46.29`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`22:47.30`	`*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`22:52.25`	`coldoutside`	`I've still been unable to find a decent troubleshooting tutorial for choppy audio, arr.`
`22:52.42`	`findlay`	`coldoutside: when you find one let me know (:`
`22:52.51`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`22:52.55`	`justinu\|laptop`	`it's called jitter and packet loss`
`22:52.57`	`coldoutside`	`my ping times wander a bit, but are 60-100ms on average`
`22:53.33`	`coldoutside`	`shouldn't the jitter buffer take over and give me clean--though delayed--audio?`
`22:53.57`	`coldoutside`	`and how do you gather stats from Asterisk on jitter and loss?`
`22:54.07`	`findlay`	`justinu\|laptop: those are the only parameters which control audio quality?`
`22:54.36`	`coldoutside`	`also- a good portion of the time It still says '0' for lost under 'iax2 netstats' during these situations`
`22:55.00`	`justinu\|laptop`	`yeah, you need a packet every 20ms (in the right sequence) for stuff to sound good`
`22:55.20`	`justinu\|laptop`	`high ping times only indicate latency, not sound quality issues`
`22:55.55`	`justinu\|laptop`	`you can't really use normal ICMP ping to test packetloss/jitter, things change when you're sending 50+ packets/second`
`22:57.46`	`*** join/#asterisk vlt (n=daniel@dslb-088-073-249-136.pools.arcor-ip.net)`
`22:57.49`	`justinu\|laptop`	`jitter buffer in asterisk isn't all that, afaik`
`22:57.59`	`justinu\|laptop`	`it only applies to IAX too, i think`
`23:00.16`	`findlay`	`the problem I'm having is the sound I send out has lots of very narrow spaces in it like the individual packets of sound aren't joind up`
`23:00.45`	`justinu\|laptop`	`make sure your ATA or whatever is set to use 20ms packetization`
`23:01.04`	`findlay`	`what's ATA?`
`23:01.39`	`justinu\|laptop`	`your SIP device, phone, ata, whatever`
`23:01.49`	`findlay`	`ok`
`23:02.17`	`*** join/#asterisk ast_freak\|Laptop (n=jesse@66-191-130-194.static.roch.mn.charter.com)`
`23:02.54`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`23:02.55`	`coldoutside`	`hmm. how do you test packetloss for the purpose of voip?`
`23:03.04`	`justinu\|laptop`	`i use ethereal/wireshark`
`23:03.31`	`justinu\|laptop`	`it has a lot of nice RTP analysis`
`23:03.33`	`justinu\|laptop`	`tools`
`23:04.12`	`coldoutside`	`hmm. I've never tested packet loss with ethereal, I'll have to give that a shot`
`23:05.47`	`*** join/#asterisk mv00 (i=HiTMaN@gateway/tor/x-fb8bf92636eb2d7c)`
`23:06.40`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`23:07.10`	`*** part/#asterisk DaneM (n=DaneM@70.135.56.182)`
`23:07.36`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`23:10.21`	`*** join/#asterisk adorah (n=admin@84.94.127.28.cable.012.net.il)`
`23:12.27`	`*** join/#asterisk vlt (n=daniel@dslb-088-073-249-136.pools.arcor-ip.net)`
`23:18.50`	`coldoutside`	`justinu\|laptop: thanks for the help, I'm compiling Wireshark now, might ask you a few more questions later`
`23:24.20`	`*** join/#asterisk Skyelar (n=planet@222-152-135-196.jetstream.xtra.co.nz)`
`23:26.23`	`*** part/#asterisk doolph (n=doo@200.46.148.58)`
`23:27.35`	`Skyelar`	`is there a way to administratively "busy out" an fxo port on a tdm400, ie. take it off hook so the exchange thinks it's busy, and keep it that way? Have a line fault on an analogue line that's part of a stepping group`
`23:28.11`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`23:31.53`	`*** join/#asterisk Schulich (n=Jazba@165.154.37.136)`
`23:32.59`	`*** join/#asterisk oQPa (i=Ftv@237.Red-83-44-33.dynamicIP.rima-tde.net)`
`23:36.00`	`*** join/#asterisk psiforce_ (n=mark@c210-49-175-128.mckinn1.vic.optusnet.com.au)`
`23:36.55`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`23:38.07`	`psiforce_`	`hi all can someone help me with faxing`
`23:39.14`	`psiforce_`	`I keep getting these warnings "unable to restore write format" when receiving faxs`
`23:40.02`	`psiforce_`	`calls are coming in from a pri connection (via sangoma card)`
`23:40.39`	`*** join/#asterisk MoutaPT (n=MoutaPT@a213-22-41-37.cpe.netcabo.pt)`
`23:41.17`	`*** join/#asterisk dir (n=dir@124.106.223.190)`
`23:41.32`	`Skyelar`	`psiforce_: how are things connected - pri -> asterisk -> ?`
`23:42.15`	`psiforce_`	`fax is correctly stored as a tiff in the correct directory but asterisk does not proceed to the next priority (sending the tiff)`
`23:42.25`	`psiforce_`	`I think that is largely due to the warnign I get`
`23:42.29`	`FuriousGeorge`	`so are the pickupgroups in sip.conf now the current implementation of sla (1.4beta)`
`23:42.59`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`23:42.59`	`*** mode/#asterisk [+o mog] by ChanServ`
`23:43.27`	`psiforce_`	`skyelar: asterisk is recieving the fax via rxfax() app`
`23:43.40`	`Skyelar`	`psiforce_: ahh, not something I've played with sorry`
`23:43.42`	`justinu\|laptop`	`Skyelar: there's no way to do what you want without convincing asterisk to take the port off hook indefinitely`
`23:44.08`	`Skyelar`	`justinu\|laptop: that's exactly what I'd like to do, providing it's reversable :)`
`23:44.15`	`justinu\|laptop`	`asterisk doesn't know about concepts like out of service, in service, etc.`
`23:47.11`	`bkw__`	`ASSERTS KICK US`
`23:47.17`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`23:47.45`	`justinu\|laptop`	`somone oughta tell this guy about asterisk`
`23:47.46`	`justinu\|laptop`	`http://www.raphnet.net/electronique/cid/cid_en.php`
`23:48.27`	`mog`	`bkw__, knock it off`
`23:48.34`	`*** join/#asterisk fifer (n=sirfifer@c-24-20-155-56.hsd1.wa.comcast.net)`
`23:48.56`	`fifer`	`I'm running into a new issue with voicemail`
`23:50.51`	`psiforce_`	`no takers on the faxing problem ?`
`23:51.38`	`fifer`	`When a VM is being recorded I get a bunch of these:`
`23:51.39`	`diclophis-work`	`psiforce_: whats the problem?`
`23:51.39`	`fifer`	`format_wav_gsm.c243 update_header Unable to find our position`
`23:52.07`	`diclophis-work`	`psiforce_: i have been unable to send faxes as well`
`23:52.11`	`diclophis-work`	`rx works well though`
`23:53.40`	`fifer`	`Then, when a user tries to listen to the VM I get: app_voicemail.c:3783 play_message: No origtime?!`
`23:54.21`	`fifer`	`VM has been working fine for many months, no issues`
`23:55.27`	`findlay`	`is it necessary to restrict remote connections to the IAX port 4569 to specific hosts?`
`23:56.20`	`findlay`	`or can I leave it open because I'm lazy (:`
`23:56.30`	`justinu\|laptop`	`s/lazy/brave`
`23:56.46`	`findlay`	`ok`

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