irclog2html for #asterisk on 20050210

`00:00.04`	`Essobi`	`and doing nothing AFAIK.`
`00:00.05`	`Nethab`	`"Crap my boat is sinking, honey quick do you have a quarter?"`
`00:00.08`	`Kirk_22`	`sure it's real ;-)`
`00:00.16`	`Sedorox`	`Essobi: what OS?`
`00:00.23`	`Essobi`	`<Essobi> Mmm. My freeBSd install of asterisk is acting wack`
`00:00.23`	`cp5`	`i am implementing a change in asterisk to allow queues to eject users when all agents are offline, but the queue does have agents in it. what is the best way for app_queue.c to access data (the agents array) from chan_agent.c`
`00:00.28`	`Essobi`	`heh`
`00:00.29`	`Sedorox`	`thought so...`
`00:00.30`	`Sedorox`	`:-p`
`00:00.33`	`Sedorox`	`is it 4.1?`
`00:00.35`	`Sedorox`	`or 5.3?`
`00:00.40`	`Essobi`	`4.2`
`00:00.42`	`Luke-Jr`	`Why would my * be giving 407 when SIPMedia is trying to give me an incoming call?`
`00:00.54`	`Sedorox`	`we had a box that had 4.1 on it.. would have 99% CPU...`
`00:00.56`	`Sedorox`	`dunno why`
`00:01.07`	`Sedorox`	`we upgraded to 5.3 when we got a FXO card.. no problems`
`00:01.07`	`terrapen`	`http://www.payphone-project.com/payphones/usa/alaska/`
`00:01.08`	`Essobi`	`It's asterisk.. looks like it's beating a tight loop.`
`00:01.13`	`terrapen`	`Along Alaska Highway,`
`00:01.13`	`terrapen`	`about 30 miles east of TOK`
`00:01.13`	`terrapen`	`(very small town),`
`00:01.13`	`terrapen`	`at a parking area -`
`00:01.22`	`Essobi`	`Hmm.`
`00:01.29`	`Sedorox`	`so.. I don't know if it was just coinicadence... or the upgrade...`
`00:01.32`	`Essobi`	`I think I'll throw up a kernel trace and see what it's doing.`
`00:01.35`	`Essobi`	`what ver?`
`00:01.37`	`Essobi`	`1.03?`
`00:01.40`	`Essobi`	`from ports?`
`00:02.12`	`Sedorox`	`1.03.. and I think from ports...`
`00:04.38`	`Essobi`	`yea`
`00:04.42`	`Luke-Jr`	`Why would my * be giving 407 back to SIPMedia when I have an incoming call?`
`00:04.43`	`Essobi`	`it's slamming a loop somewhere..`
`00:05.13`	`terrapen`	`It is at the end of a 1100km microwave link to Tennant Creek, in the middle of nowhere in outback Australia. For several months in 1998 it was home to a feral pig.`
`00:05.19`	`terrapen`	`i like this phone number.`
`00:05.21`	`freddy`	`I'm still having trouble with CALLERIDNAME, it is returning the local extension rather than the caller`
`00:05.35`	`Essobi`	`It's beating a poll to death in a loop.`
`00:05.52`	`Essobi`	`Suck. I'll setup 1.05 and see what it does, I guess.`
`00:06.15`	`terrapen`	`+61889645541`
`00:06.19`	`terrapen`	`can anyone reach that #?`
`00:07.09`	`Luke-Jr`	`terrapen: what kind of #?`
`00:07.30`	`Essobi`	`Jees, maybe not. Have you seen that damn patch set for FreeBSD?`
`00:07.33`	`terrapen`	`thats the phone booth inhabited by the ferel pig`
`00:07.42`	`ctooley`	`I'm having a problem with forwarding a call via IAX and losing the number that was called (CALLEDNUMBER) on the second machine.`
`00:07.44`	`terrapen`	`at the end of a 1100km microwave link`
`00:08.10`	`Essobi`	`Umm.. so why would I call it?`
`00:08.21`	`terrapen`	`for the hell of it, why else? :)_`
`00:08.32`	`freddy`	`terrapen: who is at the other end?`
`00:08.51`	`Nethab`	`a feral pig duh`
`00:09.04`	`*** join/#asterisk SimonR (~SimonR@dialin63.eagle.ca)`
`00:09.05`	`Nethab`	`or was that my wife`
`00:09.15`	`Nethab`	`same difference i guess`
`00:09.26`	`ctooley`	`So, call comes in via provider, asterisk answers, agi started, agi dials second server, second server gets the header with VERSION : 2 CALLED NUMBER : s`
`00:09.33`	`loud`	`terrapen, fax tone.`
`00:09.35`	`*** part/#asterisk paramahansa (~chatzilla@bzq-218-252-252.red.bezeqint.net)`
`00:09.43`	`terrapen`	`http://www.payphone-project.com/payphones/australia/`
`00:09.44`	`ctooley`	`instead of CALLED NUMBER: 2146588351`
`00:09.46`	`Nethab`	`whistle at it`
`00:10.12`	`Feral_Kid`	`Are there any Cisco 7960 user out there?`
`00:10.14`	`Essobi`	`Sedorox Ahh.. the wilcallu module is flipping out.`
`00:10.17`	`Nugget`	`<-- 7960 user`
`00:10.22`	`loud`	`<-`
`00:10.36`	`freddy`	`Feral_Kid: yeah`
`00:10.40`	`Essobi`	`Feral_Kid I got about 3K of them. :)`
`00:10.50`	`loud`	`like 60% of this room got one.`
`00:10.52`	`Luke-Jr`	`Why would my * be giving 407 back to SIPMedia when I have an incoming call?`
`00:10.56`	`Nugget`	`$3,000? That's, what, two phones? :)`
`00:10.59`	`*** join/#asterisk sabre (sabre@pcp08743188pcs.lvylok01.ar.comcast.net)`
`00:11.07`	`Sedorox`	`hehe`
`00:11.17`	`freddy`	`Nugget: or 3000 phones at $`
`00:11.23`	`Feral_Kid`	`I a using version 3 firmware, and I can not get the **# to change the settings... And of course, I don't remember the password...`
`00:11.35`	`loud`	`reset it to factory defaults.`
`00:11.36`	`Nugget`	`so don't use the version 3 firmware. that's ancient.`
`00:12.02`	`terrapen`	`and upgraded to 7.x`
`00:12.23`	`sjaak538`	`Anybody knows the zapHFC interupt problem howto solve`
`00:12.28`	`Essobi`	`Nugget 3000 phones, not $3k worth of phones.`
`00:12.50`	`Nugget`	`one of my 7960s is messed up. it locks up every once in a while. I scripted to reboot it every day at 4am which seems to allow it to avoid the problem.`
`00:12.52`	`freddy`	`Essobi: so your the reason we have so much trouble getting them`
`00:12.57`	`Nugget`	`Essobi: I know. I was making a joke.`
`00:13.06`	`Essobi`	`Feral_Kid ##* is for SCCP images, not SIP afaik.`
`00:13.14`	`Essobi`	`freddy >:)`
`00:13.22`	`Essobi`	`Buying more every week.`
`00:13.34`	`Feral_Kid`	`Essobi: Ok, what is the trick for a SIP firmware?`
`00:13.49`	`freddy`	`Essobi: we've got about 100. spread accross australia`
`00:13.52`	`Essobi`	`I don't know if there is a key combo for one.`
`00:14.08`	`*** join/#asterisk buddah (~hnic@208.179.86.5)`
`00:14.12`	`Essobi`	`freddy Mexico and US. Mostly Middleto eastern seaboard.`
`00:14.17`	`Nethab`	`how much was your 7960`
`00:14.43`	`Essobi`	`I setup two 7960's behind nat today, and the damn things won't register. :\`
`00:14.47`	`Feral_Kid`	`Essobi: Ok, can I reset the settings if I use the console cable?`
`00:14.59`	`Essobi`	`O_o`
`00:15.04`	`Essobi`	`Can you?`
`00:15.06`	`Essobi`	`SHRUG`
`00:15.16`	`Essobi`	`cisco is the default`
`00:15.17`	`Essobi`	`heh`
`00:15.32`	`Feral_Kid`	`Essobi: Yeah, that is what I am banking on...`
`00:15.33`	`Essobi`	`flash it to SCCP`
`00:15.43`	`Essobi`	`and back to SIP if you can't remember`
`00:16.42`	`loud`	`http://www.cisco.com/warp/public/788/AVVID/7960_default.pdf`
`00:16.43`	`*** join/#asterisk CpuID (~nathan@dsl-202-173-176-82.qld.westnet.com.au)`
`00:16.49`	`buddah`	`anyone know why i'm getting these error messages on inbound calls to my DIDs? http://pastebin.ca/5554`
`00:16.52`	`Essobi`	`AVVID!`
`00:16.53`	`Essobi`	`Heh`
`00:18.54`	`Nethab`	`buddah: try using SIP/<peer>/<dialednumber>`
`00:19.05`	`Nethab`	`i think you did it the other way around`
`00:19.22`	`buddah`	`exten => 2519,1,Dial(SIP/5622832519/${EXTEN}\|20,t)`
`00:19.25`	`buddah`	`thats how it is now`
`00:19.48`	`Nethab`	`is 5622... a peer in sip.conf?`
`00:20.02`	`buddah`	`no`
`00:20.03`	`buddah`	`2519 is`
`00:20.05`	`buddah`	`oh`
`00:20.11`	`buddah`	`just the 2519 part`
`00:20.17`	`buddah`	`should it be all 10?`
`00:20.18`	`Essobi`	`http://www.voip-info.org/img/wiki_up//asterisk-bsd.gif`
`00:20.22`	`Essobi`	`I need to drop that to a bmp`
`00:20.24`	`Nethab`	`yeah`
`00:20.26`	`buddah`	`k`
`00:20.27`	`Essobi`	`and slap it on my phones`
`00:20.28`	`Essobi`	`:)`
`00:20.43`	`loud`	`mm`
`00:20.54`	`*** join/#asterisk l-fy (~diana@diana.null.ro)`
`00:20.59`	`l-fy`	`hello`
`00:21.12`	`loud`	`you should have something like exten => 5622832519,1,Dial(SIP/exten,30,m)`
`00:21.18`	`l-fy`	`i have a problem with asterisk`
`00:21.21`	`l-fy`	`i get something like`
`00:21.26`	`l-fy`	`*CLI> Ouch ... error while writing audio data: : Broken pipe`
`00:21.26`	`l-fy`	`Warning, flexibel rate not heavily tested!`
`00:21.26`	`l-fy`	`Segmentation fault (core dumped)`
`00:21.54`	`CpuID`	`ok, who has had success using cisco 7940/60 phones with asterisk, with the handsets behind NAT but the server on a public ip?`
`00:21.55`	`Mavvie`	`l-fy: stop using variable rate mp3s`
`00:22.08`	`*** join/#asterisk E\|nyPRI_ (E_nste_N@S0106000bdb97681e.wp.shawcable.net)`
`00:22.09`	`l-fy`	`Mavvie > i'm not using mp3`
`00:22.17`	`l-fy`	`i was making a call to voicemailmain`
`00:22.17`	`Mavvie`	`l-fy: oh, what then?`
`00:22.26`	`hermie`	`l-fy what version?`
`00:22.38`	`l-fy`	`cvs head updated half of hour`
`00:22.45`	`l-fy`	`same problem with an older version`
`00:23.16`	`buddah`	`nethab: now its doin this http://pastebin.ca/5555`
`00:23.21`	`buddah`	`and ringing but not on their end`
`00:24.11`	`freddy`	`CpuID: you need to put nat=yes in your sip.conf`
`00:24.17`	`freddy`	`for each ext`
`00:24.50`	`l-fy`	`ok, i'm going to have some sleep`
`00:24.55`	`l-fy`	`maybe it will help`
`00:25.01`	`Nethab`	`if your peer is 562... then all you need to do is Dial(SIP/562... no need for $EXTEN`
`00:25.18`	`CpuID`	`sec, let me check if its there`
`00:25.20`	`terrapen`	`http://www.payphone-directory.org/paypn.html`
`00:25.23`	`terrapen`	`there's an expensive call`
`00:25.27`	`CpuID`	`might already have it`
`00:25.27`	`l-fy`	`ok my problem is zaptel driver`
`00:25.32`	`l-fy`	`i've hit a problem there`
`00:25.35`	`terrapen`	`$3-something a minute`
`00:26.00`	`CpuID`	`o0 it wasnt there, dang :)`
`00:26.09`	`*** join/#asterisk lilneon (~tj_r3@200.108.22.232)`
`00:26.12`	`lilneon`	`hi guys`
`00:26.27`	`CpuID`	`freddy: can i have nat=no in my general, but nat=yes in certain sip contexts?`
`00:27.27`	`loud`	`(04:28:52 pm) <freddy> for each ext`
`00:27.32`	`*** part/#asterisk l-fy (~diana@diana.null.ro)`
`00:28.25`	`CpuID`	`heh np`
`00:28.43`	`Nethab`	`buddah: try using just Dial(SIP/5622832519\|20\|t)`
`00:29.59`	`ctooley`	`if I Dial(IAX2/pplayterm:secret@in-term1/2146588351@default) and on in-term1 I allow that call, it should jump to default,2146588351,1 correct?`
`00:31.09`	`*** join/#asterisk vondax (~dax@host81-139-83-249.in-addr.btopenworld.com)`
`00:31.09`	`Blake0PS`	`I'm setup on a DUNDi so that I can place calls but incoming calls give context and "no rule in context" errors, how do I fix that`
`00:31.17`	`*** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)`
`00:31.24`	`Firestrm`	`hello`
`00:31.50`	`Luke-Jr`	`Why would my * be giving 407 back to SIPMedia when I have an incoming call?`
`00:32.15`	`lilneon`	`hey guys. only have port 22 opened up at university, how can i get a softphone (SIP or IAX) to wrk and dial my itsp so i can call?`
`00:32.41`	`loud`	`stunnel ?`
`00:32.42`	`CpuID`	`ok...weird, nat=yes still didnt help`
`00:32.48`	`CpuID`	`running sip 7.3 on these 7960s`
`00:33.02`	`CpuID`	`basically, the handset pulls all the configs by tftp fine`
`00:33.13`	`CpuID`	`and it gets the logo.bmp off the webserver, and the xml services work fine`
`00:33.28`	`CpuID`	`but the handset for line 1 (only one setup) still has the x next to it, up the top right`
`00:33.30`	`lilneon`	`loud: tried that.. gave me a network error.. actually my itsp sent my putty and a preconfigured batch file`
`00:33.34`	`CpuID`	`like its not provisioned successfully`
`00:33.37`	`Nethab`	`what is a 407`
`00:33.51`	`CpuID`	`and theres no data at all in the asterisk console about any failures, verbose is like 25+`
`00:33.55`	`Essobi`	`any chance in hell I can get a 7960 to sip register against a * bow that's nated?`
`00:33.59`	`Essobi`	`actually..`
`00:34.01`	`loud`	`only 22, not even 80 ?`
`00:34.03`	`brc_`	`yes`
`00:34.07`	`Essobi`	`the phone's behind a nat and the * is too`
`00:34.08`	`buddah`	`Nethab: asterisk said ringing, and it timed out before i even got a ring on the phone`
`00:34.12`	`brc_`	`Essobi, no`
`00:34.16`	`brc_`	`in that case`
`00:34.17`	`CpuID`	`Essobi: havent tried that scenario myself`
`00:34.27`	`brc_`	`Essobi, got a decent router?`
`00:34.38`	`Essobi`	`cisco's on each end.`
`00:34.41`	`buddah`	`ooh workin now`
`00:34.47`	`lilneon`	`loud: well that was the only port found open when i scanned.. i think 80 is cuz we are allowed to use msn.. but no voice or video cam :(`
`00:34.49`	`brc_`	`Essobi, if so, forward ports 5060, 10000 to 20000 on the nat router to the asterisk server`
`00:34.53`	`Nethab`	`buddah: yay!`
`00:34.55`	`Essobi`	`I punched holes for the sip already`
`00:35.14`	`brc_`	`Essobi, then, read /usr/src/asterisk/configs/sip.conf.sample`
`00:35.14`	`Essobi`	`the 7960's aren't registering.. I can make calls from them to the *`
`00:35.19`	`brc_`	`set`
`00:35.21`	`Nethab`	`you don't really need all 10,000 rtp ports do you`
`00:35.21`	`brc_`	`hold please`
`00:35.38`	`Sedorox`	`Nethab: I don't have them forwarded and everything works fine here...`
`00:35.39`	`brc_`	`back`
`00:35.57`	`Nethab`	`you need one rtp port per call leg in or out to your * box`
`00:36.06`	`brc_`	`Essobi, set nat=yes, set externip= as per the comments in sip.conf.sample, and set the subnet`
`00:36.19`	`loud`	`lilneon, if you can't stunnel, try something like httptunnel.com to hide non http traffic.`
`00:36.19`	`Nethab`	`and my call volume is far smaller than 10,000 simultaneous`
`00:36.35`	`brc_`	`Essobi, you got cisco's on both ends?`
`00:36.41`	`lilneon`	`loud: k.. will try that one too..`
`00:36.44`	`brc_`	`Essobi, easiest thing to do would be to setup a vpn tunnel`
`00:36.53`	`brc_`	`Essobi, you gone dude?`
`00:37.02`	`AgiNamu`	`BTW, anyone who says nufone's service isnt good has never hired them`
`00:37.09`	`brc_`	`AgiNamu!`
`00:37.11`	`loud`	`you can hide anything through 80 :)`
`00:37.12`	`CpuID`	`brc_: got any suggestions here for debugging why this one handset wont connect?`
`00:37.13`	`AgiNamu`	`they just fixed my asterisk install. so im happy`
`00:37.16`	`lilneon`	`loud: which softphone should i concentrate on? an iax phone or try xten?`
`00:37.16`	`AgiNamu`	`brc_!`
`00:37.20`	`brc_`	`longtimenosee`
`00:37.24`	`AgiNamu`	`yea`
`00:37.26`	`AgiNamu`	`where've you been?`
`00:37.28`	`AgiNamu`	`;)`
`00:37.29`	`brc_`	`how's ast_mono`
`00:37.31`	`lilneon`	`loud: that's if the httptunnel.com works`
`00:37.39`	`CpuID`	`i thought id see some kinda output on * even if it didnt work`
`00:37.42`	`AgiNamu`	`brc_, working on something else for this month. but we've hired someone else`
`00:37.45`	`CpuID`	`which makes me wonder`
`00:37.45`	`AgiNamu`	`AND I got OK to release it`
`00:37.51`	`brc_`	`hired somebody to work on it?`
`00:37.56`	`buddah`	`Nethab: thanks a bunch`
`00:37.57`	`brc_`	`yay`
`00:38.02`	`AgiNamu`	`hired someone to work on related stuff, and evnetually, they'll work on this`
`00:38.06`	`brc_`	`ahh`
`00:38.07`	`AgiNamu`	`or at least, free me up`
`00:38.29`	`brc_`	`k`
`00:38.32`	`Nethab`	`np`
`00:38.51`	`AgiNamu`	`So I guess I'd like to do a dual GPL license or something`
`00:39.15`	`brc_`	`bsd...`
`00:39.19`	`brc_`	`:p`
`00:39.24`	`lilneon`	`loud:yup right there now.. firefox is the bomb.. just type httptunnel. and it took me to www.nocrew.com`
`00:39.33`	`*** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar)`
`00:39.39`	`AgiNamu`	`yea, Bsd`
`00:39.43`	`brc_`	`:)`
`00:39.43`	`AgiNamu`	`I prefer BSD over GPL`
`00:39.47`	`Nugget`	`GPL is ucky.`
`00:39.47`	`brc_`	`me too`
`00:39.56`	`AgiNamu`	`but if it's BSD, can it go into Asterisk?`
`00:39.59`	`brc_`	`yes`
`00:40.00`	`Nugget`	`yes`
`00:40.06`	`brc_`	`bsd is compatible with (almost) everything`
`00:40.33`	`brc_`	`just don't use the deprecated one with the advertising clause`
`00:40.49`	`AgiNamu`	`oh, well, cool then.`
`00:41.00`	`AgiNamu`	`this month I have to get our VSP up and running`
`00:41.01`	`brc_`	`if it's bsd you don't even need to sign a disclaimer (iaps)`
`00:41.03`	`AgiNamu`	`So I'm 100% full with that`
`00:41.17`	`AgiNamu`	`but after that, i definately want to look into it`
`00:41.21`	`AgiNamu`	`BUT... I think there are more serious problems.`
`00:41.30`	`brc_`	`eh`
`00:41.44`	`AgiNamu`	`namely, asterisk api stability. it'll be impossible to write a 'wrapper' on top of an instable base :\`
`00:41.56`	`brc_`	`ahhhhh...`
`00:41.58`	`brc_`	`yup`
`00:42.11`	`AgiNamu`	`which brings me to the serious issues`
`00:42.17`	`AgiNamu`	`about asterisk project management and so on`
`00:42.26`	`*** join/#asterisk Legend (~legend@24.244.142.133)`
`00:45.42`	`Essobi`	`they should write a soap interface`
`00:45.50`	`Essobi`	`:)`
`00:49.51`	`CpuID`	`brc_: that externip= option in sip.conf, thats only required if the * box is behind nat, with port forwards right?`
`00:49.59`	`brc_`	`yup`
`00:50.12`	`CpuID`	`whereas, in my case, where my * box has an internet ip, but just my handset is behind a nat elsewhere, no need...`
`00:50.41`	`CpuID`	`anything other than nat=yes on the handset's context that you can think of that would be causing this handset not to provision? the configs look fine`
`00:51.16`	`*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)`
`00:51.41`	`Luke-Jr`	`Why would my * be giving 407 back to SIPMedia when I have an incoming call?`
`00:52.30`	`shido6`	`CpUID what do the configs look like?`
`00:52.33`	`shido6`	`paste them on pastebin.ca`
`00:52.42`	`shido6`	`i bet you're using "friend"`
`00:52.47`	`shido6`	`dont use friends`
`00:53.10`	`Luke-Jr`	`shido6: Why not?`
`00:53.10`	`CpuID`	`sec, let me dump my sip.conf`
`00:53.19`	`shido6`	`because friends do not always work`
`00:53.24`	`shido6`	`they are supposed to be both a user and peer`
`00:53.36`	`shido6`	`but if u break the friend into two stanza's`
`00:53.37`	`shido6`	`one user`
`00:53.38`	`shido6`	`one peer`
`00:53.47`	`shido6`	`you will have better results`
`00:53.55`	`Luke-Jr`	`why?`
`00:54.10`	`shido6`	`because users dont have hosts`
`00:54.13`	`shido6`	`and peers dont have contexts`
`00:54.21`	`Luke-Jr`	`so?`
`00:54.33`	`Essobi`	`http://www.pastebin.com/239938`
`00:54.43`	`Essobi`	`Everything in that look all kosher?`
`00:54.48`	`Essobi`	`It's my sip.conf`
`00:54.57`	`shido6`	`prime example`
`00:54.57`	`shido6`	`no`
`00:55.06`	`Luke-Jr`	`PHP?`
`00:55.30`	`Essobi`	`heh`
`00:55.34`	`iMediax`	`pastebin adds that`
`00:55.34`	`Essobi`	`it slapped php tags on it`
`00:56.32`	`Luke-Jr`	`shido6: If user/peer pairs are so much better than friend, why doesn't * just convert a friend to a pair?`
`00:56.56`	`CpuID`	`shido6: get my sip.conf there?`
`00:57.18`	`*** join/#asterisk matobago (~matobago@einstein.transtelco.com.mx)`
`00:57.22`	`CoaxD`	`Just got back from the feed store. You would not BELIEVE how hard it is to find sweet feed out here.`
`00:57.25`	`matobago`	`hello everybody`
`00:57.46`	`matobago`	`anyone knows about a good sip carrier`
`00:57.48`	`matobago`	`??/`
`00:58.09`	`Luke-Jr`	`matobago: for incoming or outgoing?`
`00:58.28`	`Wi_Fi`	`halo`
`00:58.28`	`Wi_Fi`	`hehe`
`00:58.33`	`shido6`	`try http://www.pastebin.com/239940`
`00:58.34`	`shido6`	`first`
`00:58.40`	`shido6`	`Essobi`
`00:58.42`	`matobago`	`both`
`00:58.51`	`Wi_Fi`	`freakin weird here all phones connected to asterisk now dont ring`
`00:58.57`	`Wi_Fi`	`goes to voicemail`
`00:59.13`	`shido6`	`Wi_Fi, friends or users and peers?`
`00:59.17`	`*** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net)`
`00:59.18`	`shido6`	`in sip.conf or iax.conf :)`
`00:59.21`	`Wi_Fi`	`friends`
`00:59.25`	`shido6`	`d'oh!`
`00:59.26`	`lilneon`	`loud: this only forards tcp ports.. doesn't sip use UDP?`
`00:59.28`	`Wi_Fi`	`iax`
`00:59.35`	`*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)`
`00:59.45`	`Luke-Jr`	`shido6: What is the difference, technically?`
`01:00.25`	`matobago`	`Luke-Jr: I need incoming and outgoing`
`01:00.45`	`Luke-Jr`	`matobago: Voipjet has good outgoing`
`01:00.57`	`Luke-Jr`	`matobago: SIPPhone for outgoing toll-free`
`01:01.16`	`Luke-Jr`	`matobago: Can't comment on incoming; both the ones available w/ local DIDs suck`
`01:03.17`	`matobago`	`thanks very much Luke i'm gonna read about them`
`01:03.34`	`Inv_arp`	`matobago: i use livevoip for incoming DID works nice :)`
`01:03.38`	`Luke-Jr`	`I haven't had VoIP for more than a few days, but Voipjet prices are good, at least :)`
`01:03.51`	`Luke-Jr`	`Inv_arp: Haven't seen/heard of that one... URL?`
`01:04.01`	`*** join/#asterisk Kumbang (~ecvs@dsp.paume.itb.ac.id)`
`01:04.21`	`Inv_arp`	`Luke-Jr: www.livevoip.com limited DID's tho may have one in your area`
`01:06.06`	`Inv_arp`	`use voipjet for outgoing suits my tastes also :)`
`01:06.59`	`matobago`	`anymore???`
`01:07.47`	`*** join/#asterisk diablopico (~russ@ip68-8-227-12.sd.sd.cox.net)`
`01:07.52`	`diablopico`	`hello all`
`01:07.56`	`Moc`	`hi`
`01:08.06`	`diablopico`	`can i get some help setting the cdr file`
`01:08.41`	`diablopico`	`what i have is the cdr goes to a file thsat indicates tha account # plus the extension .csv`
`01:08.59`	`matobago`	`does anyone know what happend with icenet???`
`01:09.02`	`diablopico`	`what i want is for all records to tgo to Master.csv`
`01:09.03`	`bjohnson`	`anyone have bluetooth on linux experience? I'm trying to connect a headset to a usb dongle.`
`01:09.16`	`Nugget`	`linux is poo`
`01:10.40`	`twisted`	`linux is not poo`
`01:10.43`	`*** join/#asterisk criptos (~criptos@dsl-200-78-109-139.prod-infinitum.com.mx)`
`01:10.49`	`criptos`	`hell o`
`01:10.56`	`diablopico`	`can i get some help setting the cdr file`
`01:11.13`	`twisted`	`diablopico, they do by default.`
`01:11.24`	`diablopico`	`can it be changed ?`
`01:11.46`	`twisted`	`not unless you modify cdr.c`
`01:11.51`	`bjohnson`	`diablopico: they go there too`
`01:12.02`	`bjohnson`	`<diablopico> what i want is for all records to tgo to Master.csv`
`01:12.03`	`bjohnson`	`diablopico: they go there too`
`01:12.26`	`diablopico`	`i seem to be getting one or the other ,, not both`
`01:12.41`	`bjohnson`	`any Broadvox users here?`
`01:13.16`	`*** part/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it)`
`01:14.03`	`criptos`	`I�ve been loking for into.. I have a tdm40p card, 4 fxo ports attached to a analog phoen switch... 3 of the 4 ports works ok, the last one port, works only at the very first beginning, later, it no loinger answer the incomming calls...`
`01:14.09`	`*** join/#asterisk Damascene (Damascene@pcp0011401420pcs.ebrnsw01.nj.comcast.net)`
`01:15.04`	`kram`	`criptos: support@digium.com`
`01:15.06`	`*** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl)`
`01:15.12`	`*** join/#asterisk ptblank (~MURDER1@ca-yorbalnd-cuda2-c3c-202.anhmca.adelphia.net)`
`01:16.47`	`*** join/#asterisk [hC] (~turnerd@216.22.114.136)`
`01:16.55`	`diablopico`	`please , i get acct#.csv i do not have Master.csv anywhere on the drive ...`
`01:16.55`	`[hC]`	`Is there anyway to terminate a single channel without using stop now?`
`01:16.59`	`[hC]`	`I want to hang up a call`
`01:18.50`	`CoaxD`	`hC: What type interface`
`01:19.00`	`CoaxD`	`hC: There's a way to do it with zaptel; not sure about sip`
`01:19.44`	`diablopico`	`what i want is for all records to to go to Master.csv right now i have no Master.csv`
`01:19.49`	`diablopico`	`plese help`
`01:20.43`	`shido6`	`zap destroy`
`01:21.03`	`[hC]`	`Peer User/ANR Call ID Seq (Tx/Rx) Format`
`01:21.03`	`[hC]`	`192.168.193.27 1015 000f9054-32 00103/00104 ULAW`
`01:21.07`	`[hC]`	`thats what it came back with.`
`01:21.08`	`shido6`	`actuall`
`01:21.09`	`shido6`	`y`
`01:21.13`	`shido6`	`u know what you COULD do`
`01:21.48`	`shido6`	`add exten => 010101,1,Hangup to the context you have access to and use the manager interface to xxfer them to that h extension`
`01:21.54`	`shido6`	`err`
`01:21.58`	`shido6`	`hangup extension`
`01:22.12`	`CoaxD`	`shido6: They'd know something happened tho`
`01:22.35`	`diablopico`	`they will know anyway when the call ends`
`01:22.46`	`CoaxD`	`diablopico: But it could've just been a "technical problem"`
`01:23.12`	`[hC]`	`the thing is, i think its a stale call`
`01:23.19`	`[hC]`	`that is generating a udp flood`
`01:23.21`	`[hC]`	`heh.`
`01:23.32`	`CoaxD`	`well, it wouldn't be a flood. it'd be a 64k stream`
`01:24.10`	`[hC]`	`nod. im just trying to figure out some strange latency on that network`
`01:24.49`	`Luke-Jr`	`Why would my * be giving 407 Proxy Auth back to SIPMedia when I have an incoming call?`
`01:25.42`	`Kirk_22`	`noob question-do ATA devices normally use G711 because it needs less CPU power whereas an * box can take advantage of GSM because it has more power?`
`01:26.11`	`Luke-Jr`	`Is there a reason to use anything besides ulaw?`
`01:26.24`	`diablopico`	`bandwith`
`01:26.29`	`samueltc`	`hi, anyone bought stuff from iaxtalk.com?`
`01:26.36`	`Luke-Jr`	`ok`
`01:28.23`	`diablopico`	`can i get some help setting the cdr file`
`01:28.27`	`diablopico`	`i know what you said`
`01:28.37`	`diablopico`	`but i dont have Master.csv anywhere`
`01:28.46`	`diablopico`	`so it is not going to both .`
`01:28.57`	`diablopico`	`how do i get it to go to both again`
`01:29.01`	`diablopico`	`???????`
`01:29.22`	`Luke-Jr`	`diablopico: make the dir it goes in`
`01:29.56`	`diablopico`	`what should i make it ,,, i have /var/logs/asteris/cdr-csv`
`01:30.11`	`*** part/#asterisk JamesXYZ (~JamesB@CPE000625ee6776-CM000f9f7b2d50.cpe.net.cable.rogers.com)`
`01:30.16`	`Luke-Jr`	`diablopico: Can * write to it?`
