irclog2html for #asterisk on 20060127

`00:02.16`	`*** join/#asterisk AlexCTI (i=AlexCTI@139.sub-70-219-78.myvzw.com)`
`00:04.30`	`[TK]D-Fender`	`UGH... any RTP/audio bugs reported with 1.2.3? I've got 2 servers I'm trying to tied together with one acting as peer, the other as friend (SIP)`
`00:04.56`	`[TK]D-Fender`	`No NAT, all public IP's, server to server`
`00:05.14`	`[av]bani`	`public ips? party time for hax0ring!!11!`
`00:05.42`	`Seldon1975`	`hey`
`00:06.13`	`Seldon1975`	`i just tried to 'make install' asterisk 1.2.3 from source but I get a message: "make: *** [cleantest] Error 1"`
`00:06.18`	`[TK]D-Fender`	`After all the help I give to this channel I think I deserve a bit more thank you....`
`00:06.20`	`Seldon1975`	`please help`
`00:06.28`	`Seldon1975`	`my pbx is down at the moment`
`00:06.43`	`*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)`
`00:07.07`	`[TK]D-Fender`	`Seldon1975 : Do a "make clean" first then "make", then "make install"`
`00:07.29`	`Seldon1975`	`D-Fender: same result`
`00:07.34`	`[TK]D-Fender`	`:/`
`00:08.54`	`QbY`	`I just built * 1.2.3 -- I want to build it from the ground up (.confs) -- wheres the best documentation for it?`
`00:09.21`	`QbY`	`[TK]D-Fender -- Thank you..`
`00:09.23`	`QuAd\|Haudrauf`	`concerning the gxp2000, can someone confirm this?: - it's not possible to delete letters and numbers from phonebook entries, number, name etc... a delete key is missing!`
`00:09.32`	`Seldon1975`	`D-Fender: this is my output: http://pastebin.com/524874`
`00:10.18`	`[TK]D-Fender`	`QbY : What kind of phones are you running?`
`00:10.38`	`Seldon1975`	`i just tried to 'make install' asterisk 1.2.3 from source but I get a message: "make: *** [cleantest] Error 1"`
`00:10.41`	`Seldon1975`	`anyone?`
`00:10.43`	`QbY`	`SIP.. and I have IAX trunks`
`00:11.30`	`[TK]D-Fender`	`QbY : Ok, are your phones set up? "sip.conf" part that is.`
`00:11.52`	`QbY`	`nope.. i'm going to build it from scratch.. this is a dummy box -- so i can play/learn`
`00:12.05`	`*** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com)`
`00:12.17`	`*** join/#asterisk SibrPhrek (i=SibrPhre@user-12lccke.cable.mindspring.com)`
`00:12.41`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`00:12.45`	`Alric`	`SetCallerID(num\|a) used to set ANI as well. With the new system, does Set(CALLERID(number)=num\|a) work as well?`
`00:13.04`	`[TK]D-Fender`	`QbY : What were you using before?`
`00:14.00`	`QbY`	`[TK]D-Fender -- I have an Asterisk/AMP installation--but I got tired of being "limited" so I stopped using the AMP.`
`00:14.19`	`*** join/#asterisk kokostark (n=kokos@toronto-HSE-ppp4282177.sympatico.ca)`
`00:14.19`	`QbY`	`But with it being once an AMP install, I get so damn confused with how the call flows.. So I want to build this box from scratch`
`00:14.26`	`QbY`	`and replace my other box`
`00:14.38`	`EksilAndyCap`	`QbY: going from amp to writing config from scratch is a bit much? http://www.voip-info.org/wiki/view/Asterisk+config+files`
`00:15.22`	`[TK]D-Fender`	`QbY, ok, do you still have your AMP leftovers?`
`00:15.26`	`AlexCTI`	`How can I check my Asterisk version?`
`00:15.34`	`jaike`	`show version`
`00:15.34`	`rene-`	`QBY i have found that editing the *_custom.conf files work best most of the time`
`00:15.38`	`rene-`	`s/best/well`
`00:15.41`	`AlexCTI`	`ok, thanks`
`00:15.58`	`QbY`	`EksilAndyCap: I've been tinkering currently.. But I'm tired of asking questions.. So I want to build this box and do it..`
`00:16.11`	`eieiyo`	`QbY, i found this site pretty good... if you just read the comments you will understand how it links up http://www.loligo.com/asterisk/current/`
`00:16.20`	`rene-`	`in debian you can install amp super easy`
`00:16.20`	`*** join/#asterisk fdgfd (n=fdgfd@adsl-16-121.37-151.net24.it)`
`00:16.23`	`jaike`	`how bout the sample conf files with "make samples"`
`00:16.28`	`fdgfd`	`hi`
`00:16.31`	`QbY`	`cools`
`00:16.34`	`Seldon1975`	D-Fender: here is the error in my make process: cp: cannot stat `.cleancount': No such file or directory
`00:16.42`	`Seldon1975`	`D-Fender: does that make any sense?`
`00:16.48`	`[TK]D-Fender`	`Seldon1975 : Where did you DL it from?`
`00:17.01`	`Seldon1975`	`svn.digium.com/asterisk/trunk`
`00:17.31`	`[TK]D-Fender`	`jaike "make samples" creates a psychotic pile of crap setup that takes forever to clean out.`
`00:17.33`	`rene-`	`if you are relatively new to asterisk then i recommend to scrap the sample files and wirte them in your edito so you can learn what everything means, you can then add anything you need, and it looks neat, professional and it is easy to maintain`
`00:17.41`	`[TK]D-Fender`	`Seldon1975 : DL from FTP direct from digium.com`
`00:17.51`	`eieiyo`	`does anybody know how to change the port for sip that asterisk uses? do you just change it in the configuration files under the [general] context.... port=2000 instead of port 5060. will this work?`
`00:18.01`	`rene-`	`it is best to have limited functionality but to have a grasp on what your * box does and why`
`00:18.10`	`Seldon1975`	`D-Fender: ok, but isnt /trunk meant to be latest stable 1.2.3?`
`00:18.12`	`fdgfd`	`I'm new with asterisk: can somebody tell me if it can act as SIP registrar? and also what can I do if my asterisk is separed from phones by a NAT?`
`00:18.29`	`justinu`	`yes, and yes (for the most part)`
`00:18.37`	`[TK]D-Fender`	`QbY : list me the extensions you'd like to have and I'll give you a sample extension.conf to start with`
`00:18.41`	`jaike`	`tkd-fender: its got documentation in it so i think its a start for beginners`
`00:18.52`	`rene-`	`fdgfd: yes`
`00:18.54`	`[TK]D-Fender`	`Seldon1975 : I don't know... I'm just going with what seems to work.`
`00:18.58`	`rene-`	`and yes`
`00:19.00`	`rene-`	`haha`
`00:19.12`	`Seldon1975`	`D-Fender thanks`
`00:19.27`	`[TK]D-Fender`	`jaike : there's jsut so much randon stuff in there it leaces you asking "why" beacuse it isn't coherent. the parts don't add up... its just PARTS.`
`00:19.31`	`rene-`	`if asterisk is outside nat it will be easy to make your sip behing nat phones work`
`00:19.39`	`rene-`	`s/behing/behind`
`00:19.46`	`fdgfd`	`ok, and how can I solve NAT problem? if I have VOIP phone - NAT - Asterisk, the voip phone MUST speackIAX ?`
`00:19.59`	`rene-`	`that would be a way`
`00:20.04`	`rene-`	`but for the most part`
`00:20.13`	`justinu`	`SIP works over nat, just set nat=yes in sip.conf`
`00:20.19`	`Drew_____`	`what is natted asterisk or the phones?`
`00:20.20`	`rene-`	`you just need to add a nat=yes directive in your asterisk sip config for the device`
`00:20.33`	`fdgfd`	`and he do like a Session Border Controller?`
`00:20.35`	`[TK]D-Fender`	`fdask : No... * can work fine with NAT using SIP if you set it right`
`00:20.37`	`jaike`	`justinu: ive learned its not always the case....sip and nat has always given me problems`
`00:20.47`	`rene-`	`asterisk does sit in the audio path`
`00:20.51`	`justinu`	`it can be tricky, but if you know what you're doing you can make it work`
`00:20.58`	`AlexCTI`	`jaike: which version is it: Asterisk SVN-trunk-r7230?`
`00:21.04`	`[TK]D-Fender`	`Drew_____ has learned all sorts of things today about NAT hasn't he? :)`
`00:21.16`	`Drew_____`	`yes :)`
`00:21.29`	`rene-`	`so you can think of it as a sbc in that regard`
`00:21.42`	`fdgfd`	`SIP and NAT can work (STUN & co) but the real problem is the media`
`00:21.49`	`fdgfd`	`yea`
`00:21.59`	`Drew_____`	`well acutally i learnt about asterisk - the concept of NAT is clear ;)`
`00:22.06`	`fdgfd`	`asterisk can sit in the audio path yous saying nat = yes ? :)`
`00:22.12`	`[TK]D-Fender`	`fdask : I've worked nat every which way and have never had a problem....`
`00:22.23`	`rene-`	`yes`
`00:22.33`	`rene-`	`you dont even need stun`
`00:22.43`	`rene-`	`in some cases`
`00:22.46`	`[TK]D-Fender`	`fdask : For * to stay in the middle at least one leg of the call must be "canreinvite=no"`
`00:23.10`	`jaike`	`alexcti: dunno...mine shows Asterisk 1.2.1 built by root @ pbx-5 on a i686 running Linux on 2005-12-20 05:58:14 UTC`
`00:23.17`	`rene-`	`D-Fender is right`
`00:23.17`	`[av]bani`	`[TK]D-Fender: our polycoms shipped with boot 2.6.0 / rom 3.1.0.0269 / sip 1.6.2.0041, should we upgrade?`
`00:23.40`	`[TK]D-Fender`	`the versions don't match....`
`00:23.48`	`[av]bani`	`??`
`00:23.50`	`fdgfd`	`D-Fender: what do you meant with "one leg" sorry?`
`00:24.06`	`[TK]D-Fender`	`2.6 ione major version, 3.1 is a completely different version. can you clarify these?`
`00:24.11`	`MstlyHrmls`	`[av]bani: boot and rom aren't seperate version. It's a BootROM`
`00:24.16`	`[TK]D-Fender`	`thaose are both BR versions`
`00:24.22`	`*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)`
`00:24.27`	`rene-`	`one of the sides of the conversation other than asterisk`
`00:24.34`	`[av]bani`	`im reading it off the phone's 'status' page`
`00:24.38`	`rene-`	`or asterisk if the call ends in asterisk (e.g. voicemail)`
`00:24.56`	`[TK]D-Fender`	`MstlyHrmls : Wait.. nt entirely. there is the bootBlock.`
`00:25.13`	`MstlyHrmls`	`[TK]D-Fender: that's true`
`00:25.16`	`[TK]D-Fender`	`[av]bani : Ok, you've got near-bleeding-edge firmware on it. no need to upgrade anything`
`00:25.27`	`*** part/#asterisk jaike (n=a@203.131.137.76)`
`00:25.34`	`justinu`	`damn eyebeam softphone! doesn't look at the DNS SRV record priorities`
`00:25.35`	`MstlyHrmls`	`[TK]D-Fender: but you don't generally upgrade that :-)`
`00:25.42`	`[TK]D-Fender`	`MstlyHrmls : Never saw any meaningful docs on BootBlock vs BootROM though. Unsure of the implications.`
`00:25.43`	`fdgfd`	`canreinvite I suppose that is for optimize the Session Border Controller that is to re-invite client to put in contact without triangular routing isn't it?`
`00:25.45`	`[av]bani`	`super annoying: polycom web has no status page`
`00:25.50`	`[av]bani`	`only settings`
`00:26.07`	`_Sam--`	`chalk 2 up for the gxp2000!`
`00:26.09`	`[TK]D-Fender`	`My IP 600 here is on BB 2.4.0, BR 2.6.1, SIP 1.5.2`
`00:26.20`	`[av]bani`	`o_O`
`00:26.21`	`[TK]D-Fender`	`[av]bani : web?`
`00:26.24`	`jbalcomb`	`_Sam-- eh?`
`00:26.25`	`[av]bani`	`webadmin`
`00:26.40`	`rene-`	`polycoms are annoying`
`00:26.41`	`[av]bani`	`doesnt tell you anything about the phone status, only lets you change settings`
`00:26.52`	`[av]bani`	`sipura, grandstream etc all tell you full status`
`00:26.52`	`[TK]D-Fender`	`[av]bani : Kep away from the web-adnim if you know whats good for you :)`
`00:26.55`	`_Sam--`	`sorry...[av]bani earlier said the gxp2000 config was much easier...now he said the polycom has no status page (the gxp2000 does)...thats why i said chalk 2 up`
`00:27.00`	`*** join/#asterisk franck (n=franck@tikiwiki/franck)`
`00:27.02`	`[av]bani`	`[TK]D-Fender: thats what i'm slowly figuring out :/`
`00:27.04`	`rene-`	`canreinvite=yes means asterisk will try to stay out of the conversation`
`00:27.16`	`MstlyHrmls`	`[TK]D-Fender: hard to find info on the BootBlock`
`00:27.16`	`fdgfd`	`ok`
`00:27.19`	`[TK]D-Fender`	`[av]bani : You don't trust me after all this time? I'm hurt...`
`00:27.20`	`[TK]D-Fender`	`heh`
`00:27.27`	`rene-`	`eg asterisk will issue a reinvite to calling party`
`00:27.28`	`fdgfd`	`thank you guys you're very kindly`
`00:27.30`	`MstlyHrmls`	`[TK]D-Fender: it's primarially all about BootROM and app`
`00:27.30`	`jbalcomb`	`_Sam-- ah, too bad everything else about the GXP-2000 blows. :/`
`00:27.35`	`[TK]D-Fender`	`[av]bani : Got the SIP 1.6.2 SIP package?`
`00:27.44`	`[av]bani`	`?`
`00:27.49`	`fdgfd`	`I'll choose * for my next work as SBC`
`00:27.58`	`rene-`	`grandtream is easy to setup but poor quality, polycom is the opposite`
`00:28.02`	`[TK]D-Fender`	`[av]bani : the firmware comes with a pile of sample configs to start you off...`
`00:28.13`	`[av]bani`	`[TK]D-Fender: ours shipped that way`
`00:28.14`	`jbalcomb`	`how do you configure IRQs in the Linux?`
`00:28.16`	`[av]bani`	`out of the box`
`00:28.34`	`[TK]D-Fender`	`jbalcomb : You don't really.. you need to go into your boios to move stuff around.`
`00:28.34`	`EksilAndyCap`	`jbalcomb: in the bios?`
`00:28.40`	`[TK]D-Fender`	`jbalcomb : tha means down-time`
`00:28.50`	`jbalcomb`	`I heart downtime.`
`00:29.03`	`[av]bani`	`jbalcomb: you dont. (you dont in windows either, really. the 'irq routing' is really pseudo-routing)`
`00:29.05`	`EksilAndyCap`	`jbalcomb: and depending on the bios it might not work so you have to swap cards around`
`00:29.22`	`Mr-packet`	`any suggestions what hardware works well to connect to analog lines on the PTSN?`
`00:29.33`	`jbalcomb`	`I just need to get the four other devices to stop sharing an IRQ with my Digium TE411P`
`00:29.40`	`MstlyHrmls`	`[TK]D-Fender: the main thing I've heard is that the BootBlock is there to make the BootROM upgrades fault tolerant`
`00:29.54`	`EksilAndyCap`	`jbalcomb: since the actual irq lines A,B,C and D move around move in some fancy weave pattern`
`00:30.12`	`Drew_____`	`is there any way of controlling the volume of playback in * ?`
`00:30.23`	`AlexCTI`	`Some one can tell me which is my asterisk version: Asterisk SVN-trunk-r7230 built by root @ VoIpSrv on a i686 running Linux on 2006-01-09 15:50:45 UTC`
`00:30.49`	`Drew_____`	`alex - type "show version" in *-console`
`00:31.03`	`AlexCTI`	`thats what i did..`
`00:31.11`	`alx_`	`fugitivo: jepp .. they banned my ip to use their service .. changed the IP and its all working again ;)`
`00:32.42`	`[TK]D-Fender`	`jbalcomb : Just take it down and start poking!`
`00:33.00`	`AlexCTI`	`VoIpSrv*CLI> show version`
`00:33.00`	`AlexCTI`	`Asterisk SVN-trunk-r7230 built by root @ VoIpSrv on a i686 running Linux on 2006-01-09 15:50:45 UTC`
`00:34.06`	`EksilAndyCap`	`AlexCTI: it's a svn revision not a release. :-P so I guess you can look at the revision history in svn and see what release you're closes too`
`00:34.40`	`[TK]D-Fender`	`[av]bani : Download this, make an account on your server for your Polycom's and extract it into the /home folder for that account - http://www.freedomphones.net/polycom/files/SoundPoint_IP_SIP_1_6_2.zip`
`00:35.12`	`tehdely`	`i think i have a newer release sitting around`
`00:35.40`	`[TK]D-Fender`	`tehdely : I was just tyeing to give him the version that matched his phones first.`
`00:35.46`	`tehdely`	`ah`
`00:35.55`	`tehdely`	`i still haven't moved to central provisioning`
`00:35.58`	`[TK]D-Fender`	`Start slow, tweak it in and 1.6.3 is mostly bug-fixes...`
`00:36.02`	`tehdely`	`only have a few of the darn things`
`00:36.09`	`[TK]D-Fender`	`tehdely : I do it at work & home.`
`00:36.09`	`*** join/#asterisk Seldon19751 (n=someone@199.243.101.131)`
`00:36.36`	`[TK]D-Fender`	`(having taken an IP 600 home it provisions for home use whie here, and back to normal in the office)`
`00:36.43`	`tehdely`	`nice`
`00:36.50`	`tehdely`	`i have a 501 on my desk but i just spent a few minutes in the web panel`
`00:36.54`	`tehdely`	`lazy i suppose :>`
`00:36.59`	`malverian[work]`	`Hey guys, quick question...`
`00:37.14`	`malverian[work]`	`If I have exten => _XXXX,1,........`
`00:37.20`	`malverian[work]`	`And then I include => somecontext`
`00:37.28`	`[TK]D-Fender`	`_XXXX = evil`
`00:37.30`	`malverian[work]`	`And somecontext has exten => 1234,1,.....`
`00:37.40`	`[TK]D-Fender`	`1234 should take precedence`
`00:37.49`	`malverian[work]`	`Yeah.. that's what I thought, but it isn't for some reason..`
`00:37.50`	`malverian[work]`	`With 1.2.1`
`00:38.19`	`[TK]D-Fender`	`Hmmm... could be "order of occurence". Try swapping the calls`
`00:38.23`	`Seldon19751`	`is there a specific version of Zaptel to use with Asterisk1.2.3? Is svn/zaptel/trunk ok?`
`00:38.38`	`malverian[work]`	`[TK]D-Fender, I did.`
`00:38.39`	`[TK]D-Fender`	`Seldon19751 : 1.2.2 is still the latest.`
`00:38.55`	`Seldon19751`	`ok, is that what I get at zaptel/trunk`
`00:38.55`	`[TK]D-Fender`	`It was an emergency SIP update, not because of Zaptel`
`00:39.50`	`*** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.57.165.Dial1.Chicago1.Level3.net)`
`00:40.00`	`malverian[work]`	`:(`
`00:40.02`	`malverian[work]`	`I'm confused...`
`00:41.27`	`[TK]D-Fender`	`Maybe its pure alpha and _ beats 1. Check the WIKI.`
`00:42.03`	`QuAd\|Haudrauf`	`nn`
`00:43.15`	`malverian[work]`	`Gah.. this sucks.`
`00:43.19`	`malverian[work]`	`I swear this worked in 1.2.0`
`00:44.51`	`[av]bani`	`i hate how * sorts extensions`
`00:44.56`	`[av]bani`	`i want them in the order i put them in, dammit`
`00:45.07`	`mogorman`	`?`
`00:45.16`	`*** join/#asterisk pengyong (n=lala@222.188.134.60)`
`00:46.05`	`malverian[work]`	`Yeah.. ugh..`
`00:46.11`	`malverian[work]`	`I really thought this worked before..`
`00:46.45`	`*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`00:51.09`	`*** join/#asterisk troyb1 (n=troy@64.39.162.84)`
`00:51.47`	`[TK]D-Fender`	`Drew_____ : They're worth every penny I paid for them.....`
`00:52.35`	`Drew_____`	`sure - but you can get music unter the cc-licence on ie. garageband.com`
`00:53.51`	`Drew_____`	`ok i agree - cc isnt gpl but it will do`
`00:54.33`	`*** join/#asterisk Damin_PDA (n=pocketir@49.sub-70-199-159.myvzw.com)`
`00:56.30`	`Zodiacal-`	`for those of you that run SIP. do your ext.'s ring on the first ring that * gets?`
`00:57.23`	`Zodiacal-`	`im wondering if my ext' ring delay issue is related to sip or my analog lines and/or fxo modules..`
`00:59.04`	`Zodiacal-`	`any and all input would be greatly appreciated`
`01:00.47`	`Zodiacal-`	`i guess everyones sleepin`
`01:01.03`	`AlexCTI`	`Anyone knows how can i make that asterisk take the ANI and send it to thr carrier, so I'm using asterisk as VoIP gateway.`
`01:01.04`	`malverian[work]`	`Anyone have a decent ring tone that will work with snom320?`
`01:01.23`	`wunderkin`	`Zodiacal-, hold onto your panties, it is because it is waiting for caller id`
`01:01.33`	`wunderkin`	`that is because of the analog lines yes`
`01:01.42`	`Zodiacal-`	`wunderkin i set callerid to no`
`01:01.42`	`mzo`	`is there some limit of what kinds of songs you can put in? it will take every format? :P`
`01:02.41`	`wunderkin`	`Zodiacal-, so you have an analog card connected to your phone line and a sip phone to *, the sip phone doesn't ring until after a few rings from the pstn?`
`01:02.47`	`reza`	`can i make a call using the * cli? if so, what's the syntax or where is it documented?`
`01:03.09`	`Zodiacal-`	`wunderkin exactly.. its usualy the second or third ring`
`01:03.20`	`Zodiacal-`	`softphones and hardphones`
`01:03.37`	`malverian[work]`	`Anyone have a place to download normal phone type ring tones?`
`01:03.42`	`Zodiacal-`	`also dialing out takes about 15 seconds before the remote party side rings`
`01:04.20`	`mzo`	`i have pstn with analog cards and basically first ring from an incoming caller, has a two second pause before it rings internally, and then it works fine. outgoing calls have a simliar pause before you hear ringing`
`01:04.27`	`wunderkin`	`Zodiacal-, usecallerid=no in /etc/asterisk/zapata.conf, you must not have reloaded it.. maybe something else is wrong then`
`01:04.37`	`Zodiacal-`	`mzo so its just normal?`
`01:04.57`	`Zodiacal-`	`wunderkin yeah thats what i use. sorry i didn't give the right tag usercallerid=no but thats what i have..`
`01:05.14`	`Zodiacal-`	`it got reloaded too`
`01:05.25`	`mzo`	`i have no idea. I just use it, i never thought it was a problem, i thought it was processor related but it happens on my newer hardware too shrug`
`01:05.38`	`Zodiacal-`	`yeah i even tried on a faster pc, same thing..`
`01:05.46`	`Zodiacal-`	`mzo what speed do you run?`
`01:05.57`	`Zodiacal-`	`i tried it on a PIII 550mhz 384MB's ram`
`01:06.01`	`mzo`	`p3850, smp with 512mb`
`01:06.06`	`mzo`	`er, 850mhz :P`
`01:06.14`	`Zodiacal-`	`:/`
`01:06.24`	`Zodiacal-`	`mzo do you use usecallerid=yes or no?`
`01:06.26`	`mzo`	`i thought maybe it's a processor thing, i dunno.`
`01:06.28`	`mzo`	`i have it turned on`
`01:06.41`	`mzo`	`i need it to ignore people who i dont' want to speak too, like school, and citibank`
`01:06.50`	`malverian[work]`	`MUST HAVE NEW RINGTONE! :-P`
`01:07.05`	`Zodiacal-`	`mzo welp from what i have read, your delay is due to you useing callerid`
`01:07.17`	`Zodiacal-`	`but i don't have it with my phone co. and i don't need it, but it seems like its still waiting for it or somthin`
`01:07.35`	`malverian[work]`	`No one knows of a website for downloading normal ring tones? Eg.. not music ones for cell phones.`
`01:07.45`	`Zodiacal-`	`mzo does your * detect the ring instatly? mine does i can see it detect it, but it doesn't ring my ext. until after the first ring`
`01:07.45`	`mzo`	`i need caller id :P`
`01:07.57`	`mzo`	`i have no idea if it sees it insntantly. I think it does`
`01:08.21`	`[TK]D-Fender`	`ok, nasty problem I could use a hand with. I'm bridging 2 system together through SIP. A reg's to B like a phone but I'm not getting audio bother ways. Both are public IP's w/o filtering. on tying to dial this is the last message AI see fromt he calling side : -- Attempting native bridge of SIP/2039-2ae9 and SIP/199-008d`
`01:08.59`	`iCEBrkr`	`OMG!`
`01:09.05`	`[TK]D-Fender`	`After that message I see the application in hte receiing ends dial-plan roll through, but no audio is making its way IN to their server. when I hit Comedian mail I get audio, but they don't hear me.`
`01:09.06`	`iCEBrkr`	`[TK]D-Fender: You're stumped?`
`01:09.18`	`[TK]D-Fender`	`iCEBrkr : Merely mortal! Shocking, no?`
`01:09.53`	`iCEBrkr`	`[TK]D-Fender: I still find it odd.`
`01:10.27`	`Zodiacal-`	`tkd-fender is that that audio problem they were talking about with version 1.2.2?`
`01:11.01`	`Drew_____`	`sounds very simmilar to the timebomb`
`01:11.25`	`franck`	`sometimes I get a Unknown RTP codec 112 received in the asterisk log... and then the audio works only one way`
`01:11.48`	`*** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.57.165.Dial1.Chicago1.Level3.net)`
`01:12.20`	`franck`	`"Unknown RTP codec 112 received" <- what does it mean?`
`01:13.11`	`mzo`	`codec 112 in unkonwn? :P`
`01:13.49`	`franck`	`mzo: hahaha`
`01:13.56`	`franck`	`what is rtp codec 112?`
`01:18.10`	`SibrPhrek`	`hey can someone help me with music on hold?`
`01:18.56`	`SibrPhrek`	`i added the extention for it - but i get no sounds`
`01:19.21`	`*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)`
`01:19.49`	`*** join/#asterisk elg (n=fugalh@falcon.fugal.net)`
`01:22.50`	`*** join/#asterisk FarrisG (n=farris@c-24-1-176-16.hsd1.tx.comcast.net)`
`01:23.55`	`FarrisG`	`Is there a quick way to setup two software sip clients with very reliable backbone connections and configure a high-quality codec, to sort of setup an ISDN-like voice connection for remote radio interviews?`
`01:24.34`	`SibrPhrek`	`i have an extention to go to music on hold but i don't hear anything`
`01:25.24`	`*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`01:26.53`	`SibrPhrek`	`i get a 603 declined`
`01:26.53`	`SibrPhrek`	`No application 'WaitMusicOnHold' for extension (internal, 300, 1)`
`01:26.53`	`SibrPhrek`	`<PROTECTED>`
`01:27.51`	`*** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net)`
`01:29.31`	`*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)`
`01:30.12`	`*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`01:33.25`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`01:33.51`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`01:35.34`	`slan`	`How can I tell Asterisk to re-read its configuration files? Source filename?`
`01:36.12`	`inv_Arp`	`reload`
`01:36.18`	`slan`	`That's a command?`
`01:37.02`	`newl`	`yes`
`01:37.08`	`*** join/#asterisk argentas_ (n=martin@212.84.168.69)`
`01:37.10`	`SibrPhrek`	`argh`
`01:37.13`	`slan`	`inv_Arp: Thanks very much - I'll go try it.`
`01:37.16`	`SibrPhrek`	`this Musiconhold is annoying`
`01:37.39`	`[TK]D-Fender`	`Zodiacal- : No, both sides are 1.2.3.`
`01:37.57`	`mzo`	`it's as annoying to be on hold! :P`
`01:38.02`	`[TK]D-Fender`	`iCEBrkr : I think the tech on the other side had it firewalled transparently.....`
`01:38.08`	`*** part/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net)`
`01:38.19`	`Zodiacal-`	`tkd-fender just a thought :P`
`01:39.09`	`[TK]D-Fender`	`Zodiacal- : I jsut helped the other side FIX that problem a day ago...`
`01:39.16`	`Zodiacal-`	`mzo do you think u could spare a min to disable callerid with fxo to see if it speeds up ringing for you? im at a loss...`
`01:39.50`	`mzo`	`i don't have access to it atm. I'll have to wait until later on`
`01:40.17`	`Zodiacal-`	`mzo is your hardware digium?`
`01:40.42`	`iCEBrkr`	`[TK]D-Fender: Ahh`
`01:40.42`	`mzo`	`i have no idea.`
`01:40.50`	`mzo`	`i inherited it.`
`01:42.22`	`Zodiacal-`	`mzo ok np..`
`01:42.36`	`[TK]D-Fender`	`iCEBrkr : I swear I was freaking out.... I've never had a proble until people put up silly firewalls. Esp the ones where the router dishes out "public" (routed) IP's, but you DON'T get to see whats being filtered becuase thats upstream....`
`01:43.04`	`[TK]D-Fender`	`iCEBrkr : So I'm still immortal! Muahahahaha!!!! cough`
`01:45.46`	`rpm`	`how do i make it so when a user hits '*' it asks them for their voicemail password?`
`01:45.50`	`fubster`	`hey rpm`
`01:46.17`	`rpm`	`http://pastebin.ca/38750 is my dialplan, im not sure where to put it in`
`01:46.35`	`iCEBrkr`	`LOL`
`01:47.11`	`[TK]D-Fender`	`rpm : See my example : http://pastebin.com/524998`
`01:47.26`	`*** part/#asterisk mroth_imm[workin (n=mroth@63.65.26.220)`
`01:49.20`	`[TK]D-Fender`	`rpm : You just need an "a" exten in the same context as the one holding the Voicemail call.`
`01:50.02`	`rpm`	`[TK]D-Fender, after or before launching the voicemail?`
`01:50.31`	`[TK]D-Fender`	`rpm : It will jump to "a" if they hit "*"`
`01:50.39`	`rpm`	`ah`
`01:52.21`	`[TK]D-Fender`	`rpm : Doesn't really matter what order it appears in the context.`
`01:53.05`	`tainted-`	`anyone seen dolecmo`
`01:53.25`	`tainted-`	`~seen dolcemo`
`01:53.27`	`jbot`	`i haven't seen 'dolcemo', tainted-`
`01:53.36`	`tainted-`	`~seen dolecmo`
`01:53.38`	`jbot`	`i haven't seen 'dolecmo', tainted-`
`01:54.21`	`*** join/#asterisk copantl_ (n=copantl@205.240.200.107)`
`01:55.05`	`*** part/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net)`
`01:55.50`	`*** join/#asterisk kio (n=kio@ool-4577a5a5.dyn.optonline.net)`
`01:56.08`	`*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)`
`02:00.14`	`Drew_____`	`isnt "#1" a legal extension name?`
`02:00.39`	`SkramX`	`I know "`
`02:00.43`	`SkramX`	`#" is..`
`02:00.58`	`SkramX`	`Never tried "# 1" does it not work, or are you just asking?`
`02:01.06`	`fugitivo`	`~seen docelmo`
`02:01.08`	`jbot`	`docelmo <n=docelmo@55-65.126-70.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 2d 1h 1m 7s ago, saying: '1300 I paid 1800 a year ago for it.'.`
`02:01.11`	`Drew_____`	`#1 doesnt work for me`
`02:01.13`	`denon`	`it'd break if you have Tt`
`02:01.23`	`fugitivo`	`tainted-: that's the right nick :)`
`02:02.13`	`Drew_____`	`i have defined extensions named #1, #2, #3 etc in my dialplan but i get a 404 when trying to dial them`
`02:02.22`	`[TK]D-Fender`	`DTMF "features" = broken idea`
`02:02.36`	`[TK]D-Fender`	`Drew_____ : What is your phone?`
`02:02.42`	`Drew_____`	`Xlite`
`02:03.05`	`Drew_____`	`gxp2k is in the mail... ;)`
`02:03.47`	`Drew_____`	`could be a problem with xlite because there is no console output of the call to #1`
`02:06.59`	`*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)`
`02:07.55`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`02:09.05`	`*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)`
`02:14.19`	`SibrPhrek`	`damnit -i can't get mpg123 to get install on OS X`
`02:14.54`	`*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-253.msy.bellsouth.net)`
`02:15.39`	`mzo`	`is there a tiny memory leak in asterisk on smp boxes? It's slowly using up memory every few minutes?`
`02:16.18`	`*** join/#asterisk justin1 (n=Justin@cpe-72-129-86-208.socal.res.rr.com)`
`02:16.30`	`zigman`	`SibrPhrek what do you need mpg123 for ?`
`02:16.34`	`zigman`	`try format_mp3`
`02:16.42`	`mzo`	`mpg123 is depreciated now right?`
`02:16.47`	`zigman`	`yeah`
`02:16.54`	`mzo`	`yay i was right! :P`
`02:16.58`	`SibrPhrek`	`zigman - i'm trying to get music on hold to work -format_mp3 not working, and i wanna do streaming from itunes`
`02:17.03`	`*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`02:17.36`	`[TK]D-Fender`	`Drew_____ : Make sure your phone's dialplan is not interfereing with *'s one`
`02:18.35`	`mzo`	`itunes streaming sounds complicated :p`
`02:19.44`	`jbalcomb`	`how is my Asterisk server configured to require the single PRI card to be in?`
`02:20.08`	`jbalcomb`	`I took it out and the quad PRI card wouldn't go green`
`02:20.33`	`jbalcomb`	`Asterisk worked fine for internal calls but reported congestion/could find channel zap`
`02:20.49`	`[TK]D-Fender`	`jbalcomb : Timing source <-`
`02:21.09`	`[TK]D-Fender`	`jbalcomb : Make sure you're taking clocking from the PRi on the proper port and passing it on to the others.`
`02:21.16`	`*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)`
`02:22.05`	`*** join/#asterisk ham (i=HamYai@125.24.3.163)`
`02:24.39`	`rpm`	`aww my 'o' extension doesn't work :)`
`02:24.46`	`mzo`	`i need a sex extension`
`02:24.54`	`mzo`	`and have it text-to-speed random porn`
`02:24.56`	`*** join/#asterisk User97 (n=User97@64-42-204-178.mb.skyweb.ca)`
`02:25.23`	`[TK]D-Fender`	`rpm : Did you define it in voicemail.conf?`
`02:25.50`	`[TK]D-Fender`	`\|operator=yes`
`02:25.52`	`rpm`	`define the 'o' extension?`
`02:25.54`	`rpm`	`ooh`
`02:26.01`	`[TK]D-Fender`	`rpm : you need both sides`
`02:26.11`	`*** join/#asterisk benjk (n=benjamin@70.103.248.130)`
`02:26.12`	`rpm`	`yes, operator=yes`
`02:26.23`	`rpm`	`exten => o,1,GoTo(${ALBERTA},1)`
`02:26.26`	`[TK]D-Fender`	`My nex gen of STDEXTEN will integrate that as well.`
`02:26.51`	`[TK]D-Fender`	`rpm : ${ALBERTA} is an exten?`
`02:26.57`	`[av]bani`	`[TK]D-Fender: polycom 601 would look about 6 billion times better with backlight`
`02:27.05`	`rpm`	`ALBERTA=4036681593`
`02:27.06`	`[av]bani`	`the beautiful lcd is almost wasted`
`02:27.13`	`rpm`	`its a variable at the top of my extensions.conf`
`02:27.21`	`[TK]D-Fender`	`[av]bani : Same goes for most phones.....`
`02:27.42`	`[TK]D-Fender`	`[av]bani : have you made the account like I mentioned and extracted the provisioning stuff into it?`
`02:27.52`	`[av]bani`	`?`
`02:27.59`	`[TK]D-Fender`	`[av]bani : I'd pay a few extra $$$ for ti to be backlit...`
`02:28.58`	`Drew_____`	`how about building your own backlight?`
`02:29.05`	`[av]bani`	`[TK]D-Fender: yeah, how much can a backlight possibly cost, $0.50 ?`
`02:29.09`	`[TK]D-Fender`	`http://www.freedomphones.net/polycom/files/SoundPoint_IP_SIP_1_6_2.zip`
`02:29.15`	`Drew_____`	`get a couple of LEDs and gluegun them to the LCD ;-)`
`02:29.43`	`[TK]D-Fender`	`so you can start getting ready to provision it.`
`02:29.49`	`[av]bani`	`[TK]D-Fender: er, the phone already says sip 1.6.2`
`02:30.14`	`[TK]D-Fender`	`[av]bani : its not the firmware file in there thats of interest, its the XML configs!`
`02:30.24`	`[av]bani`	`hmm`
`02:30.26`	`[TK]D-Fender`	`Samples come along-with`
`02:30.33`	`[av]bani`	`i already configed it by hand though :/`
`02:30.36`	`[av]bani`	`what pain lol`
`02:30.45`	`jbalcomb`	`[TK]D-Fender where do I specifiy the timing source and how/where to pass it?`
`02:30.49`	`[TK]D-Fender`	`make an account on your box and get ready to provision it through FTP using that account`
`02:30.50`	`[av]bani`	`polycom are cracksmoking whores, they should make reboot optional`
`02:31.17`	`[av]bani`	`its like 'gee nice phone, but the programmers need to be clubbed over the head'`
`02:31.21`	`[TK]D-Fender`	`[av]bani : I've done nearly 30 of mine and I gave blitzrage a setup that worked "out-of-the-box"`
`02:31.38`	`[av]bani`	`[TK]D-Fender: i'll prolly mangle it into an autoprovisioner like i did for sipura`
`02:31.41`	`[av]bani`	`:)`
`02:31.46`	`[TK]D-Fender`	`AGAIN : Stop using the damn web interface and start doing it the right way!`
`02:32.31`	`[av]bani`	`[TK]D-Fender: can it provision via http?`
`02:32.32`	`[TK]D-Fender`	`The experience is much better when you do it right from the start.`
`02:32.48`	`[TK]D-Fender`	`I believe so..... HTTPS I know, https I'm unsure`
`02:33.37`	`jbalcomb`	`[TK]D-Fender where do I specifiy the timing source and how/where to pass it?`
`02:33.38`	`[TK]D-Fender`	`FTP is your friend.....`
`02:34.03`	`[TK]D-Fender`	`jbalcomb : Double check what port is what not that you pulled acard. make sure things didn't get renumberd behind your back.`
`02:34.15`	`[TK]D-Fender`	`then make sure to take timing from PRI port and pass it to the others`
`02:34.38`	`jbalcomb`	`ok, yeah, i dont know how to do this that your suggesting though`
`02:35.47`	`jbalcomb`	`the timing that i know is just in the /etc/zaptel.conf`
`02:36.04`	`*** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)`
`02:36.04`	`jbalcomb`	`i dont know how to check on port renumbering though`
`02:36.20`	`[TK]D-Fender`	`jbalcomb : PM`
`02:36.24`	`jbalcomb`	`according to what i read the ports on the pri are 1-4 and will always initialize 1-4`
`02:36.29`	`*** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com)`
`02:36.47`	`[TK]D-Fender`	`jbalcomb : unless your now removed single port took #1`
`02:37.18`	`*** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com)`
`02:40.27`	`*** join/#asterisk ivanfm (n=ivanfm@201-1-164-43.dsl.telesp.net.br)`
`02:47.03`	`*** join/#asterisk yyoio (n=admin@68-119-68-89.dhcp.mtgm.al.charter.com)`
`02:48.09`	`*** part/#asterisk yyoio (n=admin@68-119-68-89.dhcp.mtgm.al.charter.com)`
`02:48.31`	`*** join/#asterisk yyoio (n=admin@68-119-68-89.dhcp.mtgm.al.charter.com)`
`02:49.49`	`*** join/#asterisk libila (n=vye@ip68-8-174-154.sd.sd.cox.net)`
`02:50.29`	`libila`	`I'm following a doc that shows how to install asterisk on a FC machine, when doing the cvs checkout it says it's the latest development version. Where can I get the stable version?`
`02:50.31`	`harryvv`	`wow, i was wondering why i was getting jitter break up. asterisk celeron 300a running for a year and lately its been running kinda crappy and noticed though TOP im running multiple instances of asterisk.`
`02:50.44`	`yyoio`	`the default port for sip is 5060. is that correct?`
`02:51.59`	`*** join/#asterisk franck (n=franck@tikiwiki/franck)`
`02:52.14`	`Err`	`$ grep sip /etc/services \| head 1`
`02:52.18`	`Err`	`sip5060/tcp# Session Initiation Protocol`
`02:52.46`	`franck`	`I have issues with RTP codec 112 what is it?`
`02:53.16`	`yyoio`	`is there anyway to change the default port number that sip uses in asterisk?`
`02:53.53`	`yyoio`	`where it says port = 5060. i can just change it to whatever i want?`
`02:54.08`	`iCEBrkr`	`yyoio: Not sure why you'd do that, but sure`
`02:54.21`	`Err`	`as long as the port's free, and if you're !root it must be >=1024`
`02:54.39`	`yyoio`	`ok, thanks`
`02:55.00`	`SibrPhrek`	`anyone here know anything about DarwinPorts?`
`02:56.29`	`*** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net)`
`02:57.03`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com)`
`02:57.28`	`BlueDevi1`	`libila: http://atrpms.net/`
`02:57.44`	`libila`	`BlueDevi1: thnx`
`02:58.54`	`libila`	`BlueDevi1: 1.2.3 tree is stable?`
`02:59.41`	`brockj49464`	`If you are running aah2.3 you might want to look at http://sourceforge.net/forum/message.php?msg_id=3542026 for help in upgrading to a working * 1.2.3`
`03:03.04`	`*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)`
`03:03.12`	`BlueDevi1`	`libila: 1.2.3 is the last stable version....take a look at www.asterisk.org`
`03:05.41`	`*** join/#asterisk jef_ (i=fischer@p54844B22.dip.t-dialin.net)`
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`03:07.06`	`*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)`
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`03:10.40`	`*** join/#asterisk bkw__ (n=brian@70.103.248.130)`
`03:11.09`	`mdave`	`welp, just setup asterisk for the first time, and after a bit of confusion, im rather impressed`
`03:11.20`	`mdave`	`im sure more confusion to come tho :P`
`03:11.44`	`mdave`	`becuase I am wanting to do something its possible no one else has done.. (or if they have done it, then maybe there wont be so much confusion)`
`03:12.02`	`mdave`	`popped in here to see if anyone knows if its been done, or if its possible`
`03:12.42`	`mdave`	`wondering if I should just wax on about my idea or wait to see if anyone is listening`
`03:14.01`	`mdave`	`well i suppose im not interrupting anyone at least.. and at worst im just wasting my time typing`
`03:14.16`	`mdave`	`ok.. you may be aware that some cellphone companies offer 'free incoming calls'`
`03:14.45`	`mdave`	`and surely that many/most/all voip providers offer free outbound calls (to the US at least, often to other countries as well)`
`03:15.15`	`_Sam--`	`who offers free outbound voip to pstn?`
`03:15.17`	`_Sam--`	`sign me up`
`03:15.20`	`mdave`	`so heres my idea - get totally free calls from your cellphone to anywhere in the (US, or whereever your voip provider allows)`
`03:15.27`	`mdave`	`well, not completely free`
`03:15.28`	`mdave`	`but unmeasured`
`03:15.31`	`mdave`	`no 'per minutes'`
`03:15.34`	`mdave`	`all-you-can-eat`
`03:15.35`	`_Sam--`	`no such animal`
`03:15.35`	`jbalcomb`	`iCEBrkr you wanna do some tech support/`
`03:15.40`	`_Sam--`	`all the all you can eat is measured`
`03:15.45`	`_Sam--`	`they have 'softcaps'`
`03:16.00`	`mdave`	`some of them, perhaps.. but the key thing is they dont charge you per-minute`
`03:16.04`	`_Sam--`	`but you could write a call back script that would have astyerisk call you on your cell phone`
`03:16.08`	`_Sam--`	`so all your calls would be incoming`
`03:16.14`	`mdave`	`but to make calls from your cellphone, even 'local' calls`
`03:16.16`	`yyoio`	`mdave, you can buy a unit that hooks up a cell phone into your asterisk box... then use disa to give yourself a second line when it calls your phone`
`03:16.16`	`_Sam--`	`you call asterisk, dial the number you want to call`
`03:16.17`	`mdave`	`they charge you airtime`
`03:16.20`	`iCEBrkr`	`do I?`
`03:16.21`	`iCEBrkr`	`LOL`
`03:16.22`	`mdave`	`exactly`
`03:16.23`	`_Sam--`	`it calls you back, and connects the call`
`03:16.39`	`mdave`	`but.. being a complete asterisk newbie`
`03:16.44`	`_Sam--`	`mdave: most of the voip stuff ultimately is still per minute`
`03:16.50`	`mdave`	`im not entirely sure where to even begin setting up such a thing`
`03:16.50`	`jbalcomb`	`iCEBrkr I got RED on my 'zap show status'`
`03:16.54`	`_Sam--`	`even though they say its unlimited or whatever`
`03:17.10`	`jbalcomb`	`iCEBrkr i could be in some heavy horseshit`
`03:17.11`	`iCEBrkr`	`jbalcomb: you on site?`
`03:17.16`	`jbalcomb`	`yes'm`
`03:17.20`	`mdave`	`ive had a broadvoice account for 6 months.. havent seen a per minute charge yet`
`03:17.35`	`_Sam--`	`they just base their monthly service charge on so many minutes`
`03:17.37`	`iCEBrkr`	`jbalcomb: I'd check the shelf and see what the PRI looks like`
`03:17.42`	`*** join/#asterisk Corydon76-hm (i=one@pcp09181629pcs.nash01.tn.comcast.net)`
`03:17.42`	`jbalcomb`	`iCEBrkr tried taking out single PRI card`
`03:17.46`	`mdave`	`but my object here is not to make tons of calls - its to avoid the airtime charges on the cell`
`03:17.55`	`yyoio`	`its called dock-n-talk cell phone station... store.voxilla.com`
`03:18.17`	`yyoio`	`allows cell phone to be tied into your asterisk box`
`03:18.24`	`jbalcomb`	`iCEBrkr looks fine on the wall`
`03:18.28`	`mdave`	`no, i dont want to use the cellphone to make calls`
`03:18.32`	`mdave`	`I want to be out-and-about,`
`03:18.41`	`mdave`	`call my voip number, have asterisk see the callerid, and not answer`
`03:18.42`	`_Sam--`	`and use asterisk as a call-back server`
`03:18.46`	`mdave`	`and have it call me back`
`03:18.47`	`mdave`	`yes`
`03:18.50`	`iCEBrkr`	`jbalcomb: Lemme go grab a cup of coffee and park my character...`
`03:18.52`	`mdave`	`then give me a dialtone`
`03:19.01`	`mdave`	`and use the 3-way/conference to put me thru`
`03:19.03`	`yyoio`	`that can be done`
`03:19.03`	`_Sam--`	`you are thinking about it wrong, but its doable`
`03:19.06`	`yyoio`	`its called DISA`
`03:19.13`	`jbalcomb`	`iCEBrkr is there something that would require libpri and zaptel?`
`03:19.15`	`_Sam--`	`you call asterisk, you tell it what number you want to call, you hang up....`
`03:19.15`	`iCEBrkr`	`jbalcomb: What'd you muck with? lol`
`03:19.20`	`iCEBrkr`	`jbalcomb: For sure!!!`
`03:19.21`	`_Sam--`	`asterisk calls you back (incoming call)...`
`03:19.23`	`mdave`	`a dock sitting at home wouldnt work, i want to use this while im away`
`03:19.24`	`_Sam--`	`then it calls the other party`
`03:19.26`	`jbalcomb`	`iCEBrkr i mean require them to be recompiled`
`03:19.35`	`iCEBrkr`	`jbalcomb: It's possible.. Depends`
`03:19.38`	`mdave`	`i dont want asterisk to ever answer the call from my cell`
`03:19.44`	`mdave`	`i want it to see the callerid of my cell, then call my cell back`
`03:19.56`	`mdave`	`to allow me to tell it where to cal`
`03:19.57`	`mdave`	`l`
`03:20.01`	`_Sam--`	`possible for sure.`
`03:20.11`	`mdave`	`possible with a secret pin or something`
`03:20.14`	`mdave`	`possiblY`
`03:20.21`	`_Sam--`	`it wouldnt give you a dialtone (you could make it) but it would just wait for digit presses`
`03:20.29`	`mdave`	`but ive no idea where/how to tell asterisk to do that`
`03:20.43`	`_Sam--`	`start learning`
`03:20.46`	`*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)`
`03:20.54`	`*** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net)`
`03:20.57`	`_Sam--`	`aint nobody here gonna do it for ya`
`03:20.58`	`mdave`	`it just took me two days to get it installed and running, where my spa2000 connects to it instead of BV, and asterisks connects to BV`
`03:21.02`	`_Sam--`	`well, maybe someone would`
`03:21.04`	`mdave`	`i wasnt asking for someone to do it`
`03:21.09`	`harryvv`	`i have it setup as a extention It ask for a password then give me a dial tone. my own personall calling card box :)`
`03:21.10`	`mdave`	`just - where do I start?`
`03:21.34`	`_Sam--`	`if it took you two days to setup asterisk and connect to broadvoice...it will take at least 2 months to learn what you need`