`01:30.26`	`diablopico`	`how do i find where it is supposed to go .. or is that it`
`01:30.49`	`diablopico`	`yes asterisk can write to it ,, it is writing the files withthe acc# as the file name`
`01:31.00`	`AgiNamu`	`Anyone interested in USB phones that work with FireFly?`
`01:31.01`	`*** join/#asterisk syslod (~sysglod@65.114.15.70)`
`01:31.02`	`AgiNamu`	`I got them cheap`
`01:31.08`	`syslod`	`Hello ppl.`
`01:31.15`	`[hC]`	`what can i do if asterisk doesnt respond to a restart now or shutdown now`
`01:31.25`	`Luke-Jr`	`[hC]: killall -9 asterisk`
`01:31.26`	`[hC]`	`i mean it just says nothing back, and subsequent calls do not work`
`01:31.56`	`shido6`	`killall -9 safe_asterisk asterisk mpg123`
`01:32.47`	`syslod`	`Anyone know why a 4 port T1 digum card would hang in red alarm after a hard reboot?`
`01:33.00`	`[hC]`	`Thanks`
`01:33.08`	`shido6`	`yeah`
`01:33.09`	`Mavvie`	`syslod: a knight-rider light or flashing light?`
`01:33.12`	`shido6`	`seat it properly?`
`01:33.30`	`shido6`	`then reboot again :) unless u see it in cat /proc/pci/zaptel/*`
`01:33.33`	`shido6`	`err`
`01:33.38`	`shido6`	`I need some coffee`
`01:34.09`	`syslod`	`knight rider`
`01:34.16`	`Mavvie`	`syslod: then the driver isn't loaded.`
`01:35.02`	`*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)`
`01:35.08`	`ariel_`	`hello everyone`
`01:35.36`	`syslod`	`Mavvie: Why would the driver suddenly stop loading? Any way to check to see if driver loaded?`
`01:35.45`	`Mavvie`	`lsmod and /var/log/messages`
`01:36.04`	`syslod`	`I see zaptel and wt4xxx in lsmod`
`01:36.22`	`Mavvie`	`and it's still doing the knight-rider thing?`
`01:36.26`	`syslod`	`Yep`
`01:36.39`	`syslod`	`Also dmesg shows the card being detected and going to yellow alarm.`
`01:36.48`	`Mavvie`	`hmm... normally it stop doing the knight-rider thing there.`
`01:36.57`	`shido6`	`whats lsmod say`
`01:37.00`	`shido6`	`zapata.conf`
`01:37.02`	`shido6`	`and zaptel.conf`
`01:37.06`	`shido6`	`pastebin.ca`
`01:37.46`	`Mavvie`	`there need to be made some debugging sounds like "calling via SIP" and "calling via PSTN". debugging dialplans is very hard.`
`01:37.59`	`syslod`	`http://pastebin.ca/5565`
`01:38.00`	`diablopico`	`thanks for the help`
`01:38.33`	`shido6`	`zaptel.conf , zapata.conf now`
`01:39.19`	`syslod`	`http://pastebin.ca/5565`
`01:39.32`	`Luke-Jr`	`Why would my * be giving 407 Proxy Auth back to SIPMedia when I have an incoming call?`
`01:39.43`	`syslod`	`It worked an hour ago. Then keyboard was unpluged and I had to reset.`
`01:39.51`	`*** join/#asterisk Total-Net (~pbudden@user-11fa2mu.dsl.mindspring.com)`
`01:40.06`	`Total-Net`	`Hi All`
`01:41.35`	`*** join/#asterisk ionix (ionix@MTL-HSE-ppp200024.qc.sympatico.ca)`
`01:41.48`	`Total-Net`	`does anyone know anything about a Tundo Teleport-16 ?`
`01:44.01`	`Total-Net`	`I can't find any info on the net.. it says it is something to do with VoIP`
`01:45.12`	`syslod`	`Anyone have any ideas on the T1 card?`
`01:46.29`	`syslod`	`Second reseat worked.`
`01:46.36`	`*** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net)`
`01:46.41`	`MrEntropy`	`yo`
`01:46.42`	`syslod`	`Wierd? Guess its pretty fragile`
`01:48.58`	`*** join/#asterisk jarrod (jarrod@dipole.informationwave.net)`
`01:49.08`	`jarrod`	`hey how do i disable the callmanager features on a cisco 7960`
`01:49.11`	`liquidno2`	`is the rxgain and txgain for agjusting the sidetone for a specific channel?`
`01:49.25`	`jarrod`	`it gets hung on configuring the cm list with callmanager name dns error`
`01:49.37`	`tzanger`	`stevekstevek: around?`
`01:52.17`	`*** join/#asterisk fwc (~fwc@206.127.78.199)`
`01:52.35`	`srt`	`jarrod: install the sip image? ;)`
`01:52.45`	`jarrod`	`i am i downloaded it and its pulling it from tftp`
`01:53.04`	`jarrod`	`P0S3-07-3-00.zip`
`01:53.09`	`srt`	`yep`
`01:53.17`	`fwc`	`Is there a way to see which channels are in which groups? I.E. to help me debug why I'm only able to use 1 channel on a PRI.`
`01:53.21`	`srt`	`dit it install correctly?`
`01:53.33`	`jarrod`	`i put P0S3-07-3-00 in the OS79xx.TXT file and it is pulling it from my tftp logs`
`01:53.44`	`srt`	`thats wrong`
`01:54.01`	`srt`	`must have P003-07-3-00 in OS79XX.TXT`
`01:54.14`	`jarrod`	`oh`
`01:54.15`	`jarrod`	`yes`
`01:54.17`	`jarrod`	`that is whats in there`
`01:54.19`	`jarrod`	`im sorry`
`01:54.28`	`jarrod`	`P003-07-3-00`
`01:54.32`	`srt`	`and image_version: "P0S3-07-3-00" in SIP<MAC>.cnf`
`01:55.09`	`jarrod`	`now I do not see that`
`01:56.12`	`srt`	`hm?`
`01:56.20`	`jarrod`	`i did not have that line`
`01:56.21`	`jarrod`	`in there`
`01:56.29`	`srt`	`so put it there`
`01:56.32`	`jarrod`	`i did`
`01:56.57`	`jarrod`	`i dont show it pulling the P003... images from tftp`
`01:57.36`	`shido6`	`ramen time`
`01:57.38`	`shido6`	`bbl`
`01:57.56`	`srt`	`what does it get from then?`
`01:58.07`	`jarrod`	`why is it trying to get SEPblahblah`
`01:58.10`	`jarrod`	`and not SIP`
`01:58.49`	`*** part/#asterisk criptos (~criptos@dsl-200-78-109-139.prod-infinitum.com.mx)`
`01:59.07`	`srt`	`the bootloader tries to get everything`
`01:59.20`	`srt`	`CTLSEP, SEP, SIP, ...`
`01:59.22`	`jarrod`	`tftpd[5443]: tftpd: trying to get file: SEP0008218C078B.cnf`
`01:59.26`	`jarrod`	`but it never tries SIP`
`01:59.41`	`srt`	`do u have a SEP0008218C078B.cnf there?`
`01:59.45`	`jarrod`	`i just created it`
`02:00.09`	`srt`	`no it shouldn't be there`
`02:00.41`	`jarrod`	`oh and image_version: P0S3-07-3-00 was in SIPDefault`
`02:00.47`	`jarrod`	`it was not in my SIP<mac>`
`02:01.03`	`srt`	`that often works but not always ;)`
`02:01.14`	`jarrod`	`ok i removed SEP`
`02:01.30`	`srt`	`k`
`02:01.51`	`jarrod`	`its still trying to just pull SEP`
`02:01.53`	`forrestc`	`jarrod: It shouldn't even try SEP.`
`02:02.05`	`jarrod`	`well wtf`
`02:02.09`	`forrestc`	`jarrod: It should ONLY pull the OS79XX.TXT file and then the P0S bfile.`
`02:02.16`	`Mavvie`	`a SEP is exactly what it is: not mine!`
`02:02.17`	`srt`	`did it pull the OS79XX.TXT?`
`02:02.25`	`jarrod`	`why tftpd: trying to get file: OS79XX.TXT`
`02:02.26`	`jarrod`	`yes`
`02:02.41`	`srt`	`forrestc: do u have 7-03?`
`02:02.43`	`forrestc`	`Are you SURE the OS79XX.TXT file is correct?`
`02:03.07`	`forrestc`	`I think it's 7.03. Just did this the other day... So whatever is on the cisco site.`
`02:03.25`	`jarrod`	`cat OS79XX.TXT`
`02:03.25`	`jarrod`	`P003-07-3-00`
`02:03.30`	`forrestc`	`There is a OS79XX.TXT file which comes with the firmware (I think I remember correctly).`
`02:03.38`	`forrestc`	`Hold on, let me look at my tftp server.`
`02:03.41`	`srt`	`no leading spaces or stuff like that? ;)`
`02:04.01`	`jarrod`	`no completely empty`
`02:04.15`	`srt`	`forrestc: but the 7-03 bootloader tries loads SEP first...`
`02:05.08`	`shido6`	`then u need to put what version ur using in there`
`02:05.09`	`shido6`	`ok`
`02:05.10`	`forrestc`	`srt: I've only paid attention when I've been upgrading a phone, so I probably didn't notice that.`
`02:05.13`	`shido6`	`be back ina bit`
`02:05.36`	`*** part/#asterisk Kirk_22 (~Administr@pool-151-199-159-52.norf.east.verizon.net)`
`02:06.38`	`forrestc`	`jarrod: does it actually RETRIEVE it or just tries?`
`02:06.48`	`*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com)`
`02:07.08`	`forrestc`	`jarrod: is the tftp server on a public IP?`
`02:07.13`	`jarrod`	`yes`
`02:07.19`	`jarrod`	`tftpd: serving file from /tftp`
`02:08.00`	`forrestc`	`jarrod: Is the tftp server open so I could fetch a copy of the OS79XX.TXT file?`
`02:08.35`	`forrestc`	`I.E. on a public IP address...`
`02:09.23`	`jarrod`	`yes`
`02:10.39`	`forrestc`	`Bizzare... DIFF says the files differ.`
`02:10.58`	`srt`	`dos vs unix format?`
`02:11.18`	`jarrod`	`mine is unix`
`02:11.24`	`jarrod`	`ah but you knew that`
`02:11.24`	`jarrod`	`hmm`
`02:11.27`	`jarrod`	`weird`
`02:11.35`	`forrestc`	`possibly. I'm on linux here.. Jarrod: do a chmod 777 OS* so I can put mine up there.`
`02:11.36`	`srt`	`mine, too ;)`
`02:11.50`	`jarrod`	`done`
`02:11.58`	`srt`	`btw. what firmware are you currently running on the phone?`
`02:12.00`	`*** join/#asterisk jollyr (jollyr@210.16.20.2)`
`02:12.26`	`forrestc`	`done.`
`02:13.02`	`forrestc`	`try it now.`
`02:13.02`	`jarrod`	`phone still not working properly`
`02:13.09`	`jarrod`	`Feb 9 21:15:31 asterisk tftpd[5610]: tftpd: trying to get file: OS79XX.TXT`
`02:13.09`	`jarrod`	`Feb 9 21:15:31 asterisk tftpd[5610]: tftpd: serving file from /tftp`
`02:13.09`	`jarrod`	`Feb 9 21:15:31 asterisk in.tftpd[5611]: connect from 192.168.1.191`
`02:13.09`	`jarrod`	`Feb 9 21:15:31 asterisk tftpd[5612]: tftpd: trying to get file: SEP0008218C078B.cnf`
`02:13.09`	`jarrod`	`Feb 9 21:15:31 asterisk tftpd[5612]: tftpd: serving file from /tftp`
`02:13.09`	`jarrod`	`Feb 9 21:15:31 asterisk in.tftpd[5613]: connect from 192.168.1.191`
`02:13.11`	`jarrod`	`Feb 9 21:15:31 asterisk tftpd[5614]: tftpd: trying to get file: SEPDefault.cnf`
`02:13.13`	`jarrod`	`Feb 9 21:15:31 asterisk tftpd[5614]: tftpd: serving file from /tftp`
`02:13.55`	`forrestc`	`jarrod: what do my logs look like..`
`02:14.03`	`jarrod`	`same`
`02:14.03`	`forrestc`	`The get that is..`
`02:14.20`	`forrestc`	`How many bytes is the OS79XX.TXT now?`
`02:14.20`	`*** join/#asterisk Ahewes (~rsb@209.81.2.58)`
`02:14.30`	`jarrod`	`15`
`02:14.54`	`forrestc`	`jarrod: what version IS on the phone?`
`02:15.00`	`jarrod`	`cisco 7960`
`02:15.11`	`srt`	`firmware version......`
`02:15.35`	`srt`	`is it the original sccp one that the phone shipped with?`
`02:15.40`	`jarrod`	`yes`
`02:15.44`	`*** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com)`
`02:15.46`	`srt`	`ok.......`
`02:15.56`	`jarrod`	`P003AM30`
`02:15.56`	`srt`	`http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm helps...`
`02:16.25`	`forrestc`	`so does http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip3`
`02:16.34`	`jarrod`	`im on that page`
`02:17.15`	`srt`	`try to put a load_information: "P0S3-07-3-00" into SEP0008218C078B.cnf`
`02:18.42`	`liquidno2`	`why am I getting a Dial() to my FXO port after I hang up a previously completed call?`
`02:18.58`	`liquidno2`	`Do I need a h extension in the current context for clean up?`
`02:19.12`	`forrestc`	`liquidno2: Do you mean it rings again after you hang up?`
`02:19.31`	`liquidno2`	`it doesn't ring I don't think... Zap4 is FXO port anyways... so I wouldn't hear it anyways`
`02:19.49`	`liquidno2`	`I just get some werid error after completing a call`
`02:20.00`	`liquidno2`	`-- Executing Dial("Zap/1-1", "Zap/4/h\|70\|Tt") in new stack`
`02:20.14`	`liquidno2`	`I get that after I get a hangup on Zap/4-1`
`02:20.55`	`forrestc`	`liquidno2: probably something dialplan related. I had something similar once and had missed adding a Congestion after a dial.`
`02:20.55`	`liquidno2`	`the methods are s, t, h, and fax right?`
`02:21.07`	`liquidno2`	`That might be it`
`02:21.25`	`file[laptop]`	`your using a bad bad dialplan pattern matching entry`
`02:21.31`	`file[laptop]`	`like _.`
`02:21.36`	`liquidno2`	`I am using _.`
`02:21.38`	`liquidno2`	`:)`
`02:21.39`	`file[laptop]`	`so it's matching h, and dialing it out the zap channel`
`02:21.46`	`file[laptop]`	`yeah well learn the proper way :p`
`02:21.55`	`forrestc`	`liquidno2: Use _X. instead then.`
`02:22.11`	`liquidno2`	`but will that catch international as well?`
`02:22.26`	`file[laptop]`	`I put separate entries myself for that stuff...`
`02:22.38`	`*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4117547.sympatico.ca)`
`02:22.41`	`file[laptop]`	`an entry for 11 digit dialing, 10 digit, and international`
`02:22.42`	`forrestc`	`liquidno2: it will catch anything which has at least one digit. But I'd put a separate entry in there also.`
`02:22.53`	`file[laptop]`	`plus 7 personally...`
`02:22.54`	`forrestc`	`or instead.`
`02:23.02`	`file[laptop]`	`but that's just me`
`02:23.18`	`NormAst`	`Hi all.`
`02:23.45`	`ariel_`	`file[laptop], is it now a little early to sleep.`
`02:23.58`	`liquidno2`	`When I look at it... it makes perfect sense now...`
`02:24.17`	`liquidno2`	`the _. was matching anything including the hangup state`
`02:24.28`	`forrestc`	`jarrod: Do you know how you're running tftpd?`
`02:24.29`	`liquidno2`	`damn... it is always the obvious shit`
`02:24.40`	`forrestc`	`jarrod: Is it from xinetd/inetd or as a daemon?`
`02:25.15`	`forrestc`	`jarrod: Unless you fixed it, I'd love to see a few -v -v -v's added to make sure the phone is actually fetching the full file.`
`02:25.55`	`tzanger`	`hmm`
`02:26.25`	`tzanger`	`asterisk -rc does not like linux, screen or vt100 for TERM vars to give me colour, yet asterisk -gc gives me full colour`
`02:26.30`	`jarrod`	`its inetd`
`02:26.59`	`NormAst`	`Anyone know of a good 1U server`
`02:27.56`	`ariel_`	`NormAst, I like supermicro's.`
`02:28.17`	`*** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net)`
`02:28.36`	`jarrod`	`doesnt werk`
`02:28.40`	`tzanger`	`NormAst: you've got a ton of room, what do you want 1U for? :-)`
`02:28.47`	`jarrod`	`it continues to try to pull SEPxxxx`
`02:28.50`	`jarrod`	`and not SIPxxxx`
`02:28.53`	`NormAst`	`I have ton's now...`
`02:29.15`	`NormAst`	`but laster I might now have so much room.`
`02:29.20`	`tzanger`	`NormAst true enough, I imagine you'd just terminate DS3s to MaxTNTs (two per half rack) once you got to that level of density`
`02:30.05`	`Mavvie`	`is it just me, or does EnumLookup() not take the priority field of the NAPTR record into account?`
`02:31.00`	`Mavvie`	`is it just me, or does it only work for one (1) NAPTR record?`
`02:31.12`	`zimdog`	`Has anyone done a statically linked compile of *`
`02:31.31`	`tzanger`	`zimdog: what's the advantage of it`
`02:31.41`	`NormAst`	`Tzanger: True... But think of what I will be selling off when I go DS3`
`02:32.05`	`Mavvie`	`tzanger: as usual: no need for dynamic libraries.`
`02:32.22`	`tzanger`	`Mavvie: but asterisk is built on the premise of dynamically loadable modules`
`02:33.13`	`NormAst`	`Tzanger: You said at one point you are getting some Echo Can. Equipment... Any idea's yet?`
`02:33.14`	`zimdog`	`tzanger: Trying to setup a demo on my dedicated webserver because of all the nat issues I am having. I do not have all the required software on the sever to compile it locally. And do not want to lose support from the hosting company`
`02:33.29`	`Mavvie`	`tzanger: but they're added by the server, not by the rtld.`
`02:33.30`	`tzanger`	`NormAst: just tellabs is the name I know, I've seen some others`
`02:33.50`	`NormAst`	`Tzanger: You get alot of echo?`
`02:34.00`	`tzanger`	`Mavvie: yeah but then you have to try and do the same with all the modules...`
`02:34.03`	`tzanger`	`NormAst: at times, yes`
`02:34.05`	`tzanger`	`not happy with it`
`02:34.12`	`tzanger`	`zap's echo can just doesn't cut it`
`02:34.40`	`Mavvie`	`tzanger: he only asked for a statically linked version, not for a version which didn't need other modules.`
`02:34.57`	`NormAst`	`Tzanger: I get a little bit of echo bug I turned the Gain down a bit.`
`02:35.17`	`tzanger`	`Mavvie: hmm`
`02:35.20`	`forrestc`	`jarrod: can you add some -v -v -v 's in inetd.conf`
`02:35.30`	`tzanger`	`NormAst: my pri is already a little on the quiet side`
`02:35.36`	`jarrod`	`no`
`02:35.39`	`jarrod`	`not with this in.tftpd`
`02:35.46`	`NormAst`	`rtzanger: I am running -1 on both.`
`02:36.04`	`forrestc`	`jarrod: ohh.`
`02:36.11`	`forrestc`	`jarrod: stupid me.`
`02:36.26`	`forrestc`	`jarrod: is there a "increased verbosity logging" option?`
`02:36.32`	`jarrod`	`no`
`02:36.35`	`jarrod`	`i did a man :(`
`02:36.42`	`tzanger`	`bah`
`02:36.43`	`forrestc`	`jarrod: What are you running?`
`02:36.43`	`jarrod`	`i dont wanna load another tftpd`
`02:36.43`	`jarrod`	`hah`
`02:36.47`	`zimdog`	`Thi server wil not have any digium hardware. I want it to be able to connect sip phones and then use aix to tunnel to the internal server. Or I am open to other ways arount this?`
`02:36.49`	`tzanger`	`gotta run home and get my damn TE405P`
`02:37.05`	`*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)`
`02:37.26`	`forrestc`	`jarrod: got tcpdump?`
`02:37.28`	`NormAst`	`tzanger: I think I will be selling my TE405P off.`
`02:37.53`	`tzanger`	`NormAst: yeah? in favour of the sangoma card?`
`02:38.16`	`jarrod`	`http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_field_notice09186a008009451b.shtml`
`02:38.19`	`jarrod`	`that is my problem`
`02:38.43`	`NormAst`	`tzanger: yea... I am picking up there Quad card tomorrow..`
`02:38.49`	`tzanger`	`nods`
`02:38.54`	`sivana`	`NormAst: turning down the gaim help?`
`02:38.54`	`tzanger`	`be back in 10`
`02:39.02`	`sivana`	`s/gaim/gain`
`02:39.35`	`shido6`	`.`
`02:39.36`	`NormAst`	`sivana: yea.. It will help if you are getting alot of echo.. Are you getting alot of echo?`
`02:39.51`	`sivana`	`NormAst: yea, I do out my PRI`
`02:40.23`	`sivana`	`NormAst: what do you have it set at?`
`02:40.56`	`NormAst`	`sivana: Where are you getting the echo? Caller Calls into the PRI to SIP phone gets the echo?`
`02:41.04`	`forrestc`	`jarrod: Still, the phone SHOULD read the file, realize it isn't running the right code, and then fetch the new file.`
`02:41.21`	`sivana`	`NormAst: basically, any outbound device on the PRI`
`02:41.24`	`forrestc`	`jarrod: at least that's what every phone I've dealt with will do.`
`02:41.36`	`NormAst`	`sivana: So the sip phone users is hearing the echo?`
`02:41.41`	`sivana`	`NormAst: yes`
`02:42.03`	`NormAst`	`It's your SIP Device and not asterisk. Try turning down the TXgain on the sip device.`
`02:42.08`	`forrestc`	`jarrod: the bug looks like it just doesn't finish retrieving the load because it is too big.`
`02:42.14`	`jarrod`	`yea`
`02:42.20`	`jarrod`	`thats it`
`02:42.25`	`jarrod`	`im upgrading to another version`
`02:42.26`	`sivana`	`NormAst: its anything going out the PRI`
`02:42.28`	`jarrod`	`its working`
`02:42.40`	`sivana`	`NormAst: I need to put an echo can between * and the pRI`
`02:43.01`	`sivana`	`NormAst: doesn't happen on VoIP LD calls`
`02:43.13`	`NormAst`	`You should be running echo can on *.`
`02:43.23`	`sivana`	`NormAst: I am.. it sucks :)`
`02:43.40`	`NormAst`	`sivana: It's your sip box that you are getting the echo from...`
`02:43.41`	`forrestc`	`jarrod: what did you do?`
`02:44.03`	`NormAst`	`sivana: try another brand of box with better echo canceller in it.`
`02:44.12`	`sivana`	`NormAst: I'm using Sipura`
`02:44.43`	`NormAst`	`sivana: Asterisk only cancells the echo from the PSTN to the SIP User.`
`02:44.46`	`sivana`	`NormAst: I just need to put it inline the network and be done with it :)`
`02:44.58`	`Mavvie`	`hmmm... enumlookup does explicitely want tel:+, while the RFC says that it can be without a + for local extensions.`
`02:45.17`	`NormAst`	`sivana: Let me know what equipment you are going to buy and how well it works.`
`02:45.48`	`sivana`	`NormAst: yeah`
`02:46.38`	`NormAst`	`sivana: Try calling a cell phone user and see if you get echo.`
`02:46.51`	`NormAst`	`sivana: Cell company has alot of echo can on the line.`
`02:48.18`	`sivana`	`NormAst: I sent an email to www.valiantcom.com on their echo can`
`02:49.00`	`forrestc`	`sivana: how long is the echo you're getting?`
`02:49.06`	`Damascene`	`have any of you guys tried supporting asterisk for a small business environment? or does anyone know if there is official asterisk support? ;)`
`02:49.32`	`forrestc`	`sivana: Looks like the valiant cancellers only do 128ms.`
`02:50.23`	`tzanger`	`remember though that even cell echo cans are typically only 64ms`
`02:50.31`	`tzanger`	`more taps = more time to converge`
`02:50.40`	`tzanger`	`which = more time that you hear echo`
`02:50.44`	`NormAst`	`Bigger load on the system.`
`02:51.02`	`NormAst`	`Don't use 256 taps... Not a good thing.`
`02:51.12`	`sivana`	`forrestc: I have no idea how long the echo is`
`02:51.16`	`*** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net)`
`02:51.46`	`forrestc`	`sivana: does it come immediate back, or is it wait 1/2 second before it comes back.`
`02:51.56`	`sivana`	`forrestc: it's immediate`
`02:52.19`	`forrestc`	`forrestc: and it's outbound on a PRI on all calls?`
`02:52.34`	`sivana`	`forrestc: ya, most calls... always outbound`
`02:52.45`	`sivana`	`forrestc: it's a bit inconsistent though`
`02:53.55`	`forrestc`	`more often than not though?`
`02:53.58`	`sivana`	`yup`
`02:54.01`	`forrestc`	`Is this a local PRI?`
`02:54.05`	`sivana`	`ya`
`02:54.16`	`forrestc`	`Who is your telco?`
`02:54.31`	`sivana`	`it's a local CLEC`
`02:55.46`	`Mavvie`	`enum randomness....`
`02:55.48`	`Mavvie`	`what a mess.`
`02:57.30`	`tzanger`	`forrestc: I have the exact same problem with Bell Canada`
`02:57.59`	`forrestc`	`Weird. I'm not sure I've ever seen echo towards a PRI.`
`02:58.13`	`forrestc`	`(or bouncing back out of a pri is a better way to put it).`
`02:58.18`	`NormAst`	`Tzanger: My sip users never get the echo.. It's always the PSTN person.`
`02:58.19`	`tzanger`	`"towards" -- no... I've never heard of a caller saying THEY heard echo`
`02:58.22`	`forrestc`	`unless there was something wrong on the far end.`
`02:58.25`	`tzanger`	`just my own voice bouncing off their hybrid`
`02:58.37`	`tzanger`	`NormAst: interesting`
`02:58.40`	`tzanger`	`complete opposite here`
`02:58.50`	`*** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net)`
`02:58.54`	`NormAst`	`I had that same thing..`
`02:59.07`	`NormAst`	`And when I switch to Debian it reversed... Weird.`
`02:59.17`	`forrestc`	`I though the telco was supposed to cancel the echo.`
`02:59.22`	`NormAst`	`Tzanger: Do you get alot of echo calling TO?`
`02:59.41`	`forrestc`	`On 4 wire interfaces that is.`
`02:59.55`	`forrestc`	`Now on a FXO/FXS interface - that's a whole different thing.`
`02:59.57`	`tzanger`	`NormAst: haven't been told of anything`
`03:00.03`	`tzanger`	`NormAst: usually it's when they call us (i.e. our PRI)`
`03:00.09`	`tzanger`	`or we call out through our pri (local # or 800#)`
`03:00.28`	`*** join/#asterisk santiago (~santiago@63.245.86.104)`
`03:00.30`	`zimdog`	`I need suggestions on how to get by the folowing problem. I have * running behind a freesco firewall. It has a TDM21B. I want to allow my partners and also the ability to demo the functionality of * from the internet. I have tried everything to get sip working with nat and the closest I have come was getting audio from a phone on th inside to an xlite client on the outside. So now I am thinking * server on the outside and then IAX in`
`03:00.30`	`zimdog`	`to the internal server. The problem I do not have the correct devel libraries like ssl and GLIB2. I do not want to install these. WHat is the best way to proceed? Statically linked , SIP proxy or stun server?(can these talk IAX to )?.`
`03:00.35`	`NormAst`	`Tzanger: Send me the 800 stuff and see if it helps.`
`03:00.47`	`tzanger`	`NormAst: interesting idea`
`03:00.55`	`*** part/#asterisk santiago (~santiago@63.245.86.104)`
`03:01.06`	`forrestc`	`If you get echo calling INTO a sip phone. I.E it's bouncing off of the sip phone, it's probably some accoustic coupling in the phone.`
`03:01.40`	`forrestc`	`I.E. someone didn't stick cotton into the handset.`
`03:01.45`	`forrestc`	`like they were supposed to.`
`03:02.15`	`sivana`	`it's about 2200 USD for the valiantcom's echo can`
`03:02.25`	`*** part/#asterisk lilneon (~tj_r3@200.108.22.232)`
`03:03.08`	`*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)`
`03:03.20`	`forrestc`	`sivana: you've played with the tuning settings and turning on AGRESSIVE_ECHO (or whatever it is exactly called)?`
`03:04.06`	`NormAst`	`sivana: I have some pricing from another company.. What's your -email.`
`03:04.15`	`NormAst`	`or I can paste bin it.`
`03:04.59`	`sivana`	`http://pastebin.ca/5575`
`03:05.08`	`sivana`	`forrestc: that's what I have in zapata.conf`
`03:07.19`	`forrestc`	`sivana: you should also look at http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml`
`03:09.27`	`sivana`	`little bit of reading there :)`
`03:10.12`	`forrestc`	`sivana: I'm assuming you mean you have echocancel=y or echocancel=128 in the zapata.conf file, in the section related to the PRI.`
`03:10.25`	`sivana`	`I do not... I'll add that`
`03:10.49`	`sivana`	`echocancellation=yes`
`03:10.51`	`sivana`	`I have that`
`03:11.03`	`forrestc`	`sivana: let me check the syntax here.`
`03:11.46`	`forrestc`	`that would be the problem.`
`03:11.49`	`forrestc`	`echocancel=yes`
`03:12.19`	`sivana`	`ok.. I updated it`
`03:13.22`	`forrestc`	`the AGRESSIVE_ECHO is a compile time option, which shouldn't be used unless you REALLY need it, and then sometimes not even then (because in a lot of cases it doesn't help but hurts).`
`03:13.45`	`forrestc`	`sivana: you trying it now?`
`03:14.06`	`sivana`	`I updated the syntax to "echocancel", I'll have to try it tomorrow at the office`
`03:14.13`	`forrestc`	`sivana: ok.`
`03:14.42`	`forrestc`	`sivana: there is also an echocancelwhenbridged option, but I don't think that applies here.`
`03:15.02`	`forrestc`	`I.E. this probably only applies when you are bridging two pri's together.`
`03:15.17`	`sivana`	`ya, or bridging a SIP device with the PRI?`
`03:15.32`	`forrestc`	`sivana: not sure. gotta love asterisk documentation.`
`03:15.37`	`NormAst`	`sivana: If you got from the First Channel in then backout channel 2 for example.`
`03:15.47`	`NormAst`	`Turn it off.`
`03:16.01`	`forrestc`	`Normast: thanks for the clarification.`
`03:16.33`	`sivana`	`I see`
`03:16.44`	`NormAst`	`If you turn it on it will burn alot of CPU power when you are bridge..`
`03:16.53`	`NormAst`	`I bridge TDM to TDM all the time and you don't need it.`
`03:17.18`	`forrestc`	`NormAst: How well does the TDM/TDM work? Does it eat cpu?`
`03:17.33`	`forrestc`	`I'm thinking about stealing some lines from the dialin pool`
`03:18.06`	`NormAst`	`forrestc: works GREAT!!... 30 channels and uptime is 0.00 0.00 0.00`
`03:18.17`	`godsmoke`	`you mean load? :)`
`03:18.22`	`NormAst`	`yea.`
`03:18.23`	`godsmoke`	`hopefully your uptime is not 0.00`
`03:18.54`	`NormAst`	`asterisk has been up for System uptime: 3 weeks, 22 hours, 51 minutes, 8 seconds`
`03:18.57`	`modulus_`	`normast, what's a vmstat show?`
`03:19.02`	`forrestc`	`normast: I've got quite a few PRI's scattered around which peak out at maybe 1 or 2 channels in use.`
`03:19.31`	`NormAst`	`procs -----------memory---------- ---swap-- -----io---- --system-- ----cpu----`
`03:19.31`	`NormAst`	`<PROTECTED>`
`03:19.31`	`NormAst`	`<PROTECTED>`
`03:19.32`	`shido6`	`heh`
`03:19.53`	`forrestc`	`Seems logical to route them through an asterisk box and then route the modem calls back to the terminal server.`
`03:21.36`	`sivana`	`forre: I'm using Asterisk to route incoming dialup to a Portmaster 3 NAS`
`03:21.57`	`forrestc`	`sivana: working pretty well?`
`03:22.18`	`sivana`	`ya, beautifully`
`03:22.30`	`*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)`
`03:22.32`	`forrestc`	`We've got a PM3 at one site, mostly USR HiPer stuff elsewhere.`
`03:22.58`	`forrestc`	`The PM3 site is Non-PRI DSS. Gonna have to see if the old switch in that CO will send 7 DID digits.`
`03:23.40`	`Essobi`	`Hmm.`
`03:23.51`	`NormAst`	`still getting PRI got event: HDLC Abort (6) on Primary D-channel of span 3.... 4 every second.`
`03:24.05`	`*** join/#asterisk Frawg (~kamy@frawg.user)`
`03:24.08`	`Frawg`	`hey all`
`03:24.16`	`forrestc`	`Maybe Qwest will actually get around to upgrading the Erricson to something recent soon... Yeah right!`
`03:24.27`	`Frawg`	`got a problem`
`03:24.31`	`sivana`	`heh`
`03:24.36`	`Frawg`	`i have 2 cisco phones connecting to my pbx`
`03:24.43`	`Frawg`	`one is on the end of a dsl line`
`03:24.50`	`Frawg`	`which has around 40ms lag from the pbx`
`03:24.53`	`Essobi`	`I got some 7960s.. Anyone drop their config somewhere that is using nat for me?`
`03:24.54`	`Frawg`	`the other has ethernet to the pbx`
`03:25.08`	`Frawg`	`slowly the 2 phones get lagged`
`03:25.14`	`Essobi`	`I'm sniffing them and I never see it sent it's SIP registration to my * box when it reboots.`
`03:25.14`	`Frawg`	`and it becomes unuseable`
`03:25.49`	`Essobi`	`Frawh increase the timeout in the sip config for them`
`03:25.50`	`Frawg`	`Essobi: set register with proxy = YES`
`03:25.59`	`Essobi`	`Ahh.`
`03:26.02`	`Frawg`	`Essobi: hrmmk`
`03:26.36`	`Frawg`	`Essobi: what is the actual command in sip.conf?`
`03:26.36`	`Blake0PS`	`I setup FWD to work on my * box, what does this mean: "socket_read Call rejected by [ip]: no such context/extension"`
`03:26.36`	`forrestc`	`Now If I can only figure out why the asterisk box I'm dealing with right now only will originate one call out the PRI at a time.`
`03:27.03`	`Essobi`	`Frawg lemme look`
`03:27.59`	`Essobi`	`;qualify=200 ; Qualify peer is no more than 200ms away`
`03:28.07`	`forrestc`	`BlakeOPS: Are you trying to connect directly or are you going through iaxtel.`
`03:28.12`	`Essobi`	`I think ther'es a default qualify`
`03:28.28`	`Essobi`	`if you don't specify.. phones are probably exceeding and unregistering.`
`03:28.40`	`Blake0PS`	`forrestc: I'm setting it up according to http://www.freeworlddialup.com/content/view/full/1501`
`03:29.17`	`Frawg`	`Essobi: no i'm on a call`
`03:29.18`	`iMediax`	`Blake thats for incoming calls right?`
`03:29.19`	`Frawg`	`as in i'm talking`
`03:29.22`	`Essobi`	`Ahhh.`
`03:29.26`	`Essobi`	`Hmm.`
`03:29.26`	`Frawg`	`and the conversation slowly get's more and more delayed`
`03:29.30`	`Frawg`	`after around 5mins`
`03:29.31`	`forrestc`	`BlakeOPS: most people just stuff the calls through iaxtel.`
`03:29.36`	`Essobi`	`Wierd..`
`03:29.42`	`Essobi`	`have you sniffed it?`
`03:29.42`	`Frawg`	`it's a 5second delay`
`03:30.38`	`tzanger`	`whee`
`03:30.43`	`Blake0PS`	`forrestc: IAXTel is very buggy and slow for me`
`03:30.43`	`Blake0PS`	`iMediax: For outgoing calls.`
`03:30.43`	`Frawg`	`just using tcpdump?`
`03:30.46`	`tzanger`	`that new jitter buffer is still broken with trunking :-)`
`03:30.55`	`forrestc`	`BlakeOPS: but the error you're getting usually means that you've got a dialplan issue.`
`03:31.43`	`bjohnson`	`the Click-to-call service at broadvoxdirect.com is interesting. Has a web gui for that already been done? description at http://voxilla.com/voxstory51.html`
`03:31.56`	`Blake0PS`	`forrestc: I set it up exactly as described on the FWD website`
`03:32.30`	`forrestc`	`BlakeOPS: Not much more I can help you with... Except to say I have run across several docs which indicate that directly connecting to FWD has at least in the past been problematic.`
`03:33.18`	`Blake0PS`	`okay thanks anyway`
`03:33.20`	`forrestc`	`BlakeOPS: you do have the exten stuff in the right context, though, right?`
`03:33.31`	`bjohnson`	`speaking of broadvoxdirect .. anyone know what they're like as a voip provider? They're one of the only ones that I can find that offer a good selection of Canadian DIDs`
`03:33.37`	`iMediax`	`Blake show us your FWD section in you extensions.conf`
`03:33.41`	`Blake0PS`	`forrestc: what context should it be`
`03:33.58`	`Blake0PS`	`i threw it in default because the FWD sites description was ambiguous`
`03:34.21`	`forrestc`	`BlakeOPS: it should be in the context for your sip phone, or a context included in that context.`
`03:34.31`	`Essobi`	`Frawg tcpdump or tethereal`
`03:34.41`	`Blake0PS`	`forrestc: I have a Cisco 12SP+ which speaks SCCP`
`03:34.45`	`Essobi`	`tethereal can make some logical decisions about what's going on.`
`03:35.00`	`Essobi`	`Blake0PS I with the SCCP support was better in *`
`03:35.13`	`Essobi`	`I have 3000 some odd SCCP phones deployed.`
`03:35.29`	`forrestc`	`BlakeOPS: Ok, subsitute SIP for SCCP in my above statement.`
`03:35.52`	`Blake0PS`	`what is the context of my sccp phone then?`
`03:36.10`	`*** join/#asterisk snom190 (~inetd@208-242-45-124.ded.btitelecom.net)`
`03:36.16`	`file[laptop]`	`bjohnson: I deal with broadvox, their stuff is just... not the best`
`03:36.23`	`file[laptop]`	`not the worst, but not the best`
`03:36.46`	`snom190`	`I've got a working * setup with snom 190's and Digium TDM cards, i'm having trouble with caller id, can somebody give me a few pointers (things not in the wiki)`
`03:37.14`	`file[laptop]`	`their DIDs act funky sometimes, their termination sometimes partially goes kaboom`
`03:37.26`	`file[laptop]`	`like last week all of Canada fell off their network ;)`
`03:37.38`	`file[laptop]`	`for part of a day, forget which one`
`03:37.44`	`file[laptop]`	`</rant>`
`03:38.21`	`brc_`	`file!!!!!!`
`03:38.22`	`forrestc`	`blakeops: you should have set this in your sccp config file.`
`03:38.24`	`file[laptop]`	`hi brc`
`03:38.38`	`Frawg`	`hrmf`
`03:38.39`	`forrestc`	`brc: don't you know you aren't supposed to yell file in a crouded room?`
`03:38.45`	`Frawg`	`it looks like the linux box might be doing the buffering`
`03:38.47`	`bjohnson`	`file: the reviews I've found are spotty too .. but London and Kitchener DIDs seem to be hard to find unless I use different providers`
`03:39.02`	`file[laptop]`	`bjohnson: when it works it's fine and dandy`
`03:39.03`	`bjohnson`	`tzanger: you still have Kitchener available?`
`03:39.09`	`shido6`	`well`
`03:39.11`	`shido6`	`bjohnson`
`03:39.17`	`shido6`	`give me a few more weeks`
`03:39.24`	`bjohnson`	`file: right .. I'm looking for stability`
`03:39.44`	`*** join/#asterisk FirstSword (~Miranda@HSE-Quebec-City-ppp129829.qc.sympatico.ca)`
`03:39.54`	`bjohnson`	`shido6: OK. you're cdn 800s are still approx $0.08 per minute?`
`03:40.23`	`FirstSword`	`hi bjohnson`
`03:40.25`	`FirstSword`	`hi all`
`03:40.27`	`shido6`	`Chatham, Guelph , Hamilton, Kingston`
`03:40.31`	`bjohnson`	`man .. I;ve been having trouble with your and you're today`
`03:40.37`	`shido6`	`Kitchener`
`03:40.39`	`shido6`	`London`
`03:40.41`	`shido6`	`Oshawa`
`03:40.47`	`bjohnson`	`cool`
`03:40.48`	`shido6`	`St Catharines`
`03:40.49`	`shido6`	`Toronto`
`03:40.50`	`NormAst`	`shido: oshawa?`
`03:40.51`	`shido6`	`and Windsor`
`03:40.53`	`shido6`	`are coming`
`03:40.56`	`shido6`	`just give me some time`
`03:41.06`	`NormAst`	`your getting a pri to oshawa?`
`03:41.08`	`NormAst`	`WHO?`
`03:41.10`	`shido6`	`already there`
`03:41.26`	`NormAst`	`I need Oshawa, Newmarket.`
`03:41.27`	`FirstSword`	`shido, you're setting up asterisk in those places?`
`03:41.34`	`shido6`	`the pops for my next project are there already`
`03:41.35`	`NormAst`	`Have hamilton, Barrie, Toronto`
`03:42.02`	`bjohnson`	`shido6: you have pricing established yet? $7.50 US like the MI ones?`
`03:42.07`	`shido6`	`I havent turned them up I have to clear it with JerJer which means i need to get a dedicated pipe to our level 5 secure facility in michigan`
`03:42.31`	`shido6`	`they say if u build it they will come`
`03:43.00`	`bjohnson`	`well .. there seems to be a good number of Cdn users on here`
`03:43.11`	`NormAst`	`shildo6: Who's provider the oshawa PRI?`
`03:43.17`	`bjohnson`	`and there isn't much available for Cdn DIDs`
`03:43.18`	`FirstSword`	`Im at cdn user^^`
`03:43.32`	`shido6`	`I am`
`03:43.43`	`shido6`	`its a long story - just hold on to your teeth`
`03:43.45`	`NormAst`	`How much for a PRI from Oshawa to TO?`
`03:43.50`	`FirstSword`	`oh.. i'm managing a PRI in richmond hill righ tnow`
`03:43.53`	`shido6`	`they will be active as soon as I can get a dedicated pipe to nufone`
`03:43.58`	`tzanger`	`bjohnson: yes`
`03:44.36`	`NormAst`	`shido6: gonna need one.`
`03:45.19`	`shido6`	`shido6@gmail.com with your request - I can put you on a list until its turned up`
`03:45.27`	`NormAst`	`K`
`03:45.35`	`shido6`	`I have to make sure Digium gets paid before I do anything new`
`03:46.01`	`liquidno2`	`what is a lightweight app to play the gsm files?`
`03:46.08`	`shido6`	`playback`
`03:46.12`	`shido6`	`:)`
`03:46.18`	`liquidno2`	`You sir are a a funny man`
`03:46.33`	`shido6`	`why reinvent the wheel no downloads`
`03:46.37`	`shido6`	`just make an extension`
`03:46.39`	`shido6`	`and playback`
`03:46.53`	`liquidno2`	`I have my asterisk server setup as a samba share`
`03:47.02`	`liquidno2`	`so I can access the MOH and other stuff quickly`
`03:47.02`	`shido6`	`sweet`
`03:47.11`	`fwc`	`Would someone tell me all the stupid things to double check as to why I can only use one channel at a time on this *(@#&$ PRI?`
`03:47.22`	`liquidno2`	`I was just looking for something to preview the sounds in windows`
`03:47.31`	`pointer-gaim`	`fwc: how are you dialing?`
`03:47.35`	`iMediax`	`winamp can`
`03:47.41`	`fwc`	`Zap/g1`
`03:47.43`	`shido6`	`fwc`
`03:47.45`	`NormAst`	`fwc: Dial(Zap/G1/NUMBER)`
`03:47.49`	`shido6`	`how many ports do you have on that thing`
`03:47.49`	`liquidno2`	`iMediax: I tried that, but it doesn't play them...`
`03:47.57`	`fwc`	`All 23.`
`03:47.57`	`shido6`	`u can always go out voip when ur ports are filled`
`03:47.58`	`pointer-gaim`	`fwc: and what channels are in g1?`
`03:48.02`	`iMediax`	`there's a plugin... check winamp site for it`
`03:48.07`	`fwc`	`I Think all 23`
`03:48.13`	`liquidno2`	`iMediax: danke that is the guidance I needed`
`03:48.20`	`snom190`	`everytime an incoming call comes in, all my phones show a call from "asterisk"`
`03:48.26`	`shido6`	`heheh`
`03:48.30`	`shido6`	`snom190 got a cheap fxo?`
`03:48.34`	`shido6`	`or callerid not set`
`03:48.35`	`shido6`	`?`
`03:48.36`	`*** join/#asterisk darien (sentry21@mtl.rackplans.net)`
`03:48.42`	`fwc`	`group = 1`
`03:48.42`	`fwc`	`channel => 1-23`
`03:48.44`	`snom190`	`i'm using the digium fxo`
`03:48.55`	`shido6`	`bought it from Digium?`
`03:49.03`	`snom190`	`local distro... but yes..`
`03:49.08`	`snom190`	`callerid=asreceived`
`03:49.11`	`shido6`	`local distro?`
`03:49.15`	`snom190`	`in zaptel.conf`
`03:49.20`	`snom190`	`err.. zapata.conf`
`03:49.27`	`shido6`	`pastebin.ca ur settings`
`03:49.32`	`snom190`	`local distributer that digium referred me to`
`03:49.35`	`liquidno2`	`sweet`
`03:49.40`	`shido6`	`good answer, snom190`
`03:49.41`	`shido6`	`:)`
`03:49.41`	`liquidno2`	`iMediax: thanks again`
`03:49.48`	`iMediax`	`np`
`03:49.48`	*** join/#asterisk andrew` (~andrew@adsl-67-119-26-96.dsl.snfc21.pacbell.net)
`03:49.52`	`pointer-gaim`	`fwc: hmm...that's what I've got`
`03:50.00`	`liquidno2`	`"Abandon all hope, ye who dial here."`
`03:50.28`	`Essobi`	`MAHAHA`
`03:50.34`	`fwc`	`some days asterisk makes me feel like my brain is all twisted into knots.`
`03:50.44`	`Essobi`	`Umm.`
`03:50.47`	`Essobi`	`It is.`
`03:50.57`	`fwc`	`Is there a command line to verify what channels are in what groups?`
`03:51.06`	`snom190`	`http://pastebin.ca/5577`
`03:51.07`	`Essobi`	`In fact, it's theorized the more knots, the smarter you are.`
`03:51.16`	`FirstSword`	`I wonder if anyone used soekris boards b4...`
`03:51.27`	`snom190`	`netxusa, fyi`
`03:51.32`	`Essobi`	`fwc Did you check help on the cli?`
`03:51.39`	`shido6`	`um`
`03:51.40`	`shido6`	`snom`
`03:51.44`	`snom190`	`...`
`03:51.46`	`shido6`	`i think you're missing some info there`
`03:51.49`	`shido6`	`can I pm you?`
`03:51.52`	`fwc`	`yep. Tried all the likely canidates.`
`03:51.53`	`snom190`	`certainly`
`03:52.33`	`fwc`	`Let me re-check everything. All that I should have to do is make sure all the channels are in the same group and that I dial Zap/G1, right?`
`03:53.28`	`NormAst`	`fwc: what's the zap show channels give you`
`03:54.06`	`bjohnson`	`ok. I've got a Bell mobility deal for the people here right now. Basically you can get a prepaid phone with headset and $30 credit for $16. No gimmics. No money to me. Anyone interested?`
`03:54.27`	`fwc`	`all 23 just listed as "1.....default"`
`03:54.29`	`file[laptop]`	`unfortunately I don't live in a Bell Mobility area`
`03:54.32`	`file[laptop]`	`please try your call again later.`
`03:54.52`	`NormAst`	`fwc: that's good.`
`03:55.05`	`fwc`	`And one outbound call works fine.`
`03:55.06`	`bjohnson`	`Firestrm: yes`
`03:55.19`	`fwc`	`The second one asterisk freaks and says all the Zap channels are busy.`
`03:56.51`	`*** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de)`
`03:59.16`	`fwc`	`It's working now and I didn't do anything.`
`03:59.20`	`fwc`	`(@#$*`
`03:59.35`	`liquidno2`	`it hates you... it told me so.`
`03:59.38`	`liquidno2`	`sorry`
`03:59.48`	`liquidno2`	`I didn't think it was any of my business`
`03:59.50`	`liquidno2`	`:)`
`04:00.08`	`fwc`	`Hmmm...`
`04:00.45`	`bjohnson`	`FirstSword: yes`
`04:01.02`	`fwc`	`I hate that.`
`04:01.53`	`fwc`	`When the problem goes away by itself, that is.`
`04:02.00`	`fwc`	`(@#$*`
`04:03.18`	`fwc`	`FirstSword: you still monitoring?`
`04:03.23`	`FirstSword`	`yes i am`
`04:03.31`	`FirstSword`	`i just got back`
`04:03.32`	`fwc`	`If so, I've used the soekris boards... not with asterisk though.`
`04:03.52`	`FirstSword`	`wat happened to ur soekris board?`
`04:03.54`	`fwc`	`Not sure I'd want to try except for maybe a single FXO or something`
`04:04.07`	`*** join/#asterisk orlon_ (~orlon@office-fw.iexec.net.au)`
`04:04.19`	`fwc`	`They work great for internet routing.`
`04:04.22`	`FirstSword`	`right.. but can't use x100p .. only support 3.3v?`
`04:04.25`	`fwc`	`I.E. wireless access stuff.`
`04:04.59`	`orlon_`	`what's the difference between 'member => Agent/@1' and 'member => Agent/:1' ??`
`04:05.03`	`fwc`	`Not sure...`
`04:05.05`	`orlon_`	`in queues.conf`
`04:05.18`	`orlon_`	`i've tried reading the source for app_queue.c and can't work out where it's used`
`04:05.31`	`FirstSword`	`which model is ur board?`
`04:05.46`	`FirstSword`	`45xx or 48xx?`
`04:05.56`	`fwc`	`firstsword: it was an early one.. Let me look at the soekris page.`
`04:06.36`	`fwc`	`4501.. I think.`
`04:06.42`	`fwc`	`It's been a while`
`04:06.52`	`fwc`	`yeah.. 4501`
`04:07.06`	`liquidno2`	`soekris rock`
`04:07.12`	`FirstSword`	`ic~ hm.. i want to buy one and try to install asterisk to it`
`04:07.27`	`FirstSword`	`im completely new ^__^`
`04:07.39`	`liquidno2`	`I don't think those are powerful enough for asterisk`
`04:07.40`	`jollyr`	`anybody here know a good termination in the us`
`04:07.46`	`AgiNamu`	`nufone`
`04:07.49`	`Moc`	`jollyr, nufone`
`04:08.01`	`bjohnson`	`check the wiki and mailing list .. I also read the hardware in the soekris boards was a little too slow to handle more than a few calls .. can't remember the details`
`04:08.08`	`jollyr`	`but nufone is not responding`
`04:08.18`	`liquidno2`	`why am I not surprised`
`04:08.22`	`bjohnson`	`jollyr: give them a second`
`04:08.25`	`denon`	`anyone awake ever use call recording in queues?`
`04:08.31`	`bjohnson`	`shido6: TALK TO jollyr !!!`
`04:08.34`	`fwc`	`IT would be interesting to try a net4826 for a small home * box.`
`04:08.42`	`fwc`	`But I wouldn't start there.`
`04:08.54`	`Moc`	`denon, Im not gay`
`04:08.56`	`Moc`	`;)`
`04:08.59`	`denon`	`hah`
`04:09.00`	`denon`	`neither am I`
`04:09.00`	`AgiNamu`	`hey, shido6 fixed my box`
`04:09.03`	`bjohnson`	`FirstSword: few people running * on that linksys router that is hackable`
`04:09.05`	`AgiNamu`	`he really knows what he's doing`
`04:09.11`	`denon`	`Moc: you use call recording on queues?`
`04:09.16`	`bjohnson`	`wifi G router or something that has linux on it`
`04:09.21`	`denon`	`wrt54gs`
`04:09.23`	`liquidno2`	`WRT54G`
`04:09.24`	`jollyr`	`i try to followup the code that they give`
`04:09.25`	`*** join/#asterisk Inv_arp (junya@adsl-8-230-175.mia.bellsouth.net)`
`04:09.37`	`denon`	`get the S, more space`
`04:09.41`	`liquidno2`	`I run one of those, but offload asterisk to a dedicated box`
`04:09.48`	`Moc`	`denon, I dont use queue at all yet`
`04:09.51`	`FirstSword`	`WRT54G? isnt this the most common one?`
`04:09.54`	`liquidno2`	`I couldn't wedge a TDM400 in there :)`
`04:09.56`	`FirstSword`	`i see it everywhere`
`04:09.59`	`denon`	`you shouls, its pretty cool :)`
`04:10.02`	`denon`	`should, even`
`04:10.03`	`denon`	`AgiNamu?`
`04:10.10`	`AgiNamu`	`yea?`
`04:10.22`	`denon`	`you use call recording on queues?`
`04:10.24`	`Moc`	`someone should make an * package on openwrt site`
`04:10.32`	`AgiNamu`	`never used queues`
`04:10.35`	`denon`	`bummer`
`04:10.36`	`fwc`	`Moc: They already have them.`
`04:10.38`	`liquidno2`	`Moc: you mean there isn't one already?`
`04:10.57`	`Moc`	`fwc, I didnt show when I tryed it last sunday`
`04:11.04`	`Inv_arp`	`hmm on the HT486 do i put Xover cables on lan side or straight thru?`
`04:11.06`	`bjohnson`	`jollyr: I guess they need a few more seconds .. he was just here`
`04:11.39`	`orlon_`	`denon: do you know what the difference bewteen 'member => Agent/@1' and 'member => Agent/:1' in queues.conf is ??`
`04:11.40`	`Moc`	`Anyway, back to my Faxing server/client solution ;)`
`04:11.46`	`*** join/#asterisk Legend (~legend@24.244.142.133)`
`04:11.49`	`liquidno2`	`Now for the utltimate test...`
`04:11.55`	`fwc`	`Moc: Guess not on the page.... but at http://nthill.free.fr/openwrt/ipkg/testing/20041213/`
`04:11.57`	`liquidno2`	`persistance through a reboot`
`04:12.01`	`FirstSword`	`bjohnson: i've read some info about soekris with asterisk in wiki, one guy said it works pretty fine .. and stay for months.. but i havent check the mailing list (i'd check it now)`
`04:12.13`	`denon`	`orlon_: isnt that the weight?`
`04:12.35`	`denon`	`I dont use queues much .. im the wrong guy to aks`
`04:12.36`	`denon`	`ask`
`04:12.47`	`fwc`	`Guess it's time for me to head home and go to bed since I ahve to be up and in to work during human hours tomorrow.`
`04:12.49`	`orlon_`	`denon: no the weight is after that, delimited by a comma`
`04:12.55`	`orlon_`	`denon: oh...bugger`
`04:12.58`	`denon`	`I want to use em more .. but without any docs on call recording ..`
`04:13.02`	`fwc`	`As opposed to the hacker's hours I normally keep.`
`04:13.15`	`orlon_`	`denon: have you looked at voip-info.org's page for that?`
`04:13.42`	`denon`	`orlon_: I see where you specify the format .. just not sure how you start and stop it .. if you're sposed to do that in the dialplan per usual or what`
`04:14.04`	`orlon_`	`denon: aah`
`04:14.04`	`Luke-Jr`	`bjohnson: * runs on WRT54G? o.O`
`04:14.19`	`Essobi`	`hey huys`
`04:14.21`	`Essobi`	`guts`
`04:14.23`	`Essobi`	`guys`
`04:14.30`	`Essobi`	`how do you get noops to display on the console?`
`04:14.32`	`Moc`	`Luke-Jr, it does`
`04:14.36`	`orlon_`	`denon: i would've thought it'd just record the whole call`
`04:14.44`	`Essobi`	`so you can see what you're doing with them in dial plans?`
`04:14.49`	`*** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net)`
`04:14.49`	`denon`	`orlon_: well, for example, where do you tell it how to name the call, and such`
`04:14.51`	`Essobi`	`set verbose 1? I can't remember`
`04:14.53`	`Essobi`	`Mmm`
`04:15.01`	`denon`	`orlon_: on a normal call, you specify it in extensions.conf`
`04:16.23`	`orlon_`	`denon: good point...`
`04:16.43`	`*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)`
`04:16.52`	`denon`	`orlon_: functionality is obviously there`
`04:17.12`	`orlon_`	`denon: from app_queue.c`
`04:17.14`	`orlon_`	`<PROTECTED>`
`04:17.14`	`orlon_`	`<PROTECTED>`
`04:17.14`	`orlon_`	`<PROTECTED>`
`04:17.52`	`orlon_`	`line 1191 or so`
`04:18.08`	`orlon_`	`denon: looks like it uses the CDR unique id if that call fails`
`04:18.17`	`*** join/#asterisk afrosheen (~afro@c-24-0-136-170.client.comcast.net)`
`04:19.02`	`zimdog`	`This server will not have any fxo hardware. I want it to be able to connect sip phones and then use IAX to tunnel to the internal server. I am open to other ways to solve this problem also.`
`04:19.04`	`liquidno2`	`sweet... nailed it all down...`
`04:19.12`	`liquidno2`	`persistance through a reboot`
`04:19.20`	`denon`	`orlon_: hmm .. might just try the usual dialplan stuff then`
`04:19.24`	`afrosheen`	`zimdog: ok`
`04:19.28`	`denon`	`orlon_: maybe that sets the filename`
`04:19.48`	`zimdog`	`sorry`
`04:19.48`	`orlon_`	`denon: yeah, looks like it`
`04:20.10`	`orlon_`	`denon: if i'm reading it right it's getting the var MONITOR_FILENAME from the channels variables`
`04:21.03`	`*** join/#asterisk datareactor (datareacto@203.81.192.33)`
`04:21.35`	`liquidno2`	`Question`
`04:21.42`	`liquidno2`	`in theory...`
`04:21.51`	`liquidno2`	`mpg123 will play a play list`
`04:22.07`	`liquidno2`	`can I pipe in a shoutcast stream for moh?`
`04:22.18`	`zimdog`	`Anyone with a solution for this setup http://www.pastebin.com/239999`
`04:22.44`	`afrosheen`	`liquidno2: : of course you can, it's in the wiki`
`04:22.46`	`denon`	`orlon_: hmm .. yeah, normal call recording doesnt seem to be doin it`
`04:23.07`	`liquidno2`	`afrosheen: sometimes I just talk outloud as I have nothing of real value to offer :)`
`04:23.08`	`Mavvie`	`liquidno2: give it a public IP address.`
`04:23.48`	`denon`	`orlon_: we usually set CALLFILENAME`
`04:24.05`	`orlon_`	`with SetVar?`
`04:24.15`	`orlon_`	`denon: try setting MONITOR_FILENAME`
`04:25.17`	`denon`	`orlon_: look like you have to stop it? or does app_queue take care of that?`
`04:25.53`	`orlon_`	`denon: i can't see a call to ast_monitor_stop in app_queue.c`
`04:26.10`	`afrosheen`	`zimdog: get the * box away from the firewall or forward ports on it, it should be exposed to the outside`
`04:26.10`	`orlon_`	`denon: maybe it's relying on Hangup or something else handling it`
`04:26.29`	`afrosheen`	`zimdog: it's easy to deal with 1 NAT side but not 2`
`04:26.43`	`denon`	`dunno, trying again`
`04:27.23`	`denon`	`ooh, think that worked`
`04:28.29`	`denon`	`pity I cant go straight to speex :)`
`04:29.17`	`*** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)`
`04:29.25`	`orlon_`	`denon: how come?`
`04:29.34`	`jdv79`	`is it normal for asterisk to eat the CPU for a snack?`
`04:29.37`	`afrosheen`	`no`
`04:29.44`	`afrosheen`	`are you sure it's asterisk doing it`
`04:29.49`	`jdv79`	`top says so`
`04:30.02`	`jdv79`	`it was a fresh checkout from y'day`
`04:30.03`	`afrosheen`	`how many simul. calls`
`04:30.05`	`afrosheen`	`oh`
`04:30.07`	`jdv79`	`0`
`04:30.09`	`afrosheen`	`well that may explain it`
`04:30.09`	`denon`	`orlon_: hmm .. well .. just assumed not, since it wasnt listed`
`04:30.10`	`jdv79`	`that's the scary thing`
`04:30.15`	`snom190`	`asterisk is HUNGRY!`
`04:30.17`	`afrosheen`	`strace it and see if it's getting hung`
`04:30.22`	`jdv79`	`ok`
`04:30.32`	`denon`	`orlon_: we usually do a batch encode to speex during offpeak anyway`
`04:30.34`	`zimdog`	`afrosheen: thanks`