`03:21.35`	`harryvv`	`ww.voip-info.org`
`03:21.37`	`_Sam--`	`ok, maybe 2 weeks`
`03:21.39`	`Trazz`	`all of the sudden i can't get my softphone or cisco phone to pass voice back and forth.. it was working and all of the sudden today its broken`
`03:21.41`	`*** join/#asterisk damomurf (n=damomurf@ppp146-73.lns3.adl2.internode.on.net)`
`03:21.44`	`mdave`	`ok, that may be`
`03:21.54`	`Trazz`	`any ideas on how to troubleshoot it`
`03:21.58`	`iCEBrkr`	`jbalcomb: Ok, I'm back`
`03:22.13`	`mdave`	`where do I tell asterisk 'if there is a call from XXX, wait a second, then call back'`
`03:22.20`	`_Sam--`	`extensions.conf`
`03:22.27`	`jbalcomb`	`iCEBrkr pm me you # if you can chat IRL for a few minutes`
`03:22.33`	`iCEBrkr`	`jbalcomb: I typically try to keep all the versions matched up-- until now since Asterisk has been moving a bit faster.`
`03:22.34`	`mdave`	`any particular directive i should be looking at`
`03:22.39`	`mdave`	`or set thereof`
`03:22.56`	`_Sam--`	`gotoif [${callerid}]`
`03:23.03`	`_Sam--`	`ask dr-linux he is a gotoif specialist`
`03:23.26`	`Trazz`	`all of the sudden i can't get my softphone or cisco phone to pass voice back and forth.. it was working and all of the sudden today its broken.. can anyone help?`
`03:23.27`	`mdave`	`alright.. i'll go do some reasing then come back if i have any q's`
`03:23.28`	`*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241)`
`03:23.32`	`mdave`	`reaDing`
`03:23.32`	`*** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net)`
`03:23.33`	`_Sam--`	`you'll be back.`
`03:23.36`	`harryvv`	`sam, I never thought about that.`
`03:24.01`	`harryvv`	`sam, ever do belcore signaling?`
`03:24.31`	`*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)`
`03:24.36`	`_Sam--`	`cant say i have`
`03:24.54`	`harryvv`	`I would rather have all phones ring then just direct a caller to one of two seperate extentions. with bell core I would tell if the call was for me or for some one else.`
`03:25.17`	`_Sam--`	`you can DIAL(SIP/1&SIP/2&SIP/3)`
`03:25.20`	`_Sam--`	`that makes them all ring`
`03:25.26`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`03:25.36`	`harryvv`	`u dont understand`
`03:25.40`	`_Sam--`	`not yet i dont`
`03:25.48`	`_Sam--`	`tell me more and i might`
`03:27.03`	`_Sam--`	`mdave: speading of reading...you'll need to READ the input from your phone (keypresses)`
`03:27.45`	`_Sam--`	`after asterisk calls you on your cell...you are going to need to read from your cell the keypresses of the number you want to dial`
`03:27.53`	`harryvv`	`I dont want to answer the other persons calls. say somone calls, thay listen to a IVR instructing them to press the extention of one of say 3 parties. in this case, instead of running to answer the call it would ring all phones a different bell core signal and then it would revert to that original extention vm if not answered. Bell core is a type if signaling standard that will make a phone ring a different way.`
`03:27.56`	`SibrPhrek`	`can someone help me with my musiconhold conf ?`
`03:28.28`	`SibrPhrek`	`http://pastebin.com/525098 <--my extentions.conf`
`03:28.37`	`[av]bani`	`_Sam--: notice any bugs with the new gxp firmware?`
`03:28.55`	`harryvv`	`so if I setup belcore to ring all phone two rings then a pause..that would be for me. if it ran once then pause, its for the other partie.`
`03:29.04`	`SibrPhrek`	`http://pastebin.com/525101 <--my musiconhold.conf`
`03:29.06`	`harryvv`	`party.`
`03:29.07`	`_Sam--`	`ravenpi: so far 1 complaint from the sales guys...at our shop we have like 10 pphones that all ring at once in a ringall type strategy....`
`03:29.08`	`SibrPhrek`	`what am i doing wrong`
`03:29.14`	`SibrPhrek`	`the phone keeps connected, but i don't hear anything`
`03:29.14`	`_Sam--`	`damn nick completion...`
`03:29.31`	`_Sam--`	`anyway...the sales guys phones all say like 30 missed calls`
`03:29.39`	`_Sam--`	`and there is no way to turn off the call log thinger`
`03:29.46`	`harryvv`	`Sam, thats because no one is answering them.`
`03:29.51`	`harryvv`	`:)`
`03:29.55`	`_Sam--`	`they are being answered`
`03:30.01`	`harryvv`	`ohh really`
`03:30.02`	`_Sam--`	`but 10 phones ring, 1 phone answers`
`03:30.10`	`_Sam--`	`9 phones say missed call`
`03:30.47`	`harryvv`	`I dont know what to say about that. so the actuall phone will say missed call in its display? this has nothing to do with asterisk right?`
`03:31.00`	`_Sam--`	`so instead of seeing a nice date/time display all day....all they see all day is "30 MIssed Calls"`
`03:31.10`	`[av]bani`	`heh`
`03:31.13`	`_Sam--`	`harryvv: not at all (to do with asterisk)...`
`03:31.17`	`_Sam--`	`we are talking about a specific phone`
`03:31.20`	`_Sam--`	`and a new upgrade`
`03:31.25`	`nassy`	`i am looking to propose an asterisk system for one of our smaller offices, but i am not very familiar with asterisk. i am looking to make sure it can match all the (important) features we currently use on our toshiba strata ctx. if it works well and as i become more familiar with it i then would introduce it to the main office. one of the features the employees in the main office like is the ability to press intercom`
`03:31.26`	`[av]bani`	`what do you mean, of course it's asterisk's fault`
`03:32.08`	`_Sam--`	`i am not sure what im going to do yet`
`03:32.11`	`harryvv`	`great...I goto reset my linux box and asterisk does not want to come up..even under safe_asterisk`
`03:32.15`	`_Sam--`	`i may implement a new ring strategy`
`03:32.25`	`_Sam--`	`throw callers into a queue then use rrmemory`
`03:33.03`	`[av]bani`	`well if ou think about it, the phones are doing what theyre supposed to do`
`03:33.11`	`[av]bani`	`but yeah, should be possible to turn off the log`
`03:33.13`	`_Sam--`	`if you want that behavior`
`03:33.26`	`_Sam--`	`we are a call cetner type mail order place`
`03:33.26`	`[av]bani`	`well, how would phones tell missed call/non missed call?`
`03:33.32`	`[av]bani`	`i dont think any phone could do that`
`03:33.38`	`_Sam--`	`they never told missed calls before :)`
`03:33.42`	`_Sam--`	`and i never cared`
`03:33.48`	`[av]bani`	`i mean, how would cisco or polycom do it any differently`
`03:34.04`	`_Sam--`	`maybe they dont show it on the primary display as the default display when you miss one`
`03:34.12`	`[av]bani`	`well i mean, its still a missed call`
`03:34.12`	`_Sam--`	`or maybe they do`
`03:34.21`	`[av]bani`	`so cisco/polycom would show 30 missed also`
`03:34.22`	`_Sam--`	`i wouldnt care if it logged the missed call and every call...`
`03:34.29`	`_Sam--`	`but i just dont need to see it in my face all day long`
`03:34.32`	`[av]bani`	`:)`
`03:34.38`	`_Sam--`	`if i want to see missed calls, i can scroll to the missed calls area`
`03:34.55`	`[av]bani`	`you should edit the gxp page and add a feature request`
`03:34.56`	`_Sam--`	`but i hear what you're saying also`
`03:34.58`	`*** join/#asterisk annonimous (n=annonimo@201.152.124.189)`
`03:35.03`	`annonimous`	`good night`
`03:35.07`	`[av]bani`	`any bugs though? thats just a misfeature :)`
`03:35.29`	`_Sam--`	`i only have 3 out 10 people on it, but so far that was the ONLY comment.`
`03:35.35`	`harryvv`	`Im getting this. I think its the mpg123 problem again? Ouch ... error while writing audio data: : Broken pipe`
`03:35.36`	`_Sam--`	`handled maybe 150 calls on them`
`03:35.49`	`[av]bani`	`the gui is very pretty now, but i would give it up for http/sendtext`
`03:35.55`	`[av]bani`	`xml or whatever`
`03:36.11`	`SibrPhrek`	`help!`
`03:36.12`	`SibrPhrek`	`- Executing SetMusicOnHold("SIP/200-0a4d", "stream") in new stack`
`03:36.12`	`SibrPhrek`	`<PROTECTED>`
`03:36.12`	`SibrPhrek`	`<PROTECTED>`
`03:36.36`	`_Sam--`	`i think this new gui may satisfy some of the gxp haters who said you couldnt configure the phone from a central provisioning server without the web interface`
`03:36.51`	`_Sam--`	`because i think right from the phone gui on the phone...you can enter the tftp parameters`
`03:36.54`	`_Sam--`	`and have it do its thing`
`03:36.57`	`_Sam--`	`i think`
`03:37.15`	`*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)`
`03:37.24`	`_Sam--`	`which people less than one week ago were saying that is a major reason why you shouldnt get a gxp`
`03:37.43`	`*** part/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com)`
`03:38.21`	`annonimous`	`anybody here knows how to setup the 537ep intel modem as a wildcard? =/`
`03:38.38`	`BlueDevi1`	`SibrPhrek: which mpg123 version ?`
`03:38.44`	`*** part/#asterisk CaT[tm] (n=cat@nessie.weebeastie.net)`
`03:38.51`	`SibrPhrek`	`59s`
`03:39.48`	`*** part/#asterisk damomurf (n=damomurf@ppp146-73.lns3.adl2.internode.on.net)`
`03:39.52`	`_Sam--`	`[av]bani: how are you rating the gxp against your new phones?`
`03:39.54`	`SibrPhrek`	`BlueDevi1: 59s`
`03:42.34`	`BlueDevi1`	`SibrPhrek: i don't find the error in your config`
`03:42.48`	`SibrPhrek`	`so then what am i doing wrong?`
`03:43.14`	`Trazz`	`all of the sudden i can't get my softphone or cisco phone to pass voice back and forth.. it was working and all of the sudden today its broken.. can anyone help?`
`03:43.23`	`_Sam--`	`Trazz: check topic`
`03:43.50`	`Trazz`	`i am running 1.2.2 yes`
`03:43.53`	`Trazz`	`so whats up with 1.2.3 ?`
`03:43.58`	`_Sam--`	`1.2.2 is broked`
`03:44.01`	`_Sam--`	`get off it immediately`
`03:44.07`	`_Sam--`	`either 1.2.1 or 1.2.3`
`03:44.43`	`Trazz`	`wow.. is 1.2.3 stable enough ?`
`03:44.53`	`_Sam--`	`dont know, i never moved from 1.2.1`
`03:45.03`	`SibrPhrek`	`yeah 1.2.3 is stable`
`03:45.04`	`SibrPhrek`	`i'm using it now`
`03:45.13`	`SibrPhrek`	`BlueDevi1: so what am i doing wrong`
`03:45.26`	`SibrPhrek`	`BlueDevi1: why do i get that Music class bullshit`
`03:45.39`	`BlueDevi1`	`SibrPhrek: i don't know`
`03:47.00`	`nassy`	`would asterisk benefit from a dual processor (or dual core) for a small office (10 employees) that might grow but slowly.`
`03:48.05`	`harryvv`	`thats to much for 10 employees`
`03:48.24`	`harryvv`	`2 ghz is even fine for 10`
`03:48.37`	`De_Mon`	`heh, 500mhz is fine for 10`
`03:48.43`	`harryvv`	`yea`
`03:48.44`	`harryvv`	`:)`
`03:48.56`	`harryvv`	`but dont go any lower then that.`
`03:49.01`	`SibrPhrek`	`BlueDevi1: new error - WARNING[14386]: res_musiconhold.c:881 local_ast_moh_start: No class: stream`
`03:49.13`	`nassy`	`what about if they do conferencing, etc`
`03:51.04`	`*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)`
`03:51.26`	`BlueDevi1`	`SibrPhrek: which asterisk version 1.0.x or 1.2.0 ?`
`03:51.31`	`SibrPhrek`	`1.2.3`
`03:52.58`	`SibrPhrek`	`BlueDevi1: CLI reports - http://pastebin.com/525123`
`03:53.26`	`nassy`	`im loooking at the following dell server. looks like i have a choice of a couple of linux diastros: SuSE or RedHat. for a company would you suggest either of these or just go with something else. it doesnt much matter to me which because im not that familar with linux. (i do like to upgrade though.)`
`03:53.34`	`nassy`	`http://tinyurl.com/br8vh`
`03:54.28`	`annonimous`	`nassy, i would suggest Redhat AS`
`03:55.23`	`nassy`	`thanks. i see ES but no AS. one sec going to look that up`
`03:55.36`	`annonimous`	`nassy, AS = Advanced Server`
`03:55.46`	`nassy`	`ah ok. thanks`
`03:56.00`	`annonimous`	`nassy, when you buy a server on dell you also can buy in dell the Redhat`
`03:56.12`	`tuxinator_linux`	`dell, ewww`
`03:56.14`	`nassy`	`yeah thats what i would probably do`
`03:56.25`	`nassy`	`we may end up getting hp though`
`03:56.26`	`Nugget`	`linux, ewwww`
`03:56.38`	`nassy`	`our windows consultants like hp so for consistency`
`03:56.46`	`tuxinator_linux`	`Nugget: as apposed to what?`
`03:56.51`	`Nugget`	`anything else.`
`03:56.55`	`nassy`	`anyone here in Pennsylvania`
`03:57.11`	`nassy`	`thats where the office is.`
`03:57.16`	`annonimous`	`nassy, if you want consultant in linux or some other things drop me a line please`
`03:57.28`	`nassy`	`yeah may need someone to help set up.`
`03:57.48`	`nassy`	`met someone here from PA who sounded interested but never gave me his info`
`03:57.50`	`BlueDevi1`	`SibrPhrek: look@http://www.orderlyq.com/asteriskqueues.html`
`03:57.58`	`nassy`	`it would be a paid job`
`03:58.22`	`annonimous`	`ok`
`03:58.32`	`BlueDevi1`	`SibrPhrek: exten => 2000,2,SetMusicOnHold(default) exten => 2000,3,WaitMusicOnHold(20)`
`03:58.55`	`nassy`	`can you msg me you info and rates`
`03:59.43`	`nassy`	`a lot of tutorials seem to use debian`
`03:59.56`	`Trazz`	`any issues with the zap 1.2.2 then ?`
`04:00.05`	`annonimous`	`nassy, k see your private`
`04:00.18`	`Nugget`	`I don't want to see nassy's privates!`
`04:00.41`	`nassy`	`lol`
`04:01.16`	`annonimous`	`Nugget, lol`
`04:01.21`	`annonimous`	`xD`
`04:01.31`	`harryvv`	`Is there is a patch for mpg123 so it does not spawn twice and generate that broken pipe error?`
`04:01.45`	`*** join/#asterisk m_a_g_o (n=maxgluck@adsl-11-58-84.mia.bellsouth.net)`
`04:02.32`	`Trazz`	`whats the most stable * version now ?`
`04:02.48`	`libila`	`I have a digium card with four FXO ports, so would fxoks=1 be changed to fxoks=1-4? or something similiar?`
`04:03.26`	`m_a_g_o`	`hi, I've been having problems with users dialing in a second stage through G.729 using RFC2833. Asterisk seems to recognize inband tones as well as out of band ones. this is pretty serious and I can't get support anywhere. would anyone care to advise please?`
`04:05.41`	`m_a_g_o`	`please?`
`04:06.45`	`SibrPhrek`	`BlueDevi1: i got the single MP3 to work (sounds choppy tho), but i can't get the stream to load`
`04:06.50`	`SibrPhrek`	`i think it could be cuz of the stream itself`
`04:08.06`	`BlueDevi1`	`SibrPhrek: i don't use mp3 streaming`
`04:08.26`	`SibrPhrek`	`maybe i shouldn't either`
`04:08.30`	`SibrPhrek`	`but it would be cool if it worked`
`04:08.41`	`BlueDevi1`	`SibrPhrek: ask google :-)`
`04:08.56`	`*** join/#asterisk brookshire[home] (n=matt@pcp01541028pcs.huntsv01.al.comcast.net)`
`04:09.06`	`*** join/#asterisk zimdog (n=zimdog@c-24-9-24-165.hsd1.co.comcast.net)`
`04:09.08`	`*** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net)`
`04:09.14`	`BlueDevi1`	`good night`
`04:09.46`	`*** join/#asterisk greskdo (n=lesgsod@Toronto-HSE-ppp3746159.sympatico.ca)`
`04:10.07`	`SibrPhrek`	`BlueDevi1: http://mundy.org/blog/index.php?p=92`
`04:10.10`	`SibrPhrek`	`that's what i used`
`04:11.34`	`*** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)`
`04:12.43`	`BlueDevi1`	`SibrPhrek: thats the config for asterisk 1.0.x`
`04:13.03`	`BlueDevi1`	`SibrPhrek: sorry i must go now....CU`
`04:13.07`	`SibrPhrek`	`later`
`04:15.01`	`*** join/#asterisk zu (n=raz@24-pool1.ras14.floca.alerondial.net)`
`04:15.28`	`harryvv`	`is there a patch for asterisk when it displays a "Ouch ... error while writing audio data: : Broken pipe`
`04:15.48`	`libila`	`What does it mean when chanconfig failed? could it be related to the fxoks=1 in my /etc/zaptel.conf? I have a digium card with 4 fxo ports. modprobe wcfxo`
`04:15.50`	`libila`	`ZT_CHANCONFIG failed on channel 1: No such device or address (6)`
`04:15.53`	`libila`	`FATAL: Error running install command for wcfxo`
`04:16.31`	`zu`	`I could probably make one harryvv for you`
`04:17.12`	`harryvv`	`okay i lost a zaptel driver...mmmm`
`04:18.25`	`*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`04:19.54`	`m_a_g_o`	`ok, great support, I'm really amazed how you can leave users unattended, I could really use some support`
`04:20.06`	`m_a_g_o`	`I'm using g.729 and RFC2833`
`04:20.08`	`SibrPhrek`	`argh!`
`04:20.11`	`SibrPhrek`	`now the music starts`
`04:20.12`	`SibrPhrek`	`and stops`
`04:20.14`	`SibrPhrek`	`like within a second`
`04:21.23`	`m_a_g_o`	`and asterisk recognizes Both inband and outband. I have been told that asterisk indeed doesn't stop listening for inband while receiving out of band tones, and I'm being unlucky enough to get tones in G.729`
`04:22.05`	`m_a_g_o`	`SO, is someone kind enough to offer support?`
`04:23.03`	`*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)`
`04:23.57`	`tuxinator_linux`	`Hey, it's Dave, from Utah !`
`04:24.15`	`tuxinator_linux`	`and he's using comcast`
`04:28.13`	`justinu`	`m_a_g_o: you know this isn't a digium sponsored support channel? anyone here is a volunteer`
`04:29.21`	`m_a_g_o`	`justinu: I'm sorry then, but as this channel is published at Digium's site I thought it was`
`04:29.24`	`Nugget`	`"Dave's not here, man!"`
`04:29.31`	`justinu`	`free support for a free product`
`04:29.40`	`fugitivo`	`or paypal for better support`
`04:29.51`	`justinu`	`yeah, many people here take it`
`04:30.24`	`m_a_g_o`	`well, If anyone knows about that problem and can fix it, paypal is all right with me`
`04:30.34`	`harryvv`	`how in the world. my kernel has the models disabled`
`04:30.43`	`fugitivo`	`models?`
`04:30.58`	`harryvv`	`sorry`
`04:30.59`	`harryvv`	`:)`
`04:31.07`	`fugitivo`	`your kernel has models?`
`04:31.08`	`harryvv`	`module`
`04:31.13`	`fugitivo`	`:)`
`04:31.21`	`fugitivo`	`m_a_g_o: what's the problem?`
`04:31.39`	`harryvv`	`I have been having some serios problems with my asterisk sound quality lately`
`04:31.41`	`brookshire[home]`	`asterisk 1.2.3 ?`
`04:31.48`	`justinu`	`m_a_g_o: you're getting RFC2833 digits /and/ inband?`
`04:31.58`	`harryvv`	`now modprobe xcfx0 and zaptel says modules are not installed`
`04:32.03`	`harryvv`	`wcfxo`
`04:32.17`	`fugitivo`	`did you upgrade your kernel?`
`04:32.31`	`harryvv`	`nothing has been touched in the year I have been using this.`
`04:32.41`	`harryvv`	`Its almost run flawless`
`04:32.42`	`zimdog`	`I am having problems with a`
`04:32.43`	`fugitivo`	`really? one year without updates?`
`04:32.47`	`harryvv`	`yea`
`04:32.48`	`harryvv`	`:)`
`04:32.57`	`*** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net)`
`04:32.58`	`*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)`
`04:33.02`	`fugitivo`	`isn't that boring? :)`
`04:33.07`	`harryvv`	`hheh`
`04:33.19`	`harryvv`	`lucky i had a regular pstn phone near by`
`04:33.33`	`Trazz`	`if i have phone unplugged and call that extension it says the person is on the phone.. heheeh .. what can i do to fix this condition?`
`04:33.44`	`harryvv`	`Ive never updated because I did reinstalls`
`04:33.50`	`harryvv`	`but its been so stable.`
`04:34.06`	`*** join/#asterisk greskdo (n=lesgsod@Toronto-HSE-ppp3746159.sympatico.ca)`
`04:38.31`	`m_a_g_o`	`justinu: yes, I believe so after seeing tethereal traces, rtp events (normally 6 per digit) and seeing the duplicate digits in asterisk's console`
`04:39.45`	`justinu`	`m_a_g_o: how is it affecting you?`
`04:40.55`	`zimdog`	`Hello all. I seem to have lost audio on my asterisk box. When I call I can hear the IVR. I can dial an extension and it will ring. WHen I pick it up I hear no audio in etiher direction. I thought I screwed up on of my polycom config files but I just tried it from xten to the polyocm and xten through a sip trunk to pstn and hear nothing either direction. Any suggestions on where to look?`
`04:41.29`	`rob0`	`zimdog: /topic ?`
`04:41.57`	`m_a_g_o`	`well, I have a second stage dial for ANI and PIN users where PIN and dialed numbers are being received with duplicate digits, thus the users can't use the service at all`
`04:43.05`	`*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)`
`04:43.11`	`brookshire[home]`	`mattf!`
`04:43.15`	`zimdog`	`rob0: what does /topic mean? I tried to type that and it said I need to be an operator?`
`04:43.31`	`fdask`	`changes channel topic`
`04:43.37`	`fdask`	`Asterisk 1.2.3 Released`
`04:43.42`	`fdask`	`thats what it says now`
`04:43.57`	`zu`	`yea I was watching as mark fixed the bug at itexpo`
`04:43.58`	`*** join/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net)`
`04:44.02`	`*** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net)`
`04:44.10`	`Uberbot`	`Hi all.`
`04:44.27`	`rob0`	`zimdog: Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update)`
`04:44.36`	`zimdog`	`rob0: SO this is an issue fixed by 1.2.3 ?`
`04:45.14`	`*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)`
`04:45.27`	`brookshire[home]`	`zimdog: sounds like it`
`04:45.50`	`joaovianna`	`Hi guys! If I have one Digium board T1 and I order one T1, how many DID's I can setup in my system ?`
`04:45.52`	`fdask`	`quick test, set the date on your box back to sometime before the 25th if you can`
`04:46.03`	`rob0`	`ah yes`
`04:46.05`	`justinu`	`joaovianna: virtually umlimited`
`04:46.09`	`fdask`	`i believe that bridging issue was all that was fixed in the latest update?`
`04:46.33`	`Uberbot`	`Can anyone tell me why this doesn't work? exten => 555, 2, playback("/directions.gsm")`
`04:46.47`	`Uberbot`	`Jan 26 21:43:24 WARNING[25086]: file.c:820 ast_streamfile: Unable to open "/directions.gsm" (format ulaw): No such file or directory`
`04:46.49`	`joaovianna`	`justinu: Thanks, but my question ? It come with my T1 ? I paid for each one ?`
`04:47.08`	`justinu`	`you usually pay for them in blocks`
`04:47.28`	`rob0`	`Uberbot: there is no file named /directions.gsm ... take out the / ?`
`04:47.57`	`Uberbot`	`I was hoping to store it in /tmp/ Thus it would be /tmp/directions.gsm. That doesn't work, either.`
`04:48.11`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`04:48.15`	`brookshire[home]`	`joaovianna: DIDs are like ips`
`04:48.19`	`joaovianna`	`justinu: Do you recomend one good company to provide me one T1. I have my server on a good co-location. 60 Hudson St.`
`04:48.23`	`zimdog`	`Looking into thanks. That will be weird. Something happened yesterday to break it.`
`04:49.01`	`brookshire[home]`	`zimdog: http://www.asterisk.org/node/55`
`04:49.03`	`justinu`	`joaovianna: sorry, i'm west coast`
`04:49.21`	`joaovianna`	`Thanks justinu:`
`04:49.30`	`Uberbot`	`Jan 26 21:47:23 WARNING[25216]: file.c:820 ast_streamfile: Unable to open "/tmp/directions.gsm" (format ulaw): No such file or directory`
`04:49.32`	`nassy`	`how much does orderlyq cost. anyone know>`
`04:49.38`	`harryvv`	`Still cannot locate module after recompiling..need help on this one.`
`04:49.48`	`joaovianna`	`brookshire[home]: Thanks.`
`04:50.31`	`pauldy`	`wow new version of the GXP2k software is looking good`
`04:51.16`	`Corydon76-home`	`joaovianna: why not just get a link from Internap?`
`04:51.24`	`joaovianna`	`I need a good source of international rates and Did's for incoming calls. Anyone ?`
`04:51.58`	`Corydon76-home`	`joaovianna: telesys.cc`
`04:52.25`	`Corydon76-home`	`joaovianna: they're hosted in the same building`
`04:53.13`	`harryvv`	`anyone here seen a case of recompilling asterisk and zaptel models dont load?`
`04:53.29`	`pauldy`	`harryvv yup`
`04:54.24`	`pauldy`	`in my case everytime it has had something to do with me updating with yum and not rebooting`
`04:54.39`	`zimdog`	`Thanks brookshire`
`04:54.47`	`pauldy`	`new kernels and kernel mods but I'm still running the old kerenl`
`04:54.59`	`joaovianna`	`Corrydon76-home: I'm taking a look in the rates... Hummmm... I'm paying less than that...`
`04:56.25`	`*** join/#asterisk tainted- (n=identd@ppp-71-134-157-119.dsl.irvnca.pacbell.net)`
`04:56.34`	`joaovianna`	`harryvv: You need to recompile zaptel and libpri too...`
`04:57.17`	`harryvv`	`its up and running now. did not need to compile libpri because i dont use PRI`
`04:58.51`	`Trazz`	`if i have phone unplugged and call that extension it says the person is on the phone.. heheeh .. what can i do to fix this condition?`
`04:59.26`	`nassy`	`ask them to hang up. :-) i dunno. im new to asterisk`
`05:00.09`	`*** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)`
`05:04.22`	`Trazz`	`well the user is not on the phone`
`05:05.35`	`*** join/#asterisk robak (n=rrak@asq30.internetdsl.tpnet.pl)`
`05:07.09`	`Corydon76-home`	`Trazz: detect CHANUNAVAIL`
`05:07.32`	`Corydon76-home`	`You're not using jumping, are you?`
`05:08.17`	`Corydon76-home`	`GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?notbusy)`
`05:08.22`	`harryvv`	`asterisk still is spawning two mpg123`
`05:08.37`	`Corydon76-home`	`harryvv: no it's not`
`05:08.38`	`*** part/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net)`
`05:08.44`	`Corydon76-home`	`harryvv: mpg123 forks`
`05:08.53`	`harryvv`	`ohh`
`05:09.29`	`*** join/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il)`
`05:10.01`	`Romik_`	`somebody can advice what this error means?Jan 27 01:07:23 WARNING[1481]: chan_zap.c:4335 __zt_exception: We're Zap/50-1, not -- Executing Cut("SIP/213.161.9.134-082257a0", "arg2=prop\|-\|3") in new stack`
`05:11.52`	`*** join/#asterisk tank10 (n=tank10_c@netblock-72-25-92-150.dslextreme.com)`
`05:11.58`	`harryvv`	`http://pastebin.ca/38762 does this look normal for top and asterisk running?`
`05:12.19`	`tank10`	`can someone help me with broadvoice setup`
`05:12.30`	`tank10`	`does not matter what i do i get this device is not registered`
`05:12.34`	`tank10`	`blah blah`
`05:12.37`	`*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`05:13.04`	`tank10`	`i have checked my * configs about a dozen times lol`
`05:14.07`	`*** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-17-120.houston.res.rr.com)`
`05:14.27`	`*** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)`
`05:15.26`	`fdask`	`harryvv: normal?`
`05:15.46`	`fdask`	`should there be that many instances of asterisk running?`
`05:15.50`	`fdask`	`or are they like child processes`
`05:15.55`	`harryvv`	`possibly`
`05:16.37`	`fdask`	`is this on a test server?`
`05:16.43`	`fdask`	`or are there calls going through right now`
`05:16.53`	`*** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.59.139.Dial1.Chicago1.Level3.net)`
`05:16.55`	`Corydon76-home`	`fdask: those are threads`
`05:17.03`	`fdask`	`ah`
`05:17.15`	`fdask`	`asterisk won't start 2 server instances on the same system will it`
`05:17.22`	`fdask`	`unless you tweak configs i guess`
`05:17.30`	`fdask`	`im still new`
`05:17.36`	`Corydon76-home`	`Not unless you compiled them differently or tweaked the configs`
`05:20.42`	`mattwj2005`	`I am curious....does anyone know if gaim is going to support to sip? voice sip....not just text?`
`05:22.57`	`Gamera`	`mattwj2005: there is a fork of gaim that does`
`05:23.19`	`Gamera`	`http://www.phonegaim.com/`
`05:23.19`	`mattwj2005`	`what is it called?`
`05:24.08`	`iCEBrkr`	`fdask: Why would you want two instances of Asterisk on one box?`
`05:24.24`	`iCEBrkr`	`fdask: and if you >>REALLY<< need to do that, you need to build a Zen box :P`
`05:24.27`	`Gamera`	`iCEBrkr: if you sell hosted pbx`
`05:24.33`	`Gamera`	`but yeah.. i know`
`05:24.37`	`iCEBrkr`	`Zen zen Zen Zen`
`05:24.38`	`fdask`	`iCEBrkr: i dont`
`05:24.45`	`Gamera`	`xen is awesone`
`05:24.47`	`fdask`	`i was just inquiring about the top harryvv posted`
`05:24.47`	`Gamera`	`awesome`
`05:24.51`	`mattwj2005`	`but you can't connect that to an Asterisk server`
`05:24.51`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`05:24.59`	`fdask`	`whats zen?`
`05:25.00`	`fdask`	`xen?`
`05:25.04`	`iCEBrkr`	`Xen or whatever`
`05:25.13`	`Gamera`	`xen is like vmware`
`05:25.14`	`iCEBrkr`	`fdask: It's a 'virtual os' type system`
`05:25.27`	`fdask`	`linuxu distro?`
`05:25.29`	`iCEBrkr`	`fdask: you can load up a bunch of virtual operating systems`
`05:25.30`	`fdask`	`linux`
`05:25.38`	`iCEBrkr`	`It's not a distro`
`05:25.41`	`mattwj2005`	`I was thinking the offical source code......because I know they are merging gaim-vv into the main program`
`05:25.46`	`fdask`	`ok so like vmware`
`05:25.48`	`fdask`	`gotcha`
`05:25.48`	`iCEBrkr`	`well, not a linux distro really.`
`05:25.50`	`iCEBrkr`	`yeah`
`05:25.52`	`iCEBrkr`	`Kinda`
`05:26.13`	`Gamera`	`http://www.xensource.com/`
`05:26.30`	`Gamera`	`you can also run fbsd, netbsd with xen`
`05:26.43`	`iCEBrkr`	`Yeah`
`05:26.51`	`Gamera`	`and the next generation of windows servers will apparently support it`
`05:26.58`	`fdask`	`ah`
`05:26.59`	`iCEBrkr`	`So if you're confused as to what distro you wanna run, load'm all up under Xen :P`
`05:27.14`	`fdask`	`i'm using usermode linux right now for that sorta thing`
`05:27.24`	`fdask`	`but its not quite the same`
`05:28.50`	`litecode`	`hmm, never heard of xen (out of the loop)`
`05:28.58`	`litecode`	`what does it have over uml?`
`05:30.13`	`Gamera`	`#xen ?`
`05:30.15`	`Gamera`	`:D`
`05:31.18`	`fdask`	`so i've got a voicetronix openline4 card`
`05:31.28`	`fdask`	`and most of the asterisk docs say its known to work`
`05:31.32`	`fdask`	`but man what a pita card`
`05:31.36`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`05:31.53`	`fdask`	`there was a bug in the driver module that caused asterisk to die when you bridged calls`
`05:32.42`	`justinu`	`nice`
`05:32.46`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`05:32.48`	`*** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.59.139.Dial1.Chicago1.Level3.net)`
`05:33.02`	`fdask`	`im going to pick up some digium hardware i think`
`05:33.11`	`rene-`	`is anyone familiar with linux vserver? see i need to give asterisk training and i cant afford to have a box for everybody who is attending, i saw somewhere that i could virtualize this one machine i have, i know that analog cards can be shared (how would you share 1port fxo cards anyway) but i would like to know things like how much ram do i need, and if i need multiple nics or i if i can share one nic with multiple IPs (one for every instance running)`
`05:33.12`	`fdask`	`instead of futzing around with this voicetronix`
`05:33.20`	`justinu`	`fdask: sangoma is also nice`
`05:33.43`	`mogorman`	`but digium is so much nicer justinu ^_^`
`05:33.57`	`fdask`	`all im really looking for is something safe`
`05:33.58`	`iCEBrkr`	`mogorman: you're smoking`
`05:33.59`	`justinu`	`they're both excellent soluitions :P`
`05:34.04`	`fdask`	`known to work, with a lot of info out on the web`
`05:34.13`	`mogorman`	`are you saying inot nicce....`
`05:34.17`	`mogorman`	`err not nice`
`05:34.27`	`iCEBrkr`	`known to have a bunch of IRQ problems.`
`05:34.28`	`mogorman`	`see i cant even say it`
`05:34.39`	`iCEBrkr`	`known to not work on a handful of Dell hardware`
`05:34.41`	`fdask`	`im dealing with a fairly easy setup too... not even using voip at this point :V`
`05:34.47`	`iCEBrkr`	`known to have issues with e1000 nics`
`05:34.48`	`mogorman`	`bah`
`05:34.59`	`mogorman`	`you live in the past`
`05:35.06`	`Gamera`	`e1000 as in intel?`
`05:35.15`	`fdask`	`who lives in the past`
`05:35.16`	`*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)`
`05:35.27`	`mogorman`	`at one point there was a driver /firmware issue`
`05:35.31`	`mogorman`	`it didnt last long`
`05:35.40`	`iCEBrkr`	`That's good to hear`
`05:35.41`	`rene-`	`ICE: what does not work on a handful of dell hardware?`
`05:35.55`	`mogorman`	`the tdm400p`
`05:36.03`	`iCEBrkr`	`rene-: The digium website claims their cards don't like Dell products very much.`
`05:36.04`	`rene-`	`i have installed many`
`05:36.17`	`rene-`	`as many people had`
`05:36.20`	`justinu`	`i put a sangoma a101 in a dell sc1425, worked like a charm`
`05:36.23`	`iCEBrkr`	`I personally find it difficult to believe, but I didn't want to risk it.`
`05:36.24`	`rene-`	`but i have had problems`
`05:36.33`	`rene-`	`sometimes an E1 card will just lock the server`
`05:36.52`	`justinu`	`i was working on a job with a te410p`
`05:36.56`	`justinu`	`err`
`05:36.59`	`justinu`	`te210p`
`05:37.04`	`justinu`	`and the machine was locking up`
`05:37.16`	`rene-`	`but digium support help me fix it and it has worked like a champ switching thousands of calle very day`
`05:37.17`	`mogorman`	`why didnt you call me /digium?`
`05:37.19`	`justinu`	`turned out the te210p does not get a long with a certain intel ethernet driver right`
`05:37.34`	`rene-`	`so has anyone virtualized asterisk?`
`05:37.38`	`mogorman`	`yes`
`05:37.42`	`justinu`	`well, my man on the scene was handling the hardware issues, and they got it resolved`
`05:37.45`	`mogorman`	`i have`
`05:37.46`	`justinu`	`so kudos`
`05:37.52`	`mogorman`	`yay!`
`05:37.54`	`justinu`	`te210p is working ok now`
`05:37.59`	`rene-`	`mogorman: had you followed this http://www.telephreak.org/papers/vpa/?`
`05:38.16`	`rene-`	`s /had/have`
`05:38.17`	`mogorman`	`no i just did it`
`05:38.23`	`mogorman`	`for the xen people`
`05:38.25`	`iCEBrkr`	`Tho I have to say that the T100P or whatever the hell it was worked just fine in our old P3 700mhz junker machine..`
`05:38.30`	`rene-`	`is it difficult?`
`05:38.35`	`iCEBrkr`	`I used a Digium card for all our initial testing`
`05:38.40`	`justinu`	`iCEBrkr: single span card?`
`05:38.42`	`mogorman`	`yay!`
`05:38.43`	`iCEBrkr`	`justinu: yea`
`05:38.53`	`justinu`	`yeah, same thing as the a101`
`05:39.17`	`iCEBrkr`	`I was actually a bit surprised there wasn't any issues as the machine was so outdated.`
`05:39.22`	`iCEBrkr`	`But it worked just fine`
`05:39.25`	`Gamera`	`rene-: that's a nice article :)`
`05:39.31`	`iCEBrkr`	`Frankenstein box`
`05:39.45`	`justinu`	`some machines are just lucky`
`05:39.51`	`justinu`	`some are doomed`
`05:39.52`	`iCEBrkr`	`I couldn't complain`
`05:40.00`	`mogorman`	`hmmm`
`05:40.17`	`mogorman`	`well i will leave the sangnoma v digium fight for another night`
`05:40.19`	`mogorman`	`gnite people`
`05:40.21`	`rene-`	`Gamera: Thanks but i cant take credit for that (not in college anymore) hehe`
`05:40.24`	`iCEBrkr`	`mogorman: later`
`05:40.26`	`iCEBrkr`	`:)`
`05:40.33`	`*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`05:40.58`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`05:41.02`	`*** join/#asterisk DyOS (n=me@ip70-176-77-132.ph.ph.cox.net)`
`05:41.33`	`DyOS`	`<PROTECTED>`
`05:41.40`	`justinu`	`mogorman is a character`
`05:43.03`	`iCEBrkr`	`haha`
`05:43.11`	`iCEBrkr`	`DyOS: I think there's a channel for AMP`
`05:43.19`	`Gamera`	`lol`
`05:43.43`	`DyOS`	`thanks icebrkr`
`05:43.44`	`*** join/#asterisk pengyong (n=lala@222.188.134.60)`
`05:44.21`	`justinu`	`~amp`
`05:44.23`	`jbot`	`amp is, like, NOT supported here! people using it should join #amportal`
`05:45.23`	`iCEBrkr`	`Perfect!`
`05:46.50`	`*** join/#asterisk trixter_ (n=trixter@65.172.209.246)`
`05:46.55`	`rene-`	`he`
`05:48.04`	`*** join/#asterisk Netslayer (n=chris@c-24-126-202-231.hsd1.ca.comcast.net)`
`05:48.46`	`Netslayer`	`i'm extremely new to asterisk and VOIP stuff. It looks cool.. any possible uses for a single person apartment :-Pp`
`05:49.17`	`*** join/#asterisk m_a_g_o (n=maxgluck@adsl-11-58-84.mia.bellsouth.net)`
`05:49.19`	`fdask`	`your own answering machine?`
`05:49.26`	`fdask`	`voip gateway?`
`05:49.51`	`watchy`	`i love you`
`05:50.21`	`Netslayer`	`heh`
`05:50.27`	`Netslayer`	`voip gateway sounds so cool`
`05:50.36`	`rene-`	`ex gf / people you owe money/ telemarketers / other random annoying ppl screening-harassing device?`
`05:50.38`	`Netslayer`	`so what i'd have to use a regular phone line to the asterisk server?`
`05:50.43`	`watchy`	`your own sex line?`
`05:50.50`	`rene-`	`hehe`
`05:50.56`	`watchy`	`a cheap $10 card from ebay`
`05:50.59`	`nassy`	`im in that situation Netslayer, my only issue is computer is very noisy`
`05:51.02`	`Netslayer`	`wouldn't i need to sound hot for my own sex line :-P`
`05:51.22`	`Netslayer`	`nassy, i'm planning a huge rackmount setup..noise is only isolated by a closet`
`05:51.34`	`rene-`	`yeah but you can place a bounty for app_turn_my_box_into_something_sexy and some clever hacker will write it for you.. for a fee`
`05:51.43`	`Netslayer`	`are there any providers of voip that integrate with asterisk? ie so i wouldn't need a regular phone line`
`05:51.56`	`watchy`	`yea alot`
`05:51.58`	`Netslayer`	`heh`
`05:51.58`	`rene-`	`there are many, nufone and voicepulse are generally recommended`
`05:53.03`	`nassy`	`hmm, i dont think i can connect a 1U server running asterisk to 4 or 5 POTs lines am i correct?`
`05:53.14`	`Netslayer`	`i can call italy for 5 cents a minute wow`
`05:53.17`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`05:53.19`	`Netslayer`	`like i have anyone to talk to`
`05:54.35`	`*** join/#asterisk L\|NUX (n=linux@202.5.145.58)`
`05:54.37`	`iCEBrkr`	`nassy: Why not?`
`05:55.10`	`nassy`	`i cant find a FXS card with enough ports to fit on digium.org`
`05:55.11`	`rene-`	`nassy: you can`
`05:55.14`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`05:55.26`	`rene-`	`there is a multiport up to 24 ports card`
`05:55.33`	`Netslayer`	`are there any cool voip wifi handsets/headsets`
`05:55.37`	`*** join/#asterisk Ciber (n=Ciber@user-0cdfe0f.cable.mindspring.com)`
`05:55.38`	`nassy`	`sweet`
`05:55.43`	`rene-`	`it has a connector to a breakout box`
`05:55.45`	`nassy`	`let me go look again`
`05:56.05`	`nassy`	`oh ok`
`05:56.05`	`rene-`	`it is called amphenol 25 port connector and you can wire that to that box or to a patch panel`
`05:56.19`	`nassy`	`thanks`
`05:56.26`	`nassy`	`id prefer a patch panel`
`05:57.51`	`*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)`
`05:57.53`	`wasim`	`Netslayer: hitachi`
`05:58.13`	`*** join/#asterisk _-_ (n=nabudoco@206.135.48.98)`
`05:59.20`	`Netslayer`	`wasim, IPC-5000 .. hrm 320 bucks hah`
`05:59.32`	`*** join/#asterisk MGSsancho (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net)`
`06:00.30`	`nassy`	`oh wait, i just read what you wrote carefully. how do i wire it to the 1U server. or if i use a patch panel how do i go from the patch panel to the 1U server`
`06:01.12`	`iCEBrkr`	`nassy: The card will have R11 jacks`
`06:01.14`	`nassy`	`is there a pci card with that connector`
`06:01.18`	`nassy`	`ok`
`06:01.55`	`nassy`	`i must have been looking at the wrong thing`
`06:02.00`	`nassy`	`thanks`
`06:02.57`	`Ciber`	`anyone have a grandstream gxp2000 and upgrade it to the latest firmware?`
`06:03.13`	`nassy`	`i see now. i was looking at just the new products from digium`
`06:03.20`	`*** join/#asterisk Tili (i=Tili@203.101.160.156)`
`06:04.57`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`06:05.04`	`nassy`	`this is really cool`
`06:05.17`	`nassy`	`stuff`
`06:06.21`	`*** join/#asterisk bkw__ (n=brian@72-254-45-30.client.stsn.net)`
`06:06.52`	`*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)`
`06:08.02`	`*** join/#asterisk ThaZZa_Work (n=me@203.80.44.200)`
`06:09.56`	`mzo`	`is there a refernce for adding fax support to asterisk?`
`06:10.27`	`iCEBrkr`	`mzo: Should be some reference you could work from on the wiki`
`06:10.38`	`mzo`	`ooh k, look there`
`06:10.46`	`iCEBrkr`	`I started trying to make it work, but gave up... Don't really need it..`
`06:12.08`	`iCEBrkr`	`mzo: Don't expect to get faxes via VoIP tho`
`06:12.44`	`*** join/#asterisk orehtsae (n=edc@58.20.32.2)`
`06:13.15`	`mzo`	`noo i mean via phone lines`
`06:13.25`	`iCEBrkr`	`Ok`
`06:14.09`	`rene-`	`is asterisk faxing in debian apt-gettable?`
`06:14.19`	`mzo`	`im finding all kinds of useless stuff :P`
`06:14.23`	`iCEBrkr`	`haha`
`06:14.31`	`iCEBrkr`	`mzo: There IS a 'fax' extension`
`06:14.53`	`iCEBrkr`	`mzo: Asterisk is supposed to land in 'fax' if it detects a fax.`
`06:14.57`	`*** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell)`
`06:15.15`	`mzo`	`it didn't answer when the fax called though, that's what i mean. it's automagic?`
`06:15.16`	`iCEBrkr`	`mzo: You'll need spandsp`
`06:15.28`	`rene-`	`yeah but you need to compile spandsp`
`06:15.34`	`iCEBrkr`	`rene-: Poor baby!`
`06:15.52`	`mzo`	`url?`
`06:16.01`	`iCEBrkr`	`google :P`
`06:16.11`	`mzo`	`got busted by the girl have to sleep =(`
`06:16.26`	`rene-`	`im poor now but wait till weekend i will be broke then`
`06:16.34`	`iCEBrkr`	`haha`
`06:16.37`	`*** part/#asterisk orehtsae (n=edc@58.20.32.2)`
`06:16.41`	`Netslayer`	`is there a big wiki somewhere.. i have so many questions`
`06:16.47`	`iCEBrkr`	`~docs`
`06:16.48`	`jbot`	`docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf 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`
`06:18.04`	`rene-`	`me thinks spandsp is available via apt-get`
`06:18.31`	`iCEBrkr`	`apt-cache search spandsp`
`06:18.53`	`Gamera`	`rene-: in unstable i think it is`
`06:19.09`	`rene-`	`asterisk-app-fax is the name in sarge`
`06:19.40`	`rene-`	`i dont know where im picking that package from, i have addes repos from xorcom amp distribution`
`06:19.44`	`rene-`	`added`
`06:20.43`	`*** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it)`
`06:21.11`	`diLLec`	`hey guy's. Is there any timeout on BYE Messages for SIP Phones configureable ?`
`06:21.19`	`*** part/#asterisk FarrisG (n=farris@c-24-1-176-16.hsd1.tx.comcast.net)`
`06:22.27`	`justinu`	`diLLec: depends on the phone`
`06:22.27`	`*** part/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-17-120.houston.res.rr.com)`
`06:22.51`	`diLLec`	`i've got SNOM 190 hardphones`
`06:23.02`	`diLLec`	`and sometimes they not respond on a BYE`
`06:23.05`	`justinu`	`not sure, but the sipuras can do it`
`06:24.05`	`justinu`	`they don't receive the bye?`
`06:24.18`	`justinu`	`asterisk retransmits them 5+ times, i believe`
`06:26.49`	`diLLec`	`it seams that they don't recieve it. showing "sip show channels" the "Last Message" tab lists "BYE"`
`06:27.04`	`diLLec`	`so the phone either don't recieve or it don't accepts it`
`06:27.34`	`justinu`	`if the phones aren't receiving byes, i'd start pinging the phones for packet loss`
`06:28.03`	`justinu`	`if it's not a network issue, you'll need sip debug traces so solve it`
`06:28.08`	`justinu`	`s/so/to/`
`06:29.27`	`iCEBrkr`	`haha`
`06:29.46`	`diLLec`	`i don't think that it is a network problem`
`06:30.21`	`*** join/#asterisk Jun (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net)`
`06:30.22`	`diLLec`	`sadly the problem is very rare and i can't drink enough coffee to constantly look on the asterisk CLI`
`06:30.29`	`justinu`	`log it`
`06:30.34`	`justinu`	`enabled full.log`
`06:30.58`	`nassy`	`if i have two offices in two separate locations and they each have an asterisk box at their locations and they are interconnected. if one office is down (say fire) can the other asterisk box take over the duties of both boxes. if so what kind of lines would that require: VoIP only, or can it be done with POTS and T1 PRI`
`06:31.32`	`justinu`	`you could do it two ways`
`06:31.37`	`iCEBrkr`	`nassy: Sure`
`06:32.16`	`iCEBrkr`	`nassy: Tho, if you're planning on saving $$$ between the offices, I'd suggest a point-to-point data T1.`