`04:31.33`	`afrosheen`	`zimdog: we wanted to do it that way as well, but double nat is a bitch`
`04:31.51`	`harryvv`	`Im getting this problem. But, I have assigned its own irq and even went as far as removing the nic card to spare a irq. There is only the x100p and vidio card installed. When modprobing the wcfxo module it comes back as the path its belongs to and its missing yet its there. The error is insmode: /lib/modules/2.4.27/misc/wcfxo.o no such device and "wcfxo: Out of space to write register 05 with 0a" It was a working system all day and been`
`04:31.54`	`jdv79`	`ha, all strace said so far is "poll("`
`04:32.02`	`afrosheen`	`jdv79: over and over right`
`04:32.07`	`jdv79`	`nah, just once`
`04:32.17`	`harryvv`	`any ideas?`
`04:32.29`	`afrosheen`	`jdv79: congratulations, you found a big bug`
`04:32.33`	`jdv79`	`:)`
`04:32.36`	`jdv79`	`i feel special`
`04:32.41`	`harryvv`	`btw, thats a bad idea to have a x100p go down and bring down asterisk to.`
`04:33.04`	`denon`	`orlon_: "no such format: speex"`
`04:33.30`	`harryvv`	`hi denon`
`04:33.34`	`orlon_`	`denon: that'd do it`
`04:33.34`	`afrosheen`	`harryvv: kernel 2.6.x?`
`04:33.40`	`harryvv`	`2.4.27`
`04:33.48`	`afrosheen`	`weird`
`04:33.52`	`denon`	`orlon_: oh well, gsm will do ok`
`04:34.02`	`afrosheen`	`I've seen odd hardware problems with some 2.6.x kernels but not 2.4`
`04:34.24`	`harryvv`	`afroshen it has come and gone but now its perm. I can change the irqs manually for those pci slots and even removed the nic card.`
`04:34.40`	`harryvv`	`well, this is a old pentium II 440BX chipset`
`04:34.52`	`afrosheen`	`harryvv: have you looked carefully at the motherboard for bad caps?`
`04:35.00`	`harryvv`	`bad caps/`
`04:35.03`	`harryvv`	`capacitors?`
`04:35.11`	`afrosheen`	`yes`
`04:35.17`	`afrosheen`	`they will cause graceful hardware failure`
`04:35.24`	`afrosheen`	`in power supplies also`
`04:35.38`	`harryvv`	`this been a working system. its a running system right now.`
`04:35.44`	`shido6`	`afrosheen`
`04:35.46`	`harryvv`	`just this issue with asterisk`
`04:36.05`	`harryvv`	`err, a issue with x100p`
`04:36.08`	`afrosheen`	`harryvv: I've seen alot of problems with caps in boards of that era`
`04:36.25`	`harryvv`	`mmmm`
`04:36.40`	`afrosheen`	`just look`
`04:36.50`	`afrosheen`	`either you'll see flat crosses or bubbles with ooze`
`04:36.52`	`harryvv`	`why thay dry out and pop ? I have two years electrionic technoligy training behind me.`
`04:37.14`	`harryvv`	`what was the symptom?`
`04:37.23`	`harryvv`	`the affect on the system`
`04:38.01`	`harryvv`	`also, would it smell like fish? we has some student that poped a couple dialectrics.`
`04:39.33`	`*** join/#asterisk OzJames79 (~James@203.208.64.29)`
`04:40.53`	`*** join/#asterisk denon (denon@synapse.subneural.net)`
`04:40.53`	`*** mode/#asterisk [+o denon] by ChanServ`
`04:41.24`	`jdv79`	`well i stoped it and start it back up again`
`04:41.36`	`jdv79`	`i'll keep an eye on things`
`04:41.54`	`orlon_`	`denon: got it. the diff between @ and : in an Agent string is in chan_agent.c:1021`
`04:42.16`	`orlon_`	`denon: ':' sets 'waitforagent'`
`04:44.03`	`orlon_`	`denon: which will keep a caller in a queue if there are no available agents`
`04:44.09`	`orlon_`	`denon: makes sense now`
`04:44.41`	`*** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net)`
`04:45.29`	`Mavvie`	`somebody here familair with enumlookup.agi ?`
`04:46.48`	`FirstSword`	`anyone knows if there's a sound file for just telephone tone?`
`04:47.09`	`FirstSword`	`like the sound u get when u pick up a telephone...`
`04:47.14`	`shido6`	`crap`
`04:47.17`	`shido6`	`theres a url for that`
`04:47.20`	`shido6`	`gimme a secon`
`04:48.05`	`shido6`	`http://www.telephonetribute.com/international.html`
`04:48.10`	`shido6`	`should do it`
`04:48.35`	`*** join/#asterisk B4 (~B4@202.69.48.245)`
`04:48.52`	`FirstSword`	`thx`
`04:49.07`	`*** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de)`
`04:50.45`	`FirstSword`	`haha. this is nice!!1 thx shido6~`
`04:51.01`	`shido6`	`you asked me a question, I will answer you as thoroughly as I can`
`04:51.07`	`eipi`	`what's wrong: app_meetme.c:1039 conf_run: Unable to write frame to channel: No such file or directory`
`04:51.21`	`JerJer`	`zaptel timing?`
`04:51.24`	`orlon_`	`eipi: you got a zap driver loaded?`
`04:51.37`	`eipi`	`yes`
`04:52.18`	`orlon_`	`eipi: are the dev nodes there? eg. /dev/zap/*`
`04:53.28`	`eipi`	`yes, but we are three guys trying to do meetme`
`04:53.47`	`eipi`	`that error appears in the meetme server`
`04:56.16`	`orlon_`	`eipi: do you get something like this?`
`04:56.16`	`orlon_`	`# ls -l /dev/zap/channel`
`04:56.17`	`orlon_`	`crw-rw-rw- 1 root root 196, 254 Jan 1 1970 /dev/zap/channel`
`04:56.19`	`JerJer`	`meetme requires zaptel`
`04:56.36`	`Essobi`	`Howdy JerJer.`
`04:56.37`	`harryvv`	`afrosheen i have another system i can use.`
`04:56.38`	`Essobi`	`:)`
`04:56.42`	`JerJer`	`mooo`
`05:00.06`	`*** join/#asterisk Inv_arp (~junya@adsl-8-230-175.mia.bellsouth.net)`
`05:01.25`	`Inv_arp`	`can the HT486 be locked ... i cant connect to its ipaddress given by its prompt over the phone`
`05:04.12`	`*** part/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)`
`05:05.13`	`*** join/#asterisk FirstSword (~Miranda@HSE-Quebec-City-ppp129829.qc.sympatico.ca)`
`05:05.32`	`FirstSword`	`Hi`
`05:06.57`	`*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)`
`05:07.02`	`ManxPower`	`~docs`
`05:07.03`	`jbot`	`Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org`
`05:09.43`	`Essobi`	`~sip`
`05:09.44`	`jbot`	`X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261)`
`05:09.53`	`Essobi`	`~7960`
`05:10.04`	`Essobi`	`:\`
`05:10.20`	`Essobi`	`~seen my ass!`
`05:10.21`	`jbot`	`i haven't seen 'my ass!', Essobi`
`05:10.29`	`Essobi`	`Ahh, good ole Jbot.`
`05:10.43`	`Essobi`	`Howdy ManxPower`
`05:12.40`	`datareactor`	`<PROTECTED>`
`05:13.02`	`datareactor`	`~packetcable`
`05:13.42`	`ManxPower`	`Hello Essobi`
`05:14.35`	`datareactor`	`~iptv`
`05:14.45`	`*** join/#asterisk Inv_arp (~junya@adsl-8-230-175.mia.bellsouth.net)`
`05:15.13`	`datareactor`	`seen ~pranav`
`05:15.53`	`*** part/#asterisk orlon_ (~orlon@office-fw.iexec.net.au)`
`05:17.47`	`ManxPower`	`The newbies must crawl back home at night.`
`05:18.02`	`Essobi`	`I guess.`
`05:18.06`	`Essobi`	`It got quiet in there.`
`05:18.23`	`Essobi`	`here, even.`
`05:18.48`	`Inv_arp`	`can the HT486 be locked ... i cant connect to its ipaddress given by its prompt over the phone not a subnet issue`
`05:19.48`	`ManxPower`	`Inv_arp, Unless you bought it as part of a service provider discount, it's not locked.`
`05:19.51`	`Essobi`	`can't ping?`
`05:20.10`	`Inv_arp`	`ManxPower: so it can be?`
`05:20.31`	`ManxPower`	`Inv_arp, You did what ever magic key sequece they use to have the device read ip's IP address to you over the phone.`
`05:20.37`	`Essobi`	`Anyways.. Hey, ManxPower .. I know you or the bot can answert his.. what's the minimum config I need in sip.conf to get a 7960 to register?`
`05:20.45`	`ManxPower`	`Inv_arp, I've never heard of a GS product being locked, but I admit it could happen.`
`05:21.13`	`ManxPower`	`I dunno, but I can put a sip.conf entry for a 7905G on pastebin.ca`
`05:21.18`	`Essobi`	`ManxPower Anything with firmware technically can be locked. given the programmers or someone has some know how to do it. ;)`
`05:21.43`	`Essobi`	`ManxPower I got a 60 but a 05 config should be all the same. I'd appreciate it.`
`05:22.21`	`ManxPower`	`BTW, this is pretty much my standard config for ALL SIP devices.`
`05:22.24`	`Essobi`	`I got my phone sending registrations finally, but * keeps kicking it out the door yelling bad ragistration`
`05:22.39`	`Essobi`	`cool.. drop my the 7905 tftp config too? :)`
`05:22.42`	`Essobi`	`me even`
`05:23.05`	`ManxPower`	`http://pastebin.ca/5581`
`05:23.22`	`ManxPower`	`I have NO idea where the 7905G config is.`
`05:23.36`	`Essobi`	`Ah, okay.`
`05:23.42`	`ManxPower`	`It's been deployed to a VP and he never uses it and it's been moved around a lot.`
`05:24.08`	`Essobi`	`Oh wait.`
`05:24.12`	`Essobi`	`It just came up.`
`05:24.13`	`Essobi`	`:)`
`05:24.19`	`ManxPower`	`There prolly should be a username=2020 in there as well, I guess.`
`05:24.21`	`Essobi`	`But thanks, anyways.`
`05:24.25`	`Essobi`	`how you been man?`
`05:25.32`	`Essobi`	`FFS, do 7960's register like a mad cow or what? I'm seeing registers over and over and over again.`
`05:25.36`	`Essobi`	`all succeeding mind you`
`05:25.55`	`ManxPower`	`BTW, all I ever do to put sip.conf entries in for SIPura SPA-2k, SPA-3k, SPA-841, Cisco 7905G, Cisco ATA-186, Polycom IP 300, Polycom IP 500, Grandstream BT101 is pretty much what I put in pastebin, changing only the relevant stuff.`
`05:26.07`	`shido6`	`u have all those lines registering, essobi?`
`05:26.14`	`ManxPower`	`I've been busy, tired, leaving for New Orlans tomorrow.`
`05:26.25`	`Essobi`	`I took the rest out. Just 1 line registering.`
`05:26.34`	`Essobi`	`Nice`
`05:26.46`	`shido6`	`wait till u get vmail , it'll be cool for a second`
`05:26.56`	`shido6`	`then u'll learn to hate the little red light`
`05:27.02`	`ManxPower`	`BTW, we use things like 2022a and 2022b for the different "lines" of devices, if they all are the same "extension" number.`
`05:27.09`	`shido6`	`and the mailbox icon in the place of your phone`
`05:27.28`	`*** join/#asterisk outsidefactor (~blah@203-206-254-16.dyn.iinet.net.au)`
`05:27.46`	`ManxPower`	`I just REALLY wish SIPura would come out with firmware that lets you set the gain.`
`05:28.58`	`*** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net)`
`05:29.44`	`ManxPower`	`Once the SPA phone firmware matures (it's pretty mature already) and they add a second ethernet port and PoE it will be a true enterprize class phone. As it is now, it's really only suitable for small deployments or for service providers with home users.`
`05:33.05`	`Frawg`	`Essobi: go into your confog`
`05:33.07`	`*** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com)`
`05:33.19`	`Frawg`	`check in sip`
`05:33.23`	`Frawg`	`there is register expires`
`05:33.26`	`Frawg`	`see what that is set to`
`05:33.34`	`Essobi`	`confogdarndidlydodid`
`05:33.35`	`Essobi`	`oh`
`05:33.46`	`Essobi`	`config. ;) I thought you were getting all ned flanders on me.`
`05:33.57`	`ManxPower`	`As long as the phone is not behind NAT you can set the registration expiration to something like 3600 or much higher.`
`05:34.55`	`Essobi`	`it is behind a nat`
`05:35.02`	`Essobi`	`I need to ask the admin what the timeouts are set to`
`05:35.07`	`Essobi`	`and get closer to it`
`05:35.22`	`ManxPower`	`You can just set qualify=something then and set the reg expires to a high number.`
`05:35.28`	`Essobi`	`right now it's just chucking those things out like a fireman in a burning orphanage`
`05:35.45`	`ManxPower`	`qualify should keep the nat mappings from timeing out.`
`05:35.56`	`Essobi`	`on the router?`
`05:36.01`	`ManxPower`	`Yeah`
`05:36.02`	`Essobi`	`or sip.cnf?`
`05:36.03`	`Essobi`	`oh`
`05:36.04`	`Essobi`	`ok`
`05:36.05`	`Essobi`	`;)`
`05:36.28`	`ManxPower`	`qualify makes Asterisk send an OPTIONS packet to the phone and wait for a response pretty often.`
`05:36.33`	`brc_`	`42`
`05:36.41`	`ManxPower`	`That will keep the nat port/address translations open.`
`05:37.07`	`Essobi`	`right right`
`05:37.31`	`ManxPower`	`Then you can set your registration expiration to pretty much anything you want.`
`05:37.33`	`Essobi`	`I got a 2 or 3 gig Xenon compaq coming up in the works`
`05:37.52`	`Essobi`	`for.. get this.. "Call recording." Jeees.`
`05:38.01`	`ManxPower`	`Tomorrow night we are doing an Stealth Asterisk Install`
`05:38.10`	`Essobi`	`"Stealth"?`
`05:38.16`	`ManxPower`	`nod nod nod`
`05:38.25`	`Essobi`	`Like all dressing up in ninja gear?`
`05:38.45`	`Essobi`	`Or dropping the install where it ain't sposed to be? :)`
`05:38.48`	`ManxPower`	`With approval of a high mucky muck, we are taking and putting an Asterisk box bewteen the T-1's and the PBX at corporate HQ`
`05:38.50`	`*** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no)`
`05:38.57`	`Essobi`	`Ahh.`
`05:39.04`	`Essobi`	`Nice.`
`05:39.14`	`Essobi`	`Two PRI interface cards?`
`05:39.22`	`ManxPower`	`LOL!`
`05:39.39`	`ManxPower`	`The T-1 comes into a CHANNEL BANK, and then analog lines into the PBX.`
`05:39.50`	`ManxPower`	`They are too cheap to buy PRI cards`
`05:40.13`	`Essobi`	`Ahh, right right... Nice. cheap bastards. So.. what're you going to use for an interface?`
`05:40.13`	`ManxPower`	`Well, at least for the PBX.`
`05:40.46`	`WilliamK`	`Manx, I had a customer who was going to order 2 analog T1s and I said no way`
`05:40.46`	`Essobi`	`A ton of FXOs?`
`05:40.49`	`ManxPower`	`We are using a TE410P in the Asterisk box T-1 -> Asterisk -> Channel Bank -> PBX`
`05:40.50`	`WilliamK`	`and put my foot down`
`05:41.00`	`Essobi`	`ahhh`
`05:41.02`	`Essobi`	`right right`
`05:41.21`	`Essobi`	`So T1 is voice or data?`
`05:41.34`	`ManxPower`	`WildPikachu, I won't manage a system that isn't PRI normally.`
`05:41.38`	`ManxPower`	`T-1 voice.`
`05:41.43`	`Essobi`	`voice I'd assume but.. SHRUG you might be switching them over.`
`05:41.52`	`Essobi`	`Wild who?`
`05:41.54`	`Essobi`	`:)`
`05:41.59`	`Essobi`	`Love that nic completiong`
`05:42.07`	`ManxPower`	`Then we just have to make sure that Asterisk ALWAYS gets calls from channel 1 on the T-1 to channel 1 on the channel bank, etc.`
`05:42.16`	`*** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu)`
`05:42.30`	`*** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)`
`05:42.40`	`Essobi`	`Ewwww.`
`05:42.52`	`bjohnson`	`Beirdo: what happened to syonex? They don't have any SPAs?`
`05:42.54`	`Essobi`	`like channel bridging.. how you going to accomplish that?`
`05:42.57`	`ManxPower`	`Once that's running for a week or two, then we'll start doing clever stuff.`
`05:43.34`	`Beirdo`	`bjohnson: I dunno, he doesn't sell Grandstream or Sipura any more it seems`
`05:43.37`	`ManxPower`	`Essobi, I'm not totally sure yet. We can get NO information about how the T-1 does DID (for the DID trunks), but we'll figure it out.`
`05:43.48`	`harryvetch`	`ManxPower you been here long enoug. Ive been troubleshooting this problem on my asterisk box for the last hour and dont think its a irq problem as the hardware has been swithing to another machine as well as the hard drive. I get messge in modprobe wcfxo of wcfxo: Out of space to write register 06 with 0e and wcfxo: Out of space to write register 0f with 00 ? Ive looked in google and came up with some possibilities being one to update`
`05:44.03`	`bjohnson`	`damn .. I was just going to send a crowd of people to his web site for the SPAs`
`05:44.06`	`Essobi`	`It's likely a static DID ringdown..`
`05:44.18`	`ManxPower`	`bjohnson, voxzilla?`
`05:44.19`	`Essobi`	`how many DID's are they serving up on that T1?`
`05:44.25`	`ManxPower`	`I believe 100`
`05:44.28`	`bjohnson`	`ManxPower: canadian source`
`05:44.32`	`ManxPower`	`Or maybe just 20.`
`05:44.43`	`Essobi`	`Jees.. they HAVE to be dropping a DNIS somehow.. the question is how.`
`05:44.45`	`ManxPower`	`No idea yet. We'll figure out that too.`
`05:45.18`	`ManxPower`	`Essobi, Usually something like WINK CALLERID*DID WINK or something like that.`
`05:45.37`	`Essobi`	`ahh.`
`05:45.42`	`harryvetch`	`x100p has has alot of problems in the past even when thay worked on the system?`
`05:45.51`	`ManxPower`	`debug mode will tell you a LOT.`
`05:45.57`	`Essobi`	`WINKCALLERIDDNIS(WINK`
`05:45.58`	`harryvetch`	`for me/`
`05:45.59`	`harryvetch`	`?`
`05:46.07`	`ManxPower`	`harryvetch, Can you just get the card replaced.`
`05:46.08`	`Beirdo`	`bjohnson: you can always try calling him tomorrow to ask`
`05:46.09`	`Essobi`	`Cool..`
`05:46.34`	`ManxPower`	`The NICE thing is that there will be NO changes on the T-1 or the channel bank or the PBX.`
`05:47.00`	`ManxPower`	`So if we fail we just plug the T-1s directly into their channel banks and everything is back to the way it was.`
`05:47.03`	`harryvetch`	`manx, yea might have to. Devkit to me seems a little to high in price considering that I could take the other hundred and place it to a ipone.`
`05:47.22`	`ManxPower`	`harryvetch, Is it a real X100P or a clone?`
`05:47.58`	`harryvetch`	`it says tigerjet`
`05:48.08`	`harryvetch`	`let me get you the lspci -v specs`
`05:48.12`	`ManxPower`	`Does it have a heat sink?`
`05:48.16`	`shido6`	`heheh`
`05:48.38`	`harryvetch`	`yes`
`05:48.52`	`ManxPower`	`harryvetch, I have no idea how to tell the X100P from a clone from lspci. You bought it you should know what it is. Did it cost about US$100?`
`05:48.54`	`harryvetch`	`its been working for a week`
`05:48.56`	`harryvetch`	`or more`
`05:49.07`	`harryvetch`	`nope`
`05:49.07`	`harryvetch`	`:)`
`05:49.15`	`ManxPower`	`Then it's a clone.`
`05:49.15`	`*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)`
`05:49.22`	`harryvetch`	`it said it was a digium in the packaging and was used though ebay.`
`05:49.29`	`shido6`	`ebay`
`05:49.30`	`ManxPower`	`Ah! Was used.`
`05:49.39`	`harryvetch`	`:)`
`05:49.51`	`harryvetch`	`but its worked fine`
`05:49.55`	`harryvetch`	`untill last night.`
`05:50.18`	`ManxPower`	`The stigma of being a cheap asshole and being a traitor to the cause if you buy a clone X100P is much less these days.`
`05:50.24`	`harryvetch`	`modprobe wcfxo does not load the wcfxo.o module`
`05:50.41`	`Essobi`	`ManxPower I remember that.. that was ugly.`
`05:50.50`	`ManxPower`	`Essobi, And partially true.`
`05:51.04`	`ManxPower`	`I certinally would not usually help a person with a clone card.`
`05:51.09`	`Essobi`	`harryvetch rebooted?`
`05:51.15`	`Essobi`	`checked the card seating?`
`05:51.19`	`harryvetch`	`manx look at it this way.. its my choice and it has expanded my`
`05:51.22`	`harryvetch`	`hold on.`
`05:51.42`	`ManxPower`	`harryvetch, Well since Digium stopped selling the X100P, I consider it a non-issue.`
`05:51.43`	`Essobi`	`expanded his hold on what? ;)`
`05:52.00`	`Essobi`	`They did?`
`05:52.05`	`ManxPower`	`Yup!`
`05:52.12`	`Essobi`	`What're they pushing for FXOs now?`
`05:52.17`	`ManxPower`	`Apparently Intel discontinued the chip.`
`05:52.19`	`Essobi`	`and FXSs?`
`05:52.23`	`Essobi`	`Ahhhh wow.`
`05:52.29`	`harryvetch`	`Manx was going to say ever by anything from wal-mark k-mart and was made in china? :)`
`05:52.47`	`ManxPower`	`So there's lots of those cards out there, but when they are gone, they are gone.`
`05:52.50`	`Essobi`	`walmart is chinas #1 exporter.`
`05:53.00`	`harryvetch`	`If i had the extra cash at that time laying around would have considered the devkit but then did not at that time.`
`05:53.08`	`Essobi`	`ManxPower Damn.. I guess I should pickup a few`
`05:53.11`	`Essobi`	`Mm..`
`05:53.15`	`Essobi`	`clones worth a damn?`
`05:53.27`	`ManxPower`	`I don't know if the rumor is TRUE, but that is what someone told me somewhere.`
`05:53.37`	`Essobi`	`right right`
`05:53.41`	`ManxPower`	`Essobi, At $9 each, but 10 clones.`
`05:54.02`	`Essobi`	`was that meant to be buy?`
`05:54.04`	`Essobi`	`Heh.`
`05:54.08`	`ManxPower`	`Yes.`
`05:54.11`	`harryvetch`	`I think there is a slight difference between the chips on the x100p's clone and digiums.`
`05:54.18`	`ManxPower`	`<- is a little intoxicated.`
`05:54.21`	`Sauron	`Dum di dum.`
`05:54.22`	`shido6`	`GOOD!`
`05:54.24`	`shido6`	`its about time`
`05:54.28`	`Essobi`	`Like what?`
`05:54.33`	`Sauron	`Manx my buddy`
`05:54.38`	`ManxPower`	`hardwire, I suspect the CHIP is the same, it's just some of the supporting stuff that may be different.`
`05:55.01`	`Essobi`	`right right.. probably some of the gain accomodating stuff is different.`
`05:55.04`	`Essobi`	`Mmm.`
`05:55.10`	`*** join/#asterisk wasim (~wasim@203.81.200.8)`
`05:55.18`	`harryvetch`	`but its a great card for dialing as a zap its clear and people on the other end hardly hear any echo`
`05:55.24`	`Sauron	`Hum. Digium doesn't sell the x100p anymore?`
`05:55.26`	`Sauron	`bummer.`
`05:55.28`	`harryvetch`	`no`
`05:55.30`	`Essobi`	`Okay.. Welp.. my phone is regestering.. I'm going to bed after I lookup some stuff on call recording. Night all.`
`05:55.37`	`ManxPower`	`Sauron, Go to the Digium hardware page and see for your self.
`05:55.42`	`Sauron	`I'm guessing they want us to buy the 4-daughtercard card`
`05:55.48`	`ManxPower`	`They may still be SELLING them, but they are not on the web site anymore.`
`05:56.11`	`ManxPower`	`Sauron, Yes. It's a small amount more than the X100P was, but is far more expandable.
`05:56.48`	`harryvetch`	`btw, what cards support existing digital multiline phones?`
`05:56.56`	`Essobi`	`Anyone used that new t.38 devicethat's 90 bucks?`
`05:56.59`	`ManxPower`	`harryvetch, none whatsoever.`
`05:57.23`	`Sauron	`t.38 is evil`
`05:57.28`	`Essobi`	`tehe`
`05:57.34`	`Essobi`	`FOID is evil.`
`05:57.35`	`harryvetch`	`Oh so then the commercial digital pbx needs to interface with asterisk box then.`
`05:57.38`	`Essobi`	`FOIP even.`
`05:57.42`	`Sauron	`I almost sold a co-worker on the $8k bounty for a * channel driver for t.38`
`05:57.44`	`harryvetch`	`if tha want to keep those older phones.`
`05:57.46`	`Essobi`	`T.38 is stomachable.`
`05:57.54`	`Sauron	`But then he couldn't find the spec documentations, and I think he gave up.`
`05:58.08`	`ManxPower`	`harryvetch, You usually either toss out the existing PBX or put Asterisk between the PBX and the telco,.`
`05:58.11`	`Essobi`	`THERES AN 8K bounty for t.38?!`
`05:58.19`	`Sauron	`Essobi: $8250`
`05:58.23`	`Essobi`	`OFFS!`
`05:58.39`	`Essobi`	`Time to crack some books and the spandsp code.`
`05:58.45`	`WilliamK`	`guess someone wants T.38 done really fast`
`05:58.45`	`WilliamK`	`=)`
`05:58.50`	`Essobi`	`FXS interface to T38?`
`05:58.53`	`Sauron	`You shouldn't need the spandsp code`
`05:58.56`	`Essobi`	`WTF is he thinking?`
`05:58.58`	`ManxPower`	`Essobi, You really need to understand DSP and modem stuff to do T.38, and there are not many people out there with those skills and invlved with Asterisk.`
`05:59.04`	`harryvetch`	`or just add on a asterisk if the company is expanding and needs more phones. Dont know if thats possible.`
`05:59.10`	`Sauron	`As far as I understand`
`05:59.22`	`Sauron	`well, you'd feed it into the txfax stuff.. ;)`
`05:59.23`	`JerJer`	`who has time to think in this day and age`
`05:59.28`	`Essobi`	`Why isn't he just doing MMFOIP?`
`05:59.35`	`ManxPower`	`You'll notice that Digium has not found someone to pay them enough to put T.38 into Asteirsk. That's what happened with ADSI.`
`06:00.00`	`Essobi`	`I mean really.. Hell, I coded some MMFOIP in * ages ago.`
`06:00.04`	`Essobi`	`ManxPower did too.`
`06:00.17`	`Sauron	`I've done FoIP, but through a proprietary method.`
`06:00.22`	`Sauron	`and store+forward, not realtime`
`06:00.30`	`Essobi`	`Well yea.`
`06:00.36`	`Sauron	`fake t.37 is way easier to implement than t.38 :)`
`06:00.43`	`ManxPower`	`Essobi, You mean my perl scripts to do e-mail -> fax and fax <- email?`
`06:00.47`	`Essobi`	`;)`
`06:00.50`	`Essobi`	`Yea.`
`06:00.58`	`Essobi`	`I expanded on those quite a bit.`
`06:01.04`	`Essobi`	`I think I'll be turning that up this spring`
`06:01.07`	`Sauron	`Manx: I have some perl scripts like that, but I wrote them myself... :)`
`06:01.10`	`Essobi`	`in production.`
`06:01.19`	`ManxPower`	`Essobi, nod Once CVS-HEAD becomes a full release I'll revisit those scripts.`
`06:01.30`	`ManxPower`	`rxfax and txfax would not work for us.`
`06:01.32`	`Essobi`	`Why CVS-HEAD?`
`06:01.35`	`Essobi`	`ahh`
`06:01.52`	`ManxPower`	`Essobi, The new txfax and rxfax are for CVS-HEAD.`
`06:01.57`	`Essobi`	`you brokey brokey?`
`06:02.01`	`Essobi`	`right right.`
`06:02.10`	`Essobi`	`I havn't looked at his sight in a while`
`06:02.11`	`Essobi`	`site`
`06:02.12`	`Essobi`	`ok`
`06:02.23`	`Essobi`	`I'm going to reaadup on call recording in the dial plan and head to bed`
`06:02.27`	`Essobi`	`yall have a good one`
`06:02.48`	`ManxPower`	`I suspect I could "backport" it to 1.0.x, but honestly, why bother. I have plenty to keep me busy while I wait for the feature to mature.`
`06:02.56`	`Essobi`	`;)`
`06:02.57`	`Sauron	`Hum`
`06:03.02`	`Sauron	`there's a /dev/$MONTHNAME?`
`06:03.05`	`Sauron	`That's funny.`
`06:03.12`	`Essobi`	`I wonder if I can get spandsp to build on FreeBSD`
`06:03.17`	`Essobi`	`that would roxors`
`06:03.29`	`Essobi`	`I should give it a whirls`
`06:04.15`	`Sauron	`I hate that ast_data can't do includes. Grrr.`
`06:04.46`	`*** join/#asterisk thepdakid (~vtrandal@c-24-8-106-135.client.comcast.net)`
`06:05.07`	`Sauron	`What's the point of having your dialplan manageable through a SQL server, when you can't configure the precedence of it.`
`06:06.15`	`WildPikachu`	`hrmmm, I was looking at the sangoma products`
`06:06.38`	`WildPikachu`	`they say they work with asterisk, but how can an unchanellized E1/T1 card work with 24/32 channels?`
`06:06.45`	`WildPikachu`	`doesn't it have to be channelized`
`06:06.57`	`Damin`	`I think the Sangoma cards are an interesting design.`
`06:07.18`	`thepdakid`	`Sauron: good point. I wouldn't call that manageable.
`06:07.31`	`Damin`	`WildPikachu: Because the "channelization" of the T1 stream is handled by libpri..`
`06:07.39`	`WildPikachu`	`aha`
`06:07.53`	`WildPikachu`	`so u can use a unchannlized card, and use libpri to channelize it?`
`06:07.56`	`Damin`	`WildPikachu: One of the benefits of having your PRI stack in Software! ;)`
`06:08.09`	`thepdakid`	`cool`
`06:08.14`	`Damin`	`WildPikachu: I suspect... don't quote me on it though..`
`06:08.47`	`WildPikachu`	`well... why would www.voipsupply.com advertise it?`
`06:08.47`	`WildPikachu`	`heh`
`06:09.59`	`wasim`	`WildPikachu: we're testing the sangoma's here, pretty good so far`
`06:10.19`	`WildPikachu`	`unchannelized?`
`06:10.23`	`WildPikachu`	`the card`
`06:10.46`	`wasim`	`WildPikachu: i really want someone to give me ISUP on it though`
`06:10.46`	`Damin`	`WildPikachu: Dude.. a T1 is a T1. ;) Channelization happens later on.. :)`
`06:11.03`	`WildPikachu`	`:)`
`06:11.06`	`WildPikachu`	`ISUP?`
`06:11.57`	`Luke-Jr`	`Has anyone ever run PPP over VoIP?`
`06:12.33`	`*** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net)`
`06:12.44`	`Mavvie`	`Luke-Jr: there was a posting about a pppd extensions for zap-interfaces a not so long time ago.`
`06:12.50`	`Sauron	`Juke: Huh?`
`06:12.53`	`Sauron	`why would you want that`
`06:12.59`	`wasim`	`WildPikachu: a necessary component for ss7`
`06:13.07`	`wasim`	`Luke-Jr: htf would you do that?`
`06:13.12`	`brc_`	`wasim`
`06:13.13`	`brc_`	`whatup`
`06:13.16`	`Luke-Jr`	`softmodem?`
`06:13.22`	`brc_`	`wasim, did the phones make it?`
`06:13.29`	`wasim`	`brc_: yep, 30 of them`
`06:13.35`	`wasim`	`brc_: 10 more going out this weekend`
`06:14.07`	`brc_`	`ahh`
`06:14.10`	`Sauron	`Manx, what's ADSI used for?`
`06:15.26`	`WildPikachu`	`wasim, u don't get any calls without ss7 hey?`
`06:15.58`	`wasim`	`WildPikachu: of course we do, but we need ss7 to do the interconnects here`
`06:16.06`	`WildPikachu`	`:(`
`06:18.58`	`Sauron	`JerJer, you around?`
`06:23.12`	`harryvetch`	`Wild, I took a quarter of telecom and basicly ss7 is to save bandwith on a network? I recall in the old days when a desitnation ss7 sends a ring code and your co is ringing your phone but in reality there is no link as of yet between you and the desitnation phone untill thay pick it up right?`
`06:24.10`	`Strom_C`	`its for more than just saving bandwidth`
`06:24.13`	`harryvetch`	`It saves bandwith on the phone network if there other end does not pickup`
`06:24.22`	`harryvetch`	`also somethign about tollcharges`
`06:24.27`	`Strom_C`	`you can do so much more with the network with ss7 than with MF`
`06:24.42`	`Strom_C`	`it's more secure and you have far more flexibility in your signaling`
`06:24.51`	`harryvetch`	`but ss7 is a network outside of a network.`
`06:24.56`	`Strom_C`	`no`
`06:25.15`	`Strom_C`	`the voice and ss7 are interconnected at key points; they run parallel and one needs the other`
`06:25.26`	`harryvetch`	`sure`
`06:25.44`	`Qwell`	`Stupid question, whats an ss7?`
`06:25.46`	`WildPikachu`	`hrmmmm`
`06:26.01`	`harryvetch`	`one is signaling on/off hook and what ever else and then there is the voice channel`
`06:26.15`	`Strom_C`	`qwel: common channel interoffice signaling system no. seven; the out-of-band signaling network that the PSTN uses`
`06:26.17`	`Sauron	`It's a protocol for telecommunication.`
`06:26.24`	`Sauron	`What Strom_C said. :)`
`06:26.24`	`Qwell`	`ahh`
`06:26.38`	`WildPikachu`	`the digium TDM card....`
`06:26.45`	`WildPikachu`	`is it modulized?`
`06:26.56`	`Strom_C`	`what do you mean "modulized"?`
`06:26.59`	`WildPikachu`	`i see somone advertising FXO and FXS modules`
`06:27.04`	`Strom_C`	`yes`
`06:27.07`	`WildPikachu`	`for like $100`
`06:27.07`	`harryvetch`	`yes`
`06:27.08`	`Qwell`	`it can have up to 4 "modules"`
`06:27.24`	`WildPikachu`	`ok.... what is the model number or part number of the card with no modules on?`
`06:27.46`	`Strom_C`	`you cant buy just the card IIRC`
`06:27.58`	`WildPikachu`	`aha`
`06:28.21`	`Sauron	`You can`
`06:28.26`	`Sauron	`it's on digium's sitd`
`06:28.27`	`Sauron	`site`
`06:28.29`	`Sauron	`or, was.`
`06:28.50`	`WildPikachu`	`cool... so if you buy a TDM* card, it will have X modules on it, of which you can chop and change as needed?`
`06:29.17`	`ManxPower`	`The TDM400P and TDMxxB products are on Digium's store, which can be found in the Hardware section of the Digium web site.`
`06:29.35`	`Qwell`	`WildPikachu: http://www.digium.com/index.php?menu=wildcard_tdm400p2`
`06:30.06`	`ManxPower`	`WilliamK, Most people find a PRI just fine. Not a lot of people need SS7 support.`
`06:30.16`	`*** join/#asterisk garyitcom (~Tech@119-114.8-67.tampabay.rr.com)`
`06:30.31`	`WildPikachu`	`cool!`
`06:30.49`	`Qwell`	`WildPikachu: They also sell the modules seperately. http://www.digium.com/index.php?menu=hardware_products`
`06:31.16`	`WildPikachu`	`nod`
`06:31.22`	`WildPikachu`	`pretty cool`
`06:31.25`	`WilliamK`	`only major difference is basically shifting the signalling from the D channel to a dedicated DS1 which really doesn't pay off unless you have a huge number of PRIs...correct?`
`06:32.10`	`ManxPower`	`WilliamK, If you have an SS7 connection you can pretty much bill your calls to any telephone number you want to, you can, in theory really screw up other telcos, etc.`
`06:32.33`	`ManxPower`	`WilliamK, Remember the stories of the telephone hackers of the 70's and 80's?`
`06:32.54`	`ManxPower`	`Well an SS7 connection would be their dream.`
`06:32.55`	`WilliamK`	`yeah faintly`
`06:33.13`	`postel`	`OMG!! whats next? Whistlin 2600 tones?`
`06:33.27`	`postel`	`</flamebait>`
`06:34.29`	`bjohnson`	`for canucks: couple of spa 2000 units on ebay from BC for $125 + $15 shipping (CDN) .. good price`
`06:34.37`	`harryvetch`	`fone phreeks. Kevin mitnic was very involved in that.`
`06:35.21`	`*** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au)`
`06:37.18`	`harryvetch`	`There is kevin mitnicks call sign and address :)`
`06:37.23`	`harryvetch`	`http://www.qrz.com/callsign/N6NHG`
`06:37.53`	`Qwell`	`hmm, his ban finally got lifted, didn't it? heh`
`06:39.01`	`*** join/#asterisk pranav (~dawda_pra@202.63.174.250)`
`06:39.23`	`*** part/#asterisk pranav (~dawda_pra@202.63.174.250)`
`06:41.50`	`*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)`
`06:42.22`	`terrapen`	`holy shit, North Korea just admitted to having nuclear weapons.`
`06:42.43`	`WilliamK`	`for some reason that doesn't surprise me`
`06:44.14`	`wasim`	`we didn't give 'em to 'em`
`06:45.13`	`postel`	`hardwire: N6 is a novice class, aint even technician level, 2m tops`
`06:45.42`	`terrapen`	`im glad im not in Seoul`
`06:49.25`	`Frawg`	`terrapen: url?`
`06:49.30`	`terrapen`	`foxnews.com`
`06:51.42`	`terrapen`	`http://www.foxnews.com/story/0,2933,146950,00.html`
`06:52.03`	`Frawg`	`yeh got it`
`06:52.26`	`terrapen`	`http://www.foxnews.com/story/0,2933,146797,00.html`
`06:52.27`	`terrapen`	`holy cow`
`06:52.30`	`terrapen`	`that teacher is hot`
`06:52.36`	`terrapen`	`and she had to rape a 13 year old!`
`06:52.48`	`*** join/#asterisk [ro]nic3try (~nic3try@p3.pub.ro)`
`06:53.40`	`[ro]nic3try`	`can anyone tell me if asterisk can be configured as a sip server ?`
`06:54.55`	`terrapen`	`sure`
`06:55.06`	`terrapen`	`sip.conf`
`06:57.31`	`[ro]nic3try`	`but why doesn't my phone register ?`
`06:57.53`	`[ro]nic3try`	`it sais that " REGISTER not implemented "`
`06:59.32`	`terrapen`	`did you read the wiki?`
`06:59.49`	`[ro]nic3try`	`no :(`
`06:59.52`	`datareactor`	`have you made user in sip.conf`
`07:00.00`	`[ro]nic3try`	`yes`
`07:00.44`	`[ro]nic3try`	`but i don't have a sip server to connect..`
`07:01.01`	`[ro]nic3try`	`i'm using just asterisk`
`07:01.09`	`dwC-`	`asterisk is a sip server ;p`
`07:01.39`	`dwC-`	`read this [ro]nic3try http://www.voip-info.org/wiki-Asterisk+config+sip.conf`
`07:03.15`	`Firestrm`	`ManxPower,bjohnson,(other knowlegable great one :) what do you know about voicemail? Is it possible to have multiple versions of the voices in different languages?`
`07:04.39`	`Firestrm`	`i have a english/french menuing system, but when they get kicked into VM its english..`
`07:04.42`	`letherglov`	`terrapen, you talking about the one in florida?`
`07:05.16`	`letherglov`	`whoa whoa`
`07:05.20`	`letherglov`	`another one`
`07:05.26`	`wasim`	`Firestrm: voicemail.conf`
`07:05.31`	`letherglov`	`jesus. i managed to get through grade school without getting laid by my teacher`
`07:05.39`	`letherglov`	`thank god too, they were hideous`
`07:06.14`	`terrapen`	`no, one in tennessee`
`07:06.29`	`Firestrm`	`wasim, ive looked through voicemail.conf.. maybe i missed it..`
`07:07.12`	`wasim`	`Firestrm: /lang will get you there`
`07:08.27`	`Luke-Jr`	`sick`
`07:15.26`	`[ro]nic3try`	`has anyone used CSPS from cisco ??`
`07:17.56`	`terrapen`	`well, i was PLANNING on writing an AGI for my weather station`
`07:18.01`	`terrapen`	`damned Urban Terror`
`07:18.11`	`terrapen`	`wasted all my time tonight on that FPS video game`
`07:18.20`	`terrapen`	`aand now its 0118`
`07:18.22`	`terrapen`	`and im tired`
`07:19.27`	`Luke-Jr`	`terrapen: Quake 1 is fun`
`07:19.48`	`Nethab`	`what's the trick to getting fwd split into a seperate user and peer instead of one friend`
`07:20.02`	`Luke-Jr`	`Nethab: why?`
`07:20.23`	`Nethab`	`cause i want to try seperating my inbound and outbound termination providers`
`07:20.39`	`Nethab`	`and i too now believe that friend == bad`
`07:21.00`	`terrapen`	`http://ruk.ca/article/2139`
`07:21.02`	`Firestrm`	`wasim, found it on the wikki buried under 300 other pages... its allways in the last place you look :P`
`07:21.02`	`terrapen`	`that's broken`
`07:21.08`	`terrapen`	`does that work for anyone else?`
`07:22.05`	`Luke-Jr`	`Nethab: if friend works, it doesn't hurt to use it, AFAICT`
`07:22.11`	`Firestrm`	`terrapen, worked for me`
`07:22.24`	`*** join/#asterisk unixgeek (~unixgeek@67.109.59.67.ptr.us.xo.net)`
`07:22.46`	`Nethab`	`the problem is that insecure = very is a bad bad hack`
`07:23.30`	`Nethab`	`terrapen you want us to call the number on that page`
`07:24.28`	`terrapen`	`well, it doesn't work`
`07:24.37`	`terrapen`	`the ISBN thing worked?`
`07:24.46`	`terrapen`	`when i press 5, it disconnects`
`07:24.49`	`Luke-Jr`	`Nethab: I don't need insecure=very for FWD...`
`07:24.53`	`terrapen`	`i can hear the weather`
`07:25.36`	`Firestrm`	`gnite all.. need sleep`
`07:25.36`	`Nethab`	`fwd was just an example, other providers tell you to use insecure = very`
`07:25.36`	`terrapen`	`night.`
`07:25.37`	`datareactor`	`can i use emta with asterisk`
`07:26.16`	`unixgeek`	`Does anyone have any experience configuring D-link's DVG-1402S/L?`
`07:27.04`	`unixgeek`	`I have logged into the advance tab and then proceeded to set the SIP settings to use my Asterisk server.`
`07:28.08`	`unixgeek`	`The only problem I am having is that the box will not send any SIP registration packets to Asterisk.`
`07:28.40`	`Mavvie`	`if that is all!`
`07:28.53`	`unixgeek`	`Sounds like this is a fairly common problem with the DVG-1402 box. Anyone know how to force it to use the new settings? Thanks.`
`07:32.48`	`Mw3`	`has anyone managed to setup a fax server which sends the incoming faxes to the recipients email in pdf and somehow it can be used as a printer to send faxes with it ?`
`07:33.00`	`Mw3`	`with isdn`
`07:34.03`	`Nethab`	`Would you like fried with that?`
`07:34.08`	`Nethab`	`fries?`
`07:34.27`	`Mw3`	`that would be nice, but not necessary :)`
`07:35.40`	`Nethab`	`if the fax server can output to a printer, it can be caught in the postscript stage, and converted to pdf`
`07:36.30`	`datareactor`	`is there any packetcable guru around ?`
`07:36.36`	`Mw3`	`yeah, the main problem is that i don't know which fax server to use with isdn`
`07:37.21`	`Nethab`	`i assume one that works with asterisk?`
`07:37.24`	`*** join/#asterisk orlon_ (~orlon@office-fw.iexec.net.au)`
`07:43.17`	`*** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk)`
`07:46.09`	`*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)`
`07:47.48`	`harryvv`	`nethab what kind of system you have`
`07:48.07`	`terrapen`	`ok, in a few minutes, i will have my asterisk weather station up for testing`
`07:48.14`	`terrapen`	`the weather station resides on my roof`
`07:48.22`	`Nethab`	`Mw3: have you read http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email`
`07:48.29`	`Mw3`	`not yet`
`07:48.30`	`terrapen`	`it has a wireless link to a little console thing on my desak`
`07:48.40`	`terrapen`	`and im pulling data from that console over the serial port`
`07:48.46`	`terrapen`	`and storing it in a Perl storable`
`07:48.54`	`terrapen`	`and my AGI will read that storable and report the weather`
`07:49.28`	`terrapen`	`(up-to-the-minute weather)`
`07:49.33`	`terrapen`	`updated every 5 secs`
`07:49.55`	`terrapen`	`i need to record some soundfiles`
`07:50.28`	`terrapen`	`north west south east, "wind is out of the", etc`
`07:51.16`	`godsmoke`	`hrm`
`07:51.29`	`godsmoke`	`is there a reason you don't use the public weather stations?`
`07:53.26`	`implicit`	`cause he wants to know HIS weather`
`07:53.29`	`implicit`	`not somebody elses`
`07:53.30`	`implicit`	`:)`
`07:54.01`	`Nethab`	`he has a weather sensor hooked up to his * box`
`07:54.04`	`terrapen`	`its the weather on my roof.`
`07:54.06`	`terrapen`	`right here.`
`07:54.07`	`terrapen`	`right now.`
`07:54.13`	`terrapen`	`we sell these stations :)`
`07:54.30`	`terrapen`	`http://weathertools.com/components/catalog/product.html?pid=1327&cat=276`
`07:54.33`	`terrapen`	`that's the station`
`07:57.13`	`terrapen`	`public stations update only ever 10-15min at best...`
`07:57.21`	`terrapen`	`this updates every 5s`
`07:57.55`	`coppice`	`should be perfect for the british climate :-)`
`07:58.25`	`JerJer`	`terrapen: up to the 5 second weather :)`
`07:58.55`	`terrapen`	`exactly :)`
`07:59.06`	`terrapen`	`i think i need to get another DID for this :P`
`07:59.26`	`terrapen`	`(210) 226-3232 has been around FOREVER`
`07:59.32`	`JerJer`	`light one up`
`07:59.39`	`terrapen`	`you could call that thing to get the current temp back in the 1970s`
`07:59.41`	`JerJer`	`how about a toll-free? :P`
`07:59.43`	`terrapen`	`i still wonder how they do it`
`07:59.55`	`terrapen`	`jerjer, already have one from ya :)`
`07:59.58`	`terrapen`	`gonna get another :P`
`08:00.09`	`JerJer`	`rock on`
`08:00.28`	`terrapen`	`hang tight, i'll have this AGI up in a couple of minutes.`
`08:01.19`	`terrapen`	`its going to sound really wierd because the voices are different`
`08:01.27`	`terrapen`	`im using SayNumber(), coupled with my own voice`
`08:02.55`	`JerJer`	`lol`
`08:03.03`	`JerJer`	`check asterisk-sounds`
`08:03.21`	`JerJer`	`i do recall seeing weather prompts`
`08:03.26`	`terrapen`	`im such a * noob...how do i have * listen for dialed DTMF and direct the caller to that extension`
`08:03.44`	`JerJer`	`Background ?`
`08:03.49`	`terrapen`	`ahh...`
`08:04.20`	`*** join/#asterisk fantomax1 (~fanto@81.208.114.250)`
`08:06.28`	`terrapen`	`ok this is SUPER cheezy`
`08:06.34`	`terrapen`	`i gotta do the sounds next`
`08:06.40`	`terrapen`	`1-866-817-7667`
`08:06.43`	`terrapen`	`dial '7'`
`08:06.53`	`terrapen`	`thats the outside temp`
`08:06.56`	`terrapen`	`followed by the wind speed`
`08:07.01`	`terrapen`	`followed by outside humidity`
`08:07.50`	`terrapen`	`someone want to test? :)`
`08:07.52`	`*** join/#asterisk denon (denon@synapse.subneural.net)`
`08:07.52`	`*** mode/#asterisk [+o denon] by ChanServ`
`08:13.33`	`*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)`
`08:13.46`	`*** join/#asterisk Joertzer (fwuser@213.61.141.198)`
`08:14.05`	`*** part/#asterisk orlon_ (~orlon@office-fw.iexec.net.au)`
`08:14.49`	`fantomax1`	`hi all`
`08:14.57`	`Joertzer`	`hi`
`08:15.03`	`fantomax1`	`hi`
`08:15.59`	`terrapen`	`HEY!`
`08:16.02`	`terrapen`	`whoever you are!`
`08:16.04`	`terrapen`	`push 7!`
`08:16.08`	`BoRiS`	`unF`
`08:16.17`	`JerJer`	`45`
`08:16.19`	`JerJer`	`5`
`08:16.22`	`JerJer`	`64`
`08:16.41`	`*** join/#asterisk techie (gus@asterisk.horizonte.us)`
`08:16.50`	`JerJer`	`a script:`
`08:17.21`	`terrapen`	`yeah, i know, its super cheezy right now`
`08:17.24`	`terrapen`	`im recording the sounds`
`08:18.22`	`JerJer`	`Conditions observed at <insert time> are partly cloudy, tempurature 45 degrees, dew point 41, winds out of the west at 5`
`08:18.45`	`JerJer`	`departing traffic use runway 28`
`08:18.50`	`JerJer`	`you have information Charley`
`08:18.50`	`JerJer`	`:)`
`08:18.58`	`terrapen`	`hahahah`
`08:20.39`	`JerJer`	`an ebay seller with their own custom IVR system AND customer support staff... killer`
`08:20.45`	`JerJer`	`:)`
`08:21.22`	`terrapen`	`:)`
`08:21.56`	`*** join/#asterisk riksta (~rick@81-178-238-229.dsl.pipex.com)`
`08:21.59`	`JerJer`	`now only if their provider had customer support staff`
`08:22.15`	`terrapen`	`:)`
`08:22.20`	`terrapen`	`i'd love to help you guys with the website`
`08:22.28`	`terrapen`	`i did bikeworld.com/weathertools.com/gpstools.com`
`08:22.32`	`terrapen`	`all the backend programming, too`
`08:23.00`	`*** join/#asterisk Vico (vico@c-5d6470d5.1011-1-64736c12.cust.bredbandsbolaget.se)`
`08:23.25`	`harryvetch`	`terrapen, some time when I have time will tell you what gps was like before it was a civilian product back in the 1980s`
`08:23.41`	`terrapen`	`are you military?`
`08:23.47`	`harryvetch`	`It was on our specialOps helicopters back then. In the mean time im off to bed.`
`08:24.13`	`terrapen`	`dang.`
`08:26.43`	`coppice`	`GPS was a serious chunk of hardware in the 80s`
`08:29.13`	`terrapen`	`ok, got my weather sounds recorded`
`08:29.16`	`terrapen`	`adding them to the AGI`
`08:29.24`	`*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)`
`08:30.56`	`coppice`	`weather sounds? let me guess what sound you use to represent the wind`
`08:31.13`	`Zeeek`	`Morning`
`08:34.05`	`terrapen`	`1-877-817-7667 x7`
`08:34.12`	`terrapen`	`ok, gotta code the wind direction`
`08:34.45`	`terrapen`	`if (($wind_dir > 0) && ($wind_dir < 30)) { ...`
`08:39.46`	`Zeeek`	`wind? did you say wind?`
`08:41.57`	`*** join/#asterisk Pulu (~chatzilla@64.200.224.158)`
`08:44.30`	`*** join/#asterisk abbas_ (nidobas@203.81.198.6)`
`08:52.26`	`*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)`
`08:52.40`	`Zeeek`	`damned`
`08:53.36`	`terrapen`	`ok try now:`
`08:53.40`	`terrapen`	`1-866-817-7667`
`08:53.42`	`terrapen`	`dial 7 on connect`
`08:56.22`	`Zeeek`	`.`
`08:58.05`	`*** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it)`
`08:58.17`	`maruz`	`i get this error: res_musiconhold.c:506 monmp3thread: Only wrote -1 of 1600 bytes to pipe`
`08:58.33`	`maruz`	`someone does know anything about?`
`08:58.52`	`Zeeek`	`never saw that`
`08:59.07`	`*** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it)`
`08:59.30`	`markit`	`anyone using this 4-port ATA? do they work with asterisk? http://ipphone.eezeephone.com/index_files/Page700.htm`
`09:00.02`	`terrapen`	`hahah`
`09:00.07`	`terrapen`	`my weather station has its own DID`
`09:00.14`	`terrapen`	`1-866-869-7359`
`09:01.33`	`terrapen`	`oops`
`09:01.36`	`terrapen`	`1-866-859-7359`
`09:02.59`	`Zeeek`	`.`
`09:04.31`	`Nethab`	`you bought another DID just for your weather station extensions`
`09:04.37`	`*** join/#asterisk Pressy (~Pressy@pD9E17C19.dip.t-dialin.net)`
`09:04.42`	`terrapen`	`yep`
`09:04.48`	`terrapen`	`its basically free with nufone`
`09:04.52`	`terrapen`	`i just pay per min`
`09:04.56`	`terrapen`	`(I HOPE!!!)`
`09:05.04`	`Zeeek`	`read the fine print!`
`09:05.10`	`*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)`
`09:05.20`	`terrapen`	`ruh roh!`
`09:05.35`	`terrapen`	`well, they haven't nailed me yet`
`09:05.55`	`Nethab`	`i wish my girlfriend could see this, then she'd know i'm not that much of a freak`
`09:06.20`	`Nethab`	`of course I'd have get a girlfriend first...`
`09:07.53`	`Zeeek`	`I doubt you'll find one here!`
`09:08.18`	`Zeeek`	`terrapin I think they're free, I have a few too`
`09:08.25`	`Zeeek`	`pen, sorry`
`09:08.40`	`markit`	`wget ftp://ftp.digium.com/pub/girlfriends/blond`
`09:08.40`	`*** join/#asterisk peeebeee (peeebeee@aylesbury.salmark.net)`
`09:08.55`	`markit`	`mmm yes, does not work for girlfriends.. :(`
`09:09.05`	`terrapen`	`my girlfriend does not understand`
`09:09.07`	`terrapen`	`nor care`
`09:09.14`	`terrapen`	`to her, this is just time spent away from her`
`09:09.18`	`terrapen`	`attention that she didn't get`
`09:09.24`	`terrapen`	`so im always in trouble :)`
`09:09.26`	`Nethab`	`at least they can't get past the "what's a DID stage"`
`09:11.01`	`Zeeek`	`I can't get past the what's a DID stage either. WHat does it stand for?`
`09:11.10`	`Zeeek`	`direct in debt?`
`09:11.39`	`terrapen`	`this guy has the wife-friendly * setup:`
`09:11.39`	`terrapen`	`http://scottstuff.net/scott/archives/cat_asterisk.html`
`09:11.39`	`terrapen`	`i got a lot of my ideas from him`
`09:11.40`	`Nethab`	`direct inward dial`
`09:11.55`	`Zeeek`	`Ray Bradbury once commented in a book that women do not care about watches that show time to the 100th of a second`
`09:12.00`	`Nethab`	`in essence it's a phone number`
`09:12.13`	`Zeeek`	`I know what it is, I just can't rememebr what it stood for`
`09:12.40`	`Nethab`	`oh, i guess i should have expected only hard core people to be up this late`
`09:13.05`	`Zeeek`	`My wife is my business partner so I had to worry about asterisk too. I marketed it by giving pour pbx a name : Phoebe`
`09:13.19`	`Zeeek`	`It happens to to be 10:13 AM here`
`09:14.09`	`modulus_`	`bleh`
`09:14.25`	`Nethab`	`your in germany or italy?`
`09:14.33`	`terrapen`	`i'd love to meet a beautiful girl with tech skills`
`09:14.35`	`terrapen`	`they do exist`
`09:14.38`	`terrapen`	`i'ev worked with a few`
`09:14.54`	`terrapen`	`but they were either a) wierd or b) had boyfriends or c) both`
`09:15.02`	`*** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz)`
`09:15.11`	`modulus_`	`d) or they were 500 lb land monsters`
`09:15.20`	`Nethab`	`there are only two types of tech girls, not ready, and taken`
`09:15.26`	`terrapen`	`well, that does not qualify as beautiful`
`09:15.56`	`terrapen`	`i need to go to bed`
`09:16.09`	`coppice`	`I used to work with a very beautiful girl who was a very capable engineer. she was also a real PITA :-)`
`09:16.16`	`Zeeek`	`Last year, I went ot buuy some capacitors and there was a beautiful one behind the counter`
`09:16.19`	`*** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)`
`09:16.28`	`Zeeek`	`I asked her if she spoke solder`
`09:16.34`	`Nethab`	`there was a beautiful capacitor behind the counter?`
`09:16.35`	`jdv79`	`why would a call be rejected due to usage limit of 1?`
`09:16.39`	`*** part/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it)`
`09:16.42`	`Zeeek`	`girl, girl`
`09:16.44`	`jdv79`	`there are no calls in progress`
`09:16.46`	`*** join/#asterisk kleptog (~kleptog@ip503db2aa.speed.planet.nl)`
`09:17.10`	`jdv79`	`maybe that has something to do with the fact that asterisk, under no load, is owning the CPU`
`09:17.36`	`*** join/#asterisk Delvar (~irc@83.146.53.34)`
`09:17.45`	`Nethab`	`maybe one is the load average`
`09:18.16`	`jdv79`	`maybe, load avg is 1.16`
`09:18.26`	`Nethab`	`if the limit is tied to load average that would do it`
`09:19.02`	`jdv79`	`but then on the console it says: chan_sip.c:1569 update_user_counter: Call from user '2222' rejected due to usage limit of 1`
`09:19.05`	`jdv79`	`so i doubt it`
`09:19.29`	`Nethab`	`maybe there's a stale call in the stack`
`09:19.33`	`Nethab`	`sip show channels`
`09:19.53`	`jdv79`	`it says one active sip chan`
`09:20.10`	`jdv79`	`there are none cause the only 2 phones on the server are right in front of me`
`09:20.32`	`Nethab`	`soft hangup SIP/channel`
`09:21.09`	`jdv79`	`what is the channel part?`
`09:21.18`	`jdv79`	`i need to read`
`09:21.38`	`Nethab`	`use tab it will autcomplete nearly anything`
`09:21.38`	`*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)`
`09:21.44`	`Nethab`	`in the CLI`
`09:22.06`	`Nethab`	`like show channel <tab>`
`09:22.15`	`jdv79`	`didn't work:)`
`09:22.23`	`jdv79`	`now there are 2 stale chans`
`09:22.43`	`kleptog`	`Hoi. Is there a minimum gcc version for Asterisk? I'm compiling chan_zap from CVS with gcc 2.95.4 and it's barfing on illegal C code on line 7943. It was added to CVS in 28 Jan. What version are people using to compile?`
`09:22.45`	`jdv79`	`i'm chalking it up to a bug in HEAD`
`09:25.53`	`jdv79`	`i used the one that came with FC3 and it was fine`
`09:26.04`	`jdv79`	`bunch of warnings but what else is new;)`
`09:27.36`	`kleptog`	`Hmm, that's 3.2 I think.... Oh well...`
`09:27.45`	`jdv79`	`Nethab, are you using v1.0?`
`09:28.08`	`jdv79`	`3.4.2 actually`
`09:28.21`	`jdv79`	`but i checked out HEAD`
`09:29.24`	`*** join/#asterisk vhatz (Antichrist@titan.physics.auth.gr)`
`09:29.25`	`jdv79`	`anyone know what the last relatively stable version was?`
`09:29.43`	`*** join/#asterisk djin (~marius@62.58.40.196)`
`09:29.53`	`vhatz`	`hello, does anyone know if * can accept calls with different signaling and RTP ip addresses now?`
`09:29.57`	`vhatz`	`in SIP?`
`09:30.31`	`kleptog`	`jdv79: I'd love to know...`