`06:32.24`	`nassy`	`are you serious. i didnt really expect it to be possible.`
`06:32.30`	`iCEBrkr`	`That way your VoIP data never leaves the 'network'`
`06:32.51`	`nassy`	`how can it be done with say T1's most of the offices are too far for point to point`
`06:32.57`	`iCEBrkr`	`nassy: Sure. You'd just have to redirect your number from the office that's on fire to the secondary office`
`06:32.59`	`kuku5`	`point to point t1's get expensive`
`06:33.00`	`nassy`	`we have one with point to point`
`06:33.21`	`justinu`	`i suggest doing it with voip and using DNS SRV to setup the failovers`
`06:33.24`	`iCEBrkr`	`kuku5: Yea I know, I guess it'd depend on the amount of interoffice calls you make`
`06:33.48`	`nassy`	`where is that redirect5 done. by me calling the T1 provider, or would the asterisk box need to be operational in the office with the fire`
`06:33.57`	`kuku5`	`there is no failover when your office goes down and you have ptp`
`06:34.03`	`kuku5`	`you need bgp`
`06:34.06`	`kuku5`	`and 2 t1's`
`06:34.18`	`nassy`	`i heard bgprouters are expensive`
`06:34.27`	`nassy`	`im not familar with them though`
`06:34.30`	`*** join/#asterisk tuxinator_linuxM (n=tuxinato@70-32-106-248.ontrca.adelphia.net)`
`06:34.34`	`iCEBrkr`	`nassy: I have an Asterisk box setup here at home that's connected to the Asterisk box at work.. I also have a phone connected to this local asterisk box which will make calls out through the asterisk box at the office.`
`06:34.34`	`justinu`	`you don't need bgp`
`06:34.34`	`kuku5`	`you can set it up with a 2600 series cisco router`
`06:35.07`	`kuku5`	`justinu: youre right - im thinking of a different scenario`
`06:35.11`	`drumkilla`	`i say if your office is burning down, you have bigger problems than getting phone calls :)`
`06:35.19`	`iCEBrkr`	`I never heard of BGP 'enabled' routers being expensive??`
`06:35.29`	`justinu`	`you just need to have your voip provider to SIP routing via DNS SRV lookup`
`06:35.29`	`iCEBrkr`	`lol`
`06:35.30`	`kuku5`	`you cant enable BGP`
`06:36.11`	`justinu`	`s/to/do`
`06:36.45`	`wasim`	`jusdinu?`
`06:36.56`	`justinu`	`you setup your phones to also register via DNS srv`
`06:37.04`	`nassy`	`ok so it soundds like it is easier to do for the VoIP numbers`
`06:37.07`	`justinu`	`if the primary machine goes down, the phones will fail over`
`06:37.23`	`wasim`	`if dhe primary machine goes town, dhe phones will fail over`
`06:37.27`	`justinu`	`the voip provider will also route to a sip uri, and you're set`
`06:37.50`	`nassy`	`but for the numbers associated with the T1 or say POTs it would be more challenging but possible`
`06:37.51`	`wasim`	`dhe voip provider will also route do a sip uri, and you're sed {ed: or awk}`
`06:37.52`	`justinu`	`asterlink is one provider that'll route to a URI`
`06:38.13`	`justinu`	`wasim: you're an odd fellow ;)`
`06:38.14`	`wasim`	`oops, missed a d in roude`
`06:39.09`	`wasim`	`jusdinu: 5 tays of a desd heating for a traw will to dhad do you`
`06:39.23`	`nassy`	`iCEBrkr: are you connected to the office via SIp or IAX?`
`06:39.29`	`iCEBrkr`	`IAX`
`06:39.31`	`iCEBrkr`	`of course`
`06:39.36`	`justinu`	`lol, a 5 day cricket game?`
`06:39.41`	`wasim`	`justinu: oui`
`06:39.45`	`justinu`	`lol`
`06:40.21`	`nassy`	`ok thanks everyone`
`06:45.18`	`*** join/#asterisk ta[i]nted (n=tainted@ppp-71-134-157-119.dsl.irvnca.pacbell.net)`
`06:45.27`	`ta[i]nted`	`~seen docelmo`
`06:45.31`	`jbot`	`docelmo <n=docelmo@55-65.126-70.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 2d 5h 45m 30s ago, saying: '1300 I paid 1800 a year ago for it.'.`
`06:45.38`	`ta[i]nted`	`~seen docelm0`
`06:45.40`	`jbot`	`docelm0 <n=docelmo@66.239.192.34.ptr.us.xo.net> was last seen on IRC in channel #asterisk, 1d 13h 7s ago, saying: 'depends on how you look at it..'.`
`06:46.28`	`*** join/#asterisk gaspiz (n=gaspiz@86.34.6.164)`
`06:47.00`	`gaspiz`	`hi, does asterisk 1.2 suport DTMF inband with g711 (ulaw?)`
`06:47.26`	`wasim`	`gaspiz: g711 is about the only thing that can handle inband, it'd be silly not to`
`06:47.46`	`gaspiz`	`so yes?`
`06:47.50`	`justinu`	`yes`
`06:48.10`	`ta[i]nted`	`anyone know docelmo?`
`06:48.16`	`gaspiz`	`how about: DTMF, RFC2833 Inband, g729a 20ms`
`06:48.28`	`justinu`	`lol, level3, right?`
`06:48.34`	`wasim`	`2833 != inband`
`06:48.38`	`gaspiz`	`got me ...`
`06:49.03`	`justinu`	`that's a level3 interop test`
`06:49.07`	`gaspiz`	`I know but that's what level3 wrote`
`06:49.28`	`gaspiz`	`belive me a lot of headakes`
`06:49.37`	`justinu`	`just ask the interop engineer what he expects to see`
`06:51.49`	`gaspiz`	`let me refrase: DTMF, RFC2833 with g729a 20 ms`
`06:51.50`	`*** join/#asterisk Lord_Drachenblut (n=Lord@12-210-115-191.client.insightBB.com)`
`06:51.55`	`gaspiz`	`should this work?`
`06:51.58`	`wasim`	`yep`
`06:52.22`	`justinu`	`wasim: you in islamabad?`
`06:52.25`	`Beave`	`can you even do inband with g729? I was under the impression that it was only a ulaw/alaw thang.`
`06:52.42`	`justinu`	`Beave: you can, but it's not 100% reliable`
`06:52.54`	`Beave`	`hrmph.`
`06:52.54`	`gaspiz`	`thanks guys`
`06:53.18`	`Beave`	`I found out the hardway that inband with IAX2 (no matter the codec) was possible.`
`06:53.24`	`Beave`	`er.`
`06:53.25`	`Beave`	`wasnt`
`06:53.50`	`Beave`	`till someone hit me with a clue-bat here.`
`06:53.52`	`wasim`	`IAX2 by definition handles DTMF oob`
`06:53.59`	`Beave`	`right.`
`06:54.10`	`wasim`	`but it can be cajoled to not to`
`06:56.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`06:56.13`	`alephcom_`	`Hi everyone.`
`06:56.45`	`alephcom_`	`Does anybody have any comments on chan_woomera? I'm trying to use it but it doesn't want to build and so now I need to figure out if it's worth going to lots of effort or not`
`06:56.57`	`wasim`	`oh hell yeah`
`06:57.10`	`wasim`	`we use woomera for ss7, rocks indeed`
`06:59.56`	`alephcom_`	`Ok, then I'll keep struggling`
`06:59.58`	`alephcom_`	`:-)`
`07:06.26`	`*** join/#asterisk dataworm (n=dataworm@modemcable192.46-130-66.mc.videotron.ca)`
`07:07.09`	`*** join/#asterisk Delvar2 (n=irc@host-83-146-53-34.bulldogdsl.com)`
`07:07.47`	`dataworm`	`I am looking a ISP to hook my Asterisk box to world, I really have no idea witch ISP are good, anyone have a suggestion? I am in Canada btw.`
`07:09.35`	`wasim`	`dataworm: check for latency and throuput`
`07:10.07`	`wasim`	`dataworm: then see who your voip2tdm provider will be (in case you want to call a POTS line in guatemala)`
`07:10.20`	`wasim`	`dataworm: check your latency and throughput to them`
`07:10.33`	`wasim`	`dataworm: anything above 300ms and you'll use over, over`
`07:10.58`	`wasim`	`dataworm: and you need a minimum of 25 kbps per channel (unless IAX2 trunking)`
`07:11.31`	`dataworm`	`Do ISP generally give you a very limitted number of channel like 1?`
`07:12.43`	`dataworm`	`I kind of want to play with the technologies, I have no need for voip except a toys... I want to explore the voip security`
`07:12.49`	`*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)`
`07:13.16`	`dataworm`	`So my requirement are't really about nice voice quality ;)`
`07:13.39`	`wasim`	`dataworm: then its just data, have fun`
`07:16.04`	`alephcom_`	`http://pastebin.ca/38767 Is the output from my chan woomera build. Any comments?`
`07:20.09`	`alephcom_`	`any takers?? would a few $$ help? :-)`
`07:20.22`	`Qwell[laptop]`	`alephcom_, it might`
`07:20.24`	`*** join/#asterisk hd420 (n=hdiwan@c-69-181-3-188.hsd1.ca.comcast.net)`
`07:20.44`	`alephcom_`	`hmmm, how much to offer.....`
`07:21.00`	`alephcom_`	`It seems like $ often helps. :-)`
`07:21.15`	`alephcom_`	`I'll start at $25 USD`
`07:21.41`	`Qwell[laptop]`	`if I knew anything about it, I'd help...`
`07:22.17`	`alephcom_`	`I believe it. I'm ok with asterisk stuff but H323 really gets to me.`
`07:22.34`	`Qwell[laptop]`	`not being able to get dns for google doesn't help either`
`07:22.41`	`alephcom_`	`:-)`
`07:22.50`	`alephcom_`	`that would be a handicap`
`07:24.31`	`bkw__`	`W O O M E R A`
`07:26.19`	`*** join/#asterisk r0d3nt\|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`07:26.42`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`07:27.14`	`harryvv`	`A Pri voip card that cost 10 grand each. Talking to one of the engineers now. http://gl.com/ultrat1.html`
`07:27.44`	`JunK-Y`	`r0d3nt\|m: yo`
`07:30.43`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`07:34.01`	`*** join/#asterisk ChrisDE (n=ChrisDE@80.187.128.244)`
`07:34.18`	`*** join/#asterisk AndrewC (n=Andrew@58.6.118.227)`
`07:34.33`	`*** join/#asterisk EriSan (n=erisan@151.8.109.84)`
`07:34.49`	`ChrisDE`	`hi. I'm having problems with agi:`
`07:34.51`	`ChrisDE`	`Jan 27 07:26:33 WARNING[15829]: res_agi.c:219 launch_script: unable to create fromast pipe: Too many open files`
`07:35.04`	`ChrisDE`	`can anyone tell me whats wrong?`
`07:35.17`	`iCEBrkr`	`Stuck in a loop?`
`07:35.31`	`iCEBrkr`	`ChrisDE: and is that on linux?`
`07:35.37`	`ChrisDE`	`yes`
`07:35.45`	`ChrisDE`	`it worked since yesterday`
`07:35.53`	`ChrisDE`	`everything was fine...`
`07:36.29`	`ChrisDE`	`So I could try restart asterisk.... but don't know if that was the solution....`
`07:36.55`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`07:38.27`	`r0d3nt\|m`	`JunK-Y, hi`
`07:40.14`	`*** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net)`
`07:41.31`	`ChrisDE`	`so restarting asterisk was a solution.. but what does this mean? "too many open files"?`
`07:42.03`	`iDunno`	`means that you've got too many open files :)`
`07:42.16`	`ChrisDE`	`thanks iDunno`
`07:42.26`	`*** join/#asterisk frade (n=frade@ip68-104-188-212.ph.ph.cox.net)`
`07:42.31`	`Netslayer`	`ok so i'm thinking of possible uses for asterisk. voicepulse -> internet -> my asterisk server - > .. now do i use normal phones or are there good priced voip phones?`
`07:42.32`	`iDunno`	`probably means that the kernel has run out of file handles, and you might want to us /proc/sys/fs/filemax (IIRC)`
`07:42.58`	`iDunno`	`<PROTECTED>`
`07:43.17`	`iDunno`	`maybe adding some sysctl foo so that it keeps the setting on reboot ;)`
`07:43.29`	`*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)`
`07:44.11`	`ChrisDE`	`file max contains the value 205856`
`07:44.34`	`ChrisDE`	`is this too less?`
`07:46.20`	`ChrisDE`	`idunno.. will i have to increas that number?`
`07:46.39`	`frade`	`What are your prefs on IP Phones? I have Poly 501, 601 and Grand 2000. I tend to like the Grandstream best so far. Any others I should look at?`
`07:46.42`	`iDunno`	`I don't know what you're using your box for :)`
`07:46.54`	`ChrisDE`	`asterisk`
`07:47.06`	`iDunno`	`nothing else?`
`07:47.11`	`ChrisDE`	`no`
`07:47.25`	`iDunno`	`x`
`07:48.05`	`iDunno`	`noise:~# cat /proc/sys/fs/file-max`
`07:48.05`	`iDunno`	`89369`
`07:48.08`	`*** part/#asterisk hd420 (n=hdiwan@c-69-181-3-188.hsd1.ca.comcast.net)`
`07:48.18`	`iDunno`	`hmm - so you've got more filehandles available than me ;)`
`07:48.26`	`iDunno`	`and mine works :)`
`07:48.33`	`ChrisDE`	`so the problem occurs when trying to start an agi script`
`07:48.56`	`ChrisDE`	`does asterisk open file handles and doesn't give them back?`
`07:49.08`	`iDunno`	`which is where files are going to get used up`
`07:49.13`	`ChrisDE`	`every time an agi script runs?`
`07:49.19`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`07:49.29`	`iDunno`	`hmmm - not sure, you should probably close things, though ;)`
`07:49.41`	`ChrisDE`	`how?`
`07:50.59`	`*** join/#asterisk shanermn (n=etel@m010f36d0.tmodns.net)`
`07:51.44`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`07:52.21`	`alephcom_`	`Well, I got ooh323c working part way. However, I'm getting one way audio. Any suggestions? I could send calls using oh323 just fine but this causes breakage.`
`07:52.23`	`ChrisDE`	`how can I see how many files are actual being used of these 205856?`
`07:52.41`	`*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)`
`07:52.54`	`ChrisDE`	`alephcom... I think you are having some kinda firewall issue`
`07:53.40`	`*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-173.claranet.co.uk)`
`07:54.06`	`ChrisDE`	`use tcpdump and see where packets are being sent to`
`07:54.26`	`alephcom_`	`I'll have a look`
`07:56.10`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`08:01.25`	`*** join/#asterisk lorinc (n=ang@caracas-0901.adsl.interware.hu)`
`08:01.49`	`alephcom_`	`ChrisDE: You're right, thanks.`
`08:01.59`	`ChrisDE`	`youre welcome`
`08:03.35`	`*** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it)`
`08:03.36`	`*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)`
`08:07.45`	`*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)`
`08:19.36`	`libila`	`What would you be able to use ldap for with asterisk? To authenticate a phone?`
`08:20.47`	`*** join/#asterisk Bambr (n=Bambr@213-35-235-26-dsl.end.estpak.ee)`
`08:25.24`	`*** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net)`
`08:30.37`	`*** join/#asterisk knight_ (n=knight@c-24-6-248-254.hsd1.ca.comcast.net)`
`08:31.51`	`knight_`	`I want to have a main asterisk server offsite, and then a local asterisk server to terminate local sip, tdm, etc... What's the best way to trunk the two together over IAX so that all calls are handled by the remote (i.e. the remote makes and receives all the calls, but passes into local)`
`08:31.54`	`knight_`	`?`
`08:32.49`	`ChrisDE`	`whats the problem?`
`08:32.56`	`*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)`
`08:33.53`	`knight_`	`Well, I'm just having a hard time figuring out how to trunk the two together to make it handle as if there was only one`
`08:34.13`	`knight_`	`Like, if I only had the local one, it would be easier to handle the outbound trunks, incoming dids, etc`
`08:34.36`	`knight_`	`but since I'm making the remote one the main server, i'm not sure how to configure my local server to handle the handsets`
`08:34.55`	`knight_`	`i almost want the local asterisk to act as proxy to the other`
`08:35.01`	`ChrisDE`	`if you have two * you have two and not one...`
`08:35.07`	`ChrisDE`	`I don't see any problem`
`08:35.17`	`knight_`	`how do i config?`
`08:35.44`	`knight_`	`local asterisk: iax friend to remote?`
`08:35.45`	`ChrisDE`	`you configure your handsets to connect to your local box and your local box to call your remote box`
`08:35.56`	`knight_`	`local asterisk register to remote?`
`08:36.05`	`ChrisDE`	`right`
`08:36.08`	`knight_`	`then set a dial pattern of _X?`
`08:36.21`	`ChrisDE`	`yes`
`08:36.22`	`knight_`	`to Dial(IAX/remote/${EXTEN}) ?`
`08:36.29`	`ChrisDE`	`whatever you want`
`08:36.36`	`knight_`	`how about inbound from remote?`
`08:36.53`	`ChrisDE`	`set a dial pattern on the remote box`
`08:36.57`	`knight_`	`uhm`
`08:37.11`	`knight_`	`seems like a lot of work`
`08:37.27`	`ChrisDE`	`you also may want to configure that local calls keep local calls`
`08:37.29`	`knight_`	`because i'll need to have it reset the caller id, etc`
`08:38.03`	`knight_`	`i have 10 DIDs coming into the remote box`
`08:38.27`	`ChrisDE`	`you want to suppress the caller id?`
`08:38.28`	`knight_`	`in each context have it Dial(IAX/local/${EXTEN}) again?`
`08:38.30`	`knight_`	`no`
`08:38.43`	`knight_`	`i want the local box to receive all the same params that the remote gets on inbound`
`08:39.01`	`knight_`	`meaning`
`08:39.14`	`ChrisDE`	`the callerid on incoming calls will be the same as on the remote box?`
`08:39.20`	`knight_`	`when the calls come into my local handsets, i dont want to see callerid show up as the remote * itself, but the actual caller`
`08:39.26`	`knight_`	`yep`
`08:39.56`	`ChrisDE`	`yes but the callerid won't change unless you tell it your remote box`
`08:40.15`	`ChrisDE`	`the remote box will pass it through`
`08:40.19`	`knight_`	`hmm`
`08:40.25`	`knight_`	`i havent seen that from practice`
`08:40.46`	`ChrisDE`	`but i have`
`08:40.46`	`knight_`	`getting blank callerid`
`08:43.42`	`libila`	`if I registered two users just like in this tutorial ( http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html ) would I be able to to type their extenstions from within my lan without having a provider?`
`08:43.43`	`*** join/#asterisk franck (n=franck@tikiwiki/franck)`
`08:44.44`	`*** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell)`
`08:44.45`	`dudes`	`you have to create an extension to dial`
`08:45.28`	`libila`	`dudes: Yeah I did, exten => 1234,1,Dial(SIP/user) is one of them.. I think it's not working cuz one of them is configured wrong. (one has a dial tone the other doesn't)`
`08:45.32`	`franck`	`I'm a little bit confused about dundi?`
`08:45.59`	`franck`	`can I set up my asterisk so that I be part of a network of asterisks and people can call me?`
`08:46.50`	`ChrisDE`	`yes franck`
`08:47.17`	`libila`	`host=xxx.xxx.xxx.xxx in sip.conf is the phone or the asterisk box?`
`08:51.13`	`*** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net)`
`08:51.21`	`franck`	`ChrisDE: how does it work? do I need to register a phone number somwhere?`
`08:52.17`	`*** part/#asterisk knight_ (n=knight@c-24-6-248-254.hsd1.ca.comcast.net)`
`08:52.32`	`*** join/#asterisk arkanis (n=test@80-219-10-141.dclient.hispeed.ch)`
`08:52.33`	`arkanis`	`hi`
`08:53.29`	`arkanis`	`how can I improve the call-quality of asterisk?`
`08:53.30`	`dudes`	`libila - it's whoever is connecting ... else, just use Dynamic`
`08:54.12`	`dudes`	`arkanis - that's really upto what the quality is like`
`08:54.56`	`arkanis`	`hm, it cracks`
`08:55.08`	`arkanis`	`sometime I don't hear the first few words`
`08:55.25`	`dudes`	`What version of * and what is your hardware`
`08:56.49`	`*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241)`
`08:56.53`	`arkanis`	`asterisk is 1.0.8`
`08:56.54`	`CMike`	`Hm. no why don't I get a busy signal when I'm executing BUSY on a pri ?`
`08:57.20`	`arkanis`	`how can I display my cpu in linux?`
`08:57.30`	`arkanis`	`RAM is 512`
`08:57.34`	`Qwell[laptop]`	`cat /proc/cpuinfo`
`08:57.35`	`joe`	`cat /proc/cpuinfo`
`08:57.39`	`arkanis`	`thx`
`08:57.52`	`joe`	`Qwell[laptop]: jinx ;P`
`08:58.04`	`arkanis`	`Celeron 2.8 ghz`
`08:58.19`	`dudes`	`SIP/ZAP/IAX?`
`08:58.27`	`arkanis`	`SIP`
`08:58.37`	`dudes`	`And what codec`
`08:58.52`	`arkanis`	`normally g711u`
`08:59.13`	`dudes`	`How much b/w do you have and do you have a decent ping to your provider?`
`08:59.40`	`arkanis`	`b/w should be 2up/2down (mbit)`
`09:00.11`	`arkanis`	`perhaps it is a problem with the clients (x-pro)`
`09:00.42`	`dudes`	`Could be. Softphones sometimes suck`
`09:00.44`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`09:00.51`	`*** part/#asterisk ChrisDE (n=ChrisDE@80.187.128.244)`
`09:01.01`	`dudes`	`Or it could be your provider, too.`
`09:01.08`	`arkanis`	`hm`
`09:01.22`	`arkanis`	`so, on asterisk-side I can't do much?`
`09:01.46`	`dudes`	`Do you have means of testing outside of a softphone?`
`09:01.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`09:02.07`	`arkanis`	`I dont understand`
`09:02.58`	`dudes`	`You could try upgrading to a newer version`
`09:02.59`	`*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)`
`09:03.52`	`arkanis`	`hm`
`09:04.03`	`dudes`	`But it's probably the softphone or the provider ...`
`09:04.44`	`arkanis`	`hm, to softphone has so many things I can adjust...`
`09:04.52`	`arkanis`	`80% of that I don't understand`
`09:05.16`	`dudes`	`Download firefly or another softphone and try using that`
`09:05.35`	`*** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net)`
`09:05.36`	`kippi`	`hey`
`09:06.07`	`kippi`	`How would I go around making a pickup group?`
`09:06.22`	`mzo`	`doesn't the webinterface have all that stuff?`
`09:06.54`	`dudes`	`http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups`
`09:08.45`	`Damin`	`Anyone alive?`
`09:09.15`	`arkanis`	`ah, firefly doesnt suit my needs`
`09:09.17`	`kippi`	`where would i need to put Callgroup=1 etc`
`09:09.26`	`arkanis`	`I have my provider an want to keep it`
`09:12.09`	`arkanis`	`of what use is "jitter buffer" in my softphone?`
`09:13.13`	`dudes`	`arkanis - Try another softphone to see if the problem is there ... I don't care about your "needs." I'm offering a suggestion to aid in resolving your "problem".`
`09:13.30`	`*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)`
`09:14.00`	`*** join/#asterisk infinity2 (n=brendon@solara.netcal.com)`
`09:14.40`	`mzo`	`hahaha.`
`09:14.45`	`mzo`	`sorry, the 'needs' thing made me laugh. :P`
`09:14.53`	`arkanis`	`very funny`
`09:15.17`	`mzo`	`it was, it's like all darth vader ish, 'your lack of listening to me, is disturbing' force choke`
`09:15.24`	`dudes`	`haha`
`09:16.30`	`arkanis`	`You should not take it personally if firefly isn't the right choice ;-)`
`09:16.45`	`arkanis`	`Iam very glad to you try to help me`
`09:16.57`	`arkanis`	`that you`
`09:17.12`	`dudes`	`I don't CARE if you use the damn phone or not ... I was saying to try it and see if it does the same as X-Pro as comparison`
`09:17.13`	`skeffling`	`kippi, Callgroup and pickupgroup are added to sip.conf on a per user basis`
`09:17.57`	`Damin`	`iCEBrkr: You should ahve come to San Francisco due..`
`09:17.59`	`Damin`	`dude..`
`09:18.07`	`iCEBrkr`	`Damin: I wanted to.`
`09:18.14`	`iCEBrkr`	`Damin: Unfortunately, I'm chained to my desk`
`09:19.28`	`Damin`	`Yeah..`
`09:19.31`	`Damin`	`Sucks..`
`09:19.50`	`iCEBrkr`	`Damin: I'm making another attempt at getting Asterisk into production by March 1st`
`09:19.56`	`dudes`	`That reminds me of the goat song when he says he's tied to the back of a pickup truck with a three foot f'n rope`
`09:20.04`	`mzo`	`kinky`
`09:20.19`	`iCEBrkr`	`Our current solution doesn't handle TimeZones -- or at least the manual says it doesn't and our initial contact with tech support says it doesn't either...`
`09:20.23`	`iCEBrkr`	`So.. Here we go again. :(`
`09:20.39`	`iCEBrkr`	`I'm supposed to be the savior since our managment and project managers can fucking do anything right`
`09:20.53`	`dudes`	`Timezone filtering, heh`
`09:26.50`	`*** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net)`
`09:26.53`	`ckruetze`	`Hi`
`09:28.22`	`*** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)`
`09:30.44`	*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
`09:32.34`	`*** join/#asterisk bozo_ (n=bozo@64.238.165.139)`
`09:33.30`	`bozo_`	`hi guys, anyone know how to play a gsm file via a music player or convert gsm to mp3/wav ?`
`09:33.51`	`iCEBrkr`	`bozo_: There's a WinAMP plugin for GSM playback`
`09:34.27`	`*** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54)`
`09:34.49`	`jozsab1`	`Hello evrybody. How can i turn off DTMF tones ?`
`09:34.57`	`iCEBrkr`	`Turn them off??`
`09:34.58`	`iCEBrkr`	`huh?`
`09:35.18`	`jozsab1`	`it is strange to me to but the guy from level3 asked me to do it`
`09:35.37`	`iCEBrkr`	`WHy would you turn them off?`
`09:35.41`	`iCEBrkr`	`What problems are you having?`
`09:35.46`	`jozsab1`	`"Please re-submit the test with G711 codec and DTMF turned off and try sending at least three pages fax."`
`09:37.06`	`dudes`	`If you're sending a fax why would you be sending DTMF?`
`09:37.22`	`jozsab1`	`:). Very good question`
`09:37.27`	`*** part/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il)`
`09:37.31`	`iDunno`	`maybe the guy the other end is being deliberately obtuse? :)`
`09:37.40`	`dudes`	`Unless you are sharing the line and hitting the keypad on your phone ...`
`09:37.43`	`[av]bani`	`i'm guessing some voip equipment might mistake fax tones for dtmf`
`09:37.58`	`[av]bani`	`i think he means turn off rfc2833`
`09:38.29`	`jozsab1`	`so i choose inband ?`
`09:38.32`	`[av]bani`	`eg g711u and inband`
`09:39.16`	`jozsab1`	`i admit i was using g729 codec but turning off dtmf is just too much for me :)`
`09:39.36`	`jozsab1`	`i will try. it will only cost me another day :)`
`09:39.39`	`*** join/#asterisk areski (n=areski@polar.es6.egwn.net)`
`09:39.58`	`[av]bani`	`iCEBrkr: whats your "current solution" ?`
`09:40.13`	`iCEBrkr`	`[av]bani: ?`
`09:40.22`	`[av]bani`	`<iCEBrkr> Our current solution doesn't handle TimeZones`
`09:40.36`	`iCEBrkr`	`[av]bani: Some PoS commerical junkware`
`09:40.41`	`[av]bani`	`voip pbx?`
`09:40.54`	`iCEBrkr`	`[av]bani: naa, it's not a PBX`
`09:41.13`	`*** join/#asterisk fenlander (n=fenlande@82.152.81.57)`
`09:41.19`	`[av]bani`	`boiler room backend`
`09:41.20`	`[av]bani`	`?`
`09:41.36`	`iCEBrkr`	`I'm not even sure why it matters.`
`09:41.40`	`*** join/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it)`
`09:41.58`	`[av]bani`	`just curious what you are supposed to be saving the day for :)`
`09:42.14`	`agx`	`Hello, asterisk 1.2 include support for single port BRI cards (ISDN HFC chipset) ? or i've to use aster.1.0.10+bristuff patches??`
`09:42.15`	`[av]bani`	`what asterisk is supposed to do that current junkware doesnt`
`09:42.27`	`iCEBrkr`	`[av]bani: Our sales people are dense. They're trying to sell a job that we can't do.`
`09:42.46`	`iCEBrkr`	`[av]bani: I can develop for Asterisk, we can't modify our current technology`
`09:42.48`	`[av]bani`	`cant do ever, or cant do right this moment`
`09:43.03`	`iCEBrkr`	`Can't do with our current setup`
`09:43.08`	`[av]bani`	`well thats what sales people are supposed to do`
`09:43.11`	`[av]bani`	`dont you read dilbert`
`09:43.12`	`iCEBrkr`	`lol`
`09:43.21`	`iCEBrkr`	`I shouldn't even care anymore`
`09:43.29`	`iCEBrkr`	`This is the 3rd time they've done this`
`09:43.31`	`bozo_`	`iCEBrkr: sox does the conversion- sox input.gsm -r 8000 -c 1 -w -s ouput.wav`
`09:43.44`	`[av]bani`	`iCEBrkr: kill all humans`
`09:43.44`	`iCEBrkr`	`bozo_: Yea, sox will.`
`09:43.57`	`iCEBrkr`	`bozo_: but there's still a GSM plugin for winamp :P`
`09:44.40`	`iCEBrkr`	`[av]bani: It'd be different if we had a few months.. But I have 'til march 1st`
`09:44.45`	`[av]bani`	`plenty of time!`
`09:44.52`	`iCEBrkr`	`ha`
`09:45.08`	`[av]bani`	`we open source hax0rs are super heroes`
`09:49.35`	`*** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net)`
`09:52.52`	`Feral_Kid`	`Question about dialrules... I have three trunks, one to allow dialing into MX, one to dial into JA, and the last one to handle all other international calls. If I start a call 011 52., how does it know to use the trunk for MX versus it going out on the generic international trunk?`
`09:53.14`	`Feral_Kid`	`Incidentally, using this with *@H`
`09:53.31`	`iCEBrkr`	`_01152XXXXXX`
`09:53.32`	`iCEBrkr`	`or something`
`09:54.36`	`iCEBrkr`	`Assign your span contexts to those trunks`
`09:55.04`	`Feral_Kid`	`iCEBrkr> Ah, the _ is what I am missing... Does that provide priority versus a simple 01152.`
`09:55.17`	`iCEBrkr`	`It provides mattern matching`
`09:55.19`	`iCEBrkr`	`err`
`09:55.21`	`iCEBrkr`	`Pattern matching`
`09:55.39`	`*** join/#asterisk Abbas (i=Abbas@203.81.200.67)`
`09:55.46`	`Feral_Kid`	`Got you... Thanks...`
`09:58.25`	`*** join/#asterisk acehunky (n=chat_jok@221-128-138-157.exatt.net)`
`09:58.39`	`acehunky`	`hello`
`09:58.45`	`acehunky`	`i have this problem`
`09:58.59`	`acehunky`	`with Grandstream BT100 and Asterisk`
`09:59.18`	`acehunky`	`i get this message on asterisk cli (on sip debug) SIP/2.0 403 Forbidden (Bad auth)`
`10:00.22`	`iCEBrkr`	`The phone doesn't register?`
`10:01.22`	`acehunky`	`naa`
`10:01.41`	`acehunky`	`SIP/2.0 403 Forbidden (Bad auth)`
`10:01.41`	`acehunky`	`Via: SIP/2.0/UDP 192.168.40.133:38398;branch=z9hG4bKe57313f99676beec;received=64.110.100.179`
`10:01.42`	`acehunky`	`From: "101" <sip:101@192.168.1.254:5060;user=phone>;tag=f7c50f029a714893`
`10:01.42`	`acehunky`	`To: <sip:101@192.168.1.254:5060;user=phone>;tag=as153280ab`
`10:01.42`	`acehunky`	`Call-ID: b841746abf70e4df@192.168.40.133`
`10:01.42`	`acehunky`	`CSeq: 326 REGISTER`
`10:01.44`	`acehunky`	`User-Agent: Asterisk PBX`
`10:01.46`	`acehunky`	`Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY`
`10:01.48`	`acehunky`	`Max-Forwards: 70`
`10:01.50`	`acehunky`	`Contact: <sip:101@192.168.1.254>`
`10:01.51`	`iCEBrkr`	`~pb`
`10:01.53`	`jbot`	`hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`10:01.54`	`acehunky`	`Content-Length: 0`
`10:01.54`	`acehunky`	`these are the register requests`
`10:01.55`	`iCEBrkr`	`OMG`
`10:01.57`	`iCEBrkr`	`STop`
`10:02.01`	`acehunky`	`oops sorry`
`10:02.07`	`iCEBrkr`	`sip show peers`
`10:02.12`	`iCEBrkr`	`Does it show up in there?`
`10:02.15`	`acehunky`	`nopes`
`10:02.34`	`iCEBrkr`	`you created an entry for it in sip.conf right?`
`10:05.40`	`*** join/#asterisk TallAndy (i=TallAndy@83.104.196.72)`
`10:06.26`	`*** join/#asterisk fulgas (n=fulgas@209.8.233.208)`
`10:06.48`	`*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)`
`10:06.53`	`agx`	`Q: asterisk 1.2 include support for single port BRI cards (ISDN HFC chipset) ? or i've to use aster.1.0.10+bristuff patches??`
`10:10.11`	`*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)`
`10:11.55`	`*** join/#asterisk Peste (i=Peste@195.230.162.134)`
`10:12.00`	`Peste`	`hello again :D`
`10:12.07`	`Peste`	`anybody here?`
`10:12.25`	`iCEBrkr`	`nope`
`10:12.33`	`Peste`	`^^`
`10:12.43`	`Peste`	`i got a question..`
`10:13.27`	`af_`	`what difference there is between host=dynamic and ip in sip.conf?`
`10:13.31`	`fulgas`	`anyone ever connected * with a alcatel 4400 thru a pra2 card ?`
`10:13.43`	`fulgas`	`and a 405p`
`10:13.48`	`*** join/#asterisk TonyM_ (n=TonyM@adsl-solo-80-168-227-12.claranet.co.uk)`
`10:13.48`	`af_`	`may be the cauese of gxp2000 randomically hang?`
`10:14.02`	`Peste`	`if i do a 'pri show span 1' i got status: Provisioned, Down, Active - what does this down mean... because i cannot phone..`
`10:14.22`	`iCEBrkr`	`Peste: Um, common sense 101 tells me the PRI is down.`
`10:15.38`	`Peste`	`yeah thougt so, but i dont know why :(`
`10:15.48`	`Peste`	`down, but active ^^`
`10:15.55`	`TallAndy`	`Hi, does anyone have any experience using phpagi with Asterisk?`
`10:15.58`	`acehunky`	`oops sorry iCEBrkr i was on the box ..`
`10:16.06`	`acehunky`	`yeah i have the entry in sip.conf`
`10:16.12`	`iCEBrkr`	`TallAndy: sure`
`10:16.22`	`acehunky`	`let me check with pastebin to paste in the entries instead of flooding here`
`10:16.40`	`TallAndy`	`Excellent I like it just working on a new project with it.`
`10:16.43`	`iCEBrkr`	`acehunky: You did a 'sip reload'`
`10:16.54`	`TallAndy`	`Using the asterisk manager phpagi class`
`10:17.01`	`acehunky`	`yes, i killed asterisk and restarted as well`
`10:17.04`	`iCEBrkr`	`ok`
`10:17.14`	`TallAndy`	`So far to invoke calls from say a phpscript`
`10:17.15`	`iCEBrkr`	`acehunky: Then I'm going to assume there's something misconfigured in the phone`
`10:17.33`	`acehunky`	`http://pastebin.com/525378`
`10:18.03`	`TallAndy`	`Making the calls using the 'originate' command works great, and sending DTMF using the SendDTMF application command.`
`10:18.25`	`iCEBrkr`	`acehunky: you're trying to hard...`
`10:18.30`	`TallAndy`	`My problem is recieveing tones back from the dialed handset`
`10:18.36`	`acehunky`	`http://pastebin.com/525379 --> includes sip.conf entry as well`
`10:18.40`	`iCEBrkr`	`acehunky: confirm Username/secret match up in the the BT100 and sip.conf`
`10:19.04`	`acehunky`	`yes .. the user and password are exact ..`
`10:19.13`	`acehunky`	`but one more interesting thing that i saw is`
`10:19.42`	`agx`	`Q: asterisk 1.2 include support for single port BRI cards (ISDN HFC chipset) ? or i've to use aster.1.0.10+bristuff patches??`
`10:19.45`	`*** join/#asterisk doszy (n=adosztal@border.albacomp.hu)`
`10:19.52`	`acehunky`	`when i use Xlite .. when i put Domain/Realm: asterisk or like iCEBrkr the phone gets registered otherwise there also i get 403 Bad Auth`
`10:20.37`	`iCEBrkr`	`In your BT100, you've set SIP UserID: 101 and Authenticate ID: 101?`
`10:20.52`	`iCEBrkr`	`TallAndy: eh?`
`10:21.31`	`TallAndy`	`The project i'm working needs a phpscript to dial a box that will recieve DTMF`
`10:21.42`	`iCEBrkr`	`agi_read()`
`10:21.48`	`TallAndy`	`And then the dialed box sends DTMF back`
`10:22.04`	`TallAndy`	`Yeah, can that be done from the Asterisk manager class?`
`10:22.16`	`iCEBrkr`	`you mean from phpagi?`
`10:22.24`	`iCEBrkr`	`$agi->agi_read();`
`10:22.25`	`iCEBrkr`	`yes`
`10:22.40`	`acehunky`	`iCEBrkr yes thats right, Userid: 101, Authenticate ID: 101, Password as mentioned in SIP.conf entry`
`10:24.01`	`TallAndy`	`My scripts are initiated from Apache, rather than /var/lib/asterisk/agi-bin/`
`10:24.06`	`acehunky`	`iCEBrkr: the deal is. I have my asterisk box behind 1-to-1 NAT and my BT Phone is behind double NAT`
`10:24.08`	`iCEBrkr`	`acehunky: hrrm, I don't have a username= in mine`
`10:24.22`	`iCEBrkr`	`TallAndy: That's not how phpagi works.`
`10:24.53`	`TallAndy`	`phpagi.php - that class is for the /agi-bin`
`10:24.53`	`iCEBrkr`	`Hrrm, tho, it might work that way. Never tried it`
`10:25.03`	`TallAndy`	`where as phpagi-asmanager.php works from apache`
`10:25.04`	`acehunky`	`oops i mean Sip UserID: 101`
`10:25.09`	`agallo`	`Q: asterisk 1.2 include support for single port BRI cards (ISDN HFC chipset) ? or i've to use aster.1.0.10+bristuff patches??`
`10:25.11`	`iCEBrkr`	`I've only used phpagi from inside Asterisk AGI() call`
`10:25.34`	`iCEBrkr`	`TallAndy: Ahh, don't know anything about -asmanager.php`
`10:25.44`	`TallAndy`	`Sure I see what your saying`
`10:25.53`	`TallAndy`	`Theres hardly any examples of using it from apache :)`
`10:37.39`	`*** part/#asterisk agallo (n=agx@ip-37-53.sn1.eutelia.it)`
`10:41.11`	`*** join/#asterisk serg_b (n=sergey@9i.ru)`
`10:42.19`	`serg_b`	`does anyone seen JerJer here ?`
`10:42.33`	`iCEBrkr`	`engrish?`
`10:42.47`	`fugitivo`	`morning`
`10:42.58`	`fugitivo`	`argh, it's 7am`
`10:43.15`	`iCEBrkr`	`Fri Jan 27 05:42:39 EST 2006`
`10:43.30`	`TallAndy`	`engrish is badly translated english :P`
`10:43.44`	`fugitivo`	`iCEBrkr: am?`
`10:43.57`	`iCEBrkr`	`Yeah`
`10:44.01`	`fugitivo`	`:/`
`10:44.04`	`*** join/#asterisk BugKham (n=lamer@gb.ja.95.227.revip.asianet.co.th)`
`10:44.21`	`iCEBrkr`	`I could also mention that I haven't been to bed yet`
`10:45.20`	`BugKham`	`I need to connect my asterisk box to an D41E card, I will need an FXS, right?`
`10:45.33`	`BugKham`	`or an FXO`
`10:46.21`	`fugitivo`	`iCEBrkr: ok, now it has a sense`
`10:47.01`	`iCEBrkr`	`I'm just gonna pull an allnighter`
`10:47.08`	`iCEBrkr`	`I'm still debating if I'm actually gonna go into the office`
`10:47.51`	`kippi`	`has anyone installed a NTP server on there redhat box so that their phones can get the correct date and time?`
`10:48.07`	`iCEBrkr`	`ntpdate`
`10:48.16`	`iCEBrkr`	`err`
`10:48.17`	`iCEBrkr`	`ntpd`
`10:48.47`	`Err`	`I've used nptd and openntpd, and they both work just fine`
`10:49.17`	`*** join/#asterisk sherbang (n=sherbang@69.182.224.2)`
`10:49.38`	`dudes`	`you can set a time server on some ATA's, too.`
`10:49.51`	`dudes`	`If you're using one anyway`
`10:49.53`	`Skumling`	`damnit, I just don't understand anything of asterisk...`
`10:50.08`	`chapeaurouge`	`any pb with asterisk on 64bits servers? (Pentium Intel Dual Core)`
`10:50.14`	`iCEBrkr`	`Skumling: Neither do we, we just guess really good`
`10:50.36`	`dudes`	`We have a magic * 8-ball and we just ask and shake and sure as it rains`
`10:51.33`	`mut`	`mornin all`
`10:52.01`	`Skumling`	`iCEBrkr: I can't have my asterisk "put" incoming SIP calls into the right context`
`10:52.08`	`mut`	`anyone know what'de cause audio coming from the pstn side to crackle a lot`
`10:52.19`	`mut`	`er going to the pstn side`
`10:52.29`	`mut`	`but audio going to the sip side sounds perfect`
`10:52.40`	`dudes`	`Skumling - They go into the context in globals. Then you tell them where to go from there (at least that's what I do)`
`10:52.43`	`Skumling`	`iCEBrkr: I've got two accounts at the same VoIP-provider, and I have configured them as peers in sip.conf and made properly register lines`
`10:52.55`	`mut`	`like i'm on my sip phone calling the zoo, i can hear them fine but all they hear is me breaking up really bad`
`10:53.13`	`mut`	`it's started happening when i switched from my cisco as5350 for trunking to a te405p`
`10:53.15`	`iCEBrkr`	`Skumling: what dudes said`
`10:53.29`	`Skumling`	`dudes: how do I tell it where to go from there?`
`10:53.42`	`iCEBrkr`	`Skumling: Based on the extension`
`10:53.42`	`dudes`	`by the user:pass@host/DID`
`10:53.47`	`dudes`	`note the "DID"`
`10:54.27`	`iCEBrkr`	`Like.. my FWD is 47191`
`10:54.35`	`iCEBrkr`	`So I have an extension 47191`
`10:54.38`	`mut`	`i'm also getting double rings 45% of the time when i dial from my voip to the pstn`
`10:54.42`	`mut`	`if that helps any?`
`10:55.07`	`iCEBrkr`	`mut: you have Ringing used in your dialplan?`
`10:55.13`	`mut`	`nope`
`10:55.41`	`Skumling`	`iCEBrkr: I've put myt sip.conf at http://pastebin.com/525421`
`10:55.52`	`dudes`	`you have a t1 and you use your SIP phone to dial?`
`10:56.10`	`mut`	`yes`
`10:56.32`	`Skumling`	`the problem is, that calls to bothn incoming numbers is landing in the incoming-klein context`
`10:56.49`	`iCEBrkr`	`Skumling: ok, so you should have exten => 36930822,1,Dial(SIP/1000) or whatever in your [default] context`
`10:56.52`	`dudes`	`Skumling - you make an extension in default according to that`
`10:56.53`	`*** join/#asterisk jabuka (n=edumatao@200.205.205.254)`
`10:57.05`	`mut`	`this all happened right after i moved to a te405p instead of my cisco`
`10:57.16`	`mut`	`before with the cisco all sip calls were passed via sip to the cisco then out the pri`
`10:57.19`	`iCEBrkr`	`err I guess it'd be Dial(SIP/klein-telsome01)`
`10:57.31`	`jabuka`	`hello. The capture call works with IAX ??`
`10:57.34`	`Skumling`	`dudes: okay... so there isn't a way to have the call placed directly into the "correct" context from the start?`
`10:57.56`	`dudes`	`Yea,`
`10:58.22`	`iCEBrkr`	`I suppose you could try /36930822@context`
`10:58.24`	`dudes`	`36930822,1,Goto(mycontext,${EXTEN},1)`
`10:58.28`	`iCEBrkr`	`lol`
`10:58.37`	`dudes`	`I don't know if the @ works`
`10:58.41`	`iCEBrkr`	`yeah, me either`
`10:58.52`	`Skumling`	`I hoped that the SIP-module was able to match the incoming number and send the call to either the incoming-klein or incoming-wmc context from the very beginning...`
`10:59.17`	`iCEBrkr`	`Skumling: It's really not important.`
`10:59.22`	`dudes`	`Just use the Goto... It really isn't that big of a deal.`
`10:59.27`	`Skumling`	`iCEBrkr: it just feels sooooo wrong ;)`
`10:59.37`	`iCEBrkr`	`Skumling: You could be over-organized :P`
`10:59.41`	`*** join/#asterisk cpm (n=Chip@border0.avitecture.net)`
`10:59.46`	`dudes`	`mod chan_sip to do that then.`
`11:00.21`	`dudes`	`If it doesn't. You could always look at the source and see where it does the register stuff too and see how it does it.`
`11:01.04`	`Skumling`	`iCEBrkr: heh :) what also seems funny is that whichever number I dial (36930916 or 36930822), the Asterisk console always reports SIP/36930822 - actually it reports the number last put into the config-file...`
`11:01.21`	`jabuka`	`transference call works with iax????`
`11:02.25`	`dudes`	`Skumling - do a sip debug and call in. It'll say something like, Found Peer (YADDA).`
`11:02.44`	`iCEBrkr`	`Need...Redbull...`
`11:02.54`	`iCEBrkr`	`Slowing....down...`
`11:02.55`	`Skumling`	`dudes: sip debug peer or sip debug ip?`
`11:03.05`	`jabuka`	`OUW, V�O SE FUDER!!`
`11:03.19`	`dudes`	`just "sip debug"`
`11:03.37`	`dudes`	`or sip debug peer wmc-telsome01`
`11:03.42`	`*** part/#asterisk jabuka (n=edumatao@200.205.205.254)`
`11:03.50`	`dudes`	`It's all upto you ;\|`
`11:04.19`	`mut`	`..`
`11:04.20`	`dudes`	`OUW, VÃO SE FUDER!! <---- what is that`
`11:04.23`	`Skumling`	`dudes: :)`
`11:04.32`	`iCEBrkr`	`dudes: j00 r teh suq`
`11:04.36`	`iCEBrkr`	`I think that's what it says :P`
`11:04.55`	`dudes`	`I don't know what that means`
`11:04.59`	`iCEBrkr`	`lol`
`11:05.23`	`enemy^x`	`I get "Ouch ... error while writing audio data: : Broken pipe" running 1.2.3 while moh... anyone else?`
`11:05.23`	`ivanfm`	`he said : "fuck you guys"`
`11:05.48`	`iCEBrkr`	`See. I was close`
`11:06.22`	`*** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net)`
`11:09.15`	`Skumling`	`Call to 36930916: http://pastebin.com/525433 - call to 36930822: http://pastebin.com/525434`
`11:10.28`	`Skumling`	`both are hitting the klein-telsome01 peer`
`11:10.38`	`*** join/#asterisk fourcheeze (n=rich@82.153.215.21)`
`11:11.53`	`fourcheeze`	`can anyone explain why I would be getting 'SIP response 489 "Bad Event"' back from some clients?`
`11:15.09`	`TallAndy`	`iCEBrkr: Do you think a call initiated from php-asmanager.php use extensions.conf? If so I could then write an agi script to be executed in there`
`11:21.18`	`mut`	`so no one has any idea on my te405 problem?`
`11:22.04`	`dudes`	`you have a PRI (T1) and you're using a SIP phone which calls out via the PRI?`
`11:22.08`	`mut`	`yea`
`11:23.43`	`dudes`	`So is it only when you call the zoo?`
`11:23.48`	`mut`	`no`
`11:23.53`	`mut`	`anything`
`11:24.00`	`mut`	`and it doesn't happen all the time`
`11:24.16`	`dudes`	`So it doesn't always happen`
`11:24.31`	`mut`	`depends`
`11:24.55`	`mut`	`some phones happen more than others`
`11:25.08`	`mut`	`like my bosses at home, he gets the double ring thing 90% of hte time`
`11:25.14`	`*** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net)`
`11:25.40`	`mut`	`like 709% here in the office`
`11:25.43`	`mut`	`70%`
`11:26.03`	`mut`	`and it's always crystal clear sound on the sip side listening`
`11:26.13`	`mut`	`but breaks up on the pstn side`
`11:26.15`	`Inkubot`	`good morning folks`
`11:26.24`	`mut`	`(if it does it)`
`11:26.59`	`dudes`	`What version of *?`
`11:27.22`	`mut`	`svn of 2 days ago`
`11:27.33`	`mut`	`er no`
`11:27.35`	`mut`	`1.2.1`
`11:27.40`	`dudes`	`And when did this start happening?`
`11:28.22`	`dudes`	`Or what happens when someone call in to the system? Into a menu /w Background or something?`
`11:28.39`	`mut`	`when i switched from using a cisco for my t1 gateway to using the te405p`
`11:29.13`	`mut`	`if i call into my voicemail or something`
`11:29.14`	`mut`	`sounds fine`
`11:29.20`	`mut`	`i still get double ring`
`11:29.24`	`mut`	`but i havn't had it breakup`
`11:31.05`	`Peste`	`hello! what can i do, if my TE110P (asterisk) and my AG4000 (voice portal) connected with a E1 (using PRI) cable are out of sync? what could be the problem?`