`09:31.38`	`jdv79`	`I guess it might be better to ask when most are awake though`
`09:32.15`	`kleptog`	`That would be in the US daytime right?`
`09:32.21`	`jdv79`	`yeah`
`09:32.28`	`jdv79`	`its 4:32 am here now`
`09:32.32`	`jdv79`	`NYC`
`09:33.05`	`jdv79`	`maybe i'll try out 1.0.5`
`09:34.43`	`vhatz`	`Does anyone know if * can accept SIP calls with different signaling and RTP ip addresses now? It seems taht * cannot acept calls from SIP proxies...`
`09:35.37`	`kleptog`	`vhatz: someone on the mailing list said they were using a SIP proxy, dunno how...`
`09:36.16`	`vhatz`	`but until now, the latest Ast. version does not support sip proxies, correct?`
`09:37.49`	`Delvar`	`we have been using SER and jasomi for a while now with no problems, SEr is a sip proxy and doesnt handle the RTP at all.`
`09:38.06`	`jdv79`	`Delvar, what version of ast?`
`09:38.24`	`Delvar`	`quite a few, including stable`
`09:38.44`	`jdv79`	`is there a short list of what stable is missing anywhere?`
`09:39.04`	`Delvar`	`its missing all the good stuff :)`
`09:39.10`	`jdv79`	`i knew it!`
`09:39.31`	`jdv79`	`that sucks cause the current is dying here:(`
`09:39.36`	`kleptog`	`Hmm, do you have to create an account on the bug tracker to submit a bug?`
`09:39.45`	`Delvar`	`yes`
`09:40.06`	`Delvar`	`jdv79: download a backdated CVS then`
`09:40.20`	`kleptog`	`Delvar: when was a stable date?`
`09:42.49`	`Delvar`	`not sure, last CVS update i did was about 2 weeks ago and that worked ok`
`09:43.27`	`*** join/#asterisk CSerpent (~me@62.49.253.91)`
`09:43.42`	`CSerpent`	`morning`
`09:44.04`	`*** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de)`
`09:44.29`	`CSerpent`	`usual lively bunch ;)`
`09:45.11`	`*** join/#asterisk r0d3nt\|m (~RatMan@4.19.77.194)`
`09:45.19`	`*** part/#asterisk r0d3nt\|m (~RatMan@4.19.77.194)`
`09:47.51`	`Delvar`	`morning`
`09:49.43`	`*** join/#asterisk meppl (~mephisto@pD9E69042.dip.t-dialin.net)`
`09:50.03`	`meppl`	`guten morgen`
`09:50.23`	`kleptog`	`Whacky, I try to download a patch from the bugtracker and it tell my browser it's a PHP script :)`
`09:52.58`	`*** join/#asterisk zeedo (~notroot@www.bsrf.org.uk)`
`09:54.18`	`sezuan`	`morning`
`09:54.40`	`modulus_`	`jbot babelfish de en guten morgen`
`09:55.12`	`modulus_`	`jbot babelfish de en guten nachte`
`09:55.18`	`modulus_`	`jbot babelfish de en guten nacht`
`09:55.28`	`modulus_`	`jbot hug me`
`09:55.30`	`jbot`	`ACTION hugs modulus_`
`09:55.54`	`Nethab`	`jbot babelfish es en mono no limpio`
`09:56.05`	`Nethab`	`dirty dirty monkey`
`09:56.33`	`modulus_`	`jbot babelfish es en que hondas muchacho i bienes te miro`
`09:56.43`	`Nethab`	`jbot babelfish es en una boca abierto coge moscas`
`09:57.41`	`datareactor`	`~jbot`
`09:57.42`	`jbot`	`methinks jbot is the shipboard computer, but you may call me eddie if it helps you relax`
`09:57.50`	`Nethab`	`why is it that most sip providers need insecure = very`
`09:57.59`	`MrEntropy`	`anyone here familiar with SER's interaction with radius?`
`09:58.39`	`*** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62)`
`09:59.37`	`CSerpent`	`aha people are alive :D`
`09:59.49`	`modulus_`	`jbot babelfish de en du bist eine dummer schweinehund`
`09:59.58`	`Nethab`	`what are people and what is this 'alive' you speak of`
`10:00.15`	`CSerpent`	`I don't know but I think it's a concept that could appear in 1.0.7`
`10:01.04`	`kleptog`	`Hmm, I love it: You must run Open H.323 v1.15.1 and PWLib v1.8.1. All other versions are not supported. Just great...`
`10:01.21`	`Nethab`	`not supported in what`
`10:01.29`	`kleptog`	`asterisk h323 driver in CVS`
`10:01.45`	`CSerpent`	`Anyone know anything about disabling the SIP 102 command being sent to a provider?`
`10:03.43`	`Nethab`	`notify?`
`10:04.04`	`CSerpent`	`options`
`10:04.22`	`Nethab`	`do you have a mailbox = in your providers channel definition`
`10:04.28`	`CSerpent`	`nope`
`10:04.28`	`Nethab`	`oh options`
`10:04.29`	`Nethab`	`hmm`
`10:04.39`	`CSerpent`	`in fact it's minimal in what I do have`
`10:04.43`	`Nethab`	`it's requesting options, or giving them`
`10:05.05`	`Nethab`	`right after register, or at regular intervals`
`10:05.08`	`CSerpent`	`requesting it seems, in which the provider's coming back with a 404`
`10:05.37`	`CSerpent`	`they haven't elaborated on that one - but my assessment says after register`
`10:06.19`	`Nethab`	`do they use qualify for keepalive with NAT`
`10:06.54`	`Nethab`	`my * never transmits OPTIONS unless i use qualify`
`10:07.07`	`CSerpent`	`aha.. I have got qualify in there`
`10:09.40`	`CSerpent`	`it was some article I'd read that advised it :\|`
`10:09.41`	`*** join/#asterisk shaZwaz (~chatzilla@203.81.196.167)`
`10:09.59`	`CSerpent`	`apparantly we're connecting it to a Sonus softswitch at the provider`
`10:10.13`	`CSerpent`	`but it's all public IPs`
`10:10.25`	`shaZwaz`	`hi room`
`10:10.44`	`CSerpent`	`<room says hello>`
`10:10.46`	`CSerpent`	`:)`
`10:12.03`	`Nethab`	`qualify's are just a way to triple check that the other end is really really still there`
`10:12.26`	`Nethab`	`but not all providers support them`
`10:13.20`	`CSerpent`	`doesn't help the provider said they had * working already.. but actually didn't :)`
`10:14.07`	`CSerpent`	`yep.. have knocked that off now`
`10:15.05`	`*** join/#asterisk Marouani (~ltostain@i01v-30-28.d4.club-internet.fr)`
`10:15.25`	`CSerpent`	`I'm actually nearly there, we just have problems with VAD frames now and some dropping out on calls`
`10:17.14`	`CSerpent`	`but otherwise the * project I'm doing's working well :) appreciate your help there Nethab`
`10:17.37`	`Nethab`	`np`
`10:17.58`	`Nethab`	`are you using silence suppression?`
`10:18.32`	`CSerpent`	`that's where I'm having trouble working out if I am or not.. I'm assuming so since we're getting the VAD frame errors on 729`
`10:19.33`	`Nethab`	`* always transmits silenceSupp = off and tells you to put it in your phone config`
`10:20.23`	`CSerpent`	`under sip.conf for each phone or?`
`10:20.24`	`*** join/#asterisk TheEmperor (TheEmperor@cki-234-75.tm.net.my)`
`10:20.51`	`Nethab`	`no as part of *'s sip invites it automatically sends silenceSupp off`
`10:21.16`	`Nethab`	`and wants you to set your soft and hard phones to disable it`
`10:21.26`	`Nethab`	`i think it doesn't work right`
`10:21.36`	`CSerpent`	`I'll have to check if it's possible on these phones :\|`
`10:21.45`	`*** join/#asterisk helios4 (~utente1@host10-52.pool8175.interbusiness.it)`
`10:22.04`	`Nethab`	`dropped calls are the first sign of that`
`10:22.27`	`CSerpent`	`well versus the call dropping, if someone counts to 20, it's lose about 40% of the numbers`
`10:22.32`	`CSerpent`	`it'll rather`
`10:22.45`	`Nethab`	`that could be from jitter too`
`10:23.00`	`*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)`
`10:23.23`	`CSerpent`	`all I know is when it's doing it, I see tons and tons of "NOTICE[12829]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end`
`10:23.26`	`Nethab`	`out of order voice packets usually get discarded`
`10:24.46`	`*** join/#asterisk _rudy (~trash@wormhole.ipcable.de)`
`10:25.04`	`CSerpent`	`hmm.. the odd thing is that it's only this provider too, never had this before`
`10:25.29`	`Nethab`	`that doesn't look like a jitter or out of order thing though`
`10:25.35`	`CSerpent`	`hmm`
`10:26.12`	`Leland`	`morning all`
`10:26.22`	`Leland`	`have a bit of a problem setting ToS in the sip.conf...`
`10:26.23`	`Nethab`	`I think it's more likely a VAD or G729 problem, asterisk doesn't do VAD well yet`
`10:26.24`	`CSerpent`	`it's set as a peer in the conf, can reinvite = no, insecure = all, and only allowing G729`
`10:26.54`	`Leland`	`I need to ensure that call traffic is flagged with DSCP CS5 (ToS 0xA0), but asterisk complains that it can't set 0xA0)`
`10:26.54`	`CSerpent`	`finding that out the hard way lol`
`10:27.15`	`Nethab`	`lol`
`10:27.23`	`CSerpent`	`H.323 <hurl> :)`
`10:27.50`	`kleptog`	`it's just rediculous`
`10:28.23`	`kleptog`	`asterisk compiles no worries and run, but try to get any kind of H323 happening and you disappear in a maze of dependancies and libraries`
`10:28.40`	`Leland`	`I had managed to get h323 to work with no problems :S`
`10:28.47`	`Leland`	`don't use it anymore though`
`10:29.49`	`Leland`	`mainly cos I've dumped callmanager express and flashed the phones to use SIP with them registering on to asterisk directly`
`10:31.17`	`kleptog`	`unfortunatly, we've got a provider who insists on H323. They say they've got a special version of asterisk that works but they've been saying that for weeks now`
`10:31.18`	`Nethab`	`cisco phones?`
`10:33.08`	`*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)`
`10:34.02`	`CSerpent`	`you might just find it easier to use a SIP or IAX provider tbh`
`10:34.17`	`*** join/#asterisk RoyK (~roy@80.239.107.80)`
`10:34.51`	`*** join/#asterisk pelotas (~alek@host-ip130-216.crowley.pl) [NETSPLIT VICTIM]`
`10:35.05`	`shaZwaz`	`howdy RoyK`
`10:35.14`	`CSerpent`	`the money you've likely lost in spending time trying to sort the H.323 will prob outweigh the benefits of going elsewhere`
`10:35.25`	`CSerpent`	`and time`
`10:35.40`	`kleptog`	`CSerpent: not my choice. We have a "special relationship" with these people and my boss wants it to work...`
`10:35.55`	`RoyK`	`~seen inspired`
`10:35.57`	`jbot`	`inspired <mikael@host-81-191-113-158.bluecom.no> was last seen on IRC in channel #asterisk, 2d 18h 51m 19s ago, saying: 'Zeeek: try turning echotraining=off in zapata.conf. it solved everything here!'.`
`10:35.57`	`CSerpent`	`oh dear`
`10:36.12`	`CSerpent`	`could try the famous Dutch phrase.. "niet mogelijk" :)`
`10:36.40`	`RoyK`	`http://news.bbc.co.uk/1/hi/england/london/4252421.stm IKEA _is_ dangerous`
`10:37.01`	`kleptog`	`CSerpent: he wouldn't understand... He's not really a good manager. I'm on an hourly rate though :)`
`10:37.22`	`CSerpent`	`yeah.. get crushed for a sofa`
`10:37.29`	`CSerpent`	`ahh now that's always good`
`10:37.35`	`*** part/#asterisk helios4 (~utente1@host10-52.pool8175.interbusiness.it)`
`10:38.52`	`kleptog`	`wow! we have an IKEA here and you don't want to go when it's busy...`
`10:39.06`	`kleptog`	`nice food though`
`10:40.35`	`RoyK`	`http://www.vidarholen.net/contents/wordcount/`
`10:41.35`	`RoyK`	`http://www.themorningnews.org/archives/how_to/the_nonexpert_ikea.php <-- IKEA walk-through`
`10:42.11`	`Nethab`	`well if they have a 'special' version of asterisk, then it's still asterisk and they can still do IAX`
`10:42.43`	`_rudy`	`Hi I am new and have a basic problem: I have a client that send a sip invite and I get basic things running (e.g. can call 1000 etc.) but now I have a client that sends a sip register and I get a 403 forbidden. I have not found any starting point in the docs so far for this setup...`
`10:43.15`	`kleptog`	`Nethab: maybe, I did ask if we could just do IAX and lose the H323, but something about cisco routers. I'm not sure I want to know...`
`10:44.04`	`Nethab`	`well they don't have to lose the h323 just let you 'trunk' in through IAX into their asterisk box`
`10:44.33`	`Nethab`	`and have their 'special`
`10:44.41`	`Nethab`	`version do the translation`
`10:45.01`	`Nethab`	`and iax only needs one port`
`10:45.06`	`Nethab`	`4569`
`10:46.10`	`Nethab`	`so does asterisk have to use a h323 gateway to talk to an h323 endpoint at their end?`
`10:52.54`	`kleptog`	`That's the plan yes`
`10:53.15`	`kleptog`	`But if they have a version that works I could get it to do the transaltion, I like that plan...`
`10:53.47`	`*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)`
`10:54.32`	`CSerpent`	`can I just check.. is the codec_g729a.so actually 729 or an annex of?`
`10:55.47`	`*** join/#asterisk shaZwaz (~chatzilla@203.81.196.167)`
`10:56.13`	`Nethab`	`i think it's version a, but it should only use that codec if * needs to do transcoding, if both endpoints support it natively * should just pass through`
`10:56.37`	`CSerpent`	`thx :)`
`10:57.52`	`*** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net)`
`10:57.58`	`Casper_UA`	`hi`
`10:58.08`	`CSerpent`	`lo`
`10:59.22`	`*** join/#asterisk [ro]nic3try (~iancu@81.181.199.33)`
`11:04.35`	`Casper_UA`	`<PROTECTED>`
`11:05.49`	`[ro]nic3try`	`i have recently installed asterisk.. and i'm trying to conect a HT286 to it`
`11:06.06`	`[ro]nic3try`	`it doesn't work`
`11:06.16`	`[ro]nic3try`	`pls help`
`11:06.28`	`Casper_UA`	`~docs`
`11:06.29`	`jbot`	`Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org`
`11:07.00`	`Casper_UA`	`what protocol uses your ht286?`
`11:07.01`	`[ro]nic3try`	`i get host <192.168.1.88> does not implement "REGISTER"`
`11:07.09`	`[ro]nic3try`	`SIP`
`11:07.21`	`[ro]nic3try`	`pe UDP`
`11:07.32`	`[ro]nic3try`	`UDP port 5060`
`11:08.37`	`[ro]nic3try`	`i have read wiki.. i did the config as it is there`
`11:08.45`	`[ro]nic3try`	`but still doesn't work`
`11:09.17`	`[ro]nic3try`	`i cannot register the grandstream device`
`11:09.19`	`Casper_UA`	`don't use register=... in sip.conf`
`11:09.33`	`*** join/#asterisk Conductor (~thomas@62.8.240.132)`
`11:09.35`	`Conductor`	`hi all`
`11:09.47`	`shaZwaz`	`howdy`
`11:09.53`	`Conductor`	`somehow Playback(filename) doesnt work here...`
`11:10.06`	`Conductor`	`Feb 10 12:20:09 WARNING[13910033]: file.c:475 ast_openstream: File /var/lib/asterisk/sounds/john.gsm does not exist in any format`
`11:10.45`	`shaZwaz`	`its not there ?`
`11:11.13`	`Conductor`	`yes it is`
`11:11.24`	`shaZwaz`	`by the name and extenion ?`
`11:11.36`	`Casper_UA`	`did you load format_gsm.so?`
`11:11.51`	`Conductor`	`oh wait...`
`11:12.10`	`Conductor`	`i have to set the filename without extension?`
`11:12.21`	`[ro]nic3try`	`i have erased the line.. but still no effect..`
`11:12.29`	`[ro]nic3try`	`the device tryes to register`
`11:12.47`	`[ro]nic3try`	`.. and asterisk refuses to do it :(`
`11:13.00`	`shaZwaz`	`Playback(john) shd work !`
`11:14.27`	`Conductor`	`shaZwaz, ok this looks better.`
`11:14.45`	`Conductor`	`how do i play a wav-file?`
`11:15.18`	`netsurfer`	`Qwell - heh.. thought as much.. he lied`
`11:15.21`	`shaZwaz`	`same as b4`
`11:15.48`	`Conductor`	`do i have to load a wav.so or so?`
`11:16.11`	`Casper_UA`	`[ro]nic3try: you should have a 'user' entry for your device and do 'sip reload' to re-read sip.conf`
`11:17.31`	`*** join/#asterisk pranav (~dawda_pra@202.63.174.250)`
`11:17.31`	`Conductor`	`format_wav.c:139 check_header: Not a wav file 7`
`11:17.45`	`Conductor`	`file.c:408 ast_filehelper: Unable to open fd on /var/lib/asterisk/sounds/extension720.wav`
`11:17.55`	`pranav`	`hello`
`11:17.59`	`Conductor`	`it's compressed with ulaw`
`11:18.37`	`Conductor`	`how can i play it then?`
`11:20.37`	`pranav`	`i am finding some problem in astguiclient`
`11:20.40`	`shaZwaz`	`convert it to gsm/wav`
`11:20.57`	`pranav`	`do you know someone to whom i can ask`
`11:21.37`	`Conductor`	`shaZwaz, ok. thx`
`11:22.41`	`pranav`	`there is no channel for astguiclient`
`11:22.50`	`[ro]nic3try`	`[7757] \| type=friend \| context=from-sip \| username=7757 \| secret=1234 \| host=192.168.1.88 \|nat=no \|dmtfmode=info`
`11:23.18`	`[ro]nic3try`	`this is how may sip.conf look for the ht286`
`11:23.24`	`RoyK`	`type=enemy`
`11:23.46`	`Delvar`	`lol`
`11:24.05`	`Delvar`	`host=dynamic else you wont be able to register`
`11:24.35`	`Luhiwu`	`hi all again`
`11:24.36`	`[ro]nic3try`	`i made that too. but no efect ..`
`11:24.41`	`Delvar`	`if you want to put the ip in you can use defaultip=192.168.1.88`
`11:24.49`	`Casper_UA`	`made what? =enemy? :)`
`11:25.03`	`[ro]nic3try`	`no.. host = dymanic`
`11:25.42`	`[ro]nic3try`	`:D`
`11:25.42`	`RoyK`	`dynamic works well`
`11:26.27`	`pranav`	`does anyone know where we find rpms`
`11:27.15`	`pranav`	`i want an rpm for net-mysql`
`11:27.41`	`pranav`	`i searched in google but i could not find t`
`11:28.07`	`netsurfer`	`pranav - rpmfind.net ?`
`11:28.07`	`pranav`	`i tried in rpmfind.net but still i could not find it`
`11:28.14`	`*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)`
`11:28.15`	`netsurfer`	`heh... ok`
`11:28.19`	`Luhiwu`	`i have a question about , SIP and Zap: I receive a call from a SIP client that wants to call an external number, get the call and dials thru Zap ok, but even if the zap is never answered i get a cdr record for the call as 'answered', is there any way to change that?`
`11:28.49`	`*** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net)`
`11:29.00`	`pranav`	`i have rpm for mysql but i need rpm for Net/MySQL`
`11:30.49`	`Delvar`	`pranav: looking for aperl module?`
`11:31.25`	`pranav`	`no see i am running astguiclient`
`11:32.21`	`MrEntropy`	`can i use sipsak or another tool to dodgily terminate a call?`
`11:32.44`	`pranav`	`and in that when i give the command /home/cron/admin_area_populate.pl it gives an error "cant locate Net/MySQL.pm`
`11:33.09`	`pranav`	`<PROTECTED>`
`11:33.34`	`pranav`	`what is this #INC`
`11:33.35`	`Delvar`	`then yes you ARE looking for a perl module`
`11:33.47`	`pranav`	`ok so then what should i do`
`11:33.58`	`Delvar`	`how noob are you?.. i can get the exact commnds if you want?`
`11:34.30`	`kleptog`	`pranav: it's in Perl?`
`11:35.12`	`pranav`	`but i have perl installed`
`11:35.43`	`pranav`	`Delvar:it would be great if you can`
`11:36.08`	`pranav`	`i have the Net::MySQL up to date`
`11:38.27`	`kleptog`	`pranav: do you know how to install any Perl module you've ever heard of?`
`11:41.48`	`Casper_UA`	`kleptog: every module has nice README`
`11:43.40`	`pranav`	`i.e from cpan prompt when i give install Net::MySQL it shows that it is up to date`
`11:45.18`	`pranav`	`i just need an rpm for Net-MySQL`
`11:46.02`	`pranav`	`i have rpm's for MySQL but not for Net-MySQL`
`11:46.57`	`datareactor`	`pranav i think its perl module`
`11:47.28`	`pranav`	`see the error is "cant locate Net/MySQL.pm in @INC`
`11:48.03`	`pranav`	`i got the similar error "cant locate Time/HiRes.pm`
`11:48.39`	`datareactor`	`i dont think Net-MYSQL comes in rpm its separate perl module`
`11:48.54`	`pranav`	`so i installed an rpm for it"per-Time-HiRes-1.55-3.i386.rpm`
`11:49.06`	`pranav`	`so then it worked`
`11:51.13`	`pranav`	`ok so if its a seperate perl module how to install that`
`11:51.45`	`*** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net)`
`11:51.55`	`datareactor`	`download http://search.cpan.org/CPAN/authors/id/O/OY/OYAMA/Net-MySQL-0.08.tar.gz`
`11:52.42`	`datareactor`	`<PROTECTED>`
`11:53.08`	`pranav`	`ok`
`11:53.12`	`datareactor`	`perl Makefile.PL`
`11:53.29`	`*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)`
`11:53.41`	`datareactor`	`make`
`11:53.41`	`datareactor`	`make test`
`11:53.41`	`datareactor`	`make install`
`11:54.14`	`Liquide`	`hmm, i've set both asterisk and my softphone to use gsm, but i get ; Capabilities: us - 0x2 (gsm), peer - audio=0xd (g723\|ulaw\|alaw)/video=0x0 (nothing), combined - 0x0 (nothing)`
`11:54.14`	`Liquide`	`Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined - 0x0 (nothing)`
`11:54.40`	`pranav`	`ok let me try`
`11:54.45`	`datareactor`	`here is link for rpm ftp://ftp.task.gda.pl/vol/3/ftp.pld-linux.org/dists/2.0/PLD/i586/PLD/RPMS/perl-Net-MySQL-0.08-1.noarch.rpm`
`11:54.47`	`Liquide`	`and i'm not able to accept calls then, whats hould i do?`
`11:55.08`	`*** part/#asterisk Conductor (~thomas@62.8.240.132)`
`11:56.37`	`datareactor`	`Liquide you define phone exten in extension.conf`
`11:57.06`	`jalsot`	`does anybody know if it is possible to get out some PRI statistics from asterisk/zaptel? (e.g. slips, errors, etc.)`
`11:57.30`	`Liquide`	`datareactor huh? that problem is not there when i use any other codecs than gsm`
`11:57.54`	`Delvar`	`anyone know a good tool for sip trafic visualisation? either from ethereal type trace or sip debug?`
`11:58.22`	`[ro]nic3try`	`ngrep .. works nice :)`
`12:02.27`	`CSerpent`	`q 4 anyone :) If I'm getting VAD framing errors and unwanted silences on Digium G729a (paid for) codec, but not getting it on G711 to a peer - any ideas on what the problem is likely to be?`
`12:02.46`	`datareactor`	`Liquide try disallow=all allow=gsm in user setting in sip.conf`
`12:03.05`	`Liquide`	`i have it under general`
`12:03.09`	`Liquide`	`in sip.conf`
`12:03.28`	`datareactor`	`try where you declared users`
`12:03.50`	`Liquide`	`remove it from general?`
`12:03.55`	`datareactor`	`yes`
`12:04.24`	`datareactor`	`or leave as it is declare under user`
`12:04.30`	`Liquide`	`nope, same problem`
`12:05.51`	`Liquide`	`wait, typo tries again`
`12:06.33`	`Liquide`	`nope, same problem`
`12:08.02`	`Liquide`	`could be some codec crash between X-Lite and asterisk.. or?`
`12:08.39`	`*** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk)`
`12:09.12`	`pranav`	`hello mr datareactor it shows an error saying "vendor_perl is needed by perl-Net-MySQL`
`12:09.13`	`DrPete`	`Does the cisco 7960 MWI work with asterisk?`
`12:12.48`	`pranav`	`hello`
`12:12.52`	`DrPete`	`postel: You there?`
`12:13.07`	`CSerpent`	`DrPete: AFAIK`
`12:13.50`	`pranav`	`datareactor: are you there`
`12:15.08`	`postel`	`DrPete: sure it does, just set the voicemail accounts in the config and you're good to go`
`12:15.34`	`CSerpent`	`postel: have you had any experience with the 7920 on *?`
`12:16.00`	`postel`	`CSerpent: not (yet) ;)`
`12:16.20`	`postel`	`but give me time, im twisting some arms to get some`
`12:16.29`	`techie`	`DrPete: yes it does.`
`12:16.58`	`*** join/#asterisk jayden (~ircatjerr@65.170.43.34)`
`12:20.22`	`DrPete`	`postel: yeah, i set them, in the config, but the light does not light up, i must have made a mistake`
`12:20.25`	`WildPikachu`	`hrmmm`
`12:20.33`	`WildPikachu`	`can u send faxes using asterisk aswell?`
`12:20.39`	`WildPikachu`	`i saw somewhere u could receive faxes`
`12:21.02`	`WildPikachu`	`or even better plug in hylafax`
`12:21.52`	`postel`	`DrPete: a nice litle config for the 79xx is set an exten in your config to go to voicemail(s<mailbox), then from Sip Configuration -- Messages URI map the msg button to the extension, voila, when you see the MWI lighting up hit the msg button and comedian mail app would start "You got yada yada.." no password, you can do the same and drop it to voicemailmain (so that you can check your voicemail from other than your own phone) or ommit the s and be asked`
`12:21.52`	`postel`	`<PROTECTED>`
`12:22.09`	`postel`	`DrPete: are the msg files generated?`
`12:22.33`	`Liquide`	`datareactor no matter what i try only ulaw works`
`12:23.00`	`DrPete`	`postel: yeah, msg files are working and getting mailed`
`12:23.18`	`Liquide`	`datareactor and alaw`
`12:23.59`	`postel`	`DrPete: dialling the extension of the voicemail does it give you the messages?`
`12:24.39`	`postel`	`(it would be really bad if its a bad phone and not a setup glitch) :/`
`12:25.00`	`DrPete`	`postel: yes`
`12:25.47`	`DrPete`	`postel: does it need to be line 1 to work?`
`12:26.32`	`postel`	`dont think so`
`12:26.36`	`pranav`	`how to get perl-base >=0:5.004 is needed`
`12:26.50`	`postel`	`DrPete: does it work at ALL? line 1 or not`
`12:27.19`	`DrPete`	`postel: no light comes up for a message on the line 2 ext, i will change to line 1 and see what happens`
`12:28.20`	`*** join/#asterisk calvinhp (~calvinhp@dhcp065-029-088-222.indy.rr.com)`
`12:28.23`	`*** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de)`
`12:28.25`	`postel`	`DrPete: * ver?`
`12:29.09`	`DrPete`	`stable I think, i just used cvs`
`12:29.12`	`DrPete`	`1.0.1`
`12:29.16`	`DrPete`	`or whatever is it`
`12:29.26`	`postel`	`1.0.5 it is`
`12:29.38`	`postel`	`topic ;)`
`12:29.39`	`DrPete`	`do i need 1.0.5?`
`12:29.53`	`DrPete`	`to make the light work`
`12:29.57`	`postel`	`DrPete: im on 1.0.3 and it works just fine with MWI`
`12:30.12`	`DrPete`	`postel: so what is the sable then, 1.0.5?`
`12:30.16`	`postel`	`yeap`
`12:30.32`	`DrPete`	`ooh, i might have to recompile or something`
`12:33.58`	`DrPete`	`postel: if I follow the instructions on the website, will it get the 1.0.5 from the CVS?`
`12:34.22`	`postel`	`no, its all a big lie`
`12:35.18`	`postel`	`yes it would`
`12:35.22`	`Luhiwu`	`i have a question about , SIP and Zap: I receive a call from a SIP client that wants to call an external number, get the call and dials thru Zap ok, but even if the zap is never answered i get a cdr record for the call as 'answered', is there any way to change that? i don't want to send 200 OK until the zap channel has been answered`
`12:35.53`	`*** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it)`
`12:36.22`	`*** join/#asterisk r1 (~erwan@www.thiscow.com)`
`12:37.44`	`DrPete`	`postel: should I remove the old SRC dirs?`
`12:37.46`	`CSerpent`	`WildPikachu: http://www.inter7.com/?page=astfax`
`12:38.03`	`Liquide`	`anyone have a good SIP softphone to recommend(for windows)? not xlite or sjphone`
`12:38.23`	`sezuan`	`Liquide: SIPPS`
`12:39.02`	`Liquide`	`its commercial`
`12:39.10`	`Liquide`	`i dont wanna pay to get one :P`
`12:39.16`	`DrPete`	`lol`
`12:39.20`	`CSerpent`	`cheapass :)`
`12:39.28`	`DrPete`	`then your gonna have a crap one then`
`12:39.40`	`DrPete`	`You get what you pay for`
`12:39.49`	`Liquide`	`hehe`
`12:39.52`	`CSerpent`	`hmm we don't pay for * ;)`
`12:39.54`	`Liquide`	`not in the OS world`