`11:31.37`	`dudes`	`I've never had that issue before. Could be a lot of things though. Sounds kind of like a NAT issue.`
`11:31.49`	`mut`	`no nat involved`
`11:32.56`	`dudes`	`Goodluck, I'm out.`
`11:33.17`	`Peste`	`can somebody help me with this?`
`11:34.10`	`dudes`	`It's probably a bad config, bad signalling, (wrong cable), not setup for E1. Hell a lot of shit`
`11:34.24`	`dudes`	`err it's/it could be ...`
`11:34.36`	`*** join/#asterisk dippo (n=cwage@quietlife.net)`
`11:36.07`	`Peste`	`i think its a bad config, but i dont know which files are involved`
`11:36.29`	`dudes`	`Then read the wiki`
`11:36.31`	`Peste`	`maybe somethink with the clocking?`
`11:36.44`	`*** join/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru)`
`11:36.44`	`Peste`	`wiki doesn't help ;_;`
`11:37.07`	`dudes`	`The wiki can answer any question --- damnit`
`11:37.08`	`dudes`	`heh`
`11:38.07`	`Peste`	`...`
`11:41.26`	`*** join/#asterisk razu (n=razu@tln-kontor.norby.ee)`
`11:44.41`	`*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)`
`11:52.03`	`doszy`	`hi. I have an Eicon Diva ISDN PCI adapter with an Asterisk@home box. When I try to set the card with the "divactrl load" command I get a "A: can't get card type for DIVA adapter number 1" response. Could you help me find a solution for this problem?`
`11:55.01`	`*** join/#asterisk kink0 (n=kinko@pluton.interec.com)`
`11:55.22`	`kink0`	`I still getting this error: Ext: 1 Cause: Unknown (100), class = Protocol Error (6)`
`11:55.28`	`kink0`	`all help will be welcome !!`
`11:55.50`	`kink0`	`this is when I connected Digium TE405 to 2N Pri Gateway`
`11:57.28`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`11:58.16`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`12:02.34`	`*** join/#asterisk jief (n=jief@184.216-78-194.adsl-fix.skynet.be)`
`12:04.12`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`12:05.14`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`12:09.30`	`puzzled`	`morning all`
`12:10.02`	`kink0`	`hi`
`12:13.59`	`X-Rob`	`wibble.`
`12:16.56`	`*** join/#asterisk coppice (n=chatzill@223.143.17.210.dyn.pacific.net.hk)`
`12:17.59`	`*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)`
`12:19.34`	`kink0`	`I still getting this error: Ext: 1 Cause: Unknown (100), class = Protocol Error (6)`
`12:22.14`	`doszy`	`still have the same problem:`
`12:22.15`	`doszy`	`I have an Eicon Diva ISDN PCI adapter with an Asterisk@home box. When I try to set the card with the "divactrl load" command I get a "A: can't get card type for DIVA adapter number 1" response. Could you help me find a solution for this problem?`
`12:24.30`	`doszy`	`the card is listed when I execute "lspci"`
`12:24.50`	`*** join/#asterisk zotz (n=zotz@24.231.47.175)`
`12:27.04`	`Falle`	`Who here is trying out the new GXP2000 firmware?`
`12:28.06`	`CMike`	`anybpdu now why applicaion BUSY don't generat busy on a PRI ?`
`12:28.12`	`CMike`	`spell`
`12:28.15`	`*** join/#asterisk pengyong (n=lala@222.185.18.133)`
`12:28.44`	`CMike`	`very strange.. I send a BUSY .. but the PRI wont indicate busy (on incoming)`
`12:28.53`	`CMike`	`anybody seen that before ?`
`12:29.10`	`*** join/#asterisk jontow (i=jontow@bsd.adminforrent.com)`
`12:29.18`	`*** join/#asterisk spunz (n=spunz@h081217096096.dyn.cm.kabsi.at)`
`12:35.36`	`kink0`	`q.931 is exclusivelly for E1, right ?`
`12:35.51`	`coppice`	`wrong`
`12:36.46`	`kink0`	`coppice: I have a trouble while sending q.931 SETUP from a Digium TE405 to my 2N Stargate PRI port`
`12:37.34`	`kink0`	`appears some variable is wrong in the dialog between Digium and 2N PRI's`
`12:38.02`	`kink0`	`<PROTECTED>`
`12:38.02`	`kink0`	`<PROTECTED>`
`12:38.02`	`kink0`	`<PROTECTED>`
`12:38.12`	`coppice`	`kink0: if you get protocol errors, you probably have the wrong switchtype selected`
`12:38.56`	`kink0`	`coppice: I select as documentation and 2N support told me, euroisdn, and I am able to send calls from 2N to Asterisk, but not from Asterisk to 2N`
`12:40.07`	`*** join/#asterisk DannyF (n=dannyf@c-f0aae455.24-0099-74657210.cust.bredbandsbolaget.se)`
`12:40.09`	`*** join/#asterisk linville (n=linville@azure.tuxdriver.com)`
`12:40.55`	`kink0`	`coppice: NT/TE concepts are not for E1 ?`
`12:41.10`	`coppice`	`yes they are`
`12:41.47`	`kink0`	`ok, then that is also ok, I will try to reverse NT/TE now... a bit desesperate after four days with this problem.`
`12:43.51`	`*** part/#asterisk doszy (n=adosztal@border.albacomp.hu)`
`12:45.25`	`prh`	`hmm`
`12:48.09`	`kippi`	`has anyone used Digium IAXy ?`
`12:50.27`	`*** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net)`
`12:56.52`	`mut`	`anyone know what'de cause a double ring on a sip phone calling pstn via zap? there is no Ringing() in the dialplan`
`12:57.44`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`12:58.08`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`13:01.58`	`Peste`	`Jan 27 15:01:45 NOTICE[7129]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1`
`13:02.10`	`Peste`	`does anybody know what does this mean?`
`13:05.01`	`*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)`
`13:05.36`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`13:06.17`	`RoyK`	`he lo`
`13:06.24`	`Martincit0`	`Peste it seems u need to adjust span at zaptel.conf`
`13:07.54`	`mut`	`i'm gettin really bad echo on my zap too`
`13:09.06`	`Peste`	`i did :/`
`13:14.13`	`cron`	`_Sam--: yeah the reboots are nasty.. said it requires like 15min + two boots but from what I caught it gets stuck in a loop on the tftp and requires the files to be moved after the first update? Hows the phone/echo issue sence?`
`13:15.26`	`*** join/#asterisk elephantMan (n=elephant@252.205.103-84.rev.gaoland.net)`
`13:15.29`	`*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)`
`13:17.45`	`Sjeemz`	`I've just upgraded from 1.0.9 to 1.2.3, but all incoming calls with blocked callerid get displayed on my cisco 7960 as '"Uknown" <Unknown>', this should not happen, because I use Set(CALLERID(name)=name) on all incoming calls. CIDName does seem to get overridden for all incoming calls that do supply a callerid`
`13:17.58`	`wasim`	`RoyK: goddag`
`13:19.39`	`RoyK`	`goddagen...`
`13:23.17`	`jbalcomb`	`Woot! I was at work until 12:30 AM!! I heart Asterisk!`
`13:23.43`	`jbalcomb`	`Hows come I can't take out my unused single PRI card? Hows come I had to recomple libpri and zaptel to get Asterisk working after I put it back in?`
`13:24.12`	`Ahrimanes`	`RoyK: in c? :D`
`13:24.18`	`*** join/#asterisk yun (n=yun@84.21.79.2)`
`13:25.54`	`[TK]D-Fender`	`jbalcomb : Is it up and running now?`
`13:26.12`	`yun`	`any body connect wellgate 38xx to asterisk with sip?`
`13:27.49`	`*** join/#asterisk rculp (n=rculp@66.173.240.20)`
`13:28.48`	`mut`	`are the mark and steve echo cancellers obsolete or what?`
`13:28.52`	`mut`	`cause i tried to use MARK2`
`13:28.57`	`jbalcomb`	`[TK]D-Fender yes'm. long hours of watching the other phone guy freak out. he threw a CD and it broke and I wanted to go home.`
`13:28.58`	`mut`	`and it wouldn't compilke`
`13:29.29`	`RoyK`	`Ahrimanes: no, just a dialplan from hell and three AGI scripts`
`13:30.30`	`[TK]D-Fender`	`jbalcomb : By the time you were done did the remaining card get its own IRQ, etc?`
`13:31.28`	`mut`	`anyone know how to get rid of echo on the sip side of a sip -> pstn call?`
`13:32.17`	`wasim`	`put an ec on the pstn side`
`13:33.07`	`*** join/#asterisk RussCC (n=face@216.157.205.211)`
`13:33.39`	`fourcheeze`	`is there a module to do dynamic routing between different * sharing the same realtime back end?`
`13:33.46`	`mut`	`wasim? a what?`
`13:34.02`	`mut`	`i'm using a te405p card..`
`13:34.35`	`RoyK`	`5V is for chickens`
`13:34.38`	`mut`	`my cisco i had to adjust attenuation but i don't see any adjustments for that`
`13:34.49`	`[TK]D-Fender`	`mut : If you can't get the software Zaptel EC to do the job its time to invest in hardware...`
`13:35.13`	`mut`	`my cisco as5350 did it i don't see why this thing can't`
`13:35.36`	`wasim`	`mut: an echo canceller`
`13:35.37`	`fourcheeze`	`or is there a way to record the host that is doing the registering using realtime?`
`13:35.58`	`wasim`	`RoyK: kinky`
`13:36.06`	`fourcheeze`	`any realtime experts around?`
`13:36.11`	`[TK]D-Fender`	`mut : I went through 2 revisions of TE405P's on my server because of EEC / timing issues.... It was truely unbearable`
`13:36.43`	`RussCC`	`Hello, Quick question how do you guys secure your asterisk systems?`
`13:36.55`	`jbalcomb`	`[TK]D-Fender no, strangely enough it ended up back on IRQ 11 along with the four other devices.`
`13:36.58`	`wasim`	`RoyK: missus is getting jealous`
`13:37.14`	`RoyK`	`lol`
`13:37.19`	`jbalcomb`	`[TK]D-Fender while we were working our way through the trouble I saw it on IRQ 5, 9, & 7.`
`13:37.52`	`jbalcomb`	`[TK]D-Fender the BIOS on the ASUS board we're using doesn't allow you to assign IRQs to specific slots`
`13:38.01`	`[TK]D-Fender`	`jbalcomb : So you tried all the different slots you could, and deactivated every non-essential device in the system?`
`13:38.11`	`EksilAndyCap`	`jbalcomb: maybe it only has one irq line? :)`
`13:38.22`	`jbalcomb`	`[TK]D-Fender additionally the PRI card is trapped because the rest of the slot are 5v`
`13:38.40`	`[TK]D-Fender`	`jbalcomb : I killed all my serial, USB, LPT, etc port and played with slots till my old TDM22B got its own. A serious pain to say the least`
`13:38.58`	`[TK]D-Fender`	`jbalcomb : You have a TE410P?`
`13:39.17`	`jbalcomb`	`[TK]D-Fender well, I had disabled all that stuff but by the time we got to get things working again we put everything back like we found it`
`13:39.42`	`jbalcomb`	`[TK]D-Fender the paint on the board says TE410P but the system lists it as TE411P`
`13:40.34`	`mut`	`um`
`13:40.55`	`[TK]D-Fender`	`jbalcomb : Do you SEE an EC module on it?`
`13:40.57`	`mut`	`what echo canceller do you use tk?`
`13:40.59`	`mut`	`in zaptel`
`13:41.42`	`[TK]D-Fender`	`mut : Otasic :D`
`13:41.42`	`jbalcomb`	`[TK]D-Fender is that the daughter board?`
`13:41.42`	`mut`	`hm?`
`13:41.42`	`[TK]D-Fender`	`jbalcomb : Yeah....`
`13:41.42`	`jbalcomb`	`[TK]D-Fender i think so`
`13:41.42`	`[TK]D-Fender`	`mut : I ditched my TE405P's, and now run a Sangoma A104d ;)`
`13:41.43`	`mut`	`oh`
`13:42.00`	`[TK]D-Fender`	`jbalcomb : Well then that would seem to say you have their EC board and could maybe try using the HWEC`
`13:42.36`	`[TK]D-Fender`	`I dun got me a reul DSP hyuk!`
`13:44.06`	`jbalcomb`	`[TK]D-Fender what the hell is HWEC?`
`13:44.17`	`[TK]D-Fender`	`HardWare Echo Cancellation`
`13:44.23`	`wasim`	`Hell Will Echo Cancel`
`13:44.52`	`cpm`	`Have Wads Extra Cash`
`13:45.05`	`jbalcomb`	`He Will Enhance you Cynicism`
`13:45.09`	`[TK]D-Fender`	`cpm : I accept Paypal ;)`
`13:45.26`	`jbalcomb`	`How Will Employees Cope?`
`13:45.35`	`mut`	`well damnit`
`13:45.44`	`mut`	`i got this card cause i figured it'de be better than the cisco box`
`13:45.51`	`jbalcomb`	`cpm: dont do it, he's Canadian. It's a scam!`
`13:45.56`	`mut`	`what a rip off`
`13:46.19`	`wasim`	`it is`
`13:46.24`	`jbalcomb`	`mut HAHA!! you thought something would be better than Cisco!! bumwhahaha...`
`13:46.36`	`cpm`	`heh`
`13:46.47`	`jbalcomb`	`All your box are belong to Cisco`
`13:46.56`	`mut`	`uh`
`13:46.58`	`mut`	`yea`
`13:48.11`	`*** join/#asterisk blkremedy (n=ur3rdeye@240M06.oasis.mediatti.net)`
`13:48.28`	`[TK]D-Fender`	`mut : A104d = 0 echo, PCI voltage agnostic, and platform independant. Its no the cheapest thing on the market, but you already have a 4-port so the diff is nominal.`
`13:48.29`	`jbalcomb`	`ok, maybe its just that my paychecks are bigger when Cisco is invlolved.`
`13:51.13`	`*** join/#asterisk }btorch{ (n=kvirc@208.63.19.172)`
`13:51.17`	`iCEBrkr`	`jbalcomb: You get green lights?`
`13:51.19`	`}btorch{`	`hello`
`13:51.31`	`tzanger`	`morning`
`13:52.25`	`jbalcomb`	`iCEBrkr yes'm. put everything back like we found it and recompiled libpri & zaptel`
`13:52.38`	`iCEBrkr`	`hrrm`
`13:52.44`	`iCEBrkr`	`I bet that 1st card is setup to do timing`
`13:53.06`	`jbalcomb`	`iCEBrkr million dollar question: how is that done and undone?`
`13:53.33`	`wasim`	`we've standardized on a104d pretty much`
`13:53.39`	`iCEBrkr`	`jbalcomb: Lemme get into work. I'll check my config there`
`13:53.51`	`jbalcomb`	`iCEBrkr oki`
`13:54.08`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`13:54.18`	`*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`13:55.46`	`[TK]D-Fender`	`iCEBrkr : Yeah I was asking about that yesterday :)`
`13:56.36`	`[TK]D-Fender`	`wasim : Great little card, isn't it? They make a very "tight" package. No frills, just serious hardware.`
`13:56.58`	`jbalcomb`	`[TK]D-Fender was that right before you dropped offline while my server was down/down? Where's the customer service?!?`
`13:57.02`	`wasim`	`i have three customers who are looking at 20 quads each in the next quarter`
`13:57.06`	`jbalcomb`	`[TK]D-Fender ;)`
`13:57.06`	`*** part/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)`
`13:57.57`	`*** join/#asterisk Tribastian (n=tribasti@62-2-138-202.business.cablecom.ch)`
`13:58.12`	`[TK]D-Fender`	`jbalcomb : .... funny I never got a penny for my help :)`
`13:58.14`	`jbalcomb`	`iCEBrkr you Asterisk@home today?`
`13:58.34`	`Tribastian`	`hallo leute, bin neu aber habe gerade ein paar wichtige fragen...`
`13:58.36`	`jbalcomb`	`[TK]D-Fender well, if you help me now maybe I can work that out for ya.. pad the hours..`
`13:58.41`	`blkremedy`	`hello everyone`
`13:58.50`	`Tribastian`	`sorry english`
`13:59.02`	`*** join/#asterisk QuAd\|Haudrauf (n=hau@port-212-202-185-252.dynamic.qsc.de)`
`13:59.04`	`*** join/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch)`
`13:59.12`	`iCEBrkr`	`jbalcomb: I'm taking my time getting ready`
`13:59.15`	`Tribastian`	`i am new but i do have some serious problems with asterisk in the company i work in`
`13:59.20`	`iCEBrkr`	`jbalcomb: I didn't get to bed until 6:45ish`
`13:59.34`	`Katty`	`mew.`
`13:59.57`	`jbalcomb`	`iCEBrkr nice.`
`14:00.24`	`tzanger`	`postgres for CDR rocks. That is all.`
`14:00.26`	`tzanger`	`# select (max(calldate)-min(calldate))::interval,(sum(billsec)/3600.00)::numeric(6,2) as hours, count(*) as calls from cdr where host='wu-ast';`
`14:00.29`	`tzanger`	`<PROTECTED>`
`14:00.31`	`tzanger`	`------------------+--------+-------`
`14:00.34`	`tzanger`	`<PROTECTED>`
`14:00.37`	`tzanger`	`(1 row)`
`14:00.53`	`*** join/#asterisk coppice (n=chatzill@223.143.17.210.dyn.pacific.net.hk)`
`14:00.57`	`[TK]D-Fender`	`jbalcomb : Well... its up and running now, and you have hardware problems. Not that much I can do there. When it comes to a serious overhaul of your PBX design (* config) that I can do....`
`14:01.20`	`}btorch{`	`has anyone here installed sphinx to work with asterisk ?`
`14:01.33`	`Tribastian`	`well we did build up the asterisk server, we did the dialplans, we did the extension.conf and we are able to phone inernal, we are able to phone put, but we are not able that someone can dial in...`
`14:01.38`	`jbalcomb`	`[TK]D-Fender yeah. I feel ya. I just need to figure how/what I gotta do to get the system working when that extra card is out.`
`14:01.52`	`blkremedy`	`Pretty new to asterisk here....After days of searching, I've hardly found any configurations with asterisk@home on a laptop. Is it not recommended?`
`14:01.54`	`Katty`	`wasim: :<`
`14:02.14`	`fourcheeze`	`blkremedy: depends what you want to do with it`
`14:02.18`	`[TK]D-Fender`	`jbalcomb : Pastebin up you zaptel / zapata / and interrupts list.`
`14:02.28`	`[TK]D-Fender`	`Katty: Mew.`
`14:02.32`	`Katty`	`[TK]D-Fender: mew.`
`14:02.33`	`jbalcomb`	`[TK]D-Fender iCEBrkr Our phone consultant email us at 8:50 to let us know he is stuck in michigan and can't make the 9 AM meeting!! i'm lovin' it.`
`14:02.36`	`*** join/#asterisk Dad (n=dad@206.125.55.168)`
`14:02.52`	`[TK]D-Fender`	`jbalcomb : Ever heard of MeetMe? :)`
`14:03.01`	`[TK]D-Fender`	`And eyeBeam?`
`14:03.01`	`Katty`	`jbalcomb: we loves it when you be smilin </mcdonalds>`
`14:04.08`	`mut`	`ok`
`14:04.10`	`jbalcomb`	`[TK]D-Fender i have heard of those things.. might be too complicated for a guy who can't even use a phone`
`14:04.16`	`mut`	`my echo comes into play when i call verizon lines`
`14:04.20`	`mut`	`if i call my cell`
`14:04.29`	`jbalcomb`	`[TK]D-Fender iCEBrkr /etc/zaptel.conf http://pastebin.com/525593`
`14:04.33`	`mut`	`no echo, call sbc, no echo, call telnet local switch, no echo`
`14:04.51`	`*** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au)`
`14:05.03`	`Tribastian`	`well we did build up the asterisk server, we did the dialplans, we did the extension.conf and we are able to phone inernal, we are able to phone put, but we are not able that someone can dial in...`
`14:05.20`	`jbalcomb`	`[TK]D-Fender iCEBrkr /etc/asterisk/zapata.conf http://pastebin.com/525597`
`14:06.03`	`jbalcomb`	`[TK]D-Fender iCEBrkr /proc/interrupts http://pastebin.com/525599`
`14:06.31`	`[TK]D-Fender`	`jbalcomb : 2 ports on a 4-port card?`
`14:06.49`	`wasim`	`Tribastian: what error do you get on *CLI?`
`14:07.12`	`Tribastian`	`just a sec, gonna check so i do not tell shit... ;-)`
`14:08.41`	`wasim`	`umm, cut/paste helps you cut down on the habit`
`14:08.42`	`*** join/#asterisk rob314 (n=root@207.58.194.55)`
`14:08.46`	`jbalcomb`	`[TK]D-Fender yes, m. room for growth..`
`14:09.01`	`Katty`	`aka, cookies.`
`14:10.14`	`iDunno`	`hmm. cookies :)`
`14:10.49`	`[TK]D-Fender`	`jbalcomb : OMG, whats with the LBO settings in zaptel.conf? How far are you from your smartjack?!`
`14:11.23`	`Tribastian`	`WARING[16640]: chan_sip.c.4045 sip_reg_timeout: ---Registration for '6266952@sipgate.de' timed out, trying again ---parse_srv:SRV mapped to host sipgate.de, port 5060`
`14:11.58`	`Tribastian`	`for seqno 4107 (Critical Request)`
`14:12.18`	`sivana`	`reload`
`14:12.21`	`sivana`	`ack`
`14:12.28`	`Tribastian`	`and then again and again only the seqno counts up`
`14:13.26`	`*** join/#asterisk pato (n=just@nat1.inalambrica.net)`
`14:15.37`	`jbalcomb`	`[TK]D-Fender yeah, i asked the new phone guy what the LBO setting was about and he says 'i dont know' and then changes it from 0 to 5`
`14:15.50`	`sivana`	`I have this in my s exten, GotoIf($[${CALLPROGRESS} > 0]?inprogress) what does it need to be to jump to a different exten?`
`14:15.58`	`jbalcomb`	`[TK]D-Fender is my smart jack that box on the wall?`
`14:16.10`	`*** join/#asterisk stegbth (n=stegbth@stegbth.sim.tronicplanet.de)`
`14:16.16`	`stegbth`	`hi all`
`14:16.21`	`*** join/#asterisk pengyong (n=lala@218.93.153.249)`
`14:16.38`	`*** join/#asterisk dorphalsig (n=dorphals@200.106.223.5)`
`14:16.40`	`dorphalsig`	`Hi`
`14:16.42`	`dorphalsig`	`Jan 27 07:02:48 WARNING[15181]: Got restart ack on channel 0/30 span 1 with owner`
`14:17.14`	`dorphalsig`	`I'm getting that warning every once in a while when my B-Channels get restarted (anyway, why do they get restarted?)`
`14:17.28`	`dorphalsig`	`Can anybody give me a hand?`
`14:17.38`	`Katty`	`i'm using both of mine.`
`14:18.13`	`[TK]D-Fender`	`jbalcomb : yEAH. yOU HAVE AN lbo SETTING FO 5 ON THERE... THAT KINDA NUTS`
`14:18.54`	`[TK]D-Fender`	`Katty: WhAt ArE yOu TaLkInG aBoUt?!`
`14:19.14`	`Ahrimanes`	`1337`
`14:19.18`	`[TK]D-Fender`	`Katty: MUMBLER! .... I can't understad a word you are saying.....`
`14:19.35`	`Ahrimanes`	`that'd be the day?`
`14:20.07`	`Katty`	`[TK]D-Fender: i am pretty soft spoken (=`
`14:20.22`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:20.28`	`*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)`
`14:21.31`	`*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)`
`14:21.32`	`[TK]D-Fender`	`Katty: Put away that big stick!`
`14:23.04`	`arkanis`	`s`
`14:24.45`	`Katty`	`[TK]D-Fender: :>`
`14:25.44`	`*** join/#asterisk gambolputty (n=root@64.74.225.131)`
`14:26.22`	`arkanis`	`every time call (outbound), I call anonymous, how can I define, that asterisk displays my number?`
`14:27.43`	`dorphalsig`	`Hey, I'm getting this message --> Jan 27 07:02:48 WARNING[15181]: Got restart ack on channel 0/30 span 1 with owner after I get my B-Channels restarted. Anybody knows what does this mean?`
`14:28.38`	`brad_mssw`	`anyone have a recommended voip provider?`
`14:29.13`	`brad_mssw`	`also, anyone have voicepulses iax server so I can do a traceroute?`
`14:29.37`	`Grubs`	`apt-get upgrade`
`14:29.43`	`Grubs`	`lol - wrong window!`
`14:30.12`	`kippi`	`hey`
`14:31.24`	`RoyK`	`Grubs: try dist-upgrade :)`
`14:31.50`	`wasim`	`dist-upgrade gentoo &`
`14:33.23`	`Grubs`	`{{{{ Debian r0x0rz @st3r1x }}}}`
`14:33.37`	`*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)`
`14:35.24`	`Tribastian`	`so one more time, please help!!! well we did build up the asterisk server, we did the dialplans, we did the extension.conf and we are able to phone inernal, we are able to phone out, but we are not able that someone can dial in...`
`14:36.02`	`Tribastian`	`the message from Asterisk is: WARING[16640]: chan_sip.c.4045 sip_reg_timeout: ---Registration for '6266952@sipgate.de' timed out, trying again ---parse_srv:SRV mapped to host sipgate.de, port 5060`
`14:36.08`	`Tribastian`	`for seqno 4107 (Critical Request)`
`14:36.27`	`brad_mssw`	`ok ... so your registration line is invalid`
`14:36.49`	`[TK]D-Fender`	`Tribastian : pastebin your sip.conf`
`14:36.52`	`[TK]D-Fender`	`~pb`
`14:36.53`	`jbot`	`well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`14:37.54`	`dorphalsig`	`Tribastian --> It seems like your SIP provider is either not responding, you have a network problem or youre password`
`14:38.32`	`Tribastian`	`ok will do that, sorry i am new`
`14:38.52`	`Tribastian`	`so my registration line is invalid, solution?`
`14:39.09`	`dorphalsig`	`paste it at pastebin`
`14:39.24`	`Err`	`(after removing whatever passwords might be in the file)`
`14:41.30`	`Tribastian`	`http://pastebin.com/525658`
`14:42.03`	`Tribastian`	`this is a line of my sip.conf`
`14:42.26`	`kippi`	`anyone got a grandstream GXP2000 ? and got the speed dials to work?`
`14:42.33`	`Tribastian`	`i will post the error message there, too, just a sec...`
`14:43.22`	`dorphalsig`	`Hey, I'm getting this message --> Jan 27 07:02:48 WARNING[15181]: Got restart ack on channel 0/30 span 1 with owner after I get my B-Channels restarted. Anybody knows what does this mean?`
`14:43.55`	`[TK]D-Fender`	`Tribastian : We need the peer setup as well!`
`14:43.57`	`brad_mssw`	`you sure that IP address is right ?? you should really be using a hostname`
`14:45.00`	`Tribastian`	`ok, ahm what is my peer setup? i mean wich file do you need`
`14:45.18`	`[TK]D-Fender`	`Tribastian : The other half of your SIP trunk setup`
`14:45.24`	`[TK]D-Fender`	`in sip.conf`
`14:45.56`	`mdave`	`hrm. if I want to run DISA(1234\|default), for an outbound call places by asterisks (/var/spool/asterisk/outgoing), what do I put for Application: and Data: ? I tried "Application: DISA" and "Data: 1234\|default" and it makes the call, takes the 1234, and gives me dialtone, but then the same dialing that works when a local phone picks up to that same context doesnt seem to work`
`14:46.19`	`mdave`	`eg, I want asterisk to call a number, and then allow the callee to dialout as if they were on a local phone`
`14:46.45`	`dorphalsig`	`Tribastian --> take a look at this file http://www.voip-info.org/wiki-Asterisk+config+sip.conf`
`14:46.58`	`*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)`
`14:47.50`	`mdave`	`it definately seems to be running DISA.. just not matching up to the context.. or something`
`14:48.22`	`*** join/#asterisk elg (n=fugalh@falcon.fugal.net)`
`14:48.35`	`Tribastian`	`ok here it is: http://pastebin.com/525666`
`14:49.20`	`dorphalsig`	`[41442005335] <-- should be [mynumber]`
`14:49.28`	`Tribastian`	`dorphalsig: thank you for the link, i was there maybe i did not understand it right...`
`14:49.32`	`*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`14:49.51`	`Tribastian`	`yes it is...`
`14:50.33`	`[TK]D-Fender`	`Tribastian : Change the fixed IP in your register line to "voipgateway.org, and make sure the passwords match. You also have it sayig "net=yes" I think you mean "nat=yes".`
`14:50.50`	`*** part/#asterisk fourcheeze (n=rich@82.153.215.21)`
`14:50.58`	`Tribastian`	`will try...`
`14:50.59`	`darkskiez`	`Does the new toshiba videophone work on asterisk? http://blog.modernmechanix.com/mags/qf/c/PopularScience/7-1964/med_video_phone.jpg`
`14:51.17`	`[TK]D-Fender`	`Tribastian : and "username=41442005335"`
`14:51.29`	`kink0`	`is normal that ZAP is not running if I set the remote extreme as NT/M and ZAP is ok if I set the extreme as TE/S ?`
`14:51.39`	`*** join/#asterisk gaspiz (n=gaspiz@86.34.6.164)`
`14:51.41`	`kink0`	`or is a problem configuring zaptel.conf/zapata.conf ?`
`14:51.55`	`*** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)`
`14:52.11`	`kink0`	`because if I set the other end as NT/M , then still as RED alarm`
`14:52.44`	`Tribastian`	`it is the number, just took it out so nobody sees it... well did not succeed because i have forgot to take it out in the header..`
`14:52.53`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`14:52.53`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:52.54`	`dorphalsig`	`hehehe`
`14:53.05`	`gaspiz`	`hi, I'm trying to dial from an asterisk another asterisk using sip. It works fine for all of my users but one. for this user the second asterisk sends back 407 Proxy Authentication Required`
`14:53.21`	`gaspiz`	`does anyone know what i'm doing wrong?`
`14:53.24`	`anthm`	`yay`
`14:53.29`	`Tribastian`	`hey it works!!!!!!!!! thank you but i do have another error, sending it soon...`
`14:53.42`	`*** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar)`
`14:53.50`	`dorphalsig`	`hehe`
`14:53.51`	`viperdude`	`gaspiaz: what does the CLI say when it trys to dial?`
`14:54.51`	`gaspiz`	`<PROTECTED>`
`14:54.52`	`gaspiz`	`Jan 27 09:54:16 NOTICE[19568]: chan_sip.c:9524 handle_response_invite: Failed to authenticate on INVITE to '"Marc White" <sip:1005@204.10.64.134>;tag=as44fd5133'`
`14:54.52`	`gaspiz`	`<PROTECTED>`
`14:55.08`	`viperdude`	`before that what is the dial command?`
`14:55.27`	`gaspiz`	`<PROTECTED>`
`14:56.01`	`*** join/#asterisk stack_ (n=stack@63.239.190.202)`
`14:56.13`	`viperdude`	`ok sounds like other server is not accepting connections from other servers`
`14:56.28`	`mdave`	`ok.. if I change the context to something nonexistent, I get fast-busy as soon as I dial a digit.. with it to the correct context, it accepts digits, but then eventually still gives fast-busy, but when I dial the same digits on a local phone, in the same context, the call goes through`
`14:56.39`	`gaspiz`	`but why does it work for the other users?`
`14:56.42`	`mdave`	`wonder if Im doing something wrong or something isnt working right`
`14:56.44`	`*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`14:56.56`	`stack_`	`I would like to set up a queue so that once it has round-robined through the queue once, it exits the queue and goes to the next priority in dialplan... is this possible?`
`14:56.57`	`*** join/#asterisk fugitivo (n=ajf@201.255.176.94)`
`14:56.58`	`viperdude`	`other users on the same server?`
`14:57.09`	`gaspiz`	`viperdude: yes`
`14:57.41`	`gaspiz`	`viperdude: both servers take the users from the same database`
`14:57.52`	`kink0`	`Aterisk/PRI pri_net ---> N2/PRI as TE/any = Zap OK , but Asterisk/PRI pri_cpe -->N2/PRI as NT/M = Zap RED alarm`
`14:57.56`	`kink0`	`is that normal ?`
`14:58.04`	`kink0`	`or I am doing something wrong ?`
`14:58.14`	`*** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net)`
`14:58.15`	`viperdude`	`ok well the IP on the dial is different to the IP on the fail to auth so not sure whats going on inbetween`
`14:58.30`	`*** join/#asterisk SplasPood (i=jwb@206.252.198.100)`
`14:59.34`	`Tribastian`	`ok back again, the error message and my a part of the conf files are in there...:http://pastebin.com/525679`
`15:00.16`	`*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)`
`15:00.18`	`gaspiz`	`viperdude: the user is logged in to server A, I'm trying todial the user on server b to transfer the call to the voicemail server`
`15:00.49`	`[TK]D-Fender`	`Tribastian : You are missing an incoming context in your SIP.CONF to tell it where to dial that number into. add "context=#`
`15:00.49`	`[TK]D-Fender`	`[telin]`
`15:01.02`	`gaspiz`	`viperdude: it's working for the other users, and I have the same settings for this user as for the others`
`15:01.07`	`Tribastian`	`i think i do have a loop somewhere, and the error is repeating 100 of times`
`15:01.10`	`[TK]D-Fender`	`"context=telin" into the peer config in sip.conf`
`15:01.24`	`Tribastian`	`tring...`
`15:01.38`	`*** join/#asterisk littleball (n=littleba@cm169.epsilon169.maxonline.com.sg)`
`15:01.49`	`pato`	`stack_: i think it is possible, you just need to play with the timeout in queues.conf and the timeout parameter when you call the queue app`
`15:02.20`	`stack_`	`pato, so I would have to add the timeouts of everyone in the queue and make the queue timeout that number?`
`15:02.36`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`15:02.56`	`pato`	`stack_: yes`
`15:03.06`	`gaspiz`	`viperdude: any ideas what to do?`
`15:03.11`	`[TK]D-Fender`	`stack_ : Sounds like you are better off doing it with direct dial-plan logic....`
`15:03.13`	`Tribastian`	`ok, it still works, we have lesser error messages but still a lot of these spawn extensions lines`
`15:03.27`	`[TK]D-Fender`	`Tribastian : pastebin your entrie sip.conf please.`
`15:03.33`	`*** join/#asterisk Cresl1n (n=matt@m595e36d0.tmodns.net)`
`15:03.56`	`Tribastian`	`ok, just a sec, need to remove the passwords...`
`15:04.08`	`stack_`	`[TK]D-Fender, I'm pretty new to asterisk and we really need the other queue options, like the least dialed, etc... I probably wouldn't have time to script something like that`
`15:05.11`	`[TK]D-Fender`	`stack_ : Ok, if you need the logic on top, but rather that try and set it to time out on 1 pass, I'd suggest using a simple time-limit in queue. I have mine set to 5 mins which then bombs out to VM.`
`15:06.03`	`*** join/#asterisk Poroto (i=raul@tesla.xmission.com)`
`15:06.16`	`*** join/#asterisk OrdeJ (i=TRAX_@62.162.228.210)`
`15:06.20`	`*** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com)`
`15:06.48`	`stack_`	`[TK]D-Fender, is there a set time it takes to find the next person in the queue, i noticed a delay...`
`15:07.13`	`*** join/#asterisk Hmmhesays (n=Neg@72.24.227.83)`
`15:07.17`	`MattB2`	`hi all.. i have the fun job of trying to hook an alarm system up to asterisk via an ATA. I've read that i need to disable echo cancellation but we have an 8-channel PRI and we need echo cancel so the audio calls are fine. is there anyway of disabling per-call or per-channel?`
`15:07.27`	`*** join/#asterisk TheGoD (n=TheGoD@adsl-70-224-56-152.dsl.sbndin.ameritech.net)`
`15:08.34`	`Dabba`	`anyone any idea why im seeing this http://pastebin.ca/raw/38791`
`15:08.38`	`Tribastian`	`D-Fender: ok, here it is...`
`15:08.40`	`Tribastian`	`http://pastebin.com/525699`
`15:08.49`	`gaspiz`	`<PROTECTED>`
`15:08.50`	`*** join/#asterisk MattB2_ (n=MattB2@mail.tricycleinc.com)`
`15:09.05`	`TheGoD`	`I'm getting errors zt_rbs: tried to set RBS hook state 0 on channel WCTDM/0/1 while span WCTDM/0 lacks rbsbits or hooksig function`
`15:09.49`	`MattB2_`	`any have suggestions on asterisk/ATA setup with an alarm?`
`15:10.19`	`Tribastian`	`D-Fender: well to say this is not all of the sip.conf, we do have many more users, but they all look more or less the same... sorry to much passwords to remove...`
`15:10.48`	`Dabba`	`is this syntax correct ? exten => _X.,9,GotoIf($[${chosenmin} > 59]?wrongminentered,s,1)`
`15:11.19`	`Hmmhesays`	`ok i'm doing something retarded here, my simple static forwarding is not working in ser`
`15:11.29`	`_Sam--`	`i think you might need "59"`
`15:11.57`	`OrdeJ`	`Hi. One stupid question - i install asterisk@home on windows and vmplayer... after i logon on root - how to switch to GUI?`
`15:12.00`	`Dabba`	`cheers _Sam--`
`15:12.09`	`_Sam--`	`and also maybe one other set`
`15:12.26`	`*** join/#asterisk jpablo (n=jpablo@200.94.130.194)`
`15:12.26`	`_Sam--`	`http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf`
`15:12.33`	`_Sam--`	`some examples`
`15:12.53`	`kink0`	`[TK]D-Fender: do you know if is normal I can not get OK my Zap (RED alarm )while the other end is NT/M ?`
`15:13.02`	`jpablo`	`hey people, recommend a good voip online store, I'm looking for a sip<->gsm or analog<->gsm gateway.`
`15:13.08`	`_Sam--`	`i dont know if it will do > either..`
`15:13.15`	`Dabba`	`exten => _X.,10,GotoIf($["${chosenmin}" = ""]?nominutesentered,s,1)`
`15:13.16`	`Dabba`	`or`
`15:13.22`	`kink0`	`jpablo: I bought recently a 2N Stargate`
`15:13.23`	`Dabba`	`exten => _X.,10,GotoIf($[${chosenmin} = ""]?nominutesentered,s,1)`
`15:13.37`	`jpablo`	`kink0: it is any good ?`
`15:13.43`	`*** join/#asterisk stse (n=stse@muedsl-82-207-250-119.citykom.de)`
`15:13.43`	`gaspiz`	`hi, how do I transfer calls from an asterisk to another asterisk during the call?`
`15:13.45`	`_Sam--`	`i think the first one you pasted is good`
`15:13.49`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`15:13.53`	`Dabba`	`k`
`15:14.14`	`stack_`	`[TK]D-Fender, what happens to the queue timing when one of the people in the queue is on DND?`
`15:14.14`	`kink0`	`jpablo: is a nice equipment, I like, but now I have some problem connecting PRI-> Asterisk PRI`
`15:14.28`	`*** join/#asterisk kpettit (n=keith@69.15.174.114)`
`15:14.35`	`iCEBrkr`	`grrrrrrrrrrrr`
`15:14.36`	`kink0`	`jpablo: is easily scalable, modular, and allow SIM servers`
`15:14.58`	`_Sam--`	`and it WILL do > < etc`
`15:15.00`	`kpettit`	`I'm having a problme on a 1.0.9 similar to the timebomb bug that happened with 1.2.2`
`15:15.17`	`stse`	`Hi! How do I tell asterisk to send the pound key to the receiver, so I can control my answering maching if I use a phone connected to the asterisk?`
`15:15.20`	`kpettit`	`Can't hear any sound. Any idea's what would cause that. System worked yesterday]`
`15:15.24`	`kink0`	`also allow several manners to administration, including telnet, terminal port, and so. Up to 32 channels, 255 SIM per 3U rack`
`15:15.30`	`jpablo`	`kink0: where did you get it ?`
`15:15.32`	`_Sam--`	`kpettit: i didnt forget about either...sorry i didnt get the disk image out yet`
`15:15.36`	`kpettit`	`Can't hear any sound. Any idea's what would cause that. System worked yesterday`
`15:15.50`	`kink0`	`jpablo: from the manufacturer, www.2n.cz`
`15:16.02`	`kpettit`	`_Sam--, no problem, thanks for remembering`
`15:16.17`	`kink0`	`there some in eBay, but the price is near the same as if you order from factory, and you have not support if you buy it from eBay`
`15:16.34`	`kink0`	`well... support is good, even I have not resolved yet my problem today.`
`15:16.44`	`*** join/#asterisk Underhand (n=gavan@frog.coolfactor.org)`
`15:16.49`	`TheGoD`	`I'm getting errors zt_rbs: tried to set RBS hook state 0 on channel WCTDM/0/1 while span WCTDM/0 lacks rbsbits or hooksig function`
`15:17.03`	`kink0`	`jpablo: have you connected your digium/clone to any E1 or PRI trunk ?`
`15:17.06`	`jpablo`	`kink0, probably that thing is to big for me.`
`15:17.18`	`jpablo`	`kink0, yes. to an alestra isdn pri`
`15:17.22`	`*** join/#asterisk Assid (n=assid@203.115.64.14)`
`15:17.22`	`kink0`	`jpablo: then, see bluestar`
`15:17.26`	`Assid`	`heya`
`15:17.28`	`Assid`	`umm..`
`15:17.41`	`kink0`	`jpablo: how much channels ?`
`15:17.48`	`Underhand`	`choppy music on hold - is that likely to be caused by timing issues?`
`15:17.52`	`jpablo`	`kink0, all 30 of them`
`15:17.54`	`Bambr`	`hi, i've got a question, is it possible to put some queue working only on some certain period of time, for example, from 9 pm till 2 am ?`
`15:18.05`	`Dabba`	`> _Sam-- http://pastebin.ca/raw/38792 seems to work :-) cheers`
`15:18.06`	`kink0`	`jpablo: then Stargate is the best I know, and cheap than Teles`
`15:18.10`	`Assid`	`if i call a macro.. and from a macro.. i use include or goto .. does the arguments of the macro exist in the included/goto scope?`
`15:18.14`	`*** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar)`
`15:18.28`	`kink0`	`Valiant is lighly cheapest, but is not very nice hard.`
`15:18.30`	`*** join/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru)`
`15:18.43`	`kink0`	`6U instead 3U rack, nothing about SIM servers, and so.`
`15:18.53`	`jpablo`	`kink0, ah, sorry i was with the e1 thing, i need almost 5 or so channels.`
`15:19.33`	`kink0`	`jpablo: then Stargate, and install just 3 modules ( 6 channels )... about 4000 Euro all.`
`15:19.47`	`Assid`	`anyone know?`
`15:19.53`	`kink0`	`and you are able to scale up to 30 channels if you need tomorrow`
`15:20.12`	`jpablo`	`kink0, i don't think my employer is willing to put that much money at this point.`
`15:20.27`	`kink0`	`jpablo: can you paste me your zaptel.conf/zapata.conf ?`
`15:20.36`	`jpablo`	`kink0, i guess i will just buy five of this http://www.voipsupply.com/product_info.php?products_id=1291`
`15:20.39`	`*** join/#asterisk muzzz_ (n=chatzill@60.48.153.162)`
`15:20.49`	`kink0`	`jpablo: well.. you can also connect lets say a moto v360 to your PC sound card`
`15:21.30`	`OrdeJ`	`<PROTECTED>`
`15:21.52`	`jpablo`	`kink0, yeah, but that wouldn't scale at all, i would need 5 pc sound cards`
`15:21.55`	`kink0`	`even multiple v360 to USB sounds cards, and go really fine`
`15:22.17`	`Dabba`	`is it possible to replace this lot with a macro ? http://pastebin.ca/raw/38794`
`15:22.23`	`stse`	`Or if I want to use to pound key on a called answering machine, how do I tell Asterisk to let them through?`
`15:23.11`	`jpablo`	`kink0, the problem is, now i have to get the v360, i don't know if your gsm provider is giving thoses for our network.`
`15:23.13`	`kink0`	`jpablo: yes, but is really cheap.`
`15:23.39`	`kink0`	`sure, I got one v360 without cost from Amena`
`15:24.00`	`kink0`	`then sound cards, about 10 E/each`
`15:24.57`	`jpablo`	`maybe i can try that, then when the company sees the value of calling our private gsm network from anywhere they can invest in a 2N thing.`
`15:24.59`	`stegbth`	`hi`
`15:25.16`	`mdave`	`ok, trying to get this outbound call connected to DISA is really making my head hurt..`
`15:25.29`	`stegbth`	`i am thinking about a voicemailbox with openhours`
`15:25.40`	`stegbth`	`i have pasted my configs here:`
`15:25.43`	`*** join/#asterisk aLeeNa (n=aleena@dsl5400E65A.pool.t-online.hu)`
`15:25.43`	`stegbth`	`<PROTECTED>`
`15:25.44`	`mdave`	`it seems to work almost all the way, but just wont recognize the dialed digits.. and I dont see anything that I am missing, reading the docs at voip-info, as bext I can make sense of them`
`15:25.45`	`aLeeNa`	`hello`
`15:25.54`	`Tribastian`	`hello`
`15:26.13`	`stegbth`	`but i think this isn't the easiest way, are there other's?`
`15:26.13`	`iCEBrkr`	`[TK]D-Fender: psst`
`15:26.14`	`kink0`	`jpablo: do you know at your E1 connection is the far end is NT or TE ? I suppuse is NT because you are ussing pri_cpe`
`15:26.16`	`*** join/#asterisk SGM (n=stoyan@213.91.216.130)`
`15:26.31`	`mdave`	`is anyone hear familiar with making outgoing calls by placing .call files, and then using the Application/Data option? and having DISA be the application?`
`15:26.48`	`mdave`	`s/hear/here`
`15:27.07`	`Dabba`	`mdave > yes`
`15:27.12`	`jpablo`	`kink0, NT.`
`15:27.23`	`mdave`	`dabba did you see what I typed earlier?`
`15:27.35`	`jpablo`	`kink0, what kind of e1 are you connecting`
`15:27.42`	`mdave`	`basically, it seems to work, but then no matter what the callee dials, they get to fast-busy`
`15:27.49`	`kink0`	`jpablo: and are you ussing Digium TExxx ?`
`15:28.02`	`Dabba`	`have u console when this happens`
`15:28.08`	`Dabba`	`no id ditn see`
`15:28.09`	`mdave`	`yes`
`15:28.15`	`mdave`	`nothing terribly interesting there`
`15:28.17`	`kink0`	`jpablo: I pretend to connect 2N PRI NT/M -> Asterisk Digium TE405`
`15:28.22`	`Dabba`	`pastebin ?`
`15:28.35`	`mdave`	`it just notes that the outgoing call is placed`
`15:28.50`	`mdave`	`it doesnt display anything after or during the callee's dialing`
`15:29.03`	`*** join/#asterisk elephantMan (n=elephant@252.205.103-84.rev.gaoland.net)`
`15:29.03`	`kink0`	`but unssuccefull, only partial success if I set TE/* at 2N side and use signalling as pri_net`
`15:29.23`	`mdave`	`Attempting call on SIP/XXXX@XXXX for application Disa(123\|default) (Retry 1)`
`15:29.26`	`kink0`	`but then I was able to send calls from 2N to Digium, and not viceverse.`
`15:29.37`	`mdave`	`and then 'call completed'`
`15:29.41`	`kink0`	`then I was pretending to alter TE/NT and try`
`15:29.59`	`*** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net)`
`15:30.01`	`kink0`	`but when I set NT at the 2N end, doesn't works for me.`
`15:30.04`	`Dabba`	`are you sending the callee into default to place calls ?`
`15:30.12`	`Dabba`	`thats no so cool`
`15:30.13`	`mdave`	`yes`
`15:30.29`	`mdave`	`not cool as in cant work, or 'insecure'`
`15:30.33`	`mdave`	`I will note the callee is me`
`15:30.35`	`kink0`	`jpablo: do not use trunkgroups ??? in your zapata.conf ?`
`15:30.38`	`Dabba`	`inseure`
`15:30.45`	`stegbth`	`and is there a way to remove the woman's after playing my busy.gsm?`
`15:30.50`	`mdave`	`ok, I understand the security implications..`
`15:30.58`	`mdave`	`but im just trying to get it to work at all`
`15:30.59`	`Dabba`	`cool`
`15:31.25`	`stegbth`	`a i meant the woman's voice ;)`
`15:31.38`	`mdave`	`the same default context works fine when a local ATA-attached phone is picked up and dials`
`15:31.49`	`jpablo`	`kink0, nope`
`15:31.55`	`Dabba`	`so your dropping a .call into outgoing and then your expecting it to dial a sip device and give dialtone upon answer`
`15:32.00`	`kink0`	`hmmmm ... voy a intentarlo asi a ver...`
`15:32.04`	`*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)`
`15:32.06`	`mdave`	`exactly, and it does get that far`
`15:32.09`	`jpablo`	`kink0, nunca supe para que eran realmente :P`
`15:32.10`	`mdave`	`it calls, I answer, I get dialtone`
`15:32.32`	`Dabba`	`then it goes wonky?`
`15:32.32`	`mdave`	`I dial the digits for the DISA password, and #, and I get the second dialtone`
`15:32.35`	`mdave`	`all as expected`
`15:32.36`	`Dabba`	`k`
`15:32.43`	`mdave`	`if I dial the wrong password it errors out, as expected`
`15:33.00`	`mdave`	`but.. once I dial the right one, and get the dialtone, then I try to dial to make a call, it goes to fast-busy instead of making the call`
`15:33.29`	`Blackthorn`	`Hello. I have been over on the fedora room. I was trying to diagnose why I have to reset my fc3 asterick box every 3 days or so. Told me about a program called "top" i show two mpg123 loaded...loaded twice i guess?`
`15:33.56`	`mdave`	`dabba woah thanks`
`15:34.06`	`stse`	`Can I escape the # key, so asterisk send it to the called person?`
`15:34.58`	`*** join/#asterisk Pinnen (i=pinnen@jultomten.luktar.bajs.nu)`
`15:35.21`	`*** join/#asterisk xianlp (n=xian_1@193.170.41.114)`
`15:36.12`	`kink0`	`jpablo: ya la pregunta mas tonta del mundo... por si las moscas... tu cable es cruzado, no ?`
`15:36.23`	`*** join/#asterisk jimmy_deanPB_ (n=jhodapp@indianalifesciences.com)`
`15:36.30`	`kink0`	`o sea 2<->3 5<->6 , no ?`
`15:36.30`	`jpablo`	`kink0, no, es recto`
`15:36.47`	`kink0`	`co�o !! como que es recto ? 1<->1 .... 8<->8 ?`
`15:36.51`	`*** join/#asterisk batphone (n=will@69.15.174.114)`
`15:37.11`	`batphone`	`after one ring my call queue hangs up on the caller`
`15:37.16`	`kink0`	`vale.. vale.. a probar con tu misma config y un cable como el tuyo, a ver si consigo ver el PRI arriba`
`15:37.17`	`Bambr`	`is it possible to put some queue working only on some certain period of time, for example, from 9 pm till 2 am ?`
`15:37.38`	`kink0`	`o sea, recto quieres decir pin a pin, no ?`
`15:37.39`	`*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-61.nas28.salt-lake-city1.ut.us.da.qwest.net)`
`15:37.46`	`jpablo`	`kink0, el cable de red del balun a la tarjeta del e1 es un cable normalito de red, con la configuraci�n B`
`15:38.14`	`malverian[work]`	`Weird...`
`15:38.21`	`malverian[work]`	`My timezone for voicemails is off....`
`15:38.32`	`malverian[work]`	`It's playing at about +5 hours.`
`15:38.36`	`malverian[work]`	`(When it lists the time)`
`15:38.45`	`jpablo`	`kink0, oye, como marco con la soluci�n del v360, le envio el dtmf por la tarjeta de sonido al tel�fono ?`
`15:39.00`	`kink0`	`noooooooooooo !!! at+d<numero>; para voz`
`15:39.31`	`kink0`	`jpablo: eso abres un sock con lo que quieras ( Perl, C, etc. ) al puerto /dev/ttyACM0 - o donde lo tengas -`
`15:39.37`	`jpablo`	`kink0, entonces lo conecto al serial ?`
`15:39.41`	`kink0`	`y le envias el PIN, lo apagas, enciendes, etc.`
`15:40.03`	`malverian[work]`	`Pssst..`
`15:40.17`	`malverian[work]`	`Any ideas why my timezone would be weird in asterisk? The time is correct on the server box..`
`15:40.30`	`malverian[work]`	`But it plays 5 hours later when I'm listening to my voicemail.`
`15:40.37`	`Err`	`are you in EST?`
`15:40.45`	`mdave`	`i think asterisk uses UTC`
`15:40.54`	`Err`	`sure sounds like it does`
`15:41.04`	`Hmmhesays`	`ok ithink i've gone retarded route { forward( 1.2.3.4, 5060 ); } should work in ser`
`15:41.05`	`mdave`	`but im a newb, so im just pontificating`
`15:41.18`	`malverian[work]`	`EST yeah.`
`15:41.19`	`Weezey`	`Anyone have cisco 7940 SCCP firmware?`
`15:41.20`	`mdave`	`no idea how you control timezones in *`
`15:41.30`	`Err`	`malverian[work]: if you're in EST, then it's definitely using GMT/UTC`
`15:41.40`	`malverian[work]`	`Can I not change the timezone somehow? I set a [zonemessages] eastern=`
`15:41.44`	`jpablo`	`kink0, entonces tengo que conectarlo a la tarjeta de sonido y al puerto serial o usb ?`
`15:41.55`	`Weezey`	`there's timezone stuff in voicemail.conf`
`15:42.01`	`kink0`	`jpablo: si, por eso lo del v360, porque son conectores separados y no tienes que manipular el cable`
`15:42.06`	`af_`	`anyone using spandsp with bristuff?`
`15:42.21`	`Err`	`malverian[work]: looking at the voicemail.conf file, it looks like you want tz=eastern`
`15:42.22`	`kink0`	`en realidad, te vale cualquier terminal, yo empec� el proyecto con un viejo ericsson t56`
`15:42.28`	`jpablo`	`kink0, ya veo.`
`15:42.30`	`malverian[work]`	`eastern=America/New_York\|'vm-received' Q 'digits/at' IMp`
`15:42.32`	`malverian[work]`	`tz=eastern`
`15:42.34`	`malverian[work]`	`Doesn't help.`
`15:42.42`	`Err`	`in general?`
`15:42.45`	`kink0`	`jpablo: a ti no se te queja tu asterisk de tu linea pridialplan=unknow ?`
`15:42.54`	`*** join/#asterisk xeet2 (n=xeet3@bwi1-br1-gig2-1.jsci.net)`
`15:42.56`	`kink0`	`a mi se me queja y revienta, porque le falta la "n" al final`
`15:43.04`	`malverian[work]`	`Err, tz is in general, the eastern= is in zonemessages`
`15:43.13`	`xeet2`	`does anyone have any TE210Ps or know where I can get one shipped out today for saturday delivery in the US?`
`15:43.38`	`Err`	`that's interesting, because that looks like what you should do`
`15:43.42`	`malverian[work]`	`Err, "show voicemail zones" works.`
`15:43.43`	`*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)`
`15:43.52`	`malverian[work]`	`Err, yeah, the wiki claimed that that was the process...`
`15:44.20`	`RoyK`	`~seen OrdeJ`
`15:44.33`	`jbot`	`ordej is currently on #asterisk (38m 17s). Has said a total of 2 messages. Is idling for 23m 3s, last said: ' Hi. I'm totaly new to linux and asterisk. One stupid question - i install asterisk@home on windows and vmplayer... after i logon on root - how to switch to GUI?'.`
`15:45.10`	`*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)`
`15:45.27`	`Err`	`or how you log into windows without a GUI :-)`
`15:45.30`	`OrdeJ`	`with virtual machine...`
`15:45.36`	`MattB2_`	`help - got an alarm system that's sending DTMF tones down via an ATA into asterisk. asterisk is setecting the tones and re-sending in its own timing. The ATA i'm using is set to inband - how do i stop asterisk doing this?`
`15:45.40`	`RoyK`	`ouch`
`15:45.48`	`RoyK`	`asterisk in vmware is NOT a good idea`
`15:45.53`	`RoyK`	`NOT NOT NOT NOT NOT`
`15:45.58`	`OrdeJ`	`just for test`
`15:46.02`	`RoyK`	`use a dedicated box without X`
`15:46.04`	`Err`	`I'm sure the timings will work very well`
`15:46.40`	`RoyK`	`yeah`
`15:46.41`	`malverian[work]`	`Okay, this is my production server...`
`15:46.48`	`malverian[work]`	`I really could use a hand trying to figure this out.`
`15:46.52`	`mdave`	`why on earth anyone would run a *nix OS inside vmware on a wintoy os box is beyond me`
`15:46.57`	`malverian[work]`	`How can I change the default time zone for voicemail?`
`15:46.57`	`OrdeJ`	`ithink gui config has to be from diferent computer`
`15:46.58`	`RoyK`	`at least within the milli^Wsecond`
`15:47.05`	`mdave`	`like installing a porshe emulator in a yugo`
`15:47.08`	`*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`15:47.21`	`Err`	`mdave: good analogy`
`15:47.23`	`jpablo`	`kink0, no, de hecho esa configuraci�n me la pidio mi proveedor. estaba mandadole el tipo de numero como national y ellos lo quer�an como unknow, si no no funcionaba`
`15:47.24`	`OrdeJ`	`yugo is good car`
`15:47.25`	`MrChimpy`	`we had vmware esx (linux based) running asterisk fine for SIP testing`
`15:47.41`	`mdave`	`that it may be, but its not even comparable to a porsche`
`15:47.41`	`RoyK`	`OrdeJ: and windows is an excellent OS`
`15:47.45`	`mdave`	`LOL`
`15:47.49`	`MrChimpy`	`obv. for production it's insanity itself, but that doesn't mean it doesn't work`
`15:47.56`	`malverian[work]`	`origdate=Fri Jan 27 10:19:47 AM EST 2006`
`15:47.56`	`malverian[work]`	`origtime=1138375187`
`15:48.09`	`malverian[work]`	`So why does it say "3:19PM" when I listen to the message?`
`15:48.13`	`malverian[work]`	`This is ludicrous..`
`15:48.23`	`mdave`	`of course i cant see any reason to run a winty OS in the first place.. so that may be shaping my perceptions`
`15:48.32`	`Err`	`yugos might be decent now - they sucked when they were imported into the US some 20 years ago`
`15:48.43`	`Err`	`or, at least, the ones imported to the US sucked - maybe not all did`
`15:48.44`	`mdave`	`i dont think yugo is made anymore`
`15:48.47`	`Err`	`sure it is`
`15:48.51`	`Err`	`by Zastava`
`15:48.55`	`mdave`	`isc`
`15:48.58`	`Underhand`	`zastava still suck.`
`15:48.59`	`Err`	`they make cars and guns`
`15:49.02`	`Underhand`	`but not as much.`
`15:49.08`	`xeet2`	`do they make cars with guns?`
`15:49.12`	`malverian[work]`	`...`
`15:49.12`	`MrChimpy`	`skodas are the ones that improved`
`15:49.13`	`Hmmhesays`	`<PROTECTED>`
`15:49.19`	`malverian[work]`	`I really regret using Asterisk sometimes.`
`15:49.19`	`Err`	`probably - they're a state-run factory :-)`
`15:49.27`	`OrdeJ`	`better guns than cars...`
`15:49.29`	`[TK]D-Fender`	`malverian[work] : You didn't set your GMT offset I'm betting...`
`15:49.32`	`MrChimpy`	`they're owned by VW now and most of the components are VW`
`15:49.33`	`Err`	`malverian[work]: you get what you pay for - if you pay somebody to fix it, it'll be fixed ;-)`
`15:49.37`	`Tribastian`	`bye`
`15:49.42`	`Underhand`	`zastava are cheap cheap cheap, and it shows.`
`15:49.43`	`*** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu)`
`15:49.45`	`xeet2`	`malverian: what is your timezone set to when you type "date" in your shell`
`15:49.50`	`cpm`	`Err, that's not always so.`
`15:50.04`	`austinnichols101`	`I'm getting echo on some calls over our PRI. Should I be looking at tuning the gain as with a POTS line? I'm having a bit of trouble getting my mind around echo + digital circuit`
`15:50.11`	`mdave`	`Dabba, you finding the same problem I did and trying to figure out the cause, or are you not getting the same problem and trying to figure out why not?`
`15:50.12`	`Err`	`cpm: I'm sure that, for some price, someone would fix the source if it were a bug, and fix the config file if it isn't`
`15:50.13`	`malverian[work]`	`[TK]D-Fender, Okay, how the F do I do that?`
`15:50.13`	`Dabba`	`>mdave your right identical behaviour`
`15:50.17`	`[TK]D-Fender`	`iCEBrkr : Sorry for the late response, wassup?`
`15:50.24`	`cpm`	`Ahh, okay,`
`15:50.31`	`[TK]D-Fender`	`malverian[work] : pastebin your voicemail.conf`
`15:50.38`	`kink0`	`xeet2: voicein@aol.com and ask for MArc, he will supply you in USA for Digium cards`
`15:50.40`	`mdave`	`Dabba, well I feel better that someone else is seeing the same thing.. now, any idea what might be causing that`
`15:50.58`	`cpm`	`but sometimes folks say they will fix something in x-amount of time, but then they have cashed the check, and you're still on your own.`
`15:51.15`	`kink0`	`jpablo: no , esto sigue sin dar zap OK, sigue RED`
`15:52.15`	`mdave`	`of course that you are seeing it too makes me wonder if this isnt a bug in asterisk itself,`
`15:52.17`	`xeet2`	`kink0: thanks`
`15:52.21`	`malverian[work]`	`[TK]D-Fender, http://pastebin.ca/38798`
`15:52.31`	`xeet2`	`malverian: what is your timezone set to when you type "date" in your shell`
`15:52.42`	`malverian[work]`	`xeet2, My timezone at shell is correct.`
`15:52.50`	`MattB2_`	`any ideas how i stop asterisk detecting and resending dtmf? :(`
`15:52.53`	`malverian[work]`	`Fri Jan 27 10:52:44 EST 2006`
`15:52.55`	`Peste`	`hello again`
`15:52.58`	`Peste`	`is kram here?`
`15:53.03`	`kink0`	`TE4/0/1 "T4XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED`
`15:53.05`	`malverian[work]`	`This was _NOT_ an issue with Asterisk-1.2.0`
`15:53.16`	`malverian[work]`	`I haven't changed my voicemail.conf at all..`
`15:53.21`	`Err`	`MattB2_: do you need to ever detect DTMF?`
`15:53.30`	`MattB2_`	`no i don't`
`15:54.00`	`Dabba`	`it is odd it doesnt give any console output`
`15:54.24`	`Peste`	`can somebody help me with this: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1`
`15:54.24`	`MattB2_`	`all our SIP phones are set to INFO, so never need to manually detect inband DTMF`
`15:55.06`	`jpablo`	`kink0, trata de invertir los cables coaxiales`
`15:55.46`	`malverian[work]`	`Unbelievable...`
`15:55.59`	`xeet2`	`anyone aware of any issues with taking an existing PRI connected to an avaya ip office pbx, plugging it in to 1 port on a TE210P, and then plugging the avaya into the other port, running pri-cpe on the pri port and pri-net on the pbx port, and passing calls between the pri and the avaya?`
`15:56.07`	`malverian[work]`	`[TK]D-Fender, Don't tell me I have to manually define tz= for each voicemail user..`
`15:56.08`	`mdave`	`Dabba, ok, just for grins, I added a dialplan to dial out to disa, and it works`
`15:56.23`	`xeet2`	`malverian: you could always call digium`
`15:56.23`	`*** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net)`
`15:56.26`	`Err`	`MattB2_: in sip.conf you have dtmfmode=info, and it still detects inband?`
`15:56.29`	`xeet2`	`malvarian: and use paid support`
`15:56.36`	`xeet2`	`malvarian: they've always been really helpful`
`15:56.42`	`Err`	`or fix the source :-)`
`15:56.43`	`kink0`	`ahhh jooo olvid� cambiar el cable !!! , tu me dices que est�s usando en el E1 un cable ethernet de red ??`
`15:56.54`	`kink0`	`un rj45 de la ethernes ?`
`15:56.56`	`Err`	`you might for kicks try a different timezone, and see if that works`
`15:56.57`	`mogorman`	`malverian[work], hows it going`
`15:57.00`	`MattB2_`	`err: i have it set to inband...`
`15:57.06`	`Err`	`interesting`
`15:57.10`	`MattB2_`	`err: ah guess that's why it's detecting lol!!!!`
`15:57.11`	`Dabba`	`mdave`
`15:57.11`	`xeet2`	`malvarian: and with the fact that you're getting functionality of a high-end pbx with asterisk, you can't complain about having to pay for support`
`15:57.13`	`Dabba`	`it was my fault`
`15:57.14`	`malverian[work]`	`mogorman, Wanting to rip my head off.`
`15:57.21`	`Dabba`	`i typo'd the context in disa`
`15:57.24`	`jpablo`	`kink0, as� es.`
`15:57.26`	`Err`	`MattB2_: that will be $5 ;-)`
`15:57.31`	`MattB2_`	`lol`
`15:57.32`	`mdave`	`Dabba, oh? yours works now?`
`15:57.36`	`Dabba`	`yes`
`15:57.39`	`mdave`	`Hrm`
`15:57.42`	`xeet2`	`malvarian: seriously, call them if you're in a pinch, they'll get it figured out right away`
`15:57.43`	`Dabba`	`wanna pastebin of it`
`15:57.50`	`mdave`	`sure`
`15:57.53`	`malverian[work]`	`xeet2, I don't need support. I just get pissed off when things change without notice and require me to manually dabble in the source code to figure out what's going on.`
`15:57.57`	`mdave`	`just for foo, i'll doublecheck my typing`
`15:58.09`	`mogorman`	`eep`
`15:58.11`	`mdave`	`i will feel exceedingly dumb if I made a typo and still hadnt found it`
`15:58.13`	`xeet2`	`malvarian: and this is different than all other open source software how?`
`15:58.28`	`[TK]D-Fender`	`malverian[work] : add "\|tz=[timezonename]" in the parameters for your people based on the entries in [zonemessages] I believe`
`15:58.32`	`Err`	`I know I get angry when something I get for free doesn't work quite right, and I deploy it without testing`
`15:58.40`	`*** join/#asterisk Jun_Wang (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net)`
`15:58.46`	`[TK]D-Fender`	`malverian[work] : Oh yeah... and "sorry" :)`
`15:59.01`	`*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`15:59.04`	`mdave`	`ok, no typo here, and still doesnt work`
`15:59.10`	`cpm`	`I hate it when software does exactly what I instructed it to do.`
`15:59.19`	`xeet2`	`cpm: it sucks doesn't it?`
`15:59.21`	`mdave`	`well perhaps i did something wrong.. maybe if I compare with yours i'll see something less obvious`
`15:59.29`	`cpm`	`it really does.`
`15:59.31`	`*** join/#asterisk bkw__ (n=brian@m010f36d0.tmodns.net)`
`15:59.32`	`*** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap)`
`15:59.32`	`xeet2`	`cpm: least you don't have to worry about that with ms code`
`15:59.35`	`Err`	`I hate it when software does what I told it, but I told it something stupid (which is often the case)`
`15:59.50`	`cpm`	`Yup, is pretty much always the case with me.`
`15:59.56`	`malverian[work]`	`[TK]D-Fender, So I have to do it for each user individually.`
`16:00.06`	`malverian[work]`	`[TK]D-Fender, How was this not a problem on asterisk 1.2.0 but it is in 1.2.1?`
`16:00.06`	`Err`	`egrep is your friend`
`16:00.13`	`*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)`
`16:00.14`	`[TK]D-Fender`	`malverian[work] : use the "default" as well`
`16:00.22`	`Blackthorn`	`Is setting the mod probe command and then calling asterisk in the rc.local file the best way to start asterisk up?`
`16:00.25`	`[TK]D-Fender`	`malverian[work] : Not sure... just going by what it says....`
`16:00.38`	`[TK]D-Fender`	`Blackthorn : Works for me....`
`16:00.51`	`Dabba`	`mdave > http://pastebin.ca/raw/38799`
`16:01.08`	`malverian[work]`	`[TK]D-Fender, Use the default?`
`16:01.14`	`xeet2`	`anyone aware of any issues with taking an existing PRI connected to an avaya ip office pbx, plugging it in to 1 port on a TE210P, and then plugging the avaya into the other port, running pri-cpe on the pri port and pri-net on the pbx port, and passing calls between the pri and the avaya?`
`16:01.20`	`xeet2`	`ie, timing issues`
`16:01.23`	`mdave`	`Dabba, hrm.. ok.. thats a bit different than the way I did it`
`16:01.30`	`xeet2`	`can I recover clock from pri, supply clock to pbx`
`16:01.37`	`malverian[work]`	`[TK]D-Fender, It seems like this option should be globally configurable somewhere...`
`16:01.54`	`Blackthorn`	`Fedner: ok thanks. i'm just going over my setup trying to figure out why i have to restart my server every 3 days. it's got plenty of spare ram and hd...`
`16:02.06`	`mdave`	`lemme try that`
`16:02.16`	`cpm`	`because you haven't set up a cron job to do it?`
`16:02.19`	`malverian[work]`	`I have about 600 voicemail accounts...This will be fun..`
`16:02.25`	`*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)`
`16:02.29`	`Hmmhesays`	`ok my radius module is farking everything up`
`16:02.58`	`*** join/#asterisk mtaht (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`16:03.44`	`cpm`	`morn'n mtaht`
`16:04.08`	`MattB2_`	`err: :(`
`16:04.38`	`Err`	`malverian[work]: use egrep`
`16:04.41`	`mdave`	`Dabba, hrm, isnt DISA supposed to play dialtone?`
`16:04.52`	`mdave`	`or do you dial 7777 blind?`
`16:04.53`	`iCEBrkr`	`[TK]D-Fender: Hey, I know it's like 'a bad thing'(tm) to run Samba on an Asterisk box. But if there's only one 'user' going to be saving to that drive, you think it'll affect it much?`
`16:05.00`	`austinnichols101`	`do PRI circuits need to be gain-tuned for echo?`
`16:05.17`	`Blackthorn`	`cpm: well i guess i could setup cron to reboot the system. But i'd rather fix the issue then to just patch. but guess if there is no answer I can defently give it a try.`
`16:05.27`	`Dabba`	`mdave i just dropped that .call into outgoing and it dials the sip 1001 and connects it to 7777`
`16:05.38`	`Dabba`	`which then gives dialtone`
`16:05.39`	`mdave`	`ok, and when you pick up 1001, then what?`
`16:05.42`	`mdave`	`oh`
`16:05.43`	`mdave`	`hrm`
`16:05.46`	`cpm`	`Blackthorn, would probably be better to find out what the problem is.`
`16:05.46`	`mdave`	`i didnt get dialtone`
`16:05.51`	`Err`	`iCEBrkr: it shouldn't be any worse than running any other server that gets frequent queries`
`16:06.00`	`iCEBrkr`	`Err: That's what I was thinking`
`16:06.03`	`cpm`	`but why reboot? is the machine hardlocking? crashing?`
`16:06.05`	`Dabba`	`well step 4 says "this si a callback" then i get dialtone :-)`
`16:06.09`	`iCEBrkr`	`Err: and really, all I'm doing is writing call files...`
`16:06.16`	`mdave`	`of course, i removed your 'background', since I dont have that file`
`16:06.26`	`malverian[work]`	`Err, Or sed..?`
`16:06.33`	`mdave`	`Dabba, yeah, i took that 'this is a callback' part out`
`16:06.37`	`Err`	`samba gets QUITE frequent queries, though, because of all of its broadcast crap - you can turn that off, but it's different for every windows version IIRC`
`16:06.40`	`Err`	`malverian[work]: sure, whatever`
`16:06.46`	`Err`	`the point is, it's not any work at all`
`16:06.52`	`mdave`	`Dabba, I assume that refers to a sound file`
`16:06.56`	`Dabba`	`mdave yes`
`16:06.56`	`mdave`	`which I dont have of course`
`16:07.02`	`mdave`	`but it should work without it, shouldnt it?`
`16:07.04`	`iCEBrkr`	`Then I'll just try to slim down Samba`
`16:07.06`	`Dabba`	`of course`
`16:07.08`	`mdave`	`hrm`
`16:07.12`	`Dabba`	`it just plays tone`
`16:07.25`	`Dabba`	`i then dialled a number and got connected`
`16:07.31`	`Dabba`	`'its good to talk'`
`16:07.43`	`*** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com)`
`16:07.43`	`mdave`	`er`
`16:07.44`	`mdave`	`oh`
`16:07.44`	`mdave`	`wait`
`16:07.50`	`mdave`	`i may have forgotten to reload extensions.conf`
`16:07.52`	`mdave`	`doh!`
`16:08.15`	`mdave`	`BINGO`
`16:08.20`	`Dabba`	`yay`
`16:08.34`	`mdave`	`now just to add a passcode to it`
`16:08.39`	`mdave`	`then`
`16:08.48`	`s34n`	`what is the best way to keep your Asterisk install up to date?`
`16:08.57`	`iCEBrkr`	`s34n: Don't...`
`16:09.09`	`*** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net)`
`16:09.10`	`Dabba`	`realise its not possible :-)`
`16:09.15`	`mdave`	`once im ready, find a way to have an incoming call (that isnt answered) trigger a script to put the .call file`
`16:09.30`	`iCEBrkr`	`Dabba: It's not that it's not possible, it's more like why would you??`
`16:09.31`	`[TK]D-Fender`	`iCEBrkr : I run X, KDE, XINE, Samba, ProFTPD, gateway routing through my S518, and my X-10 setup on my home server, so why not? :)`
`16:09.34`	*** join/#asterisk mtaht` (n=user@c-71-198-23-124.hsd1.ca.comcast.net)
`16:09.35`	`mdave`	`every 2 weeks, reformat your hd, and reinstall from scratch?`
`16:09.39`	`iCEBrkr`	`[TK]D-Fender: lol`
`16:09.52`	`iCEBrkr`	`Dabba: again, I point at the topic`
`16:10.22`	`iCEBrkr`	`Same here :)`
`16:10.44`	`iCEBrkr`	`I've always pretty much run 1 or 2 versions behind`
`16:10.44`	`mut`	`damn digium and their form mails`
`16:10.45`	`mdave`	`Dabba, anyway, I thankee for thy help muchly. And now that big blue room is calling me`
`16:10.49`	`[TK]D-Fender`	`I was on SVN somewhere between 1.2.0 and 1.2.1 I believe, then jumped to 1.2.3`
`16:10.56`	`mut`	`i just asked them about echo and everything i've done to try to correct it`
`16:11.02`	`morale`	`;exten => _XXX!,1,Macro(vmbox,${EXTEN}) <- that seems to match all phone numbers, can someone let me know if that is correct to match only 3 digits?`
`16:11.08`	`s34n`	`iCEBrkr: ok, so whats the best way to run 2 versions behind ;)`
`16:11.17`	`*** part/#asterisk mtaht (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`16:11.18`	`Dabba`	`mdave sweet`
`16:11.23`	`iCEBrkr`	`s34n: Check www.asterisk.org once in awhile?`
`16:11.28`	`mut`	`and he sends me instructions that are exactly what i've already done`
`16:11.37`	`iCEBrkr`	`s34n: read the change log..`
`16:11.53`	`[TK]D-Fender`	`morale : remove the "!"`
`16:12.05`	`*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj1h.dialup.mindspring.com)`
`16:12.59`	`s34n`	`I don't want to have to re-download the whole thing and recompile. I would prefer incremental upgrades.`
`16:13.07`	`kink0`	`jpablo: podrias hacer un pri intense debug span 1 ?`
`16:13.14`	`iCEBrkr`	`s34n: This isn't windows and you really DON'T want that.`
`16:13.16`	`kink0`	`no sale esto ni de co�a, ya he probado con tres cables y lo mismo`
`16:13.28`	`iCEBrkr`	`s34n: how hard is it to: make && make install?`
`16:13.41`	`DrDeke`	`Yeah, if you aren't running it on a Sparcstation 20 or something, it should not take all that long.`
`16:14.01`	`Err`	`yeah, use an IPX - they're smaller`
`16:14.07`	`DrDeke`	`:)`
`16:14.15`	`JonR800`	`how hard is it to download and recompile? unless this is on a Pentum 166 or something.`
`16:14.15`	`*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj1h.dialup.mindspring.com)`
`16:14.20`	mtaht`	`IPX was a great monitor stand.`
`16:14.29`	`Err`	`heh, mine is holding up an alarm clock`
`16:14.30`	`MrChimpy`	`guy I know did the work to get asterisk on solaris`
`16:14.40`	`s34n`	`iCEBrkr: you forgot 'make uninstall && make clean' ...`
`16:14.46`	`*** join/#asterisk masonf_ (n=masonf@dungle.vineyard.net)`
`16:14.53`	`iCEBrkr`	`s34n: oh, no. no that....: \|`
`16:15.06`	`batphone`	`but no rule 't' in context 'apptqueue'`
`16:15.07`	`JonR800`	`lol`
`16:15.08`	`batphone`	`wtf man..`
`16:15.10`	`batphone`	`just hangs up?`
`16:15.26`	`iCEBrkr`	`s34n: heaven forbid you actually have to maintain something`
`16:15.38`	`s34n`	`iCEBrkr: exactly! :)`
`16:15.45`	`*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)`
`16:15.46`	`*** join/#asterisk santiago (n=santiago@63.245.86.155)`
`16:16.01`	`Err`	`if you don't want to fool with it, choose a distro that distributes asterisk`
`16:16.15`	`twisted[asteria]`	`ugh.`
`16:16.16`	`TheGoD`	`I'm getting errors zt_rbs: tried to set RBS hook state 0 on channel WCTDM/0/1 while span WCTDM/0 lacks rbsbits or hooksig function`
`16:16.22`	`TheGoD`	`anyone know what that could be?`
`16:16.22`	`iCEBrkr`	`s34n: apt-get install asterisk`
`16:16.33`	`twisted[asteria]`	`Err, and HOPE TO GOD they keep up with bugfixes and critical updates.`
`16:16.37`	`QbY`	`is there a function similar to "left" -- ie. i need to see if the first 3 numbers in caller id are a particular area code...?`
`16:16.41`	`iCEBrkr`	`s34n: So when shit blows up, you're screwed messing around trying to fix it... Just download and recompile.. It's the safe way`
`16:16.46`	`*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)`
`16:17.36`	`Egonis`	`When trying to modprobe zaptel, I get a series of errors such as class_simple_create -- I am using kernel-2.6.15-gentoo-r1 and udev`
`16:17.55`	`Err`	`twisted[asteria]: I didn't say it was a good idea - I just said it was the easy solution ;-)`
`16:18.00`	`[TK]D-Fender`	`s34n : Keep in mind the true meaning of "bleeding edge" <- If you went on 1.2.2 and the APT repos weren't updated.... you'd be hiding your head between you legs....`
`16:18.25`	`[TK]D-Fender`	`QbY : ${EXTEN:0:3}`
`16:18.57`	`kink0`	`jpablo: tu tienes una Digium TExxx o otra cosa ?`
`16:18.58`	`*** part/#asterisk santiago (n=santiago@63.245.86.155)`
`16:19.20`	`[TK]D-Fender`	`QbY : Or for callerid : ${${CALLERID(number)}:0:3}`
`16:19.58`	`QbY`	`[TK]D-Fender: Where can i find a good description of $EXTEN and 0:3`
`16:20.27`	`Dabba`	`README.variables`
`16:22.00`	`QbY`	`don't have README.variables`
`16:22.16`	`[TK]D-Fender`	`QbY : Check the WIKI.`
`16:22.46`	`[TK]D-Fender`	`QbY : its all very nicely described in there`
`16:23.37`	`s34n`	`iCEBrkr: so how do you clean out the last install before you go through a new install?`
`16:23.55`	`austinnichols101`	`anyone know if gain tuning is required for a PRI (I'm trying to troubleshoot an echo problem)`
`16:24.02`	`iCEBrkr`	`s34n: Dude, live by this rule "If it's not broken, don't fix it"`
`16:24.26`	`iCEBrkr`	`s34n: Really the only time I 'upgrade' is when there's a new feature I'd like to have or there's a fix for something I've been using a workaround for.`
`16:24.32`	`[TK]D-Fender`	`QbY : README.variables is in the "doc" folder in your source folder`
`16:24.36`	`_Sam--`	`unless you are bored and have a lot of extra hair to work with`
`16:25.00`	`QbY`	`[TK]D-Fender - Like ${EXTEN} it says, "The current extension." Would that be the phone laying on my desk? That would always return 203.. However, I see people scriptiong ${EXTEN} and it looks like they are talking about the number that was just dialed....`
`16:25.02`	`[TK]D-Fender`	`s34n : You don't want to auto-upgrade.... things BREAK. Depreciated features could cripple your dialplans, etc....`
`16:25.17`	`kink0`	`anybody has a digium TExxx and the other end is NT/M PRI ? With that parameters I always get RED alarm, and I have tryed with several cables also.`
`16:25.23`	`jpablo`	`kink0: una digium TE210P`
`16:25.42`	`kink0`	`jpablo: joer... la mia no se levanta ni con viagra, est� todo el rato en RED`
`16:25.51`	`TheGoD`	`I'm getting errors zt_rbs: tried to set RBS hook state 0 on channel WCTDM/0/1 while span WCTDM/0 lacks rbsbits or hooksig function, Asterisk isn't dialing the line, and it does some weird stuff when it picks up`
`16:25.52`	`[TK]D-Fender`	`QbY : ${EXTEN} is the number that was dialed. If you dial a number from a phone on your PBX, it holds that number and is what is used to match up against your phones context`
`16:26.02`	`kink0`	`jpablo: anda dime si un debug al pri te larga algo como esto:`
`16:26.05`	`Hmmhesays`	`anyone have SER working with auth_radius?`
`16:26.16`	`kink0`	`M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended)`
`16:26.27`	`Hmmhesays`	`i can't get the module to load`
`16:26.36`	`s34n`	`[TK]D-Fender: I have mucked around for many versions, but never gone production with this box yet.`
`16:26.40`	`kink0`	`es que est� como si no hubiera cable ninguno entre la digium y el 2N`
`16:27.04`	`s34n`	`[TK]D-Fender: I figure I might as well get current before I start laying out the dial plane, etc.`
`16:27.08`	`kink0`	`jpablo: por cierto, tu PRI Telco de donde viene ? de telefonica o similar ? o es otra cosa ?`
`16:27.34`	`[TK]D-Fender`	`s34n : Ok so you aren't even really running * yet?`
`16:27.39`	`s34n`	`[TK]D-Fender: So I'd like to start clean`
`16:28.05`	`s34n`	`[TK]D-Fender: It's been running as a back-closet hobby for this site.`
`16:28.39`	`Damin`	`Morning..`
`16:28.41`	`s34n`	`[TK]D-Fender: but I'd like to wipe and start clean (without wiping the OS, etc.)`
`16:28.57`	`*** join/#asterisk eKo1 (n=bernd@207.42.191.67)`
`16:30.03`	`[TK]D-Fender`	`s34n : pastebin your current extensions.conf and we'll see how big a mess you're in.`
`16:30.06`	`[TK]D-Fender`	`~pb`
`16:30.09`	`jbot`	`pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`16:30.28`	`s34n`	`[TK]D-Fender: I'm going to wipe the extensions.conf and start over`
`16:30.48`	`s34n`	`[TK]D-Fender: there is legacy PRI and SIP stuff in there that has been abandoned.`
`16:30.57`	`[TK]D-Fender`	`s34n : Ok, describe your projected setup (hardware & technologies involved)`
`16:31.15`	`*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`16:32.08`	`s34n`	`[TK]D-Fender: It is reduced to fairly simple: 1 * server, 1 soft-switch, a handful of SIP phones, and a bunch of PSTN extensions`
`16:32.34`	`s34n`	`[TK]D-Fender: I'm not too worried about re-writing it.`
`16:32.54`	`[TK]D-Fender`	`what "soft-switch", what kind of phones, and how are you bringing in PSTN phones 9not LINES, right?)`
`16:33.02`	`s34n`	`[TK]D-Fender: I just want to wipe asterisk and put up the latest version before I re-configure.`
`16:34.06`	`s34n`	`[TK]D-Fender: I have a SIP connection to a Metaswitch. The PSTN phones are accessible through the metaswitch.`
`16:34.10`	`[TK]D-Fender`	`Then just Dl stable right from digium's FTP. 1.2.3 seems to be working rather well at this point`
`16:34.39`	`[TK]D-Fender`	`s34n : And for PSTN connectivity?`
`16:34.44`	`s34n`	`[TK]D-Fender: I did dl. I want a clean uninstall before I do a new install`
`16:36.27`	`[TK]D-Fender`	`s34n : You can just compile right over without too many problems. Just un-tar, "make clean", "make", "make install", and you should be good to go from there.`
`16:36.51`	`[TK]D-Fender`	`s34n : and if you don't have a config to lose and want an aneurism "make samples"`
`16:37.50`	`*** join/#asterisk Tili (i=Tili@203.101.160.47)`
`16:39.51`	`slan`	`How can I get rid of "funny" Russian looking characters on the alt-F9 console? They are the ones in red or blue, normal text is fine.`
`16:40.32`	`eKo1`	`the what now?`
`16:41.24`	`slan`	`eKo1: The alt-F9 console is where you see progress messages in Asterisk@Home. Same on normal Asterisk?`
`16:41.49`	`eKo1`	`oh, sorry i don't use @home`
`16:41.59`	`TheGoD`	`I'm getting errors zt_rbs: tried to set RBS hook state 0 on channel WCTDM/0/1 while span WCTDM/0 lacks rbsbits or hooksig function, Asterisk isn't dialing the line, and it does some weird stuff when it picks up`
`16:42.17`	`slan`	`eKo1: You don't have an alt-F9 console to see progress messages in Asterisk?`
`16:42.22`	`Hmmhesays`	`file oh file where art thou`
`16:42.50`	`*** join/#asterisk Skkip (n=Skipper@216.160.91.91)`
`16:42.52`	`eKo1`	`slan: i just increase the verbosity and that's about it`
`16:43.07`	`file`	`WHAT`
`16:43.28`	`slan`	`eKo1: I've never messed with verbosity. What setting do you use?`
`16:43.39`	`jpablo`	`kink0, sorry, estaba en otro lado. mi e1 es de AT&T en mexico, y no, no me pone ese error que dices`
`16:43.53`	`*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`16:43.59`	`*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`16:44.06`	`*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`16:44.13`	`eKo1`	`set verbose 3`
`16:44.20`	`eKo1`	`covers everything imo`
`16:44.24`	`slan`	`eKo1: 3 - thanks.`
`16:44.54`	`slan`	`Anyone else use the alt-F9 console to see scrolling progress messages?`
`16:46.27`	`Hmmhesays`	`file have you ever used auth_radius?`
`16:46.37`	`jarrod`	`how can ser have 99% cpu free and 500meg memory free and have a load over 1.00 ?`
`16:46.43`	`file`	`Hmmhesays: no`
`16:47.00`	`Hmmhesays`	`alrighty`
`16:47.12`	`Hmmhesays`	`i cannot get it to load for the life of me`
`16:47.31`	`*** join/#asterisk elg (n=fugalh@falcon.fugal.net)`
`16:49.11`	`*** join/#asterisk bmg505 (n=leon@dsl-146-7-214.telkomadsl.co.za)`
`16:50.07`	`xachen`	`SER i have never had luck with`
`16:50.14`	`xachen`	`maybe because of lack of patience`
`16:51.00`	`file`	`Hmmhesays: what's it do?`
`16:51.13`	`file`	`xachen: lots of patience and a good understanding of SIP is very helpful`
`16:51.29`	`*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)`
`16:52.18`	`xachen`	`your telling me I need to knw the SIP forwards, backwards, inwards and every other way? :(`
`16:52.38`	`file`	`it helps ... a lot`
`16:52.38`	`Hmmhesays`	`file: all i get on the console is "error initializing module"`
`16:52.48`	`file`	`Hmmhesays: turn up the debug`
`16:53.16`	`file`	`Hmmhesays: set log_stderror to yes, and set fork to no`
`16:53.20`	`file`	`and debug to 3 or 4`
`16:54.46`	`Hmmhesays`	`<PROTECTED>`
`16:55.12`	`file`	`use what you get to debug, cause I am out of here for 20-25 minutes`
`16:55.17`	`Hmmhesays`	`WARNING: no fork mode`
`16:55.17`	`Hmmhesays`	`stateless - initializing`
`16:55.17`	`Hmmhesays`	`textops - initializing`
`16:55.17`	`Hmmhesays`	`ERROR: error while initializing modules`
`16:55.19`	`Hmmhesays`	`that is it`
`16:55.45`	`Hmmhesays`	`which is odd`
`16:56.02`	`*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)`
`16:56.07`	`*** part/#asterisk HamYaII (i=HamYai@125.24.3.163)`
`16:56.36`	`*** join/#asterisk HamYaI (i=HamYai@125.24.3.163)`
`16:56.49`	`Supaplex`	`jarrod: my box has a 3.0 load because three instances of xsane will never die (apparent deadlock waiting for a blockread)`
`16:57.01`	`*** join/#asterisk pifiu (n=myassisb@208.205.181.170)`
`16:57.12`	`*** join/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il)`
`16:57.17`	`pifiu`	`morning everyone`
`16:57.30`	`Supaplex`	`still gah :p`
`16:57.39`	`Romik_`	`somebody can advice about this problem 1.2.3? Jan 27 12:56:39 WARNING[25014]: chan_zap.c:4335 __zt_exception: We're Zap/50-1, not SIP/213.177.5.131-08265218 ?`
`16:58.21`	`xachen`	`what is with people just joining in and shoving questions in our face without a simple good morning?`
`16:58.33`	`HamYaI`	`is it necessary to specify "sip 5060/udp" in /etc/services for FC3?`
`16:58.54`	`Romik_`	`xachen: good time of day....i have evening 6:0pm`
`16:59.11`	`Underhand`	`xachen: not speaking for here in particular, but in many channels, people get flamed for saying good morning and not just getting on with the question asking.`
`16:59.30`	`xachen`	`I guess ^_^`
`16:59.33`	`MrChimpy`	`:)`
`16:59.35`	`*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)`
`16:59.42`	`rob0`	`I prefer it when people "just ask"`
`16:59.44`	`MrChimpy`	`it's evening here anyway :)`
`16:59.52`	`sulex`	`there's no way to slow down a "SAY DIGITS" agi command right? If so, better to use it in loop with some "sleep" or "noop" or modifying the sounds to have more breathe?`
`17:00.00`	`Peste`	`can somebody help me with this: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1`
`17:00.05`	`Underhand`	`if there's a concensus on this preference, might i suggest putting it in the FAQ? :)`
`17:00.12`	`s34n`	`[TK]D-Fender: "Your Asterisk modules directory contains modules that were not installed by this version of Asterisk."`
`17:00.38`	`*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`17:00.42`	`*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`17:00.57`	`arkanis`	`what can I do, when asterisk sais "loop detected"`
`17:00.58`	`s34n`	`including a bunch of chan_modem stuff`
`17:01.09`	`Romik_`	`somebody can advice about this zaptel problem? (asterisk 1.2.3) Jan 27 12:56:39 WARNING[25014]: chan_zap.c:4335 __zt_exception: We're Zap/50-1, not SIP/213.177.5.131-08265218 ?`
`17:01.39`	`*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`17:03.15`	`[TK]D-Fender`	`s34n : "rm -f /usr/lib/modules/asterisk"`
`17:03.24`	`[TK]D-Fender`	`s34n : then recompile everything`
`17:04.04`	`malverian[work]`	`WOW...`
`17:04.08`	`malverian[work]`	`I hate snom phones so much..`
`17:04.27`	`wunderkin`	`no you don't need to recompile, just do a make install, if you had any 3rd party stuff then you have to recompile/install those`
`17:04.37`	`malverian[work]`	`If you press a function button on the phone while a call is coming in it bridges the button function to the incoming call... wow.`
`17:04.45`	`malverian[work]`	`Just f*cking wow..`
`17:04.50`	`*** join/#asterisk kio (n=kio@195-11.customer.cloud9.net)`
`17:05.00`	`[TK]D-Fender`	`malverian[work] : Its CRAPTASTIC!`
`17:05.30`	`malverian[work]`	`[TK]D-Fender, You wouldn't believe how much crap I've had to go through to make these phones function in a remotely sane manner..`
`17:05.35`	`Supaplex`	`stays soggy, even in milk!`
`17:05.39`	`malverian[work]`	`And Snom's technical support is non-existant.`
`17:05.46`	`malverian[work]`	`I've waited weeks for simple support inquiries before...`
`17:06.10`	`malverian[work]`	`Then their excuse is "use support from one of our vendors" and then they can't name a vendor that provides support..`
`17:06.15`	`[TK]D-Fender`	`malverian[work] : You're right... I'm not that experienced with Snom (as in only hear-say), as I work extensively with Polycom, and Sipura right now.`
`17:06.55`	`malverian[work]`	`[TK]D-Fender, Sadly when I was choosing phones for our location, SNOM was the solution with the most programmable buttons..`
`17:07.08`	`TheGoD`	`I'm getting errors zt_rbs: tried to set RBS hook state 0 on channel WCTDM/0/1 while span WCTDM/0 lacks rbsbits or hooksig function, Asterisk isn't dialing the line, and it does some weird stuff when it picks up`
`17:07.10`	`[TK]D-Fender`	`OMG, a Grandsuck video phone is coming out!`
`17:07.12`	`malverian[work]`	`(With device hinting support)`
`17:07.45`	`*** join/#asterisk iq (n=iq@71-214-2-243.omah.qwest.net)`
`17:07.53`	`[TK]D-Fender`	`malverian[work] : Yeah, Hints are the one thing going for Snom right now. I've got a semi-crippled receptionist right now on a 601.....`
`17:08.20`	`*** join/#asterisk los415 (i=los415@los.race.com)`
`17:11.01`	`*** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it)`
`17:12.07`	`*** join/#asterisk copantl_ (n=copantl@205.240.200.93)`
`17:12.21`	`*** join/#asterisk diego_br (n=diego@200.208.241.178)`
`17:13.56`	`TheGoD`	`asterisk will not pickup my phone lines I have a tdm400 with 4 red cards. Anyone have any suggestions?`
`17:14.04`	`TheGoD`	`It SAYS it picks up`
`17:14.07`	`TheGoD`	`but really doesn't`
`17:15.36`	`TheGoD`	`It used to work, I rebooted (this was a while ago) and now it doesn't`
`17:16.07`	`cpm`	`reboot? or hardware power off, power on?`
`17:16.17`	`cpm`	`<PROTECTED>`
`17:16.18`	`cpm`	`?`
`17:16.43`	`TheGoD`	`When I call from normal line to a line on asterisk box the sip phone rings, I pick up and all I hear are sounds of more ringing, THe normal phone also still rings`
`17:16.48`	`TheGoD`	`shutdown -r now`
`17:18.01`	`*** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)`
`17:18.10`	`TheGoD`	`this is not an asterisk@home install, its a real install`
`17:18.18`	`TheGoD`	`its driving me crazy`
`17:19.17`	`*** join/#asterisk marv[work] (n=timr@64.89.118.139)`
`17:19.34`	`Romik_`	`somebody can advice about this zaptel problem? (asterisk 1.2.3) Jan 27 12:56:39 WARNING[25014]: chan_zap.c:4335 __zt_exception: We're Zap/50-1, not SIP/213.177.5.131-08265218 ?`
`17:19.49`	`justinu`	`wow, neat`
`17:20.21`	`[TK]D-Fender`	`Romik_ : Mayeb you could describe WHEN that message pops up for us....`
`17:21.45`	`Blackthorn`	`thegod: i would try a power off restart first of all`
`17:21.58`	`malverian[work]`	`[TK]D-Fender, Are there any other phones with hints?`
`17:22.00`	`*** join/#asterisk Defraz (i=t0tal@72.165.56.43)`
`17:22.06`	`malverian[work]`	`[TK]D-Fender, Looks like Seimens makes some good ones...`
`17:22.22`	`justinu`	`wow, intel and HP dump $10 billion into itanium`
`17:22.30`	`justinu`	`when will they give up?`
`17:22.49`	`Romik_`	`<PROTECTED>`
`17:23.17`	`justinu`	`romik: that sounds like a bug, more than anything.`
`17:23.17`	`[TK]D-Fender`	`malverian[work] : well Polycom is "flawed" right now. There is FOP, and other similar tools, or you could write your own script (like I do for my polycom's MicroBrowser.`
`17:25.53`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`17:29.17`	`*** join/#asterisk roulduke_ (i=xmw8vqxb@p508D2E18.dip0.t-ipconnect.de)`
`17:30.27`	`*** join/#asterisk nmsclera (n=arthurh@67-42-132-116.albq.qwest.net)`
`17:31.11`	`fndude`	`Anybody use the Grandstream IP phones?`
`17:31.32`	`badboyz`	`anyone worked w/ the * call pickup feature? pickup a ringing call from 1 extension @ another extension?`
`17:32.07`	`cron`	`justinu: your talking about total development of the itanium?`
`17:32.10`	`cron`	`grrrr`
`17:32.27`	`s34n`	`[TK]D-Fender: :) my new asterisk won't start`
`17:32.36`	`[TK]D-Fender`	`s34n : Why not?`
`17:32.59`	`[TK]D-Fender`	`s34n : You might want to have some config files present... certain modules crap out if you don't...`
`17:33.16`	`s34n`	`loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_modem.so: cannot open shared object file: No such file or directory`