`12:40.14`	`Liquide`	`CSerpent in that case you get what you pay for counts :P`
`12:40.22`	`Liquide`	`its hard to config`
`12:40.26`	`sezuan`	`Liquide: ok, but btw, is there a good SIP softphone for Linux? Commercial or not. I don't care.`
`12:40.27`	`CSerpent`	`lol true`
`12:41.17`	`Liquide`	`asterisk kills all the incomming and outgoing calls when i tell it to use gsm quality`
`12:41.27`	`Liquide`	`so you get what you pay for :P`
`12:42.42`	`CSerpent`	`can't fault Digium support tho.. always been good so worth paying for their cards`
`12:42.50`	`Liquide`	`hehe`
`12:43.03`	`Liquide`	`i'm using software only, no cards as of yet`
`12:43.59`	`CSerpent`	`ahh`
`12:44.07`	`CSerpent`	`well good fun with Zaptel when you do :)`
`12:44.20`	`CSerpent`	`actually and kapejod's BRI cards`
`12:44.25`	`Liquide`	`hehe`
`12:44.36`	`Liquide`	`even more fun chaning codecs :P`
`12:44.45`	`Liquide`	`it only likes ulaw and alaw`
`12:49.20`	`*** join/#asterisk oej_ (~Olle@apollo.webway.se)`
`12:50.19`	`CSerpent`	`oh tell me about it`
`12:50.29`	`CSerpent`	`I'm stick with peer G.729 and VAD errors at present`
`12:50.34`	`Liquide`	`you found a solution?`
`12:50.36`	`CSerpent`	`stuck*`
`12:50.37`	`CSerpent`	`nope`
`12:50.41`	`CSerpent`	`apart from ULAW`
`12:50.47`	`CSerpent`	`well both the 711s`
`12:50.54`	`Liquide`	`because i can only use alaw and ulaw, and i need to use gsm because of the BW`
`12:51.12`	`CSerpent`	`exactly why I need 729`
`12:53.05`	`Liquide`	`729 didnt work here.. propably because X-Lite didnt support it`
`12:53.48`	`RoyK`	`x-pro does`
`12:54.59`	`Liquide`	`cant find expro install files :P`
`12:54.59`	`Liquide`	`*x-pro`
`12:55.28`	`RoyK`	`x-pro is commercial`
`12:55.52`	`Liquide`	`i know`
`12:57.03`	`sezuan`	`I have an * with a registered user. Can a non-registered user call him using sip:username@asterisk.box?`
`12:57.35`	`sezuan`	`At the moment it's not working I go: Failed to authenticate user test...`
`12:58.32`	`sezuan`	`I tried in sip.conf [redirect] context=redirection;permit=0.0.0.0/0.0.0.0`
`12:58.33`	`Liquide`	`CSerpent do you use softphone for 729?`
`12:58.40`	`CSerpent`	`yep.. X-Pro :)`
`12:58.46`	`RoyK`	`why should non-registered users be allowed, sezuan?`
`12:59.10`	`Liquide`	`CSerpent bleh :P noone have put it out on torrent places or anyhting.. thats weird`
`12:59.10`	`CSerpent`	`as well as Aastra 480is, BT102s and Mitel 5055s as hardphones`
`12:59.11`	`Liquide`	`:P`
`12:59.29`	`CSerpent`	`just buy it :) it's not that expensive and it's pretty good`
`12:59.56`	`Liquide`	`60$ for a softphone.. heh, if i add another 40 i'll have a adapter so i can use a real phone :P`
`13:00.25`	`CSerpent`	`there's yer answer then :)`
`13:00.59`	`sezuan`	`RoyK: Well, I would like to give people a address like me@asterisk.box. An I connect from different places to that box. It should work like a redirection service. I guess partysip to something like that.`
`13:01.16`	`CSerpent`	`sezuan: you'd have to define an extension in incoming to do that`
`13:01.56`	`Liquide`	`CSerpent but i really dont get it why gsm wont work :S`
`13:02.01`	`CSerpent`	`e.g. exten => sezuan,1,Dial(SIP/phone1)`
`13:02.04`	`*** join/#asterisk Delvar (~irc@83.146.53.34)`
`13:02.21`	`CSerpent`	`why not try DIAX.. that uses Speex`
`13:04.07`	`sezuan`	`CSerpent: Can I define a context in sip.conf for calls from non-registered users?`
`13:04.13`	`*** join/#asterisk Joertzer (fwuser@213.61.141.198)`
`13:04.17`	`*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)`
`13:04.20`	`Joertzer`	`hi @ all`
`13:04.51`	`*** join/#asterisk vondax (~dax@host81-139-83-249.in-addr.btopenworld.com)`
`13:05.02`	`CSerpent`	`sezuan: as long as you define "context=incoming" and place that line I gave before under the [incoming] section, any non reg user calling in well follow the instructions in that section`
`13:05.09`	`CSerpent`	`in sip.conf that is`
`13:05.40`	`Joertzer`	`did anyone know how to tell asterisk that the recorded and emailed voicemails not stored in the voicemail ?`
`13:07.08`	`*** join/#asterisk zotz (~zotz@24.231.32.191)`
`13:07.40`	`CSerpent`	`sezuan: also bear in mind some providers are exceptionally unhappy about sending to an alpha numeric SIP URL.. one I know of would only dial a numeric initial part of the URL, so set up a numeric alias if you want`
`13:09.45`	`Mavvie`	`yay! asterisk with multiple NAPTR records works fine here.`
`13:12.38`	`*** join/#asterisk Kirk_22 (~Administr@pool-151-199-159-52.norf.east.verizon.net)`
`13:12.46`	`*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)`
`13:13.21`	`sezuan`	`CSerpent: I think I didn't get it. I have in sip.conf an entry: [incoming] context=redirect. In extensions.conf I have [redirect] exten sezuan => sezuan,1,Dial(SIP/phone,30)`
`13:13.45`	`*** part/#asterisk pranav (~dawda_pra@202.63.174.250)`
`13:13.59`	`Mavvie`	`exten => sezuan,1,dial(...)`
`13:14.56`	`sezuan`	`Mavvie: I have that.`
`13:15.26`	`sezuan`	`I always get "user test could not be authenticated."`
`13:15.54`	`sezuan`	`How does * find the right [...] entry in sip.conf?`
`13:15.58`	`*** join/#asterisk christo (~chris@office.enovi.com)`
`13:16.18`	`Mavvie`	`first, make sure the user is there: sip show peers`
`13:16.37`	`Mavvie`	`then, make sure the user is in the right context: sip show peer <name> and check Context (4th line)`
`13:16.39`	`Kirk_22`	`is there a precompiled zaprtc module anywhere?`
`13:17.52`	`sezuan`	`Mavvie: which user? The calling user is not-registered.`
`13:18.27`	`*** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk)`
`13:18.32`	`Mavvie`	`then don't put then in [direct] but in [default]`
`13:18.37`	`Mavvie`	`(in sip.conf)`
`13:19.02`	`DrPete`	`If i just do this to get from cvs cvs checkout zaptel libpri asterisk what verson will i get?`
`13:19.09`	`Mavvie`	`DrPete: HEAD`
`13:19.25`	`Mavvie`	`(not to be confused with sexual favours)`
`13:20.39`	`DrPete`	`lol`
`13:20.44`	`*** join/#asterisk Othello (Othello@nusnet-225-92.dynip.nus.edu.sg)`
`13:20.53`	`*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)`
`13:20.53`	`DrPete`	`its head, like unstable? or does it run well?`
`13:21.04`	`darkskiez`	`it, well, runs.`
`13:21.05`	`Mavvie`	`it's the development branch`
`13:21.11`	`MrEntropy`	`anyone got a sipsak packet file for an example?`
`13:21.13`	`*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)`
`13:21.13`	`Mavvie`	`sometimes it runs, sometimes it doesn't.`
`13:21.22`	`sezuan`	`Mavvie: I'm confused. Theres is no [(re)direct] and [default] in my sip.conf`
`13:21.53`	`shido6`	`?`
`13:21.56`	`Mavvie`	`oh. it's called general there.`
`13:22.09`	`sezuan`	`Mavvie: I thinks the problem is to tell asterisk to accept calls from non-registered users.`
`13:22.15`	`DrPete`	`Mavvie: how do i get the stable version, i asume I can go back to stable without problems`
`13:22.23`	`Mavvie`	`sezuan:`
`13:22.24`	`Mavvie`	`[general]`
`13:22.24`	`Mavvie`	`context=internet-from-sip ; Default context for incoming calls`
`13:22.36`	`shido6`	`why on earth would you want to accept calls from unregistered devices`
`13:22.48`	`Mavvie`	`shido6: think.`
`13:23.01`	`Mavvie`	`DrPete: cvs log will give you the tagged versions.`
`13:23.03`	`shido6`	`internet-from-sip in your dialplan should read something like, exten => s,1,Playback,unauthorized`
`13:23.05`	`bjohnson`	`sezuan: register just tells the server what ip address the remote has .. it is not an authentication method (it just USES authentication)`
`13:23.07`	`shido6`	`exten => s,2,Hangup`
`13:23.12`	`sezuan`	`Mavvie: the default context includes my redirect context. Sho it should be available.`
`13:23.25`	`Mavvie`	`shido6: not everybody has only only one outgoing line.`
`13:23.34`	`shido6`	`?`
`13:23.39`	`DrPete`	`Mavvie: whats the comand to get the stable version?`
`13:23.47`	`Mavvie`	`DrPete: cvs update.`
`13:24.01`	`shido6`	`so as long as i have your ip you want them to have access to your dialplan`
`13:24.03`	`CSerpent`	`sezuan: you don't want context = redirect in there`
`13:24.29`	`Mavvie`	`shido6: yes. in that context only of course.`
`13:24.39`	`shido6`	`okie dokie`
`13:24.43`	`bjohnson`	`shido6: on a small system, a guest account is useful`
`13:24.49`	`DrPete`	`Mavvie: and that will give me stable?`
`13:24.59`	`bjohnson`	`shido6: also for people just starting`
`13:25.10`	`Mavvie`	`DrPete: check with cvs log what the tagged versions are. use the latest one.`
`13:25.28`	`DrPete`	`oki`
`13:25.57`	`sezuan`	`CSerpent: So, what do I want?`
`13:26.10`	`bjohnson`	`shido6: it's a type of system where you want to make it easy to allows others to call in .. not like yours where you want to keep callers out (unless authenticated)`
`13:26.16`	`shido6`	`I dont know about just starting, just make sure your "guest" account is really what you want as everyone with access to that ip will be all over that context`
`13:26.18`	`Mavvie`	`shido6: everybody from the internet is allowed to 1. call my registered SIP users, 2. my internal PSTN ranges and 3. my internal tools (conferences).`
`13:26.29`	`CSerpent`	`sezuan.. will send in private chat`
`13:26.42`	`Mavvie`	`shido6: for the rest, they can't do anything.`
`13:26.44`	`bjohnson`	`I have guest account that rings in like any other call`
`13:27.02`	`shido6`	`Good, you've thought it through, okie dokie.`
`13:27.55`	`shido6`	`in your general do you have disallow=all, allow=somecodec, allow=someothercodec,allow=someothercodec ?`
`13:28.06`	`shido6`	`or allow=all`
`13:28.06`	`shido6`	`?`
`13:28.23`	`bjohnson`	`shido6: who are you talking to?`
`13:28.35`	`shido6`	`Mavvie?`
`13:29.02`	`Mavvie`	`shido6: I have allow ulaw,alaw,ilbc`
`13:29.05`	`shido6`	`allow=all will give a huge list of codecs and wont always work`
`13:30.03`	`bjohnson`	`what is a codec with good sound but lower bandwidth than ulaw for internet traversal? gsm?`
`13:30.21`	`shido6`	`gsm or ilbc`
`13:30.31`	`shido6`	`speex`
`13:30.39`	`RaYmAn-Bx`	`bjohnson: G729? non-free though (although SPA-2000 includes a license)`
`13:31.08`	`bjohnson`	`oh .. I guess I need to maintain use for dtmf for when connecting to other IVRs. ilbc would cause problems with that correct?`
`13:31.16`	`RoyK`	`G.729 is always non-free`
`13:31.47`	`*** join/#asterisk JunK-Y (~grepmoo@65.39.228.5)`
`13:31.51`	`bjohnson`	`RaYmAn-Bx: I might get into that. But want to avoid the complexity of tracking licenses while I'm starting out`
`13:32.30`	`bjohnson`	`I'm not sure how many voip providers provide g729. I'm really looking for codecs for that .. I use ulaw on the LAN`
`13:34.21`	`RaYmAn-Bx`	`bjohnson: as long as your calls doesn't get transcoded by asterisk you don't need licenses..it has support for passthrough`
`13:34.29`	`shido6`	`hrmm, as long as your asterisk box understands dtmf , it will pass that along to the next peer or zap device`
`13:34.33`	`RaYmAn-Bx`	`(well..or if the RTP Stream passes through asterisk)`
`13:34.47`	`Liquide`	`how did you guys get the gsm codec working? i just get a error in sip debug, and the phone is silent...`
`13:34.52`	`shido6`	`that came out wrong...`
`13:35.06`	`shido6`	`Liquide - using xlite?`
`13:35.28`	`Liquide`	`shido6 yes`
`13:35.30`	`shido6`	`what codec did u specify in your "phone" ?`
`13:35.32`	`shido6`	`err`
`13:35.35`	`shido6`	`dont use xlites gsm`
`13:35.45`	`shido6`	`use ulaw to start`
`13:35.57`	`Liquide`	`i've also tried sjphone`
`13:35.58`	`bjohnson`	`RaYmAn-Bx: my outgoing is all iax (LAN is currently all SIP)`
`13:36.00`	`Liquide`	`ulaw eats bw`
`13:36.07`	`shido6`	`and gsm doesnt work`
`13:36.15`	`shido6`	`so what codecs do u have left? :)`
`13:36.20`	`Liquide`	`what do i need to get gsm working then?`
`13:36.21`	`shido6`	`troubleshoot with ulaw first`
`13:36.26`	`shido6`	`then start messin with codecs`
`13:36.27`	`Liquide`	`ulaw works just fine`
`13:36.53`	`bjohnson`	`shido6: * understands the dtmf but I've had some calls where I cannot operate their IVR .. even when just connecting through the fxs/fxo pair on a single SPA 3k`
`13:37.01`	`*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)`
`13:37.16`	`puzzled`	`hello`
`13:37.28`	`bjohnson`	`Liquide: how did you select gsm?`
`13:37.47`	`shido6`	`bj... hrmm does inband ALWAYS work for that?`
`13:38.01`	`Liquide`	`bjohnson disallow=all`
`13:38.01`	`Liquide`	`allow=ulaw`
`13:38.05`	`bjohnson`	`I don't use inband, i use rfc2833`
`13:38.07`	`Liquide`	`ermm gsm`
`13:38.40`	`shido6`	`and u have problems with , rfc2833 ?`
`13:39.17`	`bjohnson`	`Liquide: no. why are you choosing gsm`
`13:39.55`	`Liquide`	`bjohnson because ulaw is lagging, thats only because to recieve sound from the asterisk server, it goes out to my isps router, and then back to the machine i wanna talk on`
`13:42.57`	`DrPete`	`can you leave asterisk running when you compile a new version from cvs?`
`13:43.13`	`Mavvie`	`sure`
`13:43.46`	`shido6`	`yes drpete`
`13:43.47`	`DrPete`	`oki, thats good to know. I think i have the lastest verion, i will see if it will compile`
`13:43.53`	`shido6`	`just remember you have new code compiled`
`13:44.07`	`DrPete`	`if it compiles, lol`
`13:44.49`	`ManxPower`	`1.0.x CVS always compiles`
`13:44.52`	`DrPete`	`i need a faster server`
`13:44.57`	`shido6`	`rename the working asterisk code to asterisk-works and pull down new cvs code, compile and install it , if u like it keep it if u dont u have the old working code`
`13:45.17`	`DrPete`	`thats a really good idea`
`13:45.38`	`Sauron	`Manx: y'have a good time last night? ;)`
`13:45.39`	`DrPete`	`when i get a new verion, do i need to get new zaptel, libpri etc`
`13:45.41`	`ManxPower`	`DrPete, He's not the first person to do it that way.`
`13:45.54`	`ManxPower`	`Sauron, Other than the phone call at 7:30am, yes.
`13:45.56`	`postel`	`DrPete: ohhh YES`
`13:45.58`	`DrPete`	`ManxPower: I am sure you did it first :)`
`13:46.16`	`Sauron	`hehe`
`13:46.41`	`ManxPower`	`It was prolly some programmer back in the 60's that did it first.`
`13:47.15`	`postel`	`yeah, he had a FORTRAN cvs implementation`
`13:47.30`	`ManxPower`	`postel, YEAH!`
`13:48.21`	`postel`	`and since ascii wasnt there yet and ethernet was sneakernet he was pulling it over decnet in edbic`
`13:48.45`	`ManxPower`	`Yes, the 60's may have been a little early.`
`13:48.58`	`CSerpent`	`more people awake :)`
`13:49.18`	`ManxPower`	`postel, don't be an ass before my first cup of coffee.`
`13:49.27`	`postel`	`ManxPower: i'll do my best`
`13:49.53`	`CSerpent`	`ManxPower: Can you be of any assistance with a G.729 issue please?`
`13:50.43`	`postel`	`ManxPower: try not to be a smart ass when my grid not working as it should and we gonna be just fine :>`
`13:51.33`	`shido6`	`whoa...`
`13:52.23`	`shido6`	`ManxPower isnt a smartass. Kinda like talking to Larry Brown`
`13:52.33`	`shido6`	`the coaches coach`
`13:57.00`	`Mavvie`	`if c is an int , and I give it &c to a function (thus becoming an int ), and I assign an int to *c.... that shouldn't segfault should it?`
`13:57.17`	`Mavvie`	`oh, it does if you do c=NULL just three lines before that assignment.`
`13:57.52`	`Mavvie`	`annoying that I can't run this beast in gdb :-)`
`13:58.11`	`Casper_UA`	`try to analize core`
`13:58.30`	`Mavvie`	`Casper_UA: that was the part which didn't make sense.`
`13:59.46`	`ManxPower`	`CSerpent, What assistance do you need?`
`14:00.08`	*** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it)
`14:00.13`	mAsH`	`hi all`
`14:00.18`	`*** join/#asterisk TheEmperor (TheEmperor@210.186.116.86)`
`14:02.07`	`*** part/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)`
`14:02.08`	`CSerpent`	`ManxPower: when using G.729 to a peer, I keep getting a ton of VAD frame errors and the call incoming audio dropping out... if I use 711 all is good (except bandwidth).. I wondered if you had any ideas?`
`14:02.19`	`CSerpent`	`outgoing audio is fine however`
`14:02.35`	`ManxPower`	`CSerpent, turn off VAD.`
`14:02.42`	`*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)`
`14:02.49`	`ManxPower`	`CSerpent, How many G729 licenses do you have?`
`14:02.54`	`CSerpent`	`5`
`14:03.08`	`shido6`	`odd number :)`
`14:03.15`	`Nugget`	`heh`
`14:03.31`	`CSerpent`	`was from experimental days :) never needed to rise from 5 as yet`
`14:04.12`	`*** join/#asterisk ezabi (~ezabi@82.201.194.37)`
`14:05.37`	`CSerpent`	`now.. the big q.. how do I turn off VAD, that's something I couldn't work out`
`14:05.53`	`*** join/#asterisk ToyMan (~konversat@204.8.82.238)`
`14:08.44`	`ManxPower`	`CSerpent, I don't know. That is dependent on the device you use.`
`14:09.49`	`*** join/#asterisk Othello (Othello@nusnet-225-92.dynip.nus.edu.sg)`
`14:10.20`	`CSerpent`	`it's our * box peering to a sonus networks softswitch`
`14:10.44`	`ManxPower`	`ask them to turn off VAD then`
`14:11.04`	`CSerpent`	`:) ok thx`
`14:11.18`	`DrPete`	`yey, message light works, lol`
`14:13.08`	`DrPete`	`man, these cisco phones are awsome`
`14:13.21`	`CSerpent`	`ManxPower: they're saying there's no VAD on at their end`
`14:14.53`	`postel`	`DrPete: nice, set an exten for voicemail(s<mailbox>) now and put the exten number in Sip Configuration -- Messages URI, then when the light lights up hit the envelope button`
`14:15.41`	`Nugget`	`that's "voicemailmain(s...)" not voicemail()`
`14:15.53`	`postel`	`nope, its voicemail`
`14:16.05`	`postel`	`no reason to go on top of comedian mail`
`14:16.15`	`postel`	`you just need to get dropped to your mailbox`
`14:16.33`	`Nugget`	`calling voicemail(s#) just causes it to skip the greeting.`
`14:16.45`	`postel`	`thats what you want`
`14:16.57`	`postel`	`you are local to the phone, you just want the msgs`
`14:17.05`	`Nugget`	`yes, so you want voicemailmain(s#)`
`14:17.11`	`Nugget`	`so you can skip the password entry`
`14:17.29`	`_rudy`	`hi, what do I have to do in order to answer a register request from my sip client correctly. I created an section within sip.conf but i still get a 403 forbidden?`
`14:18.10`	`postel`	`Nugget: you're right, i just double checked`
`14:19.22`	`DrPete`	`postel: ooh, i need to do that, where i put that the sip.conf?`
`14:19.32`	`postel`	`DrPete: extensions.conf`
`14:19.47`	`DrPete`	`postel: can u paste me an example?`
`14:20.29`	`Nugget`	`if the mailbox number is the same as the calling phone's extension/callerid, you can do this: exten => 2500,1,VoicemailMain2(s${CALLERIDNUM})`
`14:20.37`	`Nugget`	`just make sure it's in a safe context.`
`14:20.42`	`postel`	`thats the one`
`14:21.03`	`postel`	`and the pass 2500 to Sip Configuration -- Messages URI`
`14:21.12`	`Nugget`	`right`
`14:21.15`	`postel`	`s/the/then`
`14:21.35`	`_rudy`	`does asterisk support sip register anyways?`
`14:22.08`	`RoyK`	`yes`
`14:22.19`	`RoyK`	`of course :)`
`14:22.25`	`DrPete`	`whats the best length of extensions`
`14:23.01`	`_rudy`	`any examples anywhere? all I can get to work is an xlite which does an invite not a register`
`14:23.17`	`DrPete`	`3 numbers, 4, does it matter`
`14:23.24`	`postel`	`DrPete: that depends how you structure your dialplan`
`14:24.04`	`DrPete`	`postel: I am in the UK, i dont want ti to interfere with anything, i just want to get it right from the start, make it easier for me`
`14:24.40`	`postel`	`DrPete: maybe _2., does something else, pick something that doesnt mess up your dialplan, being in the UK has nothing to do with it, im in London btw ;)`
`14:26.14`	`DrPete`	`postel: hehe oki. I mean, i could obviously not have staring with 0`
`14:26.31`	`DrPete`	`postel: do you have 4 numner extensions`
`14:26.43`	`postel`	`DrPete: 6`
`14:27.11`	`postel`	`its funny i use 6 digit extens at home and 4 at work :P`
`14:27.39`	`Nugget`	`six seems excessive, but it really doesn't matter in the least. use what makes you happy.`
`14:27.53`	`DrPete`	`lol 6`
`14:28.10`	`Nugget`	`I use three at home and at work, but both installs are pretty small.`
`14:28.16`	`DrPete`	`how many phone you gotjot?`
`14:28.20`	`Nugget`	`plenty of room with three for me to do what I want.`
`14:28.33`	`fenlander`	`I'd go for 12: plenty of room for expansion then!`
`14:28.45`	`DrPete`	`man, when you rebot a cicso, it locks my irc, lol`
`14:28.46`	`postel`	`DrPete: 7 in production, 4 in testbeds`
`14:28.50`	`Nugget`	`be different, do them in binary!`
`14:28.56`	`Nugget`	`I'm extension 100101101`
`14:29.01`	`DrPete`	`lol`
`14:29.12`	`DrPete`	`wow, the message buttion works, lol`
`14:29.33`	`bjohnson`	`anyone have iax.cc DIDs? Do they forward callerid? Do they send in to * on an extanesion that matches the DID (to track which one is used in case you have more than one)?`
`14:30.23`	`fenlander`	`Is anyone using sipgate in the uk?`
`14:30.39`	`CSerpent`	`nope`
`14:30.50`	`CSerpent`	`I know telappliant do stuff tho`
`14:31.12`	`DrPete`	`Man, thats annyonying, on the cico, i dailed, 8010, not I want to dial 801, but it brings up 8010 and dials that`
`14:31.38`	`postel`	`DrPete: disable Auto Complete Numbers`
`14:31.39`	`fenlander`	`CSerpent: thanks`
`14:31.44`	`postel`	`Call Preferences`
`14:32.57`	`DrPete`	`postel: i quite like thatk, i change the extn, its all good`
`14:34.09`	`*** join/#asterisk dercol (~ercolani@sei.yacme.com)`
`14:35.21`	`DrPete`	`when I well, its going well so far`
`14:35.36`	`pointer-gaim`	`any bv customer around that could test a call to the UK for me?`
`14:35.38`	`DrPete`	`postel: who do you use for voip call in the UK`
`14:35.48`	`pointer-gaim`	`I can't seem to dial the numbers I've tested`
`14:35.53`	`pointer-gaim`	`but I can dial italy fine`
`14:36.17`	`pointer-gaim`	`s/numbers/UK numbers/`
`14:36.24`	`postel`	`DrPete: what do you mean in the UK? depends who you're calling`
`14:36.45`	`postel`	`DrPete: there are something like 643686483468346348468346 voip providers, check the wiki`
`14:37.14`	`*** join/#asterisk cbachman (~cbachman@129.105.7.250)`
`14:37.32`	`DrPete`	`Yeah, i see, I wondered who are good value/stable in the UK`
`14:37.54`	`postel`	`BT`
`14:37.55`	`DrPete`	`i have voiptalk, they seem ok`
`14:37.55`	`postel`	`:P`
`14:37.57`	`DrPete`	`lol`
`14:38.07`	`DrPete`	`BT are a ripoff, lol`
`14:39.40`	`*** join/#asterisk mutilator (~animenodv@65.111.201.79)`
`14:39.48`	`mutilator`	`umm`
`14:39.51`	`mutilator`	`on a sipura ata..`
`14:39.52`	`mutilator`	`<PROTECTED>`
`14:39.56`	`mutilator`	`what makes it do that?`
`14:40.08`	`mutilator`	`nvm`
`14:40.10`	`mutilator`	`got it :P`
`14:40.28`	mAsH`	`anyone use eyebeam?`
`14:41.06`	`tzafrir`	`anybody using ChanIsAvail?`
`14:42.04`	`Nugget`	`I do.`
`14:42.05`	`tzafrir`	`Any way to make it return the channel itself (I hope I got my terms right)`
`14:42.33`	`tzafrir`	`Or to get the available channel's name in a clear way?`
`14:42.49`	`tzafrir`	`For zap channels it attaches a -NUM`
`14:42.54`	`Nugget`	`If any of the requested channels are available, the next priority will be n+1,`
`14:42.54`	`Nugget`	`the channel variable ${AVAILCHAN} will be set to the name of the available channel`
`14:42.55`	`tzafrir`	`For IAX it attaches /NUM`
`14:43.09`	`Nugget`	`that's straight from "show application chanisavail"`
`14:43.36`	`tzafrir`	`read that and the voip-info page already`
`14:43.48`	`Nugget`	`isn't that the answer to your question?`
`14:44.18`	`tzafrir`	`I applied the workaround from the wiki page. But it seems to break when I try to use the same macro for both iax and zap`
`14:44.49`	`CSerpent`	`DrPete: BT have always been a ripoff :D`
`14:45.01`	`CSerpent`	`TelAppliant seem to do OK and have IAXpeering`
`14:45.30`	`DrPete`	`If I change something in zap conf, should reload apply it`
`14:45.40`	`tzafrir`	`I have asterisk 1.0.5, BTW`
`14:45.45`	`DrPete`	`CSerpent: can you recommened anyone else?`
`14:46.05`	`CSerpent`	`DrPete: there is a company called Wireless Edge, but I don't think their SIP peering is live yet`
`14:46.48`	`DrPete`	`ahh, i would like iax`
`14:49.46`	`*** join/#asterisk FirstSword (~Miranda@host6614613596.biz.tor.fcibroadband.com)`
`14:50.39`	`*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)`
`14:51.43`	`bjohnson`	`tzafrir: or you could just try to dial it to see if it is available (see the superdial macro for an example)`
`14:52.05`	`tzafrir`	`But what if the line is busy?`
`14:52.15`	`bjohnson`	`DrPete: I think zap changes require a restart`
`14:52.21`	`tzafrir`	`I want to use this for a trunk`
`14:52.46`	`bjohnson`	`tzafrir: what would you like it to do if it is busy? The framework is there for that too`
`14:53.12`	`tzafrir`	`There are two cases: a. the local line is busy, b. the target is busy`
`14:53.30`	`tzafrir`	`I'd like to handle (a). I don't care about (b)`
`14:53.41`	`bjohnson`	`right .. look at the superdial macro`
`14:53.54`	`*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)`
`14:54.20`	`tzafrir`	`I looked at it, and it retries. Which means jamming the outgoing line.`
`14:54.31`	`bjohnson`	`it is set to exit without error if the local line is busy .. you can then 1. try another outgoing channel or 2. return a message to the local caller`
`14:54.40`	`tzafrir`	`Also consider a case where every hit counts`
`14:54.44`	`Zeeek`	`hello tzfrir, you get FWD working ok?`
`14:55.14`	`tzafrir`	`No. I'm stuck with my trunk macro`
`14:56.47`	`Zeeek`	`has anyone tried these IAX hardphones yet?`