`17:33.49`	`[TK]D-Fender`	`s34n : put a "noload => chan_modem.so" into your modules.conf`
`17:34.09`	`*** join/#asterisk Assid (n=assid@203.115.64.14)`
`17:34.11`	`Assid`	`heya`
`17:34.25`	`Assid`	`anyone using voipjet? is there an issue taking place?`
`17:35.06`	`*** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)`
`17:35.23`	`*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`17:35.38`	`*** join/#asterisk muzzz_ (n=chatzill@218.111.66.117)`
`17:39.41`	`Romik_`	`justinu: What I can do to fix it?`
`17:39.50`	`junbug`	`Assid: check their site I know they chge a router or 2`
`17:40.27`	`Hmmhesays`	`voipjet is working for me`
`17:40.52`	`Romik_`	`<PROTECTED>`
`17:41.03`	`*** join/#asterisk zoa (n=kkk@pirus.securax.be)`
`17:41.45`	`Romik_`	`to whom i can contact about this zaptel problem?`
`17:42.17`	`Assid`	`hrmm.. i tried their west coast server`
`17:42.18`	`Assid`	`doesnt work`
`17:42.55`	`Hmmhesays`	`64.34.45.100`
`17:43.01`	`Hmmhesays`	`that one is working for me`
`17:43.50`	`s34n`	`any recommendations on meetme moderation software?`
`17:44.01`	`*** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net)`
`17:44.06`	`*** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net)`
`17:44.06`	`Assid`	`Hmmhesays: calls arent going through that one for me`
`17:44.07`	`*** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net)`
`17:44.17`	`FarrisG`	`Anyone know if there's a debian package available for AMP?`
`17:46.19`	`_Sam--`	`i dont think there is`
`17:46.26`	`_Sam--`	`dont waste your time anyway`
`17:46.38`	`FarrisG`	`_Sam--: Really? Not worth it?`
`17:46.50`	`_Sam--`	`the CDR part and the flash operator panel are nice`
`17:47.02`	`_Sam--`	`but the GUI to try to config stuff is bunk`
`17:47.05`	`_Sam--`	`at least, in my opinion.`
`17:47.08`	`FarrisG`	`_Sam--: Are there any other tools worth using? It's just becoming very cumbersome for me to manage asterisk config.`
`17:47.31`	`Assid`	`http://pastebin.ca/38808`
`17:48.07`	`[TK]D-Fender`	`FarrisG : How big is this deployment?`
`17:48.43`	`Assid`	`Hmmhesays: check that out`
`17:49.15`	`FarrisG`	`[TK]D-Fender: Currently 60 users, average of 2 new ones a week`
`17:49.23`	`Assid`	`doesnt voipjet have a number to contact on`
`17:49.40`	`[TK]D-Fender`	`FarrisG : What do you use your * setup for? Basic company? Anything special?`
`17:50.04`	`*** join/#asterisk justinu (n=justin@72.18.13.34)`
`17:50.18`	`_Sam--`	`it might be easiest to just use realtime for your sip/iax peers/buddies`
`17:50.24`	`_Sam--`	`throw them in an sql table`
`17:50.24`	`FarrisG`	`[TK]D-Fender: Define special? It does all of our PBX, conferencing, DID, etc...`
`17:50.28`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@m010f36d0.tmodns.net)`
`17:51.17`	`[TK]D-Fender`	`FarrisG : So MeetMe, basic extensions, PRI, VM, Queues(have any? Need special reporting?)`
`17:52.08`	`TheGoD`	`My asterisk server running on a tdm400 with 4 red cards will not pickup the phone on any line. It says it is but it doesn't actually do it. Does anyone have any suggestions`
`17:52.10`	`FarrisG`	`[TK]D-Fender: No queues at the moment. Don't need special reporting. No billing or anything, mainly just call logs, CDR. It's just for an in-house system, no billing or anything`
`17:52.11`	`[TK]D-Fender`	`FarrisG : Single unified company? (not multi-tenant)`
`17:52.36`	`FarrisG`	`[TK]D-Fender: Correct, one company. Looking to setup a second site overseas soon, but that'll be a huge nightmare`
`17:52.39`	`[TK]D-Fender`	`FarrisG : If done right I don't see why this would need AMP....`
`17:52.49`	`tainted_`	`anyone know docelmo?`
`17:53.03`	`_Sam--`	`amp WOULD work fine`
`17:53.09`	`FarrisG`	`[TK]D-Fender: I agree, but I'm trying to find a way to make managing users, etc. simpler, so that I can ramp up help we're hiring`
`17:53.24`	`_Sam--`	`but it would be easier to just put the sip users in an sql table`
`17:53.31`	`_Sam--`	`and then access the table with some easy front end`
`17:53.36`	`[TK]D-Fender`	`FarrisG : So you're running a "normal" install of * rightnow?`
`17:53.39`	`_Sam--`	`and use realtime`
`17:53.45`	`FarrisG`	`[TK]D-Fender: yes`
`17:54.05`	`badboyz`	`how do you check what version of * you are using?`
`17:54.29`	`mzo`	`asterisk -version :P`
`17:54.29`	`[TK]D-Fender`	`FarrisG : Care to pastebin your current extensions.conf for a quick peek at how you're doing it now?`
`17:55.46`	`FarrisG`	`[TK]D-Fender: How much of it?`
`17:58.28`	`badboyz`	`asterisk -V`
`17:58.28`	`badboyz`	`Asterisk`
`17:58.34`	`badboyz`	`it doesnt tell me the version =/`
`17:59.02`	`mzo`	`it should. :P`
`17:59.09`	`[TK]D-Fender`	`FarrisG : all of it... why not...`
`17:59.14`	`Assid`	`damn voipjet`
`17:59.17`	`Assid`	`i cant get through`
`17:59.24`	`Assid`	`to neither of their servers`
`17:59.32`	`mzo`	`oh it's 'show version'`
`17:59.52`	`mzo`	`you get some gibberish like this. Asterisk 1.2.3 built by root @ asterisk1.local on a i686 running Linux on 2006-01-26 22:46:24 UTC`
`17:59.52`	`mzo`	`:P`
`18:00.11`	`badboyz`	`Asterisk built by root@asterisk1.local on a i686 running Linux <<`
`18:00.13`	`badboyz`	`no version :(`
`18:00.16`	`mzo`	`really?`
`18:00.18`	`badboyz`	`yea`
`18:00.19`	`mzo`	`how weird.`
`18:00.25`	`badboyz`	`mines br0ke`
`18:00.32`	`nroej`	`hi all`
`18:00.41`	`mzo`	`when you connect to the shell it should tell you? exit out of it and do asterisk -r it should say it again?`
`18:00.45`	`benjk`	`benjk*CLI> show version`
`18:00.45`	`benjk`	`Asterisk 1.0.10`
`18:00.54`	`badboyz`	`=========================================================================`
`18:00.54`	`badboyz`	`Connected to Asterisk currently running on asterisk (pid = 25785)`
`18:01.15`	`badboyz`	`asterisk]# asterisk -r`
`18:01.15`	`badboyz`	`Asterisk , Copyright (C) 1999 - 2005 Digium.`
`18:01.20`	`mzo`	`weird!`
`18:02.07`	`nroej`	`need some help with my sirrix card,`
`18:02.08`	`nroej`	`<PROTECTED>`
`18:02.16`	`nroej`	`and then it hangs`
`18:02.30`	`nroej`	`why does it forward`
`18:03.00`	`DaPrivateer`	`can anyone recommend a windows based manager interface`
`18:03.51`	`masonf_`	`why not amp?`
`18:03.54`	`austinnichols101`	`Does rx/tx gain need to be configured with a TE110P?`
`18:03.55`	`nmsclera`	`silly question, but does a PRI circuit have to be CONNECTED to the TE100P for the channels to come up in asterisk?`
`18:04.30`	`benjk`	`badboyz: you need my patches, so you get this ...`
`18:04.31`	`benjk`	`benjk*CLI> show copyright`
`18:04.31`	`benjk`	`Asterisk 1.0.10, Copyright (C) 1999-2004 Digium and third party contributors.`
`18:04.37`	`austinnichols101`	`nmsclera: what do you mean by 'come up'?`
`18:05.03`	`*** join/#asterisk muzzz_ (n=chatzill@60.48.153.162)`
`18:05.08`	`nmsclera`	`austinnichols101: I get this error when starting asterisk (I believe everything is configured properly, but the PRI is not connected to the RJ48-45, whatever..):`
`18:05.19`	`nmsclera`	`Jan 27 06:03:06 WARNING[10250]: chan_zap.c:920 zt_open: Unable to specify channel 11: No such device or address`
`18:05.22`	`DaPrivateer`	`masonf_ - i should be more clear. im looking for something to pop up on my screen when my phone is ringing... allow me to transfer the call when im on it, stuff like that`
`18:05.58`	`badboyz`	`benjk: i dont get a copyright :( how do i install your patch?`
`18:06.01`	`nmsclera`	`austinnichols101: Wait. I just may simply be stupid. Checking.`
`18:07.10`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`18:07.30`	`nroej`	`noone with a sirrix card here?`
`18:07.34`	`benjk`	`badboyz: stock version of Asterisk does not comply with the GPLl, so I made some changes to make it compliant`
`18:08.03`	`benjk`	`the sources are available with my Asterisk for OSX build`
`18:08.11`	`*** join/#asterisk FlipZZZ (n=FlipZZZ@216.138.184.74)`
`18:08.19`	`FlipZZZ`	`hello all`
`18:08.23`	`benjk`	`however, since you probably only want the added CLI commands, you may not want the whole shebang`
`18:08.26`	`Peste`	`can somebody help me with this: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1`
`18:08.41`	`benjk`	`I can send you just the CLI changes`
`18:09.20`	`FlipZZZ`	`any ideas on the "underwater" sounding calls that people i call get sometimes, even though they are clear to me?`
`18:09.22`	`[TK]D-Fender`	`OMG, Sangoma is working on a 8 pot T1 card :D`
`18:09.26`	`[TK]D-Fender`	`port*`
`18:09.45`	`eKo1`	`smoking the sheeba i see`
`18:09.54`	`austinnichols101`	`anyone using an Echo Canceling PRI card?`
`18:10.29`	`*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)`
`18:11.00`	`nmsclera`	`austinnichols101: Scratch that. I'm an idiot. Apparently I can't count. Thanks, anyway ;)`
`18:11.06`	`austinnichols101`	`np`
`18:11.07`	`[TK]D-Fender`	`austinnichols101 : I am.`
`18:11.08`	`iCEBrkr`	`POT!`
`18:11.12`	`iCEBrkr`	`_\\|/_`
`18:11.17`	`iCEBrkr`	`<PROTECTED>`
`18:11.20`	`mzo`	`no weed please, im trying to work :P`
`18:11.22`	`iCEBrkr`	`Oops, I missed.`
`18:11.27`	`iCEBrkr`	`Damn stem is bent`
`18:11.40`	`TheGoD`	`My asterisk server running on a tdm400 with 4 red cards will not pickup the phone on any line. It says it is but it doesn't actually do it. Does anyone have any suggestions`
`18:11.45`	`austinnichols101`	`TK]D-Fender: is the advantage just in CPU cycles or does it actually do better cancellation?`
`18:12.00`	`[TK]D-Fender`	`austinnichols101 : Both`
`18:12.32`	`austinnichols101`	`TK]D-Fender: I'm fighting an echo problem with my TE110P and I'm starting to think it may just be easier to swap cards`
`18:12.40`	`FlipZZZ`	`have to love and hate asterisk`
`18:12.56`	`*** part/#asterisk stegbth (n=stegbth@stegbth.sim.tronicplanet.de)`
`18:13.01`	`[TK]D-Fender`	`austinnichols101 : I'd say so.....`
`18:13.21`	`austinnichols101`	`TK]D-Fender: which card you using?`
`18:13.23`	`sevrdseot`	`question, So even though I've changed console=no in safe_asterisk when I do asterisk -r I get Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)`
`18:13.23`	`*** join/#asterisk QuAd\|Haudrauf (n=hau@dslb-084-057-255-045.pools.arcor-ip.net)`
`18:13.39`	`[TK]D-Fender`	`austinnichols101 : Sangoma A104d`
`18:14.08`	`austinnichols101`	`[TK]D-Fender: Anything special about installing one of those?`
`18:15.16`	`Skkip`	`anyone else ever have a problem getting zaptel to start upon bootup? running 'service zaptel start' works fine, just from boot up sometimes it will not load`
`18:15.32`	`[TK]D-Fender`	`austinnichols101 : A few things to know : You need to compile all the the * stuff first, then Wanpipe (the Sangoma driver), then recompile Zaptel again. From there you need to run "wancfg" to set up the card, and exsure that you start it in your boot sequence before everything else. Thats it basically.`
`18:15.38`	`FlipZZZ`	`anyone have any ideas on the "underwater" sounding calls?`
`18:16.16`	`austinnichols101`	`[TK]D-Fender: tks`
`18:16.37`	`[TK]D-Fender`	`austinnichols101 : np`
`18:16.51`	`Assid`	`is it me .. or is there some kinda isues on the voip network`
`18:17.12`	`DaPrivateer`	`Right so any recomendations on a manager interface that can be used on win32 to give a receptionist the ability to see what's going on?`
`18:17.24`	`puzzled`	`FOP`
`18:17.33`	`iCEBrkr`	`DaPrivateer: Good luck :P`
`18:17.33`	`DaPrivateer`	`lol`
`18:17.33`	`iCEBrkr`	`FOP is gay. but it works :-/`
`18:17.35`	`[TK]D-Fender`	`DaPrivateer : FOP or IPSwitchboard.`
`18:17.45`	`[TK]D-Fender`	`IPSwitchboard is pretty good`
`18:17.49`	`*** join/#asterisk gaspiz (n=gaspiz@86.34.6.164)`
`18:18.00`	`iCEBrkr`	`Someone kick me in the ass so I start working on my damn Asterisk Call Manager.`
`18:18.35`	`zoa`	`or this one: http://www.asteriskguru.com/tools/switchboard.php`
`18:19.09`	`\|vinsik\|`	`hope it help`
`18:19.09`	`iCEBrkr`	`OoofAh!!`
`18:19.10`	`\|vinsik\|`	`s`
`18:19.12`	`iCEBrkr`	`:)`
`18:19.14`	`\|vinsik\|`	`:D`
`18:19.16`	`gaspiz`	`hi why is it asking for an extension if I use: voicemailmain(1005@company_5) I have an extension 1005 in context company_5 in voicemail.conf`
`18:19.32`	`gaspiz`	`I use asterisk 1.2.1`
`18:19.41`	`tainted_`	`when i call an ATA, i don't get the ringing tone`
`18:19.46`	`tainted_`	`does anyone know why that is?`
`18:19.47`	`DaPrivateer`	`ahh yes`
`18:19.53`	`DaPrivateer`	`FOP is what i played with before`
`18:20.00`	`\|vinsik\|`	`tainted: what ata?`
`18:20.03`	`\|vinsik\|`	`tainted: brand`
`18:20.04`	`DaPrivateer`	`and managed to make phones all over trhe place start ringing for no reason`
`18:20.04`	`tainted_`	`i thought asterisk natively adds ringing w/o the 'r' parameter in dial`
`18:20.09`	`austinnichols101`	`[TK]D-Fender: One last question - do you ever have to mess with rx/tx gain on your PRI? I'm wondering if it's necessary to make adjustments for echo...`
`18:20.13`	`tainted_`	`\|vinsik\| grandstream 488`
`18:20.29`	`*** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net)`
`18:20.41`	`\|vinsik\|`	`tainted: hah.. have the same problem with grandstreams.. they are dumb machines..`
`18:20.54`	`tainted_`	`\|vinsik\| what was your fix?`
`18:20.59`	`gaspiz`	`does anyone know about this voicemailmain issue?`
`18:21.10`	`\|vinsik\|`	`tainted: i added 'r'`
`18:21.13`	`tainted_`	`hmm`
`18:21.15`	`tainted_`	`k`
`18:21.18`	`\|vinsik\|`	`tainted: but is the device ringing?`
`18:21.21`	`tainted_`	`yes`
`18:21.30`	`\|vinsik\|`	`tainted: well 'r' worked for me anyways..`
`18:21.58`	`tainted_`	`stange thing is, i have 'r' enabled!?`
`18:22.08`	`*** join/#asterisk TarAm (n=mmm@218.111.179.96)`
`18:22.13`	`\|vinsik\|`	`Dial(SIP/phone,20,r) ?`
`18:22.20`	`tainted_`	`yes`
`18:22.33`	`\|vinsik\|`	`try removing it..`
`18:22.52`	`tainted_`	`was your ATA ringing?`
`18:22.56`	`\|vinsik\|`	`yes`
`18:22.59`	`\|vinsik\|`	`allways`
`18:23.01`	`tainted_`	`must be same issue then`
`18:23.05`	`\|vinsik\|`	`hmm`
`18:23.06`	`nroej`	`Forwarding SIP/4098883-d6e3 to 'Local/@internal' (thanks to Srx/gint10-081662a8) <- where does this message come from?`
`18:23.10`	`tainted_`	`i thought asterisk natively added rining`
`18:23.24`	`tainted_`	`ringing`
`18:23.34`	`\|vinsik\|`	`gaspiz: could you explain once again.. i didnt get it.. how do u route it from exten?`
`18:23.45`	`puzzled`	`tainted: if you use the r option in the Dial statement`
`18:24.05`	`[TK]D-Fender`	`austinnichols101 : I never have to play with ANYTHING with this card. Gain's are 0.0 for both and 0 echo.`
`18:24.14`	`tainted_`	`puzzled i do have the r option.. but it still doesn't ring`
`18:24.17`	`\|vinsik\|`	`puzzled: he has the 'r' option but it does not ring.`
`18:24.20`	`austinnichols101`	`[TK]D-Fender: tks again!`
`18:24.29`	`*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`18:24.37`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`18:24.47`	`*** join/#asterisk TarAm (n=mmm@218.111.179.96)`
`18:24.54`	`puzzled`	`tainted_: not sure but maybe the channel needs to be answered before it can send back rining`
`18:24.58`	`puzzled`	`ringing even`
`18:24.59`	`[TK]D-Fender`	`zoa : That AsteriskGuru one is pretty neat.... where does it store its server list?`
`18:25.03`	`TarAm`	`hai hello`
`18:25.04`	`\|vinsik\|`	`tainted: maybe you mispelled? :D does asterisk say -- Executing Dial(SIP/jadda\|20\|r) <?`
`18:25.08`	`SibRw0rk`	`has anyone successfully gotten musiconhold to work with streaming music?`
`18:25.09`	`DaPrivateer`	`ok ipswitchboard looks kewl, just a pain in the ass to set up :-p`
`18:25.17`	`TarAm`	`somebudy hellp me`
`18:25.20`	`_Sam--`	`idefisk?`
`18:26.00`	`TarAm`	`helllo`
`18:26.07`	`TarAm`	`any budy hom e`
`18:26.14`	`_Sam--`	`everyones eating lunch`
`18:26.21`	`TarAm`	`sam`
`18:26.24`	`TarAm`	`hai`
`18:26.28`	`_Sam--`	`high`
`18:26.38`	`TarAm`	`sam can you help me`
`18:26.44`	`*** part/#asterisk bkw__ (n=brian@m010f36d0.tmodns.net)`
`18:26.46`	`_Sam--`	`probably not but at least im honest`
`18:26.50`	`puzzled`	`TarAm: stop asking for help and just state your problem`
`18:26.51`	`TarAm`	`hemm i ave problem with my asterisk conf.`
`18:27.39`	`TheGoD`	`My asterisk server running on a tdm400 with 4 red cards will not pickup the phone on any line. It says it is but it doesn't actually do it. Does anyone have any suggestions?`
`18:27.56`	`_Sam--`	`TheGoD: what asterisk version`
`18:28.09`	`TarAm`	`puzzled i just want no that asterisk can use in touch tones system registration`
`18:28.25`	`TarAm`	`if can how i can do taht system`
`18:28.30`	`puzzled`	`TarAm: I have no idea what you are saying`
`18:28.36`	`TheGoD`	`1.2.0`
`18:28.36`	`\|vinsik\|`	`TarAm: drunk?`
`18:28.42`	`\|vinsik\|`	`:D`
`18:28.45`	`TarAm`	`heheehe`
`18:28.46`	`*** join/#asterisk cianhughes (n=cian@87.192.36.98)`
`18:28.52`	`TarAm`	`my english not good hemm`
`18:29.05`	`TarAm`	`and i a hew guy at this channel`
`18:29.15`	`TarAm`	`i join after i buy the asterisk book`
`18:29.19`	`_Sam--`	`TarAm: need the READ command`
`18:29.30`	`TarAm`	`VoIP telephony with asterisk`
`18:29.34`	`*** join/#asterisk kimosabe (n=kimosabe@201.133.216.51)`
`18:29.36`	`cianhughes`	`hey, firstly I'm in a country where G729 patents don't apply, I'm trying to update the free codec_g729.so to work with asterisk 1.2.1 on FreeBSD, anyone else tried this?`
`18:29.46`	`puzzled`	`nope`
`18:29.59`	`cianhughes`	`ok I am getting this error /usr/local/lib/asterisk/modules/codec_g729.so: Undefined symbol "USC_G729FP_Fxns"`
`18:30.09`	`TarAm`	`so what command i can read`
`18:30.34`	`*** join/#asterisk bertd (n=admin@adsl-220-179-181.mob.bellsouth.net)`
`18:30.36`	`_Sam--`	`READ gets pressed DTMF tones`
`18:30.47`	`\|vinsik\|`	`oh.. now i got what is he asking. :)`
`18:31.06`	`_Sam--`	`TarAm: http://www.voip-info.org/wiki-Asterisk+cmd+Read`
`18:31.12`	`cianhughes`	`but I don't think the symbol is undefined because of this: nm /usr/local/lib/asterisk/modules/codec_g729.so \| grep USC_G729FP_Fxns`
`18:31.15`	`cianhughes`	`000323c0 D USC_G729FP_Fxns`
`18:31.22`	`TarAm`	`:D`
`18:31.45`	`bertd`	`Hi folks. First time on asterisk channel.`
`18:31.54`	`\|vinsik\|`	`bertd: welcome`
`18:32.05`	`Mark_Halverson`	`I have several SIP providers, any AGIs out there that will select a random provider? I am wanting to keep my dialplan as small as possible`
`18:32.07`	`TarAm`	`hai all :D`
`18:32.36`	`puzzled`	`Mark_Halverson: iirc there is a built-in random function you could use`
`18:32.50`	`TarAm`	`vinsik`
`18:32.55`	`TarAm`	`sam`
`18:32.56`	`_Sam--`	`http://www.voip-info.org/wiki/view/Asterisk+cmd+Random`
`18:32.58`	`\|vinsik\|`	`TarAm`
`18:33.07`	`Mark_Halverson`	`hey but that requires at least two lines for each provider....i have like 60 SIP accounts`
`18:33.19`	`TarAm`	`vinsik can i PM you`
`18:33.37`	`\|vinsik\|`	`TarAm: im leaving work in 5min.. so state quickly.`
`18:33.46`	`_Sam--`	`you could use the random command with 60 sip providers`
`18:33.50`	`Mark_Halverson`	`i am using the random cmd now -- wanting something different`
`18:34.13`	`\|vinsik\|`	`Mark_Halverson: why dont u build MySQL random ?`
`18:34.14`	`Mark_Halverson`	`the dialplan gets to bulky - trying to simplify it`
`18:34.16`	`TarAm`	`ok vinsik`
`18:34.19`	`_Sam--`	`you could put them in an sql table somhow`
`18:34.36`	`\|vinsik\|`	`mark: with mysql .. ;)`
`18:34.45`	`\|vinsik\|`	`mark: write a shell script or something..`
`18:34.49`	`_Sam--`	`or php`
`18:34.50`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:34.51`	`TarAm`	`i just know that asterisk can do touch tones registration system`
`18:34.52`	`Mark_Halverson`	`ah...the somehow....that's where i am at...lol....i can't write a program to save my life`
`18:35.06`	`TarAm`	`sorry my english noot good`
`18:35.38`	`TarAm`	`but really want learn about asterisk`
`18:35.44`	`Blackthorn`	`somone was talking about hearing a dtfm tone ealier in this chat.. what is dtfm tone? Reason why I ask is that I have a voip user that we get a low beeping tone every 20 seconds when he's on the phone..`
`18:36.07`	`_Sam--`	`hmmm are you using chanspy?`
`18:36.12`	`\|vinsik\|`	`TarAm: build mysql database install res_mysql.so support for asterisk. make extensions to read 4 digit pin.. query it by mysql and if it checks out register the phone to asterisk database.`
`18:36.32`	`bertd`	`I built my first asterisk box recently and demoed it last week. I had some weird problems with the first 1/2 second of the voice mail prompt getting lost. Instead of "comedian mail" it sounds like "ian main". I added a playback of 1 second of silence, and that fixed the problem. Any better fix? Anybody else have this problem?`
`18:37.03`	`puzzled`	`bertd: first answer the line, then put in a Wait(1) then go on to voicemail`
`18:37.09`	`Mark_Halverson`	`1,answer`
`18:37.12`	`_Sam--`	`bertd: wait(1) is pretty benign...dont let it bother ya`
`18:37.13`	`Mark_Halverson`	`2, wait(1)`
`18:37.17`	`\|vinsik\|`	`Mark_Halverson: create a file with 60 providers.. make a shell script to parse it and choose randomly one.`
`18:37.19`	`puzzled`	`bertd: give the box a bit of time to get stuff sorted out`
`18:37.47`	`\|vinsik\|`	`TarAm: is that what u are after?`
`18:37.49`	`gaspiz`	`does any of you know a problem with voicemail-> voicemailmain in Asterisk 1.2.1?`
`18:37.55`	`TarAm`	`so where i can get res_mysql.so`
`18:38.01`	`_Sam--`	`asterisk-addons`
`18:38.04`	`[TK]D-Fender`	`Mark_Halverson : Forget the script <- Put the list into the ASTDB, and doa random & grab it with DB. Much less overhead.`
`18:38.06`	`\|vinsik\|`	`TarAm: asterisk-addons`
`18:38.12`	`nroej`	`Forwarding SIP/4098883-d6e3 to 'Local/@internal' (thanks to Srx/gint10-081662a8) <- which asterisk module throws this message?`
`18:38.28`	`\|vinsik\|`	`gaspiz: what is the problem?`
`18:38.55`	`\|vinsik\|`	`D-fender: does asterisk database support alot of data?`
`18:39.14`	`TarAm`	`thanks all sam and vinsik i try do first if i fill`
`18:39.21`	`[TK]D-Fender`	`\|vinsik\| : plenty`
`18:39.22`	`\|vinsik\|`	`TarAm: and an answer to your next q. is www.asterisk.org`
`18:39.23`	`gaspiz`	`vinsik: it does not recognize my users`
`18:39.24`	`TarAm`	`i as you again ok`
`18:39.34`	`[TK]D-Fender`	`\|vinsik\| : for his need anyways...`
`18:39.39`	`\|vinsik\|`	`gaspiz: what command do u give from extensions?`
`18:39.52`	`TarAm`	`i already go tu forum asterisk .org`
`18:40.05`	`gaspiz`	`voicemailmain(1005@mycontext)`
`18:40.07`	`\|vinsik\|`	`D-Fender: ok..`
`18:40.24`	`bertd`	`Ok guys. Thanks for help. Glad to hear it is normal to have to "wait(1)" before voicemail.`
`18:40.47`	`\|vinsik\|`	`gaspiz: and the box says unknown extension?`
`18:41.04`	`gaspiz`	`vinsik: yes`
`18:41.17`	`gaspiz`	`vinsik: I have my users in a mysql database, this is working couse I have the sip users in another table`
`18:41.39`	`gaspiz`	`vinsik: it was working great with 1.0`
`18:42.00`	`*** join/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda)`
`18:42.08`	`gaspiz`	`vinsik: today I upgraded it to 1.2.1 and crashed`
`18:42.19`	`\|vinsik\|`	`gaspiz: upgrade to 1.2.3`
`18:42.28`	`_Sam--`	`1.2.1 is fine`
`18:42.42`	`_Sam--`	`but 1.2.3 may be finer for you`
`18:42.44`	`mzo`	`no 1.2.3. si bettar! :P`
`18:42.44`	`\|vinsik\|`	`gaspiz: do u use voicemail from sql?`
`18:43.00`	`*** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com)`
`18:43.07`	`\|vinsik\|`	`gaspiz: i mean do u have voicemail config in sql`
`18:43.15`	`\|vinsik\|`	`shit`
`18:43.17`	`\|vinsik\|`	`i gatto go..`
`18:43.20`	`\|vinsik\|`	`cya guys`
`18:43.27`	`gaspiz`	`yes`
`18:43.33`	`gaspiz`	`I have the users in mysql`
`18:43.49`	`gaspiz`	`does anyone else have an idea?`
`18:44.50`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`18:45.31`	`*** join/#asterisk Cyon (n=cyon@216.179.31.166)`
`18:45.47`	`enemy^x`	`I get "Ouch ... error while writing audio data: : Broken pipe" running 1.2.3 while moh... anyone else?`
`18:45.52`	`bertd`	`One question down. Here is next one. ------- In a previous life I was a hardware designer. I just read the O'Reilly book on T1. At the bit level it is really simple. It seems like it should be possible to design a cheap (about $20 parts cost) board to take the bits off the T1 line and dump them into ethernet packets. This has got to be better than the $500+ digium board. Of course, it would need a non-trivial driver in linux to ma`
`18:45.54`	`*** part/#asterisk pato (n=just@nat1.inalambrica.net)`
`18:46.05`	`Cyon`	`enemy^x: Yeah, it's common, and I think you can google to locate the fix`
`18:46.38`	`enemy^x`	`cyan: I had it in 1.2.2 also, but non crash, just silence suppression`
`18:46.47`	`enemy^x`	`rolled over the screen`
`18:46.58`	`enemy^x`	`have also googled for the fix, but still no ok`
`18:47.08`	`Cyon`	`enemy^x: Ah, it's crashing on you? No idea then.`
`18:47.29`	`Cyon`	`bertd: Well above my abilities. :-P`
`18:47.41`	`*** join/#asterisk bob-b (n=bob@Jade.NetSurf.Net)`
`18:47.42`	`FarrisG`	`If I make changes to zapata.conf, do I have to stop *, unload modules, reload them, and the restart?`
`18:48.08`	`Cyon`	`FarrisG: I think so, yes.`
`18:48.10`	`MasterYoda`	`FarrisG: no, depending on your changes you may need to restart Asterisk though`
`18:48.26`	`Cyon`	`FarrisG: I recant my statement. :-P`
`18:48.31`	`MasterYoda`	`FarrisG: you should not have to unload any kernel modules for changes to zapata.conf`
`18:48.48`	`*** join/#asterisk zotz (n=zotz@24.231.47.175)`
`18:48.50`	`[TK]D-Fender`	`Wow I just found a place thats a few bucks cheap on Polycom IP 601's than Atacomm.....`
`18:49.09`	`iCEBrkr`	`jbalcomb: DOOD`
`18:49.19`	`FarrisG`	`I'm just having trouble getting caller ID to work. It works for the two analog lines we have, which have callerID explicitly set for their group... But no other outbound calls send any caller id data`
`18:49.20`	`[TK]D-Fender`	`FarrisG : Just do a "restart gracefully".`
`18:49.28`	`bob-b`	`New to IRC, please be gentle... I have an asterisk question that I can't find any prior info on the mailing lists.`
`18:49.38`	`MasterYoda`	`FarrisG: do your lines support it?`
`18:49.42`	`FarrisG`	`[TK]D-Fender: Ah, that answers my question. That's what I did, and still no change`
`18:49.46`	`FarrisG`	`MasterYoda: Yes`
`18:49.55`	`Cyon`	`bob-b: Ask`
`18:50.15`	`MasterYoda`	`FarrisG: does it work if you do a Set(CALLERID(num)=01234567889) before dialing?`
`18:50.45`	`bob-b`	`Is it possible to ring a sip phone from an incoming call on an Analog line WITHOUT taking the analog line off-hook until the SIP call is answered?`
`18:50.48`	`[TK]D-Fender`	`FarrisG : For you that'd be "SetCallerIDNum(0123456789)"`
`18:50.51`	`FarrisG`	`MasterYoda: Where does that go? extensions.conf?`
`18:50.57`	`MasterYoda`	`bob-b: you are starting off on the wrong foot. Rule number #1, don't ask to ask.`
`18:51.06`	`bob-b`	`:-) sorry`
`18:51.15`	`FarrisG`	`I guess I'm just confused... Do I have to do that for every extension?`
`18:51.18`	`MasterYoda`	`FarrisG: in extensions.conf before you dial statement`
`18:52.09`	`[TK]D-Fender`	`FarrisG : Just call it before your Dial-line out to Zap`
`18:52.47`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`18:52.59`	`FarrisG`	`[TK]D-Fender: And where is that? You've seen my extensions.conf, can you point me to where in that file it should go?`
`18:53.36`	`bob-b`	`I can explain why I want to do this if anyone is interested.`
`18:53.49`	`*** join/#asterisk doug (i=doug@zaxxon.telerama.com)`
`18:53.57`	`*** join/#asterisk LenOK (n=ln@66.193.84.181)`
`18:56.45`	`[TK]D-Fender`	`FarrisG : You'd have to di it like 20 places.... everywhere right before you dial in your [trunk(whatever)] contexts.... thats the price you pay for lack of abstraction :)`
`18:57.19`	`[TK]D-Fender`	`bob-b : YES`
`18:57.22`	`FarrisG`	`[TK]D-Fender: Lack of abstraction?`
`18:57.43`	`FarrisG`	`[TK]D-Fender: Again, I didn't write this .conf file :)`
`18:58.31`	`Cyon`	`bob-b: Yeah, should be feasible..just don't Answer() and Hangup() if the sip device doesn't answer.`
`18:58.37`	`Cyon`	`bob-b: But I've never tried, just guessing`
`18:58.48`	`TheGoD`	`My asterisk server running on a tdm400 with 4 red cards will not pickup the phone on any line. It says it is but it doesn't actually do it. Does anyone have any suggestions?`
`18:58.48`	`[TK]D-Fender`	`FarrisG : You have a ton of "dial" lines that directly reference the PRI like "Dial(Zap/g1/${EXTEN:1}). If you had a macro that was called based on the pattern match you would only have to chang it in 1 place.`
`18:59.03`	`[TK]D-Fender`	`FarrisG : I'm not blaming YOU, I'm blaming your CONFIG :D`
`18:59.26`	`[TK]D-Fender`	`bob-b : I've done it, yes its that easy.`
`18:59.51`	`*** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net)`
`19:00.06`	`bob-b`	`The reason I want to do this is to have a Y splitter on a POTS line where my desk phone rings and my sip phone rings and whoever answers first gets the call.`
`19:00.17`	`bob-b`	`Is there an example somewhere you could direct me to?`
`19:00.17`	`FarrisG`	`[TK]D-Fender: So if instead of defining "TRUNK=Zap/yaddayadda" I defined "DIALTRUNK=SetCallerID(kdjfhskdfh),Dial(Zap/yasyfds)" it would be cleaner?`
`19:00.26`	`[TK]D-Fender`	`bob-b : just make it Dial before any other playback style command or anything that touches sound on the call. if the Dial to SIP phone doesn't end up with an answer then the Zap call would just keep riniging.`
`19:00.39`	`*** join/#asterisk Flusher- (i=flusher@filer.euroserv.com)`
`19:00.55`	`Peste`	`can somebody help me with this: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1`
`19:01.24`	`Peste`	`i use a TE110P card from digium and a AG4000 from nms on the other side`
`19:01.25`	`[TK]D-Fender`	`FarrisG : Instead of calling "Dial" in 20 places, call a macro that does the "dirty work". That way you only change it in one place.`
`19:01.40`	`gaspiz`	`I have a realtime configuration problem for the voicemail . I'm using 1.2.1`
`19:01.54`	`wunderkin`	`Peste, erm.. omg.. have you emailed digium support? how often do you get that error? at least as often as you keep asking here? check for irq sharing and using proper timing`
`19:01.57`	`jbalcomb`	`iCEBrkr DOOD!!`
`19:02.24`	`iCEBrkr`	`jbalcomb: You figure anything out?`
`19:03.02`	`gaspiz`	`does any of you use realtime voicemail? with 1.2.1`
`19:03.22`	`jbalcomb`	`iCEBrkr not yet. i was just told i'm not to work on it anymore until get a backup server because of how much trouble we had last night`
`19:04.01`	`iCEBrkr`	`Makes sense`
`19:04.10`	`jbalcomb`	`iCEBrkr I just submitted a PO for a $7,000+ system including RH 4 ES, Asterisk Business, and additional TE411P`
`19:04.18`	`iCEBrkr`	`Geesh`
`19:04.32`	`jbalcomb`	`iCEBrkr Dual Dual Core Zeon 2.8 Ghz =)`
`19:04.33`	`iCEBrkr`	`Tho, I think that's how much ours cost.. about..`
`19:04.40`	`bob-b`	`cool. thanks guys.`
`19:04.45`	`iCEBrkr`	`Actually, I think ours was about $3500`
`19:05.05`	`MasterYoda`	`gaspiz: I have before...`
`19:05.21`	`jbalcomb`	`iCEBrkr 2GB RAM, (3) 36 GB Cheetahs RAID 5, redundant power supplies, etc.`
`19:05.42`	`shido`	`someone is getting a nice phat check, jbalcomb`
`19:06.02`	`wunderkin`	`jbalcomb, and what is this being used for?`
`19:06.12`	`jbalcomb`	`iCEBrkr all told it was $6,500 but dell is giving 1,100 off for systems over 4,000`
`19:06.25`	`jbalcomb`	`shido not me. :(`
`19:06.30`	`iCEBrkr`	`hehe`
`19:06.42`	`iCEBrkr`	`jbalcomb: You're getting the Sangoma card right?`
`19:06.50`	`*** part/#asterisk bob-b (n=bob@Jade.NetSurf.Net)`
`19:07.00`	`jbalcomb`	`wunderkin Asterisk, 120 phones, 100 DIDs, 150 800s, 2 PRIs, 16 simultaneous calls`
`19:07.05`	`jbalcomb`	`iCEBrkr no....`
`19:07.09`	`_Sam--`	`jbalcomb: you couldhave built the same thing for 2000`
`19:07.22`	`FarrisG`	`I'm so friggin' confused now`
`19:07.24`	`tainted_`	`anyone know docelmo? i bought a server from him and don't know if he's reliable...`
`19:07.25`	`nmsclera`	`iCEBrkr: are the Sangoma cards better than the Digium or Zaptel?`
`19:07.29`	`_Sam--`	`you bought a dell voipserv model?`
`19:07.36`	`iCEBrkr`	`jbalcomb: You realize you may have IRQ issues?`
`19:07.37`	`mdave`	`Dabba, you still around?`
`19:07.56`	`mdave`	`ah. I see not`
`19:08.01`	`jbalcomb`	`iCEBrkr yeah but this setup is 100% digium compliant`
`19:08.14`	`jbalcomb`	`iCEBrkr they wont support us with the sangoma card`
`19:08.37`	`iCEBrkr`	`jbalcomb: http://www.digium.com/index.php?menu=compatibility`
`19:08.42`	`jbalcomb`	`iCEBrkr the boss wont let me pull the digium card out of the current box and just move it over either.. FUD.`
`19:08.46`	`mogorman`	`iCEBrkr we gonna start the sangoma v digium argument again`
`19:08.48`	`MasterYoda`	`FarrisG: oh ok, I just re read your problem, no oubbound callerid right...`
`19:08.53`	`mdave`	`well I got callout-delivered-disa working, but it only works when the destination is a local ATA, not when I dialout thru broadvoice.. or rather, it works, but while I get a dialtone when I answer the local set, i dont get a dialtone when I answer the pots line called via broadvoice`
`19:09.03`	`MasterYoda`	`FarrisG: set it in zapata.conf for the lines that it is not working on`
`19:09.03`	`iCEBrkr`	`mogorman: I'm not arguing what I find on your guys site..`
`19:09.15`	`iCEBrkr`	`mogorman: It's stated on Digium's site what the issues are..`
`19:09.18`	`jbalcomb`	`iCEBrkr what am i looking at?`
`19:09.29`	`iCEBrkr`	`jbalcomb: The part where it said Dell Poweredge..`
`19:10.02`	`jbalcomb`	`iCEBrkr the PowerEdge 2850 is /the/ recommended system though`
`19:10.21`	`wunderkin`	`_Sam--, heh.. no kidding..`
`19:10.33`	`*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)`
`19:10.38`	`_Sam--`	`maybe even 1500 :)`
`19:10.50`	`iCEBrkr`	`jbalcomb: Well, you'll have more time to monkey around with it if it doesn't work.. I didn't/don't have that option.`
`19:10.51`	`jbalcomb`	`http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE&tab=compatibility`
`19:11.13`	`FarrisG`	`MasterYoda: I believe I tried that, which is why I asked if I needed to restart anything special. My fxo_ls lines are doing callerid Properly but my pri_cpe lines are not`
`19:11.14`	`jbalcomb`	`iCEBrkr i did also call and confirm that the new release is tested with RH ES 4`
`19:11.50`	`iCEBrkr`	`jbalcomb: Like I said, everything around here is 'fire fighting mode', so I couldn't chance a failure or problems...`
`19:11.56`	`tainted_`	`jbalcomb why's that`
`19:12.16`	`MasterYoda`	`FarrisG: well make sure it is enabled in zapata.conf and configured for those lines (or in the dialplan). Also make sure your provider support sending it.`
`19:12.28`	`MasterYoda`	`FarrisG: you could also call Digium Support`
`19:12.47`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`19:13.12`	`FarrisG`	`MasterYoda: Not sure I want to pay Digium Support just to show me which config line I'm missing.`
`19:13.13`	`*** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net)`
`19:13.27`	`mogorman`	`heh we cover zapata for free`
`19:13.34`	`FarrisG`	`MasterYoda: I know the provider supports it, as it works for two of the lines`
`19:13.38`	`jbalcomb`	`tainted_ whys what?`
`19:13.38`	`FlipZZZ`	`anyone have any ideas on the "underwater" sounding calls?`
`19:13.43`	`*** join/#asterisk dpryo (i=hn@donatello.nesland.net)`
`19:13.44`	`MasterYoda`	`FarrisG: but those are not your pri lines`
`19:14.05`	`tainted_`	`jbalcomb about the 2850`
`19:14.14`	`rob0`	`FlipZZZ: trying to drink water and talk at the same time? <g,d,r>`
`19:14.19`	`[TK]D-Fender`	`jbalcomb : PM`
`19:14.25`	`jbalcomb`	`tainted_ ah, that is digiums recommendation`
`19:14.30`	`FlipZZZ`	`rob0: i so wish it was that easy LOL`
`19:14.32`	`tainted_`	`oh`
`19:14.46`	`*** join/#asterisk zock (n=zock@p54B1ADE0.dip0.t-ipconnect.de)`
`19:14.47`	`TheGoD`	`My asterisk server running on a tdm400 with 4 red cards will not pickup the phone on any line. It says it is but it doesn't actually do it. Does anyone have any suggestions?`
`19:14.48`	`jbalcomb`	`tainted_ its on the bottom of the page i posted above`
`19:14.56`	`FlipZZZ`	`breaking up line, garbled sounds.`
`19:14.57`	`TheGoD`	`anyone?`
`19:15.01`	`FarrisG`	`MasterYoda: Ok, let me rephrase. It worked for all lines a few months ago, with no major changes since then. Some tiny piece has been changed somewhere, and now it only works for two lines. Also, those two lines it works for dial out in the same manner that the other lines (for which caller id is not working) do`
`19:15.46`	`zock`	`Hi.`
`19:16.07`	`*** join/#asterisk jalsot_ (n=tamas@abacus.eworldcom.hu)`
`19:18.57`	`*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)`
`19:19.09`	`*** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com)`
`19:19.45`	`Blackthorn`	`somone was talking about hearing a dtfm tone ealier in this chat.. what is dtfm tone? Reason why I ask is that I have a voip user that we get a low beeping tone every 20 seconds when he's on the phone..`
`19:19.47`	`*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)`
`19:20.18`	`Zeeek`	`~dtmf`
`19:20.19`	`jbot`	`DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency.`
`19:20.23`	`mdave`	`ok.. even on a call-in, disa doesnt play a dialtone to a channel coming from broadvoice.. so, im wondering are their some options I need to set controlling generation of dialtone over that sort of connection`
`19:21.04`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`19:21.48`	`Zeeek`	`mdave are you sure disa is executing?`
`19:22.48`	`MasterYoda`	`FarrisG: well if it's not your upstream provider, Digium can propbably fix it in like 5 mins`
`19:22.55`	`mdave`	`hrm`
`19:23.00`	`*** join/#asterisk acidblood (n=acidbloo@201.47.33.249)`
`19:23.04`	`mdave`	`lemme check, i think it shows on the console`
`19:23.41`	`mdave`	`Yes, the console says 'executing DISA....'`
`19:23.47`	`mdave`	`but i never hear a dialtone`
`19:24.11`	`Zeeek`	`I've never seen that using zap, iax2 or sip`
`19:24.28`	`mdave`	`perhaps jobot should mention that 'touch tone dialing' is traditionnaly DTMF`
`19:24.48`	`mdave`	`i may have something configured wrong`
`19:24.50`	`acidblood`	`Hi, I'm setting up Asterisk for the first time and I'd like pointers to a HOWTO to help me do exactly what I want.`
`19:24.50`	`mdave`	`this is a new setup`
`19:25.03`	`Zeeek`	`mdave my point with jbot was stuff like that is easily looked up`
`19:25.08`	`mdave`	`basically I have a SPA-2000 configured, and the bv account`
`19:25.27`	`mdave`	`yeah, i know.. i was just suggesting an addition to that particular entry`
`19:25.39`	`acidblood`	`Namely, I have SIP service with Broadvoice and want to connect an ATA plus a softphone on my LAN (Broadvoice doesn't let you run two devices at once). I need to set up extensions for the ATA and softphone and have Asterisk talk to Broadvoice's servers.`
`19:25.50`	`mdave`	`i can call out from the phone on the spa to bv, and inbound bv rings to the spa phone`
`19:25.53`	`Zeeek`	`mdave whatya see after the DISA exec?`
`19:26.03`	`mdave`	`i can use a .call file to tell * to call the spa phone, and I get dialtone`
`19:26.08`	`mdave`	`nothing else`
`19:26.12`	`*** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda)`
`19:26.25`	`mdave`	`<PROTECTED>`
`19:26.29`	`mdave`	`thats the last message`
`19:26.41`	`mdave`	`although I think eventually it times out and shows an 'ended' message`
`19:26.47`	`mdave`	`doesnt matter if I try to dial or not`
`19:26.48`	`Zeeek`	`to both of you, I'v been around here for almost two years and one problematic SIP provider name comes up all the time.`
`19:26.52`	`Zeeek`	`BV!`
`19:27.02`	`Zeeek`	`tons of messages on the ML`
`19:27.05`	`mdave`	`yeah, ive pretty much come to the conclusion that the suck`
`19:27.18`	`mdave`	`but for the moment, its what I have, and I dont think they are the issue`
`19:27.19`	`*** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu)`
`19:27.21`	`Zeeek`	`they're seemingly the most asterisk unfriendly provider`
`19:27.46`	`*** join/#asterisk jello333 (i=ghento@CPE0011d8a291a6-CM00111ae4684c.cpe.net.cable.rogers.com)`
`19:28.03`	`mdave`	`becuase everything else works - my question is - is the way a dialtone is generated to a phone attached to a SPA different than the way it would be provided to a remote SIP proxy, and is there something in the sip.conf I perhaps should have set`
`19:28.09`	`mdave`	`eg, does the SPA generate the dialtone itself,`
`19:28.17`	`mdave`	`and for bv * has to generate it internally?`
`19:28.31`	`Zeeek`	`mdave it does when you pick up but I think NOT when you access disa`
`19:28.37`	`jello333`	`Hi there. I have a VOIP line hooked up, SIP..coming into my asterisk box. How can I define the number of continuous incoming calls that asterisk will accept`
`19:28.38`	`justinu`	`sip devices generally play their own dialtone`
`19:29.05`	`Zeeek`	`mdave looks like a case where you want to look at sip debug`
`19:29.07`	`TheGoD`	`My asterisk server running on a tdm400 with 4 red cards will not pickup the phone on any line. It says it is but it doesn't actually do it. Does anyone have any suggestions?`
`19:29.11`	`mdave`	`justinu, so how does * expect to play a dialtone to a remote caller? is there a special setting I need in sip.conf?`
`19:29.56`	`justinu`	`you mean for DISA?`
`19:31.05`	`*** part/#asterisk LenOK (n=ln@66.193.84.181)`
`19:31.25`	`[TK]D-Fender`	`TheGoD : Pastebin your zapata.conf and extensions.conf. And where are you located?`
`19:31.35`	`Zeeek`	`TheGoD is the tdm the only thing connected to the lines?`
`19:31.40`	`iCEBrkr`	`cl`
`19:31.42`	`iCEBrkr`	`err`
`19:32.24`	`TheGoD`	`USA, hold i'll pastebin`
`19:33.06`	`*** join/#asterisk Flauto (n=zhao@71.194.194.48)`
`19:33.21`	`Flauto`	`hi people`
`19:33.46`	`acidblood`	`�Registration`
`19:33.46`	`acidblood`	`In the [general] section of the config file create a line like this:`
`19:33.46`	`acidblood`	`�register => <phonenumber>@sip.broadvoice.com:<password>:<phonenumber>@sip.broadvoice.com/<extension>`
`19:33.46`	`acidblood`	`�`
`19:33.46`	`acidblood`	`�Replace phonenumber with your account phone number,`
`19:33.48`	`acidblood`	`�Replace password with your password`
`19:33.50`	`acidblood`	`�Replace extension with one of your accessible extensions in the dial plan.`
`19:34.12`	`acidblood`	I still don't understand what `extension' means here.
`19:34.13`	`[TK]D-Fender`	`acidblood : PASTEBIN!`
`19:34.16`	`[TK]D-Fender`	`~pb`
`19:34.17`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`19:34.27`	`acidblood`	`[TK]D-Fend, oops, sorry.`
`19:34.31`	`iCEBrkr`	`pastbin or DIE`
`19:34.51`	`Zeeek`	`if (!pastebin) die();`
`19:35.20`	`TheGoD`	`pastebin.com/526126`
`19:35.40`	`Katty`	`ok, now that we've all yelled pastebin 100 times....`
`19:35.53`	`Katty`	`let's have a muffin.`
`19:35.54`	`Katty`	`and chill.`
`19:36.14`	`Zeeek`	`I didn't hear you yelling?`
`19:36.37`	`Zeeek`	`I biught some soy milk`
`19:36.40`	`Katty`	`:>`
`19:37.40`	`[TK]D-Fender`	`Zeeek : that should be "if (!pastebin) die(horribly);"`
`19:38.22`	`[TK]D-Fender`	`TheGoD : Get rid of "immediate=yes" !!!!!`
`19:38.23`	`Zeeek`	`I wrote my own function: foad();`
`19:38.44`	`Zeeek`	`or change to immediategratification=yes`
`19:38.45`	`[TK]D-Fender`	`and busydetect while you're at it. Oh and stop using AMP!`
`19:39.01`	`Zeeek`	`great bowling alley equipment, AMP`
`19:39.28`	`TheGoD`	`I set it to no.. Same thing`
`19:39.31`	`TheGoD`	`lol`
`19:39.40`	`iCEBrkr`	`o/~ Dead AMP! Dead AMP! DeadAMPDeadAMPDeadAMP DeaaaaaaaadAMP o/~`
`19:39.54`	`Zeeek`	`useaah=no`
`19:39.56`	`iCEBrkr`	`( sung to the tune of The Pink Panther )`
`19:39.58`	`Zeeek`	`useAMP=no`
`19:40.24`	`Zeeek`	`insecure=windows`
`19:40.37`	`tuxinator_linux`	`Katty: I want a muffin`
`19:40.43`	`TheGoD`	`Same problem with both of them set to no. I used amp to learn on. No I don't use @home, its on a slackware box`
`19:40.47`	`Zeeek`	`bindport=prefer_sherry`
`19:40.54`	`Katty`	`tuxinator_linux: come get it.`
`19:41.14`	`Zeeek`	`TheGoD you restared ?`
`19:41.33`	`hypnox`	`is AMP considered bad? I've tried it out recently and it does seem like hacky mess to be honest..`
`19:41.44`	`Katty`	`Zeeek!`
`19:41.52`	`Zeeek`	`none of them are bad, you just don't learn about * using them`
`19:41.56`	`Katty`	`Zeeek: more muffinery, less insulting.`
`19:42.00`	`[TK]D-Fender`	`All * is evil. A necessary evil for some, but largely NOT.`
`19:42.08`	`[TK]D-Fender`	`GUI's that is!`
`19:42.22`	`hypnox`	`yeah, of course you wont learn asterisk using one`
`19:42.23`	`Zeeek`	`Katty what insult? Soy milk?`
`19:42.55`	`Katty`	`Zeeek: rice milk (=`
`19:42.58`	`Zeeek`	`IMO they cloud the issues so even if you aren't trying to learn it doesn't make it easy for someone else to help`
`19:43.00`	`Katty`	`Zeeek: it's actually better.`
`19:43.14`	`Zeeek`	`Katty I'll try it if they have it next time`
`19:43.36`	`Katty`	`((=`
`19:43.39`	`TheGoD`	`yes, I used restart now`
`19:43.50`	`Zeeek`	`may have to reload zaptel`
`19:43.59`	`[TK]D-Fender`	`I like my red meat (blue & seared), real milk, and the rest of what makes living and breathing worthwhile!`
`19:44.24`	`Zeeek`	`42`
`19:44.55`	`TheGoD`	`well, i'm getting, No such module 'zaptel'`
`19:45.07`	`Zeeek`	`that would keep it from answering`
`19:45.08`	`*** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net)`
`19:45.14`	`lo_tech`	`FarrisG: have you done a 'pri debug span X' to check the presentation on the PRI circuit?`
`19:45.14`	`mover`	`hi all`
`19:45.21`	`TheGoD`	`your talkin bout in the cli right?`
`19:45.25`	`Zeeek`	`TheGoD no`
`19:45.34`	`TheGoD`	`your talking modprobe zaptel?`
`19:45.41`	`Zeeek`	`but I don't know how to reload zaptel on amp other than a reboot`
`19:45.43`	`Zeeek`	`ya`
`19:45.49`	`Zeeek`	`frist you have to rmmod`
`19:45.55`	`TheGoD`	`i'm doing most of it via command line`
`19:46.03`	`Zeeek`	`what modiles you have? all FXO?`
`19:46.04`	`TheGoD`	`I just used amp to setup the extensions`
`19:46.09`	`TheGoD`	`yea fxo`
`19:46.12`	`TheGoD`	`all of them`
`19:46.24`	`Zeeek`	`rmmod wctdm`
`19:46.30`	`Zeeek`	`rmmod zaptel`
`19:46.35`	`mover`	`anyone of the sip guru here alive?`
`19:46.45`	`Zeeek`	`then put them back and ztcfg -vvv`
`19:46.48`	`mover`	`i have a strange problem again :-)`
`19:46.52`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`19:47.05`	`mover`	`but its public and many ppl have it`
`19:47.25`	`*** part/#asterisk jello333 (i=ghento@CPE0011d8a291a6-CM00111ae4684c.cpe.net.cable.rogers.com)`
`19:47.28`	`Zeeek`	`please expand?`
`19:47.28`	`TheGoD`	`done and done`
`19:47.39`	`Zeeek`	`and still no joy?`
`19:47.49`	`TheGoD`	`nope, no joy`
`19:47.51`	`mover`	`if a sip ua do a prefetch register without contact asterisk will fail this register`
`19:48.08`	`TheGoD`	`it did work at one time, actually it was easy to setup`
`19:48.40`	`Zeeek`	`the fxo are working otherwise?`
`19:49.37`	`TheGoD`	`they arn't picking up the phone, when I dial or call the lines. asterisk says it picks up but it doesn't actually do it`
`19:50.04`	`Zeeek`	`show one of the four lines when you ztcfg ?`
`19:50.10`	`Zeeek`	`(they should all be the same excpt channel no)`
`19:50.15`	`*** join/#asterisk r0d3nt\|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`19:50.22`	`Flauto`	`what is the major difference of 1.2.3?`
`19:50.36`	`lo_tech`	`Flauto: bug fixes`
`19:50.36`	`Zeeek`	`it fixes 1.2.2`
`19:51.02`	`Flauto`	`okay`
`19:51.03`	`TheGoD`	`CHANNEL 01: FXS Kewlstart (Default) (Slackes: 01)`
`19:51.04`	`*** join/#asterisk razu (n=razu@213-35-173-39-dsl.prn.estpak.ee)`
`19:51.04`	`eKo1`	`that says it all does it`
`19:51.07`	`TheGoD`	`it configs fine`
`19:51.10`	`Flauto`	`it is better then`
`19:51.35`	`Zeeek`	`TheGoD and you're in the USA?`
`19:51.38`	`TheGoD`	`yeap`
`19:51.47`	`Zeeek`	`well move to russia!`
`19:51.50`	`TheGoD`	`lol`
`19:51.55`	`Zeeek`	`I bet FXO work great there`
`19:52.05`	`Zeeek`	`never need to pickup, someone is already listening :)`
`19:52.16`	`cron`	`:)`
`19:52.21`	`Zeeek`	`can you dial out?`
`19:52.26`	`TheGoD`	`actually its hooked into analog pbx. but it WAS working fine until reboot`
`19:52.28`	`TheGoD`	`nope I can;t`
`19:52.32`	`Zeeek`	`AHA`
`19:52.35`	`TheGoD`	`HAHA funny, russia`
`19:52.39`	`Zeeek`	`"hooked into...."`
`19:52.45`	`Zeeek`	`DANGER`
`19:52.47`	`TheGoD`	`lol`
`19:52.52`	`tuxinator_linux`	`Russia still around?`
`19:52.55`	`TheGoD`	`I had the thing working fine though`
`19:53.02`	`Flauto`	`what is the best way to download asterisk now`
`19:53.05`	`Zeeek`	`till you burned out the modules?`
`19:53.14`	`TheGoD`	`and yes, I have control over the pbx..`
`19:53.14`	`tuxinator_linux`	`Flauto: subversion`
`19:53.20`	`TheGoD`	`zeeek OH?`
`19:53.33`	`Zeeek`	`joking, sort of`
`19:53.37`	`TheGoD`	`heh`
`19:53.45`	`Flauto`	`tuxinator, but it does not have addons and sounds`
`19:53.45`	`Zeeek`	`The FXo is designed to connect to a phone line`
`19:53.59`	`Flauto`	`it is only show zaptel libpri and asterisk`
`19:54.13`	`Zeeek`	`it's possible it can't tell its state connected to something else (i dunno but that's possible)`
`19:54.17`	`TheGoD`	`bah, so are fax machines heh`
`19:54.32`	`Zeeek`	`yeah but their so dumb they don't know`
`19:54.32`	`TheGoD`	`I have not changed anything on the analog pbx ports`
`19:54.48`	`Zeeek`	`the best test wouldbe to hook one to a phone line and see`
`19:55.02`	`Zeeek`	`or, try the 11,000 variations of settings in zaptel.conf`
`19:55.03`	`TheGoD`	`the only thing that changed from working to not working was a reboot`
`19:55.16`	`TheGoD`	`what other settings can I change in zaptel.conf?`
`19:55.18`	`tuxinator_linux`	`Flauto: http://www.asterisk.org/asterisk-converts-to-subversion http://svn.digium.com/view`
`19:55.19`	`Zeeek`	`reboot? You should never reboot an asterisk box`
`19:55.34`	`Zeeek`	`they should always run as soon as the install is done`
`19:55.40`	`TheGoD`	`lol`
`19:55.56`	`TheGoD`	`120 days uptime`
`19:56.00`	`TheGoD`	`it was a sad day I know`
`19:56.02`	`Zeeek`	`take a look in the sample file, there are a lot of settings commented`
`19:56.04`	`*** join/#asterisk FastJack (i=fastjack@p5091E83E.dip.t-dialin.net)`
`19:56.08`	`TheGoD`	`but I had to move it to a rack`
`19:56.31`	`Zeeek`	`it isn't always obvious which technology they are for, but the wiki has some good stuff too`
`19:56.46`	`Zeeek`	`it is odd that is worked and now doesn't I agree`
`19:56.47`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`19:58.05`	`TheGoD`	`damn.. welp i'l have to continue workin on it next week.. Time to bust people for dloading pr0n`
`19:58.30`	`Zeeek`	`haha`
`19:58.32`	`TheGoD`	`THanks for your help, msg me if anything else comes to mind zeeek`
`19:58.34`	`*** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu)`
`19:58.38`	`Zeeek`	`ok`
`19:59.28`	`*** join/#asterisk r0d3nt_m (i=r0d3nt@tinfoilhat.net)`
`20:01.09`	`Mark_Halverson`	`anyone having problems with IAX on SVN? since installing SVN all my IAX dials are adding @ to the end of the number and thus failing`
`20:01.25`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`20:01.27`	`[av]bani`	`O RLY`
`20:01.56`	`harryvv`	`Anyone by any chance know what I need to do to flash over a incomming call on caller waiting on my ip500?`
`20:02.20`	`Zeeek`	`softbutton`
`20:02.28`	`Mark_Halverson`	`Both IAXcomm and IDEfisk fail...reinstalled IAXcomm same prob`
`20:02.47`	`Mark_Halverson`	`Jan 27 11:59:55 NOTICE[24678]: chan_iax2.c:6769 socket_read: Rejected connect attempt from 67.139.119.152, who was trying to reach '18006396111@'`
`20:02.53`	`[TK]D-Fender`	`harryvv : The IP 500 doesn't have an video functionality so "flashing" the caller won't have much effect :)`
`20:03.04`	`justinu`	`heh`
`20:03.08`	`Zeeek`	`TKD LOL`
`20:03.34`	`Mark_Halverson`	`just forward busy to a second line`
`20:03.54`	`tuxinator_linux`	`[TK]D-Fender needs to get back to working so he can support the wife and kids`
`20:03.57`	`*** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`20:04.17`	`Mark_Halverson`	`i need to get rid of the wife and kids so i can stop working`
`20:04.29`	`Zeeek`	`Mark_Halverson the ip500 presents a soft key "Accept" or something without doing that`
`20:04.33`	`[TK]D-Fender`	`tuxinator_linux : that'd be "non-commital SO", and I'm earning $ just sitting here :)`
`20:04.35`	`[av]bani`	`[TK]D-Fender: needs to try decaf. there are many brands on the market that are almost as tasty as the real thing!`
`20:04.37`	`tuxinator_linux`	`I like my wife, no kids yet`
`20:05.05`	`[TK]D-Fender`	`[av]bani : And will cause food-poisoning over a short period of time!`
`20:05.11`	`[av]bani`	`yay!`
`20:05.12`	`Mark_Halverson`	`lol....just playin...today's our anniversary....as for kids...their god's punishment for having sex....lol`
`20:05.23`	`[TK]D-Fender`	`[av]bani : Speaking of which, did you get that new toy of yours provisioned yet?`
`20:05.33`	`[av]bani`	`not autoprovisioned`
`20:05.55`	`[TK]D-Fender`	`[av]bani : DOH! Get off your butt and do it!`
`20:06.04`	`[av]bani`	`besides, you cant deny me my right to whine and grouse about expensive phones`
`20:06.31`	`[TK]D-Fender`	`[av]bani : Right.... you're in enough denial as it is :)`
`20:06.34`	`lo_tech`	`Mark_Halverson: does your dial command have a dial context set? i.e. exten => _1800.,1,Dial(${SomeTrunk}/${EXTEN}@long_distance)`
`20:06.43`	`[av]bani`	`btw, i figured out why the polycoms "sound better"`
`20:06.49`	`[av]bani`	`they do heavy filtering`
`20:07.41`	`Mark_Halverson`	`lo_tech: sure does...was working before...not sure what happened`
`20:08.05`	`[TK]D-Fender`	`[av]bani : Great jitter buffers, hardware, etc...`
`20:08.20`	`[av]bani`	`[TK]D-Fender: the hardware isnt any better, at least not the speakers/mic`
`20:08.27`	`[av]bani`	`they just filter the bejeesus out of the audio`
`20:08.37`	`Zeeek`	`"was working before" what version you running?`
`20:08.40`	`Zeeek`	`1.2.2 ?`
`20:08.45`	`Mark_Halverson`	`yeep`
`20:08.56`	`[av]bani`	`they do seem to have good and deep jitter buffers though`
`20:08.57`	`Zeeek`	`and you saw the release 1.2.3?`
`20:09.02`	`Mark_Halverson`	`then the audio prob the other day and went to SVN`
`20:09.11`	`lo_tech`	`Mark_Halverson: hmm... looks like it's not sending the proper context... you check the iax.conf for 'context='?`
`20:09.25`	`[TK]D-Fender`	`[av]bani : I found a place selling IP601's 10$ cheaper than Atacomm......`
`20:09.31`	`[av]bani`	`:o`
`20:09.37`	`Mark_Halverson`	`if that's it...i'm gonna slam my head in the door....`
`20:09.39`	`[av]bani`	`oen thing annoying about atacomm is their shipping`
`20:09.44`	`[av]bani`	`like $50 shipping for a single phone`
`20:09.54`	`lo_tech`	`Mark_Halverson`
`20:10.00`	`Zeeek`	`i had a huge problem with atacomm once`
`20:10.20`	`[av]bani`	`[TK]D-Fender: interestingly enough, they didnt give us any shit about our spa3k order`
`20:10.22`	`Zeeek`	`made a $500 order on their site`
`20:10.32`	`lo_tech`	`Head-Slamming is a Monday sport, Fridays are reserved for something else`
`20:10.39`	`[av]bani`	`so that 'no spa3k sales' must be a loose rule, or they arent enforcing it quite yet`
`20:10.42`	`acidblood`	`Would anyone please help me?`
`20:10.55`	`burton`	`hello, anyone know what this error mean: > CAPI INFO 0x34e5: Message not compatible with call state`
`20:11.46`	`acidblood`	`Typical open source helpfulness.`
`20:11.57`	`Zeeek`	`acidblood just ask`
`20:12.09`	`*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)`
`20:12.10`	`lo_tech`	`acidblood: you didnt mention what you needed!`
`20:12.18`	`acidblood`	`Actually I've asked two questions back then and nobody answered either.`
`20:12.29`	`Zeeek`	`well then maybe they didn'y know`
`20:12.42`	`burton`	`hello, anyone know what this error mean: > CAPI INFO 0x34e5: Message not compatible with call state ... and than == ISDN1: CAPI Hangingup`
`20:12.43`	`Mark_Halverson`	`lo_tech: nope it's point to the correct context`
`20:12.46`	`Mark_Halverson`	`strange`
`20:12.47`	`lo_tech`	`cant speak for what happens when I'm not online... but are we gonna chat ettiquette or get to issues?`
`20:12.52`	`acidblood`	`I've found my way around those, but I'm guessing if it's worth the effort of writing up my current question, since probably nobody is going to answer it anyway.`
`20:13.17`	`lo_tech`	`Marvin from HHGTG comes to mind...`
`20:13.24`	`Zeeek`	`heh`
`20:14.19`	`*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)`
`20:15.05`	`lo_tech`	`Mark_Halverson: 'iax2 debug' on the box, make a test call and pastebin the first 10 lines or so of the results?`
`20:16.00`	`*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)`
`20:16.26`	`acidblood`	`OK, so I'll try. I'm following the instructions here: http://www.broadvoice.com/support_install_asterisk.html and I've gotten a softphone to log in to Asterisk, and dialing 600 works, but when I try to dial 011xx... it says 404 not found.`
`20:16.27`	`generalhan`	`whats going on everyone ?!`
`20:16.48`	`acidblood`	`So do I need to add 0 or 1 or 9 or whatever in front of the 011xx... number to get it to dial?`
`20:17.37`	`generalhan`	`Can some one explain to me the possible reasons for an error :: unable to create channel type 'SIP' :: why might i be getting this ?`
`20:18.16`	`*** join/#asterisk riksta (n=rick@62.6.163.81)`
`20:18.21`	`junbug`	`generalhan: when do ya get it?`
`20:18.43`	`generalhan`	`junbug: give me a sec to explain what i just did !!`
`20:20.01`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`20:20.16`	`generalhan`	`We just got another office space next door. so i ran 2 cables from the patch pannel in the current office (one for voice one for data) then i threw that into a switch in the new office, connected all the phones to it, and set up all the sip info. but when i try to extension dial to those phones it goes straight to VM and says in the CLI that it cannot create the channel type 'SIP'`
`20:20.39`	`*** join/#asterisk intensedr (n=scolson@209.172.11.52)`
`20:20.53`	`*** join/#asterisk Lord_Drachenblut (n=Lord@12-210-115-191.client.insightBB.com)`
`20:20.55`	`generalhan`	`and it works in either direction, if i try to do an extension dial from those phones to a phone in the new office it says the same thing`
`20:21.25`	`Zeeek`	`generalhan show the command line with Dial() in it`
`20:21.59`	`acidblood`	`See, that's the typical open source helpfulness I'm talking about.`
`20:22.08`	`acidblood`	`This is the third question I ask that has gone unnoticed.`
`20:22.14`	`acidblood`	`I might as well read the poorly written manuals.`
`20:22.16`	`acidblood`	`Goodbye.`
`20:22.25`	`Zeeek`	`what an AH`
`20:22.31`	`eKo1`	`hahaha`
`20:22.40`	`[av]bani`	`i think he should demand a refund`
`20:22.47`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`20:22.47`	`[TK]D-Fender`	`Zeeek : And I was about to help him :)`
`20:22.49`	`justinu`	`lol`
`20:22.53`	`Zeeek`	`yes, indeed for life!`
`20:22.55`	`justinu`	`what a dork`
`20:23.06`	`[TK]D-Fender`	`schmuck thats right... we're all out here to serve YOU....`
`20:23.07`	`Zeeek`	`you fucking open source guys, shit`
`20:23.12`	`justinu`	`complaining about a free product`
`20:23.13`	`lo_tech`	`look, Acidblood... if nobody has any experience with Broadvoice, we'd just be bullshitting you anyway... so maybe you should check with your provider since it's specific to them?`
`20:23.19`	`intensedr`	`you there SWK?`
`20:23.24`	`[TK]D-Fender`	`lo_tech : he LEFT`
`20:23.31`	`generalhan`	`geez`
`20:23.44`	`generalhan`	`i have quesions all the time and i get help ... if you just have some freaking patience`
`20:23.46`	`Zeeek`	`general you didn't answer`
`20:23.57`	`Zeeek`	`the dial command line?`
`20:23.59`	`lo_tech`	`[TK]D-Fender: bumma, and I was just getting around to witty ridicule`
`20:24.13`	`justinu`	`generalhan: impossible, all of us open source people are typically unhelpful :P`
`20:24.18`	`Zeeek`	`yeah the opportunities are so limited here :)`
`20:24.44`	`Zeeek`	`ok, Dial(SIP/nowaythisdevice_exists) will do that`
`20:25.34`	`generalhan`	`zeek`
`20:25.35`	`*** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu)`
`20:25.38`	`sevrdseot`	`Zeeek: I've seen that syntax before, as an asterisk nub, are you entering that into the asterisk CLI?`
`20:25.39`	`generalhan`	`im pastebin'ing it right now`
`20:25.47`	`Zeeek`	`k`
`20:26.10`	`generalhan`	`http://generalhan.pastebin.ca/38834 :: this is my extensions.conf a little bit of each context invovled`
`20:26.26`	`wunderkin`	`speaking of broadvoice problems, caller id is working! yey!`
`20:26.51`	`Zeeek`	`generalhan but what is the console line when you dial?`
`20:26.57`	`Zeeek`	`the one before the error?`
`20:27.30`	`generalhan`	`ohh ... hang on`
`20:28.18`	`generalhan`	`zeeek: http://generalhan.pastebin.ca/38836`
`20:28.53`	`Zeeek`	`looks like 7206 doesn't exist`
`20:28.55`	`[TK]D-Fender`	`generalhan : thats a little backwards...The "extensions" in your [extension-dial] context should be in [internal], and [extension-dial] should "include => internal"`
`20:29.31`	`generalhan`	`[TK]: how come ? i just have the include => extension dial in the [internal] is that not good ?`
`20:29.44`	`*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)`
`20:30.16`	`[TK]D-Fender`	`generalhan : pastebin "sip show peers"`
`20:30.22`	`generalhan`	`roger`
`20:30.28`	`rpm`	`wilco`
`20:30.59`	`generalhan`	`[TK]: you could have just told me to do that ! LOL`
`20:31.01`	`Dr-Linux`	`question, what application asterisk use to sense DTMF digits and pass to next pirority i.e. AGI ... ?`
`20:31.03`	`generalhan`	`I see the issue now !`
`20:31.14`	`generalhan`	`all the new phones have a port number of " 0 "`
`20:31.15`	`s34n`	`My SIP trunk was working. Now incoming calls are ok, outgoing calls error:`
`20:31.17`	`s34n`	`<PROTECTED>`
`20:31.22`	`[TK]D-Fender`	`generalhan : you are mixing ideas. a menu shouldn't contain the direct extensions of people, it should include it from another context. contexts should inherit from SMALLER contexts, not larger.`
`20:31.58`	`s34n`	`the change between working and not working is upgrading from 1.2.0 to 1.2.3`
`20:32.19`	`*** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)`
`20:33.01`	`generalhan`	`[TK]: so i should really put all the Macro calls for each extension in the [internal] context ? or put [extension_dial] in [internal] ?`
`20:33.26`	`[TK]D-Fender`	`generalhan : .... pastebin the SIP show peers plz.....`
`20:33.49`	`generalhan`	`http://generalhan.pastebin.ca/38837`
`20:34.16`	`[av]bani`	`[TK]D-Fender: polycom's phone UI is better than snom's.`
`20:34.19`	`generalhan`	`[TK]: for some reason they are showing the IPs as "unspecified" too ? wth is that all about ?`
`20:34.50`	`[TK]D-Fender`	`7206 doesn't look very "alive" to me ... 7206/7206 (Unspecified) D 255.255.255.255 0 Unmonitored`
`20:35.14`	`generalhan`	`[TK]: yea whats that all about ?>`
`20:35.18`	`[TK]D-Fender`	`[av]bani : Yup.... sure the XML set is pretty big and scary at first, but the user experience is much better...`
`20:35.27`	`[TK]D-Fender`	`generalhan : 7206 = unregistered`
`20:35.40`	`[av]bani`	`[TK]D-Fender: but... snom offers an AMAZING amount of customizability for the programmable buttons`
`20:35.41`	`generalhan`	`[TK]: how can i make it manually register ?`
`20:35.47`	`[TK]D-Fender`	`its just not "talking" for some reason or another...`
`20:35.49`	`[av]bani`	`[TK]D-Fender: you can remap _every_ key on the phone if you want`
`20:35.52`	`generalhan`	`great ....`
`20:36.06`	`Dr-Linux`	`anybody know how to pass DTMF digits to AGI script?`
`20:36.06`	`[TK]D-Fender`	`[av]bani : You can remap stuff on Poly's now a certain amount.`
`20:36.13`	`[av]bani`	`[TK]D-Fender: you can remap _everything_ on snom`
`20:36.24`	`generalhan`	`i know that they have IPs cause im on their webportals right now ... so why wont they register with asterisk ?`
`20:36.27`	`[TK]D-Fender`	`[av]bani : then again... how often do you want to screw around with "common" hard buttons?`
`20:36.49`	`[av]bani`	`[TK]D-Fender: better not to second guess what customers will want to do... something i learned from writing software`
`20:37.04`	`*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)`
`20:37.39`	`[TK]D-Fender`	`[av]bani : And sometimes its better not to let customers fuck it up so bad you can't tell why its not working :)`
`20:38.08`	`[TK]D-Fender`	`[av]bani : "Buy I am pressing "HOLD" !!!!!! (not that it could possibly matter that that isn't what it does any more.......`
`20:38.16`	`[av]bani`	`[TK]D-Fender: ahh you take the apple approach, prevent the customers from doing something useful because they might do something stupid`
`20:38.32`	`[TK]D-Fender`	`[av]bani : Only a little.... it is a PHONE FFS.`
`20:38.32`	`[av]bani`	`[TK]D-Fender: you rather your customers operate wearing straitjackets :)`
`20:38.54`	`FuriousGeorge`	`do all these packet concealment and jitter buffer options in eyebeam do more harm then goo when im on the same network?`
`20:39.00`	`FuriousGeorge`	`good*`
`20:39.06`	`*** join/#asterisk Skarmeth (n=Skarmeth@201009024115.user.veloxzone.com.br)`
`20:39.09`	`generalhan`	`[av]bani: users are stupid .. i have cheap little Aastra phones that you cant do anything with and i have to reset peoples phones ALL the time`
`20:39.12`	`[av]bani`	`whats a good sip softphone?`
`20:39.12`	`[TK]D-Fender`	`[av]bani : I'm already a pretty "liberal" kind of guy tech-wise, but some customizability I'd trade off... not MUCH, but there are limits.`
`20:39.23`	`[TK]D-Fender`	`[av]bani : eyeBeam.`
`20:39.31`	`[av]bani`	`[TK]D-Fender: that the only one?`
`20:39.34`	`*** join/#asterisk angom_h (n=angom@red-corp-200.38.16.10.telnor.net)`
`20:40.04`	`generalhan`	`[TK]D-Fender: any ideas on how to get those phones to register manually to * ?? i have to get this operational by monday morning! lol`
`20:40.06`	`[TK]D-Fender`	`[av]bani : Well X-Pro is about the same without video. All the others I've seen mostly suck. Lacking features, etc.`
`20:40.13`	`[av]bani`	`[TK]D-Fender: as a programmer, i find it's best not to second guess what customers might want to do.`
`20:40.29`	`[TK]D-Fender`	`generalhan : Dunno... you haven't even offered a clue as to their CONDITION or MODEL.`
`20:40.39`	`[av]bani`	`[TK]D-Fender: just remember to give them an easy way to reset the phone :)`
`20:41.21`	`generalhan`	`[TK]: sorry man ... these are Aastra 9112i SIP phones, they seem (from the phone) to be registered because their exten number and display names come up just fine, so im not really sure why its not all working`
`20:41.50`	`[TK]D-Fender`	`[av]bani : In theory yes, but in practice these are standard phones..... Can you change the key-caps on either of these 2 phones? If I press an "envelpoe" button, that should mean DIAL right? And the one with the "mic" on it... thats "redial" right?`
`20:41.55`	`iCEBrkr`	`I'm trying to figure out what crack the programmers are smoking that wrote most of the softphones out there.`
`20:42.05`	`iCEBrkr`	`it's like HEY! Lets make it skinable and take up 75% of your desktop!!!`
`20:42.12`	`iCEBrkr`	`DUMB!`
`20:42.30`	`wunderkin`	`but its pretty!`
`20:42.39`	`[av]bani`	`iCEBrkr: its not the programmers at fault, its management dictating UI design based on focus groups`
`20:43.02`	`[av]bani`	`the programmers are just doing what theyre told: make a shit ui`
`20:43.04`	`iCEBrkr`	`[av]bani: I doubt they spent the money on a focus group.`
`20:43.25`	`[av]bani`	`iCEBrkr: focus groups are the new fad`
`20:43.30`	`[av]bani`	`and management _loves_ fads`
`20:43.34`	`iCEBrkr`	`Why you need a life size phone on your screen, is beyond me.`
`20:43.46`	`[av]bani`	`as long as they read somthing in a management magazine, they NEED IT NOW`
`20:43.52`	`s34n`	`[TK]D-Fender: WARNING[4308]: chan_sip.c:9532 handle_response_invite: Forbidden - wrong password on authentication for INVITE`
`20:43.52`	`iCEBrkr`	`haha`
`20:43.54`	`iCEBrkr`	`How true`
`20:43.58`	`QbY`	`I'm shopping for phones. The SoundPoint IP 601 looks awesome.. But can we fully utilize that big screen, or is it just over kill?`
`20:44.08`	`[TK]D-Fender`	`iCEBrkr : I haven't seen a REALLY nice soft-phone yet.... All lacking the control I see elsewher on hard-phones...`
`20:44.17`	`s34n`	`[TK]D-Fender: that's just since upgrading to 1.2.3`
`20:44.18`	`iCEBrkr`	`[TK]D-Fender: Agreed`
`20:44.21`	`iCEBrkr`	`same here`
`20:44.25`	`[TK]D-Fender`	`s34n : What is says <- bad friggen pass or user!`
`20:44.30`	`[av]bani`	`iCEBrkr: you've never had a boss run to you, all huffing out of breath, pointing to some retarded article in a magazine, and going "CAN WE DO THIS!!@!#@!$"`
`20:44.47`	`iCEBrkr`	`Hell. DialPad has a better interface than any softphone I've seen and it's a damn webpage`
`20:44.56`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`20:44.57`	`Skarmeth`	`someone know good docs about the costs of channels banks/atas/pots vs ip phones?`
`20:45.00`	`[TK]D-Fender`	`[av]bani : If you're used to "jump when I say jump regardless of how reasonable the request is" then fine :)`
`20:45.07`	`generalhan`	`i used a softphone by IP Blue software that was GREAT`
`20:45.10`	`iCEBrkr`	`[av]bani: Sure I have, and I told him he's retarded for reading into that trade-rag`
`20:45.20`	`*** join/#asterisk hroarke (n=whois@216.190.144.90)`
`20:45.25`	`[av]bani`	`[TK]D-Fender: welcome to corporate amerika`
`20:45.27`	`jbalcomb`	`are we to be using zaptel 1.2.2 with asterisk 1.2.3?`
`20:45.31`	`generalhan`	`the skin was a Cisco 7960 w/ all the features of it, so i liked it a lot !`
`20:45.49`	`[av]bani`	`hmm.. eyebeam is osx only?`
`20:46.06`	`iCEBrkr`	`Why can't I have a 100x200 dial pad with mini LCD display with option buttons below it?`
`20:46.06`	`[TK]D-Fender`	`[av]bani : You're just its "whore" :)`
`20:46.15`	`[TK]D-Fender`	`[av]bani : Win/Lin/OSX`
`20:46.31`	`iCEBrkr`	`[av]bani: I have no qualms about telling my boss he's a retard and his ideas are shitty`
`20:46.34`	`[TK]D-Fender`	`iCEBrkr : I've seen one like that actually....`
`20:47.05`	`[TK]D-Fender`	`jbalcomb : yup`
`20:47.24`	`[TK]D-Fender`	`jbalcomb : it was an emergency release for no zaptel problems`
`20:47.28`	`generalhan`	`Can anyone teach me how to force a phone to register with * ?? i cant get 8 of my Aastra 9112i SIP phones to register they are all comming back "unspecified" for the IPs :: http://generalhan.pastebin.ca/38837 ::`
`20:47.41`	`jbalcomb`	`[TK]D-Fender ok. i saw the quick jump from 1.2.2 to 1.2.3 so i figured`
`20:48.03`	`iCEBrkr`	`jbalcomb: Welcome to last week :P`
`20:48.21`	`jbalcomb`	`iCEBrkr ;)`
`20:48.27`	`[TK]D-Fender`	`generalhan : force them to register? Maybe you should watch as one ATTEMPTS to and see where it goes wrong....`
`20:48.28`	`iCEBrkr`	`generalhan: Sounds like DHCP isn't working or something.`
`20:48.30`	`*** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar)`
`20:48.32`	`*** join/#asterisk [Atlas] (n=whois@216.190.144.90)`
`20:49.29`	`generalhan`	`Ok that sounds like a good idea ! lol !`
`20:49.47`	`generalhan`	`iCEBrkr: I dont use DHCP: i manually put in their IPs`
`20:50.21`	`[TK]D-Fender`	`generalhan : EW!`
`20:50.41`	`[TK]D-Fender`	`FUGLY!`
`20:51.07`	`generalhan`	`[TK]: it makes it so much easier for me to administer the phones this way. they were getting new ip addresses every week when i was using DHCP and this way i know what the IP is for every extension in case i need to fix something fast ... or from home even`
`20:51.18`	`jbalcomb`	`[TK]D-Fender iCEBrkr you think maybe removing that card didn't work last night cause i didn't do a 'ztcfg -vv'?`
`20:51.49`	`iCEBrkr`	`jbalcomb: It should do that upon startup`
`20:51.54`	`[TK]D-Fender`	`generalhan : "asterisk -rx sip show peers\|grep 7206" <- there`
`20:52.15`	`jbalcomb`	`iCEBrkr asterisk will or somewhere in an etc config?`
`20:52.26`	`[TK]D-Fender`	`jbalcomb : Could be. Before pulling you should ahve confirmed all the timing suces and port order.`
`20:52.32`	`iCEBrkr`	`jbalcomb: Well, it gets run when you reboot that machine...`
`20:52.52`	`iCEBrkr`	`[TK]D-Fender: Almost seems like a mis-placed dchan: option or something`
`20:53.02`	`jbalcomb`	`iCEBrkr this? modprobe.d/zaptel:install wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg`
`20:53.05`	`[TK]D-Fender`	`DHCP saves you from screwing up and double-allocating, etc.... now THATS a nightmare.. when you have to debug IP's, not phones...`
`20:53.34`	`iCEBrkr`	`jbalcomb: yeah`
`20:53.43`	`jbalcomb`	`[TK]D-Fender yeah, you keep talking about timing sources and port order but i dont know how to find or configure this information..`
`20:53.59`	`iCEBrkr`	`[TK]D-Fender: Yea, I'm still not sure where people come up with these ideas about administration and implementation..`
`20:54.07`	`jbalcomb`	`iCEBrkr ok, well, i guess thats in check then. im still hoping to find /something/`
`20:54.12`	`[TK]D-Fender`	`jbalcomb : Well we'd need to see the original zaptel / zapata.`
`20:54.27`	`jbalcomb`	`[TK]D-Fender i pastebin'd them earlier..`
`20:54.32`	`[TK]D-Fender`	`jbalcomb : I use`
`20:54.37`	`[TK]D-Fender`	`"/clear" a lot`
`20:54.41`	`*** join/#asterisk zotz (n=zotz@24.231.47.175)`
`20:54.45`	`iCEBrkr`	`hehe me too`
`20:54.55`	`iCEBrkr`	`So know where new content starts on my screen.`
`20:54.57`	`[TK]D-Fender`	`every time I think I won't need to look back.`
`20:55.00`	`jbalcomb`	`[TK]D-Fender iCEBrkr haha.. i just used it for the first time`
`20:55.28`	`iCEBrkr`	`I use /clear everytime I'm done reading stuff on the screen. So I can easily tell there's new content on channel`
`20:55.31`	`iCEBrkr`	`I'm such a nerd`
`20:56.01`	`[TK]D-Fender`	`iCEBrkr : I use it much the same... also make sure my writing stands out to see how much is new`
`20:56.15`	`iCEBrkr`	`irsii does that`
`20:56.28`	`iCEBrkr`	`It'll hilite your name and things directed at you`
`20:56.32`	`[TK]D-Fender`	`So does mIRC :) I'm in a Windows world here...`
`20:56.39`	`wunderkin`	`heh me too!`
`20:56.42`	`drumkilla`	`irssi rocks :)`
`20:56.49`	`iCEBrkr`	`8 o`
`20:56.56`	`jbalcomb`	`[TK]D-Fender iCEBrkr http://pastebin.com/526293`
`20:57.22`	`[TK]D-Fender`	`[av]bani : Just saw your SPA-3000 thread on the mailing list :)`
`20:57.39`	`iCEBrkr`	`#`
`20:57.39`	`iCEBrkr`	`rxgain=-4.5`
`20:57.39`	`iCEBrkr`	`#`
`20:57.39`	`iCEBrkr`	`txgain=-16`
`20:57.43`	`iCEBrkr`	`HOLYSHIT`
`20:58.07`	`[TK]D-Fender`	`jb, you were using your TE110 and 1 port of your TE411P?`
`20:58.09`	`generalhan`	`[TK]: "asterisk -rx sip show peers\|grep 7206" what was that suposed to do ? it gave me no output at all`
`20:58.20`	`[TK]D-Fender`	`iCEBrkr : thats what I said too.... serious echo problems...`
`20:58.52`	`FuriousGeorge`	`if i wanted to make a queue where the caller just heard music while * alternated between ringing some extensions and waiting, i would do that with dialplan logic (not with agents.conf, queue.conf, and the queue app) right?`
`20:59.20`	`FuriousGeorge`	`i could have a context queue, and just do the right thing?`
`20:59.41`	`[TK]D-Fender`	`generalhan - asterisk -rx "sip show peers"\|grep 7206`
`20:59.47`	`[TK]D-Fender`	`missed the quotes`
`21:00.07`	`generalhan`	`ohh ok ! thanks !`
`21:00.23`	`[TK]D-Fender`	`FuriousGeorge : Could work either way`
`21:00.54`	`generalhan`	`[`
`21:01.40`	`*** join/#asterisk TheGoD (n=TheGoD@adsl-70-224-56-152.dsl.sbndin.ameritech.net)`
`21:02.02`	`FuriousGeorge`	`[TK]D-Fender: i dont want people to log in and out as agents to do it, if i have no agents can i use the dialplan queue apps?`
`21:02.19`	`*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)`
`21:02.25`	`[TK]D-Fender`	`FuriousGeorge : Sure, just use static agents`
`21:02.26`	`iCEBrkr`	`jbalcomb: span=1,1,5,esf,b8zs`
`21:02.29`	`iCEBrkr`	`that seems odd.. 5?`
`21:02.32`	`*** join/#asterisk kimc (n=freenode@c-68-43-224-10.hsd1.mi.comcast.net)`
`21:02.36`	`FuriousGeorge`	`[TK]D-Fender: thanks`
`21:02.49`	`[TK]D-Fender`	`iCEBrkr : I asked about that too.... his last tech just sorta "shoved it in" :)`
`21:02.58`	`iCEBrkr`	`:-/`
`21:03.13`	`[TK]D-Fender`	`FuriousGeorge : And don't give them "PauseQueueAgent" access`
`21:03.36`	`generalhan`	`[TK]D-Fender: i just restarted the phone and asterisk -rx "sip show peers"\|grep 7209 just keeps saying the same thing "unspecified" ??`
`21:03.39`	`[TK]D-Fender`	`errr PauseQueueMember rather`
`21:03.49`	`jbalcomb`	`iCEBrkr yeah, it aint my business at this point. the people at verizon couldnt tell me what LBO was or what it should be set to.`
`21:03.52`	`[TK]D-Fender`	`generalhan : then maybe your PHONE is screwed up`
`21:04.08`	`iCEBrkr`	`jbalcomb: I'd set it to 0.. but that's just me.`
`21:04.11`	`[TK]D-Fender`	`jbalcomb : "0" if you can see your SmartJack from your server.....`
`21:04.17`	`jbalcomb`	`iCEBrkr I would assume since the little box on the wall is only 10 feet away a 0 would suffice`
`21:04.22`	`[TK]D-Fender`	`YES`
`21:04.25`	`iCEBrkr`	`Exactly`
`21:04.28`	`jbalcomb`	`[TK]D-Fender iCEBrkr agreed.`
`21:04.32`	`generalhan`	`[TK]D-Fender: i dont know how its possible that ALL 8 of the phones dont work`
`21:04.37`	`jbalcomb`	`[TK]D-Fender iCEBrkr how do I convince 'Paul`
`21:04.45`	`iCEBrkr`	`jbalcomb: I wonder if that's why you got your echo problems.`
`21:04.51`	`jbalcomb`	`[TK]D-Fender iCEBrkr ' the other phone guy to leave shit alone?`
`21:04.51`	`[TK]D-Fender`	`generalhan : Fighting for IP's perhaps? Who's to say...`
`21:05.02`	`jbalcomb`	`iCEBrkr no, he changed it during the 'trouble shooting'`
`21:05.09`	`[TK]D-Fender`	`jbalcomb : High voltage :D`
`21:05.29`	`jbalcomb`	`[TK]D-Fender :D that might be part of your contract..`
`21:05.36`	`iCEBrkr`	`Bzzzzzzzzzzzot!`
`21:05.40`	`generalhan`	`[TK]D-Fender: ok everyone hates that i dont use DHCP so i will ... haha`
`21:05.51`	`*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)`
`21:05.52`	`iCEBrkr`	`generalhan: It'll make your life easier.. Trust us`
`21:06.05`	`[TK]D-Fender`	`"cyanide, TNT,.... HIGH VOLTAGE!"`
`21:06.36`	`[TK]D-Fender`	`Dirty deeds and they're done dirt cheap......`
`21:06.38`	`jbalcomb`	`generalhan I think DHCP is a pain in the nuts personally but if you support it with an automated system identification logging system (IE a web page with user=IP) then its alright.`
`21:06.51`	`*** join/#asterisk FuLg0r3 (n=fulgas@a81-84-117-84.cpe.netcabo.pt)`
`21:07.23`	`*** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net)`
`21:07.37`	`[TK]D-Fender`	`jbalcomb : He's jsut wondering how to debug a phone is you don't know its IP... well thats pretty easy if its registered, and easier still if you "grep" and nmap and cross reference with a MAC list...`
`21:07.52`	`[TK]D-Fender`	`Hell I could do it and I SUCK at Linux!`
`21:08.30`	`generalhan`	`Well everyone .... im need to go destroy something ... im getting pissed off. even with DHCP enabled on the phone it STILL wont freaking register`
`21:08.40`	`jbalcomb`	`[TK]D-Fender its in the DB too aint it by ext.?`
`21:08.47`	`*** join/#asterisk hardwire (n=hardwire@209-112-208-243-cdsl-rb1.nwc.acsalaska.net)`
`21:08.51`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`21:08.52`	`jbalcomb`	`[TK]D-Fender we have a php page that has that info`
`21:09.15`	`*** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net)`
`21:09.20`	`jbalcomb`	`generalhan destory something? stop by my office and ask for Paul`
`21:09.43`	`generalhan`	`haha sure ... what State ?`
`21:09.51`	`lo_tech`	`Altered`
`21:10.04`	`[TK]D-Fender`	`jbalcomb : could be... until your setup, I never looked :)`
`21:11.20`	`generalhan`	`i just dont know what else to do besides breaking something ! LOL these phones were SOOO easy to set up in this office, what is the big deal about being next door that these phones dont like ?`
`21:11.48`	`}btorch{`	`is there a sphinx channel ?`
`21:13.17`	`jbalcomb`	`generalhan you running VLANs?`
`21:13.17`	`jbalcomb`	`generalhan different subnet?`
`21:13.17`	`Dr-Linux`	`anybody know how to pass DTMF digits to AGI script? does asterisk us any application for this?`
`21:13.19`	`jbalcomb`	`generalhan is everything else working next door working?`
`21:13.20`	`jbalcomb`	`Dr-Linux maybe.. dial()?`
`21:13.20`	`jbalcomb`	`:/`
`21:13.20`	`*** join/#asterisk tony__ (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)`
`21:13.25`	`generalhan`	`jbalcomb: no VLANs and they are the same subnet same network`
`21:13.57`	`generalhan`	`??`
`21:14.29`	`generalhan`	`well the data network is the same as the voice for right now, so the fact that i can log them in on their computers over there tells me that its all wired correctly at least`
`21:14.42`	`Dr-Linux`	`jbalcomb: Dial()?`
`21:15.06`	`*** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM]`
`21:15.06`	`*** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) [NETSPLIT VICTIM]`
`21:15.06`	`*** join/#asterisk Cazper (n=cazper@c5100A229.sdsl.catch.no) [NETSPLIT VICTIM]`
`21:15.06`	`*** join/#asterisk Brumle (n=brumle@brumle.com) [NETSPLIT VICTIM]`
`21:15.34`	`Dr-Linux`	`jbalcomb: i mean, if they caller press 16 DTMF digits and goes to AGI external, how they script will know that what digits caller hit ?`
`21:16.42`	`iCEBrkr`	`Dr-Linux: Read(16digits,soundprompt)`
`21:17.08`	`*** join/#asterisk st3v (n=st3v@netblock-66-218-41-231.dslextreme.com)`
`21:17.49`	`lo_tech`	`Dr-Linux: we use AGI->get_variable('EXTEN');`
`21:18.01`	`st3v`	`We have 4 POTS lines with 3-way calling. Is it possible to have asterisk connect two outside people on each line (so up to 8 people on 4 lines)`
`21:18.43`	`Dr-Linux`	`iCEBrkr: in this case next pirority will be AGI(script.agi), how it will know what digits came from caller?`
`21:18.46`	`*** join/#asterisk gaspiz (n=gaspiz@86.34.6.164)`
`21:19.00`	`*** join/#asterisk dsfr (n=dsfr@gateway.digium.com)`
`21:19.36`	`gaspiz`	`hi, I'm using 1.2.1 and I'm having problems with same extension defined in more contexts`
`21:19.57`	`gaspiz`	`it's not taking the password corectly (not finding the user?)`
`21:20.19`	`iCEBrkr`	`Dr-Linux: Cuz when you Read() it stuffs the digits in the variable..`
`21:20.22`	`*** join/#asterisk tainted_ (n=somewher@mail.k2usa.com)`
`21:20.30`	`iCEBrkr`	`Dr-Linux: From your script.agi, you GET VARIABLE`
`21:20.36`	`tainted_`	`anyone have problems with grandstream ATAs not hanging up?`
`21:21.14`	`iCEBrkr`	`gaspiz: Huh, your problem is self explainatory`
`21:21.19`	`*** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net)`
`21:21.32`	`nroej`	`can anyone recommend the iaxy im planning to buy one`
`21:21.34`	`nroej`	`?!`
`21:21.54`	`gaspiz`	`not quite 1001@context1 not equal to 1001@context2`
`21:22.30`	`[av]bani`	`any good win32 softphones?`
`21:22.46`	`*** join/#asterisk m0narch (n=r3b3l@melloyello.mmi.net)`
`21:23.01`	`*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)`
`21:23.04`	`iCEBrkr`	`gaspiz: That wouldn't give you an error.`
`21:23.27`	`gaspiz`	`but still doesn't work`
`21:23.33`	`gaspiz`	`any ideas?`
`21:23.36`	`iCEBrkr`	`gaspiz: I have a feeling your dialplan is hosed a little`
`21:24.15`	`gaspiz`	`icebrkr: -- AGI Script Executing Application: (voicemailmain) Options: (1001@company_7)`
`21:24.23`	`iCEBrkr`	`Because I'm sure I have exten => 1,s in multiple contexts and my system doesn't complain`
`21:24.40`	`Dr-Linux`	`iCEBrkr: so in this case i need both pirority1 Read() and pirority2 AGI(file)?`
`21:25.35`	`Dr-Linux`	`iCEBrkr: or i need only AGI(file.agi) and the script will sense the DTMF digits?`
`21:26.16`	`Mother`	`greetings`
`21:26.21`	`Mother`	`do I read critical update?`
`21:26.29`	`iCEBrkr`	`Dr-Linux: Personally, I'd try to keep all that I can in the dialplan and only use the AGI() Stuff as a helper`
`21:26.35`	`[TK]D-Fender`	`;lok, time to go home! Later all!`
`21:26.37`	`Mother`	`will boxes running 1.2.2 spontaneously combust?`
`21:26.50`	`iCEBrkr`	`Mother: yea, 2days ago they should have`
`21:27.02`	`iCEBrkr`	`gaspiz: Ok, what about that AGI?`
`21:27.23`	`Mother`	`iCEBrkr: lol OK`
`21:27.33`	`*** join/#asterisk zwi (n=chirsch@66.7.170.26)`
`21:28.24`	`*** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu)`
`21:28.35`	`jbalcomb`	`st3v you want asterisk to implement the three way to call another party and join them on the line?`
`21:30.39`	`s34n`	`my sip trunk allows incoming calls, but throws authentication errors on outgoing calls`
`21:31.01`	`s34n`	`their is no authetication on the far side of the trunk`
`21:31.04`	`iCEBrkr`	`s34n: I think that may be a peer vs. friend issue`
`21:31.22`	`iCEBrkr`	`anyhow, I'm going hom`
`21:31.23`	`s34n`	`it was working in 1.2.0, broken in 1.2.3`
`21:31.24`	`iCEBrkr`	`e`
`21:31.57`	`iCEBrkr`	`So why'd ya upgrade to 1.2.3? LOL`
`21:32.13`	`*** part/#asterisk rob314 (n=root@207.58.194.55)`
`21:32.18`	`Mother`	`s34n: that's why I do upgrades by harddrive, stick one in, if it doesn't work, stick the old one back`
`21:32.24`	`*** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com)`
`21:32.32`	`Mother`	`unless of course you can afford the downtime`
`21:32.54`	`s34n`	`Mother: there's no real downtime, it wasn't in production`
`21:32.55`	`wunderkin`	`im waiting for a refi to go through, and im playing with my dialplan.. unfortunately the mortgage company decided to call in the middle of my playing so they got hung up on twice.. oops :( ive never done cid matching under exten before`
`21:33.06`	`Dr-Linux`	`iCEBrkr: actually i just want to pass DTFM digits to a AGI script that caller hits`
`21:33.24`	`Mother`	`s34n: not that bad then`
`21:33.30`	`iCEBrkr`	`Dr-Linux: Like I said...`
`21:33.37`	`iCEBrkr`	`Dr-Linux: Read() first, then AGI()`
`21:33.38`	`Zodiacal`	`anyone know why i can't get my softphone to login to asterisk? heres the error i get: chan_sip.c:10815 handle_request_register: Registration from 'user1 <sip:100@10.0.0.3>' failed for '10.0.0.2' - Username/auth name mismatch`
`21:33.39`	`TheGoD`	`zeeek are you still there?`
`21:33.48`	`TheGoD`	`bah guess not`
`21:34.01`	`Zodiacal`	`and heres my sip conf phone section: http://pastebin.com/526385`
`21:34.11`	`st3v`	`jbalcomb: yeah, I would like to use the 4 lines to conference more than 4 people`
`21:34.19`	`s34n`	`Mother: right. I do need to figure it out, though.`
`21:34.20`	`st3v`	`so I was thinking to use 3-way calling`
`21:34.23`	`st3v`	`on each line`
`21:35.20`	`st3v`	`but only when in a conference room`
`21:35.37`	`Dr-Linux`	`iCEBrkr: Read() will it pass hited dtfm digits to nex pirority i.e AGI ?`
`21:35.57`	`iCEBrkr`	`Dr-Linux: ok, dude.... I can't help you if you dont' listen.`
`21:36.19`	`jbalcomb`	`st3v i would think you might be able to set up an app to do that with some dialplan logic. ie. 91XXX-XXX-XXXX tells asterisk to three way on line 1 to the number dialed, etc.`
`21:36.26`	`Dr-Linux`	`:S ok`
`21:36.45`	`*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)`
`21:37.27`	`iCEBrkr`	`Dr-Linux: Read up how Read() works.. and then how to use the GET VARIABLE function in AGI`
`21:38.59`	`*** join/#asterisk gandhijee (n=user@pool-70-104-238-126.fred.east.verizon.net)`
`21:39.17`	`gandhijee`	`anyone knows how asterisk plays with IPv6?`
`21:39.28`	`Dr-Linux`	`iCEBrkr: ok thanks`
`21:39.30`	`gandhijee`	`i think i might try to hack one of the snom phones i have to support IPv6`
`21:40.45`	`*** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it)`
`21:46.46`	`*** join/#asterisk FlipZZZ (n=FlipZZZ@216.138.184.74)`
`21:47.09`	`FarrisG`	`http://pastebin.com/526406 <-- anyone mind taking a look at my zapata.conf and helping me figure out why callerid works for some lines and doesn't for others?`
`21:47.22`	`FlipZZZ`	`anyone have any ideas on calls sounding like they are underwater about 20% of the time?`
`21:48.54`	`*** join/#asterisk copantl (n=galel@63.245.93.138)`
`21:51.56`	`Mother`	`any comments on the S101I?`
`21:52.16`	`gaspiz`	`how do I make an .so from a new .c file (I downloaded a bug fixed version of app_voicemail.c and want to install it)?`
`21:52.30`	`Mother`	`(the new IAXy)`
`21:52.40`	`*** part/#asterisk Sjeemz (i=sjeemz@ipv6.sjeemz.nl)`
`21:54.23`	`gaspiz`	`any idea?`
`21:54.57`	`*** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it)`
`21:55.20`	`*** part/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il)`
`21:55.41`	`anthm`	`try /usr/src/asterisk/contrib/scripts/astxs -install /path/to/module_file.c`
`21:57.11`	`*** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`21:58.21`	`gaspiz`	`should I backup my old app_voicemail.conf?`
`21:58.51`	`brad_mssw`	`Mother: got one, it sucks`
`22:00.07`	`brad_mssw`	`Mother: loses sync all the time, on a local 100Mb network (haven't looked for new firmware)... hard to configure ... only supports ulaw/alaw ... expensive ... better off with a SIP ATA (like the LinkSys PAP2) for cheaper`
`22:00.08`	`*** join/#asterisk swineone (n=acidbloo@201.47.33.249)`
`22:00.17`	`swineone`	`hi i have this problem with asterisk`
`22:00.37`	`swineone`	`i configured xlite to register with asterisk but when i try to make a call it doesnt work`
`22:00.51`	`*** join/#asterisk greendisease (n=jack@fedora/greendisease)`
`22:00.54`	`*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)`
`22:01.07`	`swineone`	`now i did tcpdump port 5060 and figured that xlite is trying to connect to my external dsl ip instead of the internal network`
`22:01.28`	`swineone`	`however i never mentioned the external ip anywhere on xlite config so it must be an asterisk error`
`22:01.49`	`Weezey`	`mwahaha`
`22:01.55`	`swineone`	`what should i do?`
`22:02.02`	`Weezey`	`I'm brute forcing a Vonage PAP2`
`22:02.20`	`tzanger`	`yeah.. good luck iwht that`
`22:02.26`	`*** join/#asterisk bkw__ (n=brian@m210e36d0.tmodns.net)`
`22:02.29`	`tzanger`	`mind you it's probably something stupid like "vonagepap2"`
`22:02.34`	`swineone`	`i also get errors like WARNING[520]: Maximum retries exceeded on call C2D61AFE-8F7F-11DA-87BE-000D937A6A1C@192.168.1.5 for seqno 28078 (Non-critical Response)`
`22:02.42`	`brad_mssw`	`swineone: did you set externalip= in sip.conf and set nat=yes globally or something?`
`22:02.46`	`Weezey`	`tzanger: actually I haven't tried that one.`
`22:02.49`	`*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-61.nas28.salt-lake-city1.ut.us.da.qwest.net)`
`22:02.50`	`*** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0)`
`22:03.04`	`swineone`	`brad_mssw: i set nat=yes but no external ip because my ip changes everyday`
`22:03.16`	`brad_mssw`	`swineone: don't set nat=yes globally`
`22:03.31`	`swineone`	`hmm so i should set it per extension?`
`22:03.33`	`brad_mssw`	`swineone: only set it under the profiles that you know are connecting via nat ...`
`22:03.45`	`[TK]D-Fender`	`swineone : you need a dynamic DNS serive and use "EXTERNHOST" then`
`22:03.54`	`[TK]D-Fender`	`service*`
`22:03.54`	`*** join/#asterisk bjohnson__ (n=bjohnson@jecinc.tor.istop.com)`
`22:04.07`	`brad_mssw`	`swineone: as far as your externip= you must set that if you plan on allowing external sip connections`
`22:04.09`	`swineone`	`funny thing is it was working for a while then i got this error`
`22:04.23`	`swineone`	`brad_mssw: right now i'd be glad to dial 1234 successfully`
`22:04.44`	`Weezey`	`tzanger: the hard part was figuring out md5 authentication for HTTP.`
`22:04.59`	`*** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it)`
`22:05.57`	`swineone`	`brad_mssw: took out global nat=yes, still no dice`
`22:06.21`	`brad_mssw`	`swineone: make sure your client doesn't think it's behind a nat either ...`
`22:06.47`	`[TK]D-Fender`	`Geez, my monitor's price just dived again....`
`22:06.56`	`swineone`	`i disabled autodetect ip and forced firewall type to open ip`
`22:07.07`	`Weezey`	`[TK]D-Fender: which one?`
`22:07.28`	`Mother`	`brad_mssw: thanks`
`22:07.45`	`[TK]D-Fender`	`Weezey : Acer AL1916W 19" Wide-Screen LCD 19" 1440x900`
`22:07.50`	`swineone`	`maybe if i paste the xlite logs it could help?`
`22:08.54`	`[av]bani`	`[TK]D-Fender: point for snom: display xml url can be updated by asterisk via sip notify`
`22:08.55`	`[TK]D-Fender`	`swineone : * is behind NAT? Where is X-Lite located?`
`22:09.14`	`swineone`	`http://pastebin.ca/38845`
`22:09.20`	`swineone`	`this is me trying to dial 1234`
`22:09.24`	`SibrPhrek`	`[TK]D-Fender: x-lite?`
`22:09.29`	`SibrPhrek`	`[TK]D-Fender: what OS`
`22:09.29`	`swineone`	`xlite is on the same box asterisk's loaded`
`22:09.54`	`[TK]D-Fender`	`[av]bani : True, but you can also change what the XML page DOES according to phone by passing a parm in the URL.... which is what I do to personalize them.`
`22:10.11`	`*** join/#asterisk BladeRunner05 (n=feelme@adsl-48-220.37-151.net24.it)`
`22:10.49`	`*** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net)`
`22:10.54`	`[av]bani`	`[TK]D-Fender: it means asterisk can totally drive the snom, and instantaneously update the display instead of waiting for the phone to poll`
`22:11.19`	`*** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com)`
`22:12.05`	`generalhan`	`[TK]D-Fender: ok so those phones can call out ... but still no extension dial... if i pastebin my extensions.conf could you tell me what you were talking about when you said putting a smaller context in a large one, rather than the other way around.`
`22:12.15`	`swineone`	`so no idea what's wrong with my config?`
`22:12.21`	`[av]bani`	`[TK]D-Fender: point against: the display is huge, but lo-rez, so its like looking at a trs-80 :D`
`22:12.38`	`[av]bani`	`[TK]D-Fender: however... the ciscos are just as bad`
`22:12.47`	`*** join/#asterisk mattwj2005 (n=Matt@dialup-4.254.85.129.Dial1.Chicago1.Level3.net)`
`22:13.28`	`*** join/#asterisk jtdintulsa (n=jtdintul@lancer.mbo.net)`
`22:13.43`	`Weezey`	`swineone: I can't call you.`
`22:14.05`	`jtdintulsa`	`How do I register a Linphone ?`
`22:14.25`	`swineone`	`from looking at the logs seems like xlite is the problem`
`22:14.41`	`*** join/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru)`
`22:14.42`	`swineone`	`its figuring out the external ip and trying to connect to that`
`22:16.44`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`22:16.49`	`*** part/#asterisk SplasPood (i=jwb@206.252.198.100)`
`22:17.04`	`[TK]D-Fender`	`[av]bani : Yeah, thats an upside. There is a Polycom "request for feature" page on their site I'm going to vote for that amongst other things.`
`22:18.44`	`[TK]D-Fender`	`[av]bani : no "perfect" phone out there right now, but I think Polycom is ahead overall. Snom has better presence / price, Poly better sound / screen res, Cisco better speakerphone, screen size.`
`22:19.18`	`brad_mssw`	`do the polycoms have a backlight ?`
`22:19.29`	`sherbang`	`Cisco's speakerphone is better then the polycom's?`
`22:19.48`	`brad_mssw`	`the linksys 941 kills me when it's dark in the room`
`22:19.54`	`brad_mssw`	`since there's no backlight`
`22:20.30`	`[TK]D-Fender`	`brad_mssw : nope....`
`22:20.49`	`[TK]D-Fender`	`sherbang : thats the common concensus. Not necessariy by much. Both are good though`
`22:21.09`	`generalhan`	`Will some one please take a look at my extensions.comf file and tell me what i have setup incorrectly ? i pasted the error on the CLI at the bottom of the config file. if anyone could please take a look for me. :: http://generalhan.pastebin.ca/38847 ::`
`22:21.09`	`[TK]D-Fender`	`brad_mssw : I own a 941 which sits right next to my IP 601.`
`22:21.23`	`generalhan`	`the phones can dial out but cant extension dial or have their extensions dialed`
`22:22.06`	`sherbang`	`[TK]D-Fender: thanks, good to know.`
`22:22.41`	`SibrPhrek`	`anyone get musiconhold to work with streaming music?`
`22:22.58`	`*** join/#asterisk joshua_ (i=joshua@cl-5.chi-01.us.sixxs.net)`
`22:23.22`	`joshua_`	`hi -- I noticed that * supports ALSA sound cards. does it support ringing and taking them on and off-line?`
`22:23.26`	`joshua_`	`(i.e., alsa modems?)`
`22:23.51`	`*** join/#asterisk Seldon1975 (n=someone@199.243.101.131)`
`22:25.13`	`m0narch`	`<newbie> Can someone help a new *@home user with a REALLY basic problem? </newbie> We can take it offline`
`22:25.42`	`*** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)`
`22:25.52`	`*** join/#asterisk SplasPood (i=jwb@206.252.198.100)`
`22:26.33`	`swineone`	`ok so now i've got to the point that xlite will say 404 not found for everything i try to call`
`22:26.33`	`*** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com)`
`22:27.28`	`[TK]D-Fender`	`m0narch : try #amportal`
`22:27.41`	`swineone`	`here's what sip show peers outputs, is there anything wrong?`
`22:27.46`	`swineone`	`Name/username Host Dyn Nat ACL Mask Port Status`
`22:27.47`	`swineone`	`1337 (Unspecified) D 255.255.255.255 0 Unmonitored`
`22:27.54`	`[TK]D-Fender`	`swineone : So X-Lite is running on the same box as *?`
`22:27.56`	`*** join/#asterisk Seldon19751 (n=someone@toronto-HSE-ppp4239807.sympatico.ca)`
`22:28.03`	`swineone`	`[TK]D-Fend yeah`
`22:28.08`	`[TK]D-Fender`	`swineone : Its clearly not registered`
`22:28.26`	`generalhan`	`[TK]: lol youre getting nailed with these not registered questions today`
`22:28.27`	`swineone`	`sip show registry says it's registered with my provider`
`22:28.50`	`*** part/#asterisk mattwj2005 (n=Matt@dialup-4.254.85.129.Dial1.Chicago1.Level3.net)`
`22:28.51`	`generalhan`	`i still cant figure mine out either... it doesnt make any sense ... even with DHCP activated it still wont register with the asterisk server`
`22:29.06`	`swineone`	`i configured xlite according to this howto http://www.astmasters.net/howtos.html`
`22:29.44`	`*** join/#asterisk Drew___ (n=foo@zux221-065-169.adsl.green.ch)`
`22:30.54`	`Mother`	`http://www.theregister.co.uk/2006/01/27/bt_voip/ <- sure won't have anything to do with the 1.2.2 thing right? :)`
`22:31.25`	`generalhan`	`can some one help me make my 8 new phones register with * ?? lol . this is rediculous. :: http://generalhan.pastebin.ca/38848 ::`
`22:31.33`	`*** join/#asterisk r_evolution (i=_evoluti@12.155.106.12)`
`22:33.13`	`iCEBrkr`	`generalhan: and you created entries in sip.conf for these phones, right :P`
`22:33.37`	`generalhan`	`iCEBrkr: yes i did`
`22:33.48`	`generalhan`	`and in voicemail.conf and in extensions.conf`
`22:34.11`	`iCEBrkr`	`generalhan: Well, sip.conf is the important part here with your problem`
`22:34.59`	`Drew___`	`pastebin your sip.conf - if you have a sip provider remeber to xxxx-out the pswd's`
`22:35.01`	`[TK]D-Fender`	`swineone : Wait... you're trying to use X-Lite ON your * box to directly register with a VoIP provider?`
`22:35.35`	`SplasPood`	`Hrm.. I've noticed that when asterisk is bound to 0.0.0.0 in sip conf it doesn't seem to reply on the same IP the connections come in on.. instead preferring the primary IP of the interface... Anyone experienced this?`
`22:36.35`	`*** join/#asterisk elg (n=fugalh@falcon.fugal.net)`
`22:37.02`	`generalhan`	`http://generalhan.pastebin.ca/38849 <<<---------------- Sip.conf`
`22:37.13`	`[TK]D-Fender`	`generalhan : If the phone isn't registering it could be any number of things including the phone begin defective, not setup right, not on the right network segment, etc....`
`22:37.29`	`[TK]D-Fender`	`generalhan : that pastebin won't help anyone help you....`
`22:38.16`	`SplasPood`	`well`
`22:38.17`	`generalhan`	`well i have it set up on the same network as all the other Aastra phones i have and they all work just fine. same subnet, same gateway, everything but the register name and password is the same ... so i dont understand why it just wont work !`
`22:38.19`	`SplasPood`	`the extra space`
`22:38.25`	`SplasPood`	`after [`
`22:38.27`	`SplasPood`	`might be a problem..`
`22:38.47`	`SplasPood`	`oh nevermind`
`22:38.52`	`SplasPood`	`thats just a non monospaced font`
`22:38.56`	`swineone`	`[TK]D-Fend: no, i'm using xlite to register with asterisk which is registering with a voip provider.`
`22:39.01`	`[TK]D-Fender`	`generalhan : Are all of the phones the same model?`
`22:39.10`	`swineone`	`i want to make calls to my provider via asterisk`
`22:39.16`	`generalhan`	`all but the ciscos that are defined at the very bottom of my config`
`22:39.24`	`[TK]D-Fender`	`swineone : Well x-lite isn't successfully registering with * yet.`
`22:39.26`	`SplasPood`	`although thats a very odd looking font..`
`22:39.40`	`generalhan`	`we only have 2 models of phones here ... Aastra 9112i SIP Phones, and Cisco 7960s`
`22:39.53`	`[TK]D-Fender`	`generalhan : And do some of the 9112's work?`
`22:40.00`	`swineone`	`[TK]D-Fend any special settings i should look for?`
`22:40.04`	`generalhan`	`all of them but the 8 new ones in the new space`
`22:40.12`	`swineone`	`im using the howto here http://www.astmasters.net/howtos.html`
`22:40.13`	`[TK]D-Fender`	`swineone : pastebin your sip.conf`
`22:40.17`	`swineone`	`ok`
`22:40.36`	`r_evolution`	`hey it's TK the channel bot`
`22:40.53`	`generalhan`	`[TK] i have been using 16 of these Aastra phones perfectly for about 6 months now ... these new ones are the same model and everything just next door, and they wont work`
`22:41.34`	`swineone`	`http://pastebin.ca/38850`
`22:41.42`	`*** join/#asterisk jamig (n=jamig@adsl-215-88-196.mia.bellsouth.net)`
`22:42.13`	`*** part/#asterisk xachen (i=justin@magnum.thisgeek.com)`
`22:42.32`	`jamig`	`can anyone point me in the right direction for the astinstaller script on the asterisk site?`
`22:43.00`	`swineone`	`[TK]D-Fend so anything wrong?`
`22:43.01`	`dogtanian`	`jamig: i think asterisk@home is what your looking for`
`22:43.27`	`dogtanian`	`installs everything you need onto a clean drive`
`22:44.29`	`*** part/#asterisk Seldon19751 (n=someone@toronto-HSE-ppp4239807.sympatico.ca)`
`22:44.32`	`*** join/#asterisk Seldon19751 (n=someone@toronto-HSE-ppp4239807.sympatico.ca)`
`22:45.25`	`gandhijee`	`hey anyone know anything about those new Sangoma A200 cards?`
`22:45.43`	`gandhijee`	`like how the heck it supports 24 FXO/FXS's?`
`22:45.43`	`r_evolution`	`agghhh TK i think i'm going to hang myself some days`
`22:45.52`	`r_evolution`	`learning to use postgresql at the suggestion of justin`
`22:45.58`	`r_evolution`	`now i just have to make it run with .net for the CIO ;x`
`22:45.59`	`gandhijee`	`cuz from what i've been reading, the thing maxes at 16`
`22:46.05`	`r_evolution`	`but i think i'll just make it rock with PERL`
`22:47.17`	`jamig`	`hi dogtanian is it in the cvs lib?`
`22:48.53`	`*** join/#asterisk peace2u (n=peace2u@219.95.158.222)`
`22:49.01`	`peace2u`	`hi guys`
`22:49.08`	`peace2u`	`got some question here`
`22:49.28`	`peace2u`	`i'm tyring to get my te411p cards to work with IBM x225`
`22:50.05`	`*** join/#asterisk kio (n=kio@195-11.customer.cloud9.net)`
`22:50.39`	`*** join/#asterisk SGM (n=stoyan@home.marinov.us)`
`22:52.02`	`*** join/#asterisk cidez (n=sdgseg@modemcable159.198-81-70.mc.videotron.ca)`
`22:52.27`	`Dr-Linux`	`justinu: pokes mean? :S`
`22:52.45`	`Dr-Linux`	`good word or bad`
`22:53.41`	`eKo1`	`depends`
`22:53.59`	`r_evolution`	`hrm.`
`22:55.38`	`Dr-Linux`	`:S`
`22:56.36`	`lo_tech`	`peace2u whassa prob?`
`22:57.27`	`swineone`	`[TK]D-Fend, think a log of asterisk with sip debug enabled while i try to make a call would help?`
`22:57.52`	`r_evolution`	`you people... i swear *shakes head`
`22:58.20`	`r_evolution`	`"Go Back To Bed America... Your Government is in Control again!"`
`22:58.25`	`r_evolution`	`Here you go america`
`22:58.29`	`r_evolution`	`you are free... to do as we tell you`
`22:58.35`	`r_evolution`	`you are free... to do as we tell you!`
`22:58.39`	`r_evolution`	`'nuf said`
`22:58.56`	`r_evolution`	`agh where is justin... it was his idea i start into postgresql instead of mysql...`
`22:59.23`	`[av]bani`	`r_evolution: stop ripping off bill hicks`
`22:59.57`	`r_evolution`	`it's a vocal sample for a song by Adam Freeland`
`23:00.03`	`*** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net)`
`23:00.04`	`r_evolution`	`thanks for letting me know where it originally came from`
`23:00.10`	`r_evolution`	`you're my new hero.`
`23:00.14`	`r_evolution`	`;)`
`23:00.29`	`ctooley`	`Did something happen in a recent update to Windows Messenger to make it not have the ability to define separate accounts?`
`23:01.37`	`[Airwolf]`	`Evening, I'm trying to get Asterisk Realtime working, but It won't work. I was wondering if someone here would like to help me. I followed the guide on voip-info.org and everything seems fine, but I have put just 1 sip user in my db but it just wont dial.`
`23:01.43`	`[Airwolf]`	`The log is here: http://pastebin.com/526500`
`23:02.02`	`[Airwolf]`	`But I'm looking for some kind of realtime debug, but I can't find that either.`
`23:02.13`	`*** join/#asterisk MYing (n=Ming@mying.enta.net)`
`23:02.49`	`SplasPood`	`Hrm.. I've noticed that when asterisk is bound to 0.0.0.0 in sip conf it doesn't seem to reply on the same IP the connections come in on.. instead preferring the primary IP of the interface... Anyone experienced this?`
`23:02.56`	`r_evolution`	`hey Airwolf... try making sure * is set to log everything`
`23:03.01`	`r_evolution`	`then check the debug messages there`
`23:03.15`	`r_evolution`	`that's what helped me in trying to get * to connect to mysql`
`23:03.17`	`*** join/#asterisk zu (n=raz@29-pool1.ras14.floca.alerondial.net)`
`23:03.22`	`zu`	`hy all`
`23:03.29`	`r_evolution`	`up until justin advised using postgresql... O_O`
`23:03.57`	`SGM`	`hey`
`23:03.58`	`*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`23:03.58`	`*** join/#asterisk jaike (n=a@203.131.137.76)`
`23:04.05`	`[Airwolf]`	`r_evolution, can you tell me where to set that he logs everything ? :)`
`23:04.05`	`SGM`	`is asterisk realtime working with postgres?`
`23:04.21`	`r_evolution`	`im not sure yet sgm ;x`
`23:04.25`	`r_evolution`	`airwolf`
`23:04.29`	`r_evolution`	`go into logger.conf`
`23:04.44`	`r_evolution`	`under the asterisk dir`
`23:04.59`	`SGM`	`cause a friend of mine tried it`
`23:05.07`	`SGM`	`and decided it isn't working`
`23:05.19`	`[Airwolf]`	`<PROTECTED>`
`23:05.31`	`*** part/#asterisk jaike (n=a@203.131.137.76)`
`23:06.19`	`r_evolution`	`well i had it working with mysql`
`23:06.30`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`23:06.38`	`r_evolution`	`and I was talking with justin one day and he convinced me to try postgre`
`23:06.45`	`r_evolution`	`so now im learning postgres =\`
`23:06.50`	`justinu`	`~seen tainted`
`23:07.01`	`jbot`	`i haven't seen 'tainted', justinu`
`23:07.02`	`justinu`	`~seen tainted_`
`23:07.04`	`jbot`	`tainted_ is currently on #asterisk (1h 46m 42s). Has said a total of 1 messages. Is idling for 1h 46m 28s, last said: 'anyone have problems with grandstream ATAs not hanging up?'.`
`23:07.04`	`tainted_`	`yo`
`23:07.40`	`r_evolution`	`speaking of the devil.`
`23:07.46`	`r_evolution`	`~see the_devil`
`23:07.50`	`r_evolution`	`oops`
`23:07.59`	`r_evolution`	`trying to make a joke and i effed up`
`23:08.07`	`*** join/#asterisk Rowter (n=Rowter@201.145.5.26)`
`23:08.59`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`23:09.44`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-10-74.cybersurf.com)`
`23:11.12`	`jbroome`	`humor is a harsh mistress`
`23:11.22`	`[Airwolf]`	`r_evolution, alot of log :P`
`23:11.41`	`[Airwolf]`	`I will be spitting thru it for the next half hour.`
`23:11.45`	`[TK]D-Fender`	`back...`
`23:11.46`	`r_evolution`	`nah man`
`23:11.48`	`r_evolution`	`look at the bottom`
`23:12.00`	`r_evolution`	`yes... yes she is jbroome`
`23:12.19`	`Dr-Linux`	`is there anyway to reaload only voicemail.conf ?`
`23:12.54`	`[av]bani`	`[TK]D-Fender: how do you tell the polycom to dial *97 when you hit [messages] button?`
`23:13.01`	`*** join/#asterisk QuAd\|Haudrauf (n=hau@port-212-202-185-252.dynamic.qsc.de)`
`23:13.13`	`wunderkin`	`Dr-Linux, reload res_voicemail.so should do it`
`23:13.28`	`r_evolution`	`Airwolf`
`23:13.29`	`[TK]D-Fender`	`[av]bani : Are you provisioning them they way I told you to yet?`
`23:13.34`	`[av]bani`	`no`
`23:13.36`	`r_evolution`	`whitelist me`
`23:13.40`	`[TK]D-Fender`	`[av]bani : then you can't :)`
`23:13.41`	`*** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net)`
`23:13.42`	`[av]bani`	`how would you do it if you did use provisioning?`
`23:13.45`	`[TK]D-Fender`	`DO IT!`
`23:13.50`	`[av]bani`	`ok, i'm doing it`
`23:13.52`	`[av]bani`	`now tell me`
`23:14.03`	`[av]bani`	`(you dont know, do you :)`
`23:14.38`	`SPoon_TSX`	`[TK]D-Fender: Hi TK, May I ask you a simple question regarding the BLF?`
`23:14.55`	`Dr-Linux`	`wunderkin: I2C-PBX*CLI> reload res_voicemail.so`
`23:14.55`	`Dr-Linux`	`No such module 'res_voicemail.so'`
`23:15.12`	`Dr-Linux`	`I2C-PBX*CLI> reload re`
`23:15.12`	`Dr-Linux`	`res_adsi.so res_crypto.so res_features.so res_indications.so res_musiconhold.so`
`23:15.27`	`[TK]D-Fender`	`in the phonexxx.cfg go in <mwi msg.mwi.1.callBackmode="contact" msg.mwi.1.contact="*97"`
`23:15.35`	`[TK]D-Fender`	`SPoon_TSX : go right ahead`
`23:15.48`	`j`	`brb`
`23:15.51`	`*** part/#asterisk j (n=raz@29-pool1.ras14.floca.alerondial.net)`
`23:15.55`	`wunderkin`	`Dr-Linux, app_voicemail.so`
`23:15.57`	`[av]bani`	`[TK]D-Fender: LIES! you can do it through the web :)`
`23:16.29`	`[TK]D-Fender`	`[av]bani : really? news to me... mind you I avoid it like every other sane Polycom admin :D`
`23:17.00`	`SPoon_TSX`	`[TK]D-Fender: I am quite understand, to use BLF with Asterisk. I need to add exten => 2007,hint,SIP/2007 right? May i know if I want to monitoring the extension 2007 by extension 2008, shouldn't the dialplan should looks like exten => 2008,hint,SIP/2007?`
`23:17.21`	`*** join/#asterisk J_- (n=raz@29-pool1.ras14.floca.alerondial.net)`
`23:17.47`	`J_-`	`join #asterisk-dev`
`23:17.47`	`SPoon_TSX`	`[av]bani: I am with TK, you will only config your PolyCOM phone via the WEB IF you have a LOTS OF TIME to WAIT!!!!!!!!`
`23:17.47`	`J_-`	`errr`
`23:17.47`	`J_-`	`forgot the /`
`23:17.58`	`[TK]D-Fender`	`SPoon_TSX : No, your first sample is right. you tell * whre to looks for the info. then you have to tell the PHONE to look for it. thats up to the phone.`
`23:18.31`	`Dr-Linux`	`wunderkin: yeah works, i think this is a good approach to only reload the file that needs, rather the whole reload`
`23:19.27`	`SPoon_TSX`	`[TK]D-Fender: Mmm.... weird. I got the 480i and setup a softkey as the BLF and monitoring the extension but when I make a call with my 501i (Ext, 2007). It doesn't show it is on the phone.`
`23:19.56`	`SPoon_TSX`	`[TK]D-Fender: Even when I type show hints. My extension is still showing Idle....`
`23:20.30`	`SPoon_TSX`	`[TK]D-Fender: Should I use call-limit in my sip.conf?`
`23:21.24`	`[av]bani`	`SPoon_TSX: actually, configuring the polycom via the phone ui is even faster than web, which is sad :))`
`23:22.13`	`SPoon_TSX`	`[av]bani: If you have > 2 phones need to be config.`
`23:22.16`	`[TK]D-Fender`	`[av]bani : well wuddyaknow... I just found it in the web interface :)`
`23:24.08`	`SPoon_TSX`	`[TK]D-Fender: Any idea?`
`23:24.14`	`[av]bani`	`[TK]D-Fender: :))`
`23:24.22`	`[av]bani`	`SPoon_TSX: i have _exactly_ 2 phones :))`
`23:24.23`	`[TK]D-Fender`	`SPoon_TSX : pastebin your extensions.conf`
`23:24.30`	`cyburdine`	`hey gang... could someone clue me in on how to setup a voip client to be used for both in and out? voip-info.org seems to lead me to believe that shouldn't use type=friend`
`23:25.08`	`cyburdine`	`that's the way I have it set now and works for outbound calls`
`23:25.37`	`SPoon_TSX`	`[TK]D-Fender: http://pastebin.com/526539`
`23:26.06`	`swineone`	`im almost done, im just wondering why broadvoice is now replying with 404 not found`
`23:26.21`	`swineone`	`which in turn prompts a 486 busy here by asterisk`
`23:26.24`	`cyburdine`	`have not figured out why inbound doesn't answer, but that could be because I have not yet setup an extension... just wanted to check that I could use type=friend for both incoming and outgoing`
`23:26.37`	`[av]bani`	`[TK]D-Fender: cisco 7985g ... $3595 :))`
`23:29.25`	`SPoon_TSX`	`cyburdine: What do you mean?`
`23:30.39`	`cyburdine`	`Spoon: I want to use the same register line in sip.conf for both incoming and outgoing calls`
`23:31.00`	`cyburdine`	`just was curious if I needed a register=> for ingress and one for egress`
`23:31.58`	`cyburdine`	`and if I can use the same register=> can I use the same [voipprovider.com] section to define in and outbound calls by setting type=friend`
`23:32.36`	`cyburdine`	`right now I have one register=> line and type set to friend and it works for outgoing only...`
`23:33.02`	`cyburdine`	`when i try to dial in it just goes to my voip providers default message "the caller is not available"`
`23:33.24`	`[TK]D-Fender`	`SPoon_TSX : looks ok, I'd verify you BLF settings on the phone itself.`
`23:33.44`	`*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net)`
`23:34.09`	`SPoon_TSX`	`[TK]D-Fender: May I know if I want to setup the phone and the phone ask me the value for BLF, should I try 2007 or SIP/2007?`
`23:34.32`	`[TK]D-Fender`	`2007, and the phones need to be in the same context`
`23:36.25`	`[Atlas]`	`anyone have experience with asterisk realtime and postgres,, is it reliable enough for production?`
`23:36.57`	`SPoon_TSX`	`[TK]D-Fender: Same context? What does it means? You mean the context settings in sip.conf?`
`23:37.11`	`*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)`
`23:40.06`	`libila`	`I have 'exten => 1234,1,Dial(SIP/user1/1@192.168.7.130)' and 'exten => 4321,1,Dial(SIP/user2/1@192.168.7.131)' in my extenstions.conf those ip's are the real ip's of the phones. although when on user1 phone and I dial 4321 I get a 404.`
`23:40.56`	`*** part/#asterisk [Atlas] (n=whois@216.190.144.90)`
`23:44.29`	`cyburdine`	`Atlas: yeah I have it running here, rather well, but we don't have it in production`
`23:45.03`	`*** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com)`
`23:45.23`	`libila`	`also the part of my sip.conf file that refers to those users is here: http://rafb.net/paste/results/Ou8RNN24.html although I'm not sure if thats needed for just dialing extensions inside the network.`
`23:45.46`	`*** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net)`
`23:46.17`	`*** join/#asterisk Oryn (i=oryn@falcore.fsck.tv)`
`23:48.05`	`Oryn`	`if I've got a bunch of phones in a ring group and they start to ring whilst one person is on the phone, that person finishes his call and hangs up his phone, is there any way to make his phone ring with the group as soon as he hangs up?`
`23:48.55`	`*** join/#asterisk oceanlan\|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com)`
`23:55.59`	`*** join/#asterisk kippi1 (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com)`
`23:56.02`	`kippi1`	`Hey`
`23:57.03`	`kippi1`	`I am trying to setup e-mail out when you get a voicemail, my server can send mail out, but I am not getting the vmail info when I have left a message, any ideas`
`23:57.16`	`*** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net)`
`23:57.47`	`*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)`
`23:58.27`	`[TK]D-Fender`	`SPoon_TSX : The phone looking for hints must see the hints in the same context as it is registered to.`
`23:59.58`	`JonR800`	`Oryn: why not use queues/agents? i don't know of any easy way.`

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