`14:56.48`	`Zeeek`	`http://www.iaxtalk.com/product_info.php?products_id=36&osCsid=4864cf99cbae2b15a449910bce21cb62`
`14:57.29`	`bjohnson`	`Zeeek: you really like those particular ones/`
`14:57.40`	`Zeeek`	`there aren't 50 IAX phones you know`
`14:57.48`	`Zeeek`	`I'm waiting for a farfon now`
`14:57.57`	`Zeeek`	`we need to evaluate all the possible iAX stuff`
`14:58.43`	`*** join/#asterisk DrPete_ (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk)`
`14:58.46`	`*** join/#asterisk eKo1 (~bernd@63.245.57.70)`
`14:58.57`	`bjohnson`	`well .. someone here has a couple of the ones here: http://allatchina.com/`
`14:58.59`	`DrPete_`	`dammit, i got a kernel panic from asterisk, i had to reboot`
`14:59.04`	`*** join/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net)`
`14:59.12`	`bjohnson`	`Zeeek: also Aginamu has a couple different ones`
`14:59.41`	`bjohnson`	`there are also a couple different IAX ATAs I've seen available`
`15:00.03`	`Zeeek`	`all the chinese phone may well be one phone!`
`15:00.27`	`fwittekind`	`any1 fimilar with the problem with the wctdm driver with generating kernel NMI messages? digium support treated it as a known issue.`
`15:00.28`	`Zeeek`	`the allchina site could use a better structure`
`15:00.46`	`redder86`	`An IAX hardphone for $59?`
`15:00.58`	`Zeeek`	`plus $30-50 shipping`
`15:01.04`	`bjohnson`	`redder86: yes`
`15:01.33`	`bjohnson`	`someone here received 4 I think and the after delivery cost was about $75 each`
`15:01.37`	`redder86`	`Zeeek: international shipping on all of them?`
`15:01.41`	`bjohnson`	`can't rmember who it was though`
`15:01.42`	`Zeeek`	`I thuink these are both the same phones bj`
`15:01.57`	`Zeeek`	`$30 US, $40 EU, $50 Bahamas`
`15:02.40`	`bjohnson`	`one of the ATAs at that site looks like the ones selling on ebay for $25-$45 each (plus shipping)`
`15:02.40`	`Zeeek`	`no price given on allatchina`
`15:02.48`	`redder86`	`There isn't a distributor in the US who has these?`
`15:02.57`	`Zeeek`	`could be`
`15:03.11`	`bjohnson`	`not that I've seen that will sell in small quantities`
`15:03.41`	`redder86`	`any that will sell in large?`
`15:03.43`	`Zeeek`	`still an IAX phone (does SIP and the rest too) delivered for $90`
`15:03.48`	`Zeeek`	`not bad IF THEY WORK`
`15:03.53`	`bjohnson`	`redder86: talk to Aginamu`
`15:03.54`	`Zeeek`	`nbot easy to send back`
`15:03.56`	`redder86`	`not bad if they work well`
`15:04.07`	`bjohnson`	`redder86: I have the specs for his phone around here somewhere if you want it`
`15:04.08`	`Zeeek`	`these must be the ones with those processors`
`15:04.12`	`redder86`	`Grandstreams are still in that price range.`
`15:04.16`	`redder86`	`and they work reasonably well`
`15:04.19`	`Sauron	`spa-841`
`15:04.36`	`bjohnson`	`all the IAX stuff I've seen to date seem to use a "magic" PA168 processor`
`15:04.38`	`Zeeek`	`with shipping GS is silmilar but obviously doesn't do IAX`
`15:04.46`	`DrPete_`	`is getting a kernel panic from asterisk uncommon, what could have caused it?`
`15:04.47`	`Zeeek`	`yeah that's the one`
`15:05.12`	`Zeeek`	`DrPete why are you abusing your asterisk?`
`15:05.24`	`Zeeek`	`what are you doing to it?`
`15:05.28`	`Damascene`	`didn't you know there was a Asterisk Abusers Anonymous?`
`15:05.39`	`jaiger`	`does anyone have a couple polycom phones that you could run a simple test case for me?`
`15:05.48`	`redder86`	`IAX isn't important to my clientelle except for "remote" employees who do not work in the office. Until now they've been using IAXys for those. The IAXy is about $100 delivered.`
`15:05.53`	`Zeeek`	`I thought that was #asterisk ?`
`15:05.59`	`redder86`	`jaiger: still at it?`
`15:06.03`	`jaiger`	`redder86: yes`
`15:06.06`	`jaiger`	`8)`
`15:06.10`	`redder86`	`jaiger: I'm sorry.`
`15:06.39`	`Zeeek`	`the IAXy is great but has a few well-known small disadvantages`
`15:06.47`	`DrPete_`	`Zeeek: i not doing anything, i think it was a zap call, i kept ring and hanging up to test called id, and it went pear shaped`
`15:06.48`	`redder86`	`only a few?`
`15:06.49`	`Zeeek`	`like no DNS`
`15:07.06`	`redder86`	`like lack of full codec support`
`15:07.27`	`tzafrir`	`bjohnson, I looked at the superdial macro, I'm not sure it works. What if someone dials in? What if the line happens to be busy for some other reason?`
`15:07.28`	`Zeeek`	`DrPete you better treat astersik right cause there is an AA board that may have to come over and break your arm`
`15:07.31`	`tzafrir`	`(shit happens)`
`15:08.22`	`DrPete_`	`hehe`
`15:08.24`	`Nugget`	`Linux is poo.`
`15:08.30`	`Zeeek`	`here we go....`
`15:08.52`	`DrPete_`	`is there a way to connect to an asterisk process, but see colours?`
`15:09.06`	`Zeeek`	`One thing not many people think about is that the IAXy seem to allow no way to change volume or ring certain european phones.`
`15:09.18`	`Zeeek`	`DrPete LSD?`
`15:09.18`	`RoyK`	`<PROTECTED>`
`15:09.18`	`redder86`	`Zeeek: I've found that the single RJ-45 port phones end up being a nuisance. The extra port there is so nice.`
`15:09.26`	`DrPete_`	`Zeeek: hehe`
`15:09.30`	`redder86`	`Zeeek: especially if it's 10/100`
`15:09.35`	`Zeeek`	`redder86 the extra port on what?`
`15:09.47`	`redder86`	`Zeeek: the two-port hub on phones`
`15:09.54`	`bjohnson`	`tzafrir: I use it`
`15:10.00`	`*** part/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net)`
`15:10.00`	`Zeeek`	`oh phones. Ya I have a BT101 and the IAX phone has a hub model as well`
`15:10.11`	`redder86`	`yah`
`15:10.13`	`Zeeek`	`$10 for the extra RJ45`
`15:10.16`	`redder86`	`But that's $69`
`15:10.29`	`Zeeek`	`I'm asking them about shipping for one phone, one ATA`
`15:10.34`	`bjohnson`	`tzafrir: it is made so that you have multiple superdial calls in your context .. if one fails it tries the next. Works for both outgoing and incoming logic`
`15:10.36`	`DrPete_`	`the ciso I just got had a 10/100 switch, it works great`
`15:10.52`	`Zeeek`	`$120 plus shipping, if it's like $50 that isn't bad IF THEY WORK OK`
`15:10.55`	`nestAr`	`anyone in here doing shared registration with a polycom?`
`15:11.01`	`nestAr`	`well. 2 polycom's..`
`15:11.34`	`*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)`
`15:11.55`	`ManxPower`	`nestAr, I used to. Broke transfers. Stoped using it.`
`15:12.18`	`nestAr`	`hrmm`
`15:12.19`	`nestAr`	`ok`
`15:12.33`	`Zeeek`	`DrPete you went from being too cheap to buy an X100P to a cisco in a few weeks?`
`15:12.58`	`*** join/#asterisk LoRez (lorez@lorez.staff.freenode)`
`15:13.05`	`DrPete_`	`Zeeek: yeah, terrible init`
`15:13.20`	`DrPete_`	`I could not resist temptation`
`15:13.24`	`tzafrir`	`bjohnson, but I don't see exactly where it bails out if something is wrong with the line`
`15:13.28`	`*** join/#asterisk LoRez (lorez@lorez.staff.freenode)`
`15:13.29`	`Zeeek`	`no, typical - addictive personality disorder - we all have that here`
`15:13.40`	`DrPete_`	`Its an awsome phone. I need to play later thou,i need food now`
`15:13.45`	`DrPete_`	`bye everyon`
`15:13.50`	`Mavvie`	`so. enum works better now.`
`15:13.54`	`Zeeek`	`I'm already having a hard time explaining mine to my partner`
`15:13.56`	`*** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net)`
`15:19.49`	`bjohnson`	`tzafrir: it ends quietly and the next priority in your main context is run`
`15:20.34`	`tzafrir`	`I figure that what I need is a small wrapper around ChanIsAvail . Seems like it's time to write my first agi script`
`15:22.00`	`*** join/#asterisk Wireless (~bad@220.233.77.87)`
`15:22.28`	`adjacent`	`when i take an ATA off the hook, i get dialtone. so i believe it is successfully registered with my sip server. however wheni dial a number it pauses, then gives a fast busy.`
`15:22.45`	`Wireless`	`can someone sip 70600@dsl.badname.net for an echo test, need to do a NAT debugging.`
`15:22.47`	`adjacent`	`i dont think calls are being forwarded to my cisco voice gateway properly.`
`15:22.58`	`bjohnson`	`adjacent: dial tone means it's plugged in`
`15:23.00`	`adjacent`	`any idea how i test this?`
`15:23.19`	`adjacent`	`bjohnson: in a ata 186? im pretty sure you get no tone unless its registered`
`15:23.26`	`bjohnson`	`you've got a configuration problem somewhere`
`15:23.50`	`bjohnson`	`adjacent: check the server to see if it's registered.`
`15:23.57`	`adjacent`	`bjohnson: ok`
`15:24.37`	`adjacent`	`ill figure out how to do that =)`
`15:24.37`	`bjohnson`	`btw .. I'm not familiar with that particular hardware`
`15:24.37`	`adjacent`	`my damn voip provider got so bad we had to drop them, now im in the deep end of the pool learning to swim`
`15:24.37`	`riksta`	`has anyone been using my ADM tray?`
`15:24.51`	`*** join/#asterisk sabre (sabre@69.149.209.83)`
`15:24.53`	`bjohnson`	`riksta: what is it?`
`15:25.22`	`riksta`	`http://adm.hamnett.org`
`15:25.29`	`*** join/#asterisk oej (~oej@apollo.webway.se)`
`15:25.35`	`riksta`	`i have a new release ready, i want someone to test it`
`15:25.56`	`*** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230)`
`15:28.22`	`jetscreamer`	`heh.. the 'view documentation' link`
`15:28.46`	`jetscreamer`	`or lack thereof`
`15:29.00`	`adjacent`	`heh. ManxPower, yeh. except im not paying my SIP provider anymore so its only going to work for a little while longer ;)`
`15:29.32`	`Madkiss`	`How would I call 01234567@sip.sil.at via asterisk?`
`15:30.01`	`jetscreamer`	`looks cool though`
`15:30.17`	`riksta`	`jetscreamer: i'm working on it ;)`
`15:32.58`	`bjohnson`	`Madkiss: with the dial command`
`15:34.01`	`oej`	`madkiss: dial(SIP/01234567@sip.sil.at)`
`15:34.14`	`oej`	`madkiss: Make sure you enable the DNS srv lookup in sip.conf`
`15:34.38`	`bjohnson`	`riksta: what is it for? Each user manages their extension call forwarding, etc?`
`15:35.06`	`Madkiss`	`uhm, I see the problem. I have "'_.' => 1. Dial(SIP/${EXTEN}@freenet\|120\|t) [pbx_config]" in my dial-plan`
`15:35.46`	`liquidno2`	`Madkiss: I ran into that problem last night`
`15:35.47`	`riksta`	`bjohnson: yeah some of the features are listed on the front page, i can't be assed to type them here`
`15:35.54`	`Madkiss`	`liquidno2: How did you solve it?`
`15:36.02`	`liquidno2`	`_X.`
`15:36.11`	`Madkiss`	`_X.?`
`15:36.23`	`Nugget`	`_. is dangerous. Don't use it unless you're familiar with the dangers.`
`15:36.30`	`liquidno2`	`yeah`
`15:36.37`	`liquidno2`	`it catches everything`
`15:36.45`	`liquidno2`	`from fax to s to h to t`
`15:36.52`	`Nugget`	`if ${EXTEN} is always going to be numeric (which isn't necessarily the case) it's better to use _X.`
`15:37.19`	`Madkiss`	`liquidno2: Okay, is there anything else I have to add for special SIP calls?`
`15:38.16`	`liquidno2`	`That particular issue solved my problems last night`
`15:38.25`	`Madkiss`	`Only that one character?`
`15:38.32`	`bjohnson`	`riksta: yes. I just don't understand what most of those mean`
`15:38.44`	`riksta`	`oh well`
`15:39.06`	`JunK-Y`	`ive a gate here dialing another machine, how can that machine know what's the uniqueid of the 1st machine?`
`15:39.38`	`bjohnson`	`you mention bluetooth though and I need a mentor for hooking up a bt headset. available?`
`15:40.29`	`riksta`	`bjohnson: that's buggy at the moment`
`15:41.31`	`greg_work`	`you know, i never did find a voip provider`
`15:41.57`	`Sauron	`greg: For what?`
`15:42.06`	`*** part/#asterisk FirstSword (~Miranda@host6614613596.biz.tor.fcibroadband.com)`
`15:42.07`	`greg_work`	`its been over two weeks since i've heard from iax.cc when they told me they're still waiting for 800 numbers, and 3 weeks since i requested one`
`15:42.15`	`Sauron	`bjohnson: BT headset for *?`
`15:42.16`	`greg_work`	`nufone never even returned my email`
`15:42.31`	`*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)`
`15:42.43`	`HitTop`	`hi all~Good morning!~`
`15:42.46`	`greg_work`	`Sauron: canadian 800 number, and just decent rates to can/usa
`15:42.50`	`Sauron	`Ah, hehn.`
`15:42.56`	`Sauron	`what's up buddy`
`15:43.02`	`liquidno2`	`nada man...`
`15:43.09`	`Sauron	`greg: Ahha.`
`15:43.10`	`liquidno2`	`see you idle in here and was like no way`
`15:43.17`	`liquidno2`	`so I had to whois you to be sure :)`
`15:43.19`	`HitTop`	`I wonder how voip can save money for toll-free`
`15:43.24`	`Sauron	`greg: I forget you're a canuck`
`15:43.26`	`Sauron	`;)`
`15:43.49`	`greg_work`	`Sauron: i realize my use of "eh?" is under par. i'll try and fix that
`15:43.56`	`Sauron	`hehe`
`15:44.12`	`greg_work`	`anyways, aboot that voip provider...`
`15:44.13`	`liquidno2`	`Sauron: using * at the hospital? or just personal?
`15:44.27`	`Sauron	`personal, but we're trying to weasel it into work`
`15:44.32`	`liquidno2`	`nice`
`15:44.52`	`Sauron	`Our crapola cisco conference machine is broken, and the licensing for it is insane`
`15:45.09`	`Sauron	`I'd mocked up a conference thing on my home machine the other night as a test, and we got to talking`
`15:45.10`	`liquidno2`	`I see an in for meetme`
`15:45.20`	`Sauron	`so we'll prolly see about setting up * for meetme/conferencing`
`15:46.11`	`liquidno2`	`I keep seeing more and more "fringe" use of * in the corporate world`
`15:46.40`	`Sauron	`unfortunately, first I gotta get * to work with cisco callmanager :(`
`15:46.49`	`Sauron	`I already feel dirty.`
`15:47.07`	`liquidno2`	`yeah. I read that and got a pit in my stomach`
`15:47.21`	`bjohnson`	`Sauron: I was thinking bt headset for linux workstation. Then it would work with any softphone
`15:47.21`	`AgiNamu`	`anyone here know how asterisk negotiates codecs?`
`15:47.23`	`tzanger`	`stevekstevek: around??`
`15:47.30`	`AgiNamu`	`im having a real bitch trying to get things working`
`15:48.08`	`tzanger`	`AgiNamu: what are you looking for`
`15:48.08`	`bjohnson`	`greg_work: I haven't found a voip provider for canadian 800 numbers that beats our telco (Allstream) at $0.049 / minute`
`15:48.32`	`tzanger`	`bjohnson: do you know of any CLECs that can offer stratford DIDs?`
`15:48.44`	`AgiNamu`	`tzanger, when receiving an incoming call, it does not check who it's going to dial`
`15:48.48`	`tzanger`	`bjohnson: I'm looking for someone with tandem to the Bell switch in Stratford`
`15:48.55`	`AgiNamu`	`thus, if I have 2 clients, one in GSM and one in G729`
`15:48.57`	`bjohnson`	`tzanger: no .. but I don't know what a CLEC is`
`15:48.57`	`tzanger`	`AgiNamu: huh?`
`15:48.58`	`Sauron	`bjohnson: you installed chan_bluetooth yet?`
`15:49.16`	`AgiNamu`	`and an incoming call comes in, I'll get whatever it feels like negotiating, and then end up transcoding`
`15:49.24`	`AgiNamu`	`and that's lame.`
`15:49.25`	`bjohnson`	`Sauron: for ? No. This isn't for on my server
`15:49.29`	`Sauron	`Oh.`
`15:49.35`	`Sauron	`I thought you were talking about BT headset for *`
`15:49.49`	`bjohnson`	`tzanger: Allstream provides us with local numbers.`
`15:50.01`	`AgiNamu`	`like this: [incoming] exten => 123,1,DIAL(IAX2/GsmUser) exten => 456,1,DIAL(IAX2/G729User)`
`15:50.21`	`AgiNamu`	`when the incoming call comes in, it doesn't use GSM or G729, it uses whichever is the fisrt allow= in my iax.conf :@`
`15:50.23`	`JunK-Y`	`when transfering a call, is there a way the 2nd machine change the value of uniqueid.`
`15:50.34`	`bjohnson`	`tzanger: they have something on their web site about voip .. looks like they want to supply voip providers`
`15:50.38`	`AgiNamu`	`what I want it to do is figure out that it's calling a GSM user, and set its preferred codec to GSM`
`15:51.13`	`tzanger`	`bjohnson: allstream eh`
`15:51.21`	`bjohnson`	`Sauron: just a bt headset for my linux desktop. the dongle seems to be sensed but I can't get it to scan the headset
`15:51.23`	`tzanger`	`bjohnson: yeah I have pricing on them for a "VOIP PRI"`
`15:51.35`	`Sauron	`Oh.`
`15:51.40`	`Sauron	`hcitool scan`
`15:51.42`	`*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)`
`15:51.50`	`Sauron	`if the headset is in discoverable mode, that should find it`
`15:51.57`	`bjohnson`	`tzanger: there pricing seems to be very high .. but for some reason they have great local and ld rates`
`15:52.12`	`bjohnson`	`Sauron: yeah .. can't get that to work
`15:52.34`	`Sauron	`define " can't get that to work"`
`15:55.58`	`*** join/#asterisk mhnoyes (~mhnoyes@user-38lc0cb.dialup.mindspring.com)`
`15:56.07`	`*** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com)`
`15:56.50`	`*** join/#asterisk Hmmhesays (~Hmmmhesay@66.173.103.108)`
`15:57.02`	`HitTop`	`does anyone know if tdm04p can fit into a foekris board?`
`15:57.11`	`Hmmhesays`	`anyone using spandsp?`
`15:57.26`	`Hmmhesays`	`haha nm i think i just solved my problem`
`15:57.59`	`tzanger`	`yeah`
`15:58.19`	`*** join/#asterisk Realw5 (~dasd@adsl-69-226-156-174.dsl.chi2ca.pacbell.net)`
`15:58.21`	`tzanger`	`their "voip pri" pricing wasn't bad but as I understood it it was a "hosted PRI" and they sent you the data over the intarweb`
`15:58.46`	`Realw5`	`Hey everyone`
`15:58.56`	`*** join/#asterisk brettnem (~brettnem@208.54.232.29)`
`15:58.57`	`Realw5`	`came someone help me with a hardware questions?`
`15:59.09`	`Nugget`	`not unless you actually ask it, no.`
`15:59.37`	`HitTop`	`Hmmhesay: im using sandsp`
`16:00.25`	`bjohnson`	`HitTop: from what I read .. no. plus some soekris boards use 3.x volts not 5. also, some soekris don't have the horsepower for *`
`16:00.29`	`HitTop`	`does anyone know if tdm400p can fit into a soekris board (net 45xx or 48xx)?`
`16:01.34`	`HitTop`	`slow isn't a problem.. i think... as long as its functioning.. so.. is there no other zaptel card for 3.3v?`
`16:01.44`	`Realw5`	`heh, that might help. Will this 16 port station (Dialogic - MSI160-PCI GBL) card work with Asterisk?`
`16:02.45`	`Luhiwu`	`i have a question about , SIP and Zap: I receive a call from a SIP client that wants to call an external number, get the call and dials thru Zap ok, but even if the zap is never answered i get a cdr record for the call as 'answered', is there any way to change that? i don't want to send 200 OK until the zap channel has been answered`
`16:03.14`	`eKo1`	`Luhiwu: ResetCDR?`
`16:03.58`	`Luhiwu`	`eKo1, my main problem is that the call is coming from another sip proxy and when i return 200 OK and the call hasn't been answered on the Zap channel, the originator is billed for the call.`
`16:04.20`	`eKo1`	`Luhiwu: Hah, then it's not your problem.`
`16:04.32`	`Luhiwu`	`my problem is not the cdr, but the 200 OK that * sends before connecting the 2 legs.`
`16:05.05`	`*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)`
`16:05.05`	`*** mode/#asterisk [+o anthm] by ChanServ`
`16:05.30`	`Luhiwu`	`eKo1, do you know if it is possible to send call progress from Zap to SIP?`
`16:05.58`	`eKo1`	`Luhiwu: What you have described is standard behavior.`
`16:06.11`	`eKo1`	`It's your billing that sucks.`
`16:06.50`	`Luhiwu`	`may be, but i shouldn't send a 200 OK but instead 180 Ringing until the zap channel get answered, or am i wrong?`
`16:07.31`	`eKo1`	`Well, what is the 200 OKing?`
`16:07.47`	`*** join/#asterisk glock (~glock@rrba-146-104-36.telkomadsl.co.za)`
`16:08.16`	`Luhiwu`	`eKo1: the INVITE for a 541155555555 incoming call`
`16:08.43`	`Luhiwu`	`it sends 100 Trying and then 200 OK, and after that does the Dial(Zap/1/541155555555)`
`16:09.01`	`glock`	`Anyone please help me with fax detection? In my extensions.conf i have exten => fax,1,Dial,Zap/4 exten => s,1,Dial,Zap/1. Although it rings on zap1 even if it is a fax. any idea?`
`16:09.34`	`riksta`	`postel: alive?`
`16:09.40`	`JunK-Y`	`is there any way to change the value of my uniqueid???`
`16:09.57`	`*** join/#asterisk bkw_ (~brian@65.38.28.152)`
`16:09.57`	`*** mode/#asterisk [+o bkw_] by ChanServ`
`16:11.05`	`*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released \|\| development conf call today 2pm CST IAX2/66.250.68.194/996`
`16:11.44`	`*** join/#asterisk kore (kore@mindwipe.org)`
`16:12.28`	`eKo1`	`Oh!`
`16:12.35`	`eKo1`	`IAX conference?!`
`16:12.52`	`Nugget`	`spiffy`
`16:12.54`	`*** join/#asterisk jonp (~jon@81-178-75-91.dsl.pipex.com)`
`16:13.14`	`eKo1`	`1400 Central should be...the same time here.`
`16:14.03`	`Nugget`	`CST is UTC-5.`
`16:14.07`	`Nugget`	`er, UTC-6`
`16:14.18`	`eKo1`	`Yep.`
`16:15.11`	`jonp`	`Hi all. I'm new to this. Sorry for the interruption. I'm using attended transfer feature from CVS, and think I may have found a bug. Is anyone here familiar with this feature?`
`16:15.15`	`*** join/#asterisk eipi (~eipi@OL128-44.fibertel.com.ar)`
`16:15.20`	`Luhiwu`	`eKo1: do you know who can i ask about not sending the OK till receive an answer?`
`16:15.24`	`*** join/#asterisk WildPikachu[BAR] (~wildpikac@wildpikachu.user)`
`16:15.43`	`WildPikachu[BAR]`	`ok.... time to try get my linphone to work with asterisk`
`16:15.50`	`WildPikachu[BAR]`	`anyone got an example of how to setup sip?`
`16:15.59`	`eKo1`	`Luhiwu: Are you trying to break SIP or what?`
`16:16.04`	`*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released \|\| development conf call today 2pm CST IAX2/66.250.68.194/996 #astersk-dev`
`16:16.30`	`Luhiwu`	`eKo1: i just want to pass callprogress from Zap to SIP, is it too much?`
`16:16.59`	`Nugget`	`Luhiwu: the ability to detect answer on zap is unreliable.`
`16:17.13`	`eKo1`	`Luhiwu: Zap isn't incharge of sending call progress through SIP.`
`16:17.37`	`WildPikachu[BAR]`	`ok..... i have linphone and asterisk, now... what is the first thing to do to get an outgoing call up and running?`
`16:18.13`	`*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)`
`16:18.25`	`AgiNamu`	`is there a way to populate the asterisk DB via a .conf file?`
`16:18.40`	`eKo1`	`AgiNamu: Which DB?`
`16:18.48`	`AgiNamu`	`eko1, the asterisk internal db`
`16:19.00`	`eKo1`	`AgiNamu: eh, no.`
`16:19.07`	`brettnem`	`hey anthm, how you updated app_intercept to merge with head by any chance?? :)`
`16:19.08`	`AgiNamu`	`oh. how do you persist anything then?`
`16:19.27`	`AgiNamu`	`I want to associate a number (DID) with a codec`
`16:19.32`	`AgiNamu`	`to dynamically negotiate codecs.`
`16:19.38`	`AgiNamu`	`on incoming calls`
`16:19.53`	`*** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk)`
`16:20.14`	`brettnem`	`AgiNamu: you can have asterisk call itself with a particular sip username that has access to only one codec? sloppy.. yes`
`16:20.35`	`Luhiwu`	`eKo1: so i can't hold the 200 OK until know if the channel is answered or busy? no one makes termination of sip over zap here?`
`16:20.40`	`*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)`
`16:21.09`	`Luhiwu`	`gotta go, thanks for your help`
`16:21.21`	`anthm`	`brettnem, http://www.asterlink.com/asterisk/app_intercept.c`
`16:21.30`	`brettnem`	`wahoo.. thanks! :)`
`16:21.48`	`brettnem`	`btw, what happened to it? not enough support to include it into core?`
`16:21.52`	`BoRiS`	`anthm!!!!!`
`16:21.54`	`AgiNamu`	`brettnem, no, i cant`
`16:22.05`	`AgiNamu`	`incoming calls come from my provider, and he doesnt know which codec i want forf a specific DID`
`16:22.07`	`brettnem`	`AgiNamu: well it was just an idea... hmm`
`16:22.23`	`AgiNamu`	`and i think he gets it from some big hardware`
`16:22.27`	`brettnem`	`agiNamu app_transfer?`
`16:22.32`	`brettnem`	`to yourself`
`16:22.52`	`AgiNamu`	`the call is already negotiated by that point`
`16:22.59`	`AgiNamu`	`and i dont think transfers renegotiate....`
`16:23.02`	`brettnem`	`hm`
`16:23.10`	`AgiNamu`	`i.e., Provider SENDS a call to my IP`
`16:23.11`	`brettnem`	`yeah they don't I don't think..not without a reinvite`
`16:23.16`	`AgiNamu`	`this is IAX`
`16:23.19`	`AgiNamu`	`not SIP :)(`
`16:23.22`	`brettnem`	`OH`
`16:23.23`	`brettnem`	`hm`
`16:23.32`	`AgiNamu`	`afaik, i have to go make a patch`
`16:23.33`	`tzanger`	`I swear my zap echo canceller is disabled`
`16:23.33`	`AgiNamu`	`which is fine`
`16:23.46`	`tzanger`	`zap show channel [channelno] shows 128 taps and ON but it can't be`
`16:23.49`	`brettnem`	`blah echo cans`
`16:23.54`	`tzanger`	`brettnem: yeah yeah I know`
`16:24.03`	`brettnem`	`you sure it isn't the other side??`
`16:24.09`	`eKo1`	`Dang it, why isn't insecure=very working for me.`
`16:24.10`	`tzanger`	`brettnem: if I hear echo it is the other side`
`16:24.12`	`glock`	`Anyone please help me with fax detection? In my extensions.conf i have exten => fax,1,Dial,Zap/4 exten => s,1,Dial,Zap/1. Although it rings on zap1 even if it is a fax. I've tried exten => s,1,Answer - But then it doesnt ring anywhere. any idea?`
`16:24.15`	`tzanger`	`but the echo can should be killing that`
`16:24.16`	`brettnem`	`right`
`16:24.21`	`WildPikachu[BAR]`	`is there a howto to setup sip?`
`16:24.31`	`brettnem`	`but I don't think that's how the cans are supposed to work..`
`16:24.33`	`brettnem`	`really`
`16:24.42`	`brettnem`	`WildPikachu yes, look at the wiki`
`16:24.56`	`eKo1`	`argh argh argh`
`16:24.57`	`bjohnson`	`tzanger: mornington communications in Milverton might have DIDs you could use for Stratford`
`16:25.07`	`brettnem`	`the cans are supposed to prevent echo from ever going back onto the line.. It should be on the far end so the echo never traverses the link`
`16:25.09`	`WildPikachu[BAR]`	`brettnem, i'm there now`
`16:25.12`	`tzanger`	`bjohnson: hmm they're the enemy :-)`
`16:25.15`	`brettnem`	`granted, it'd work on oth`
`16:25.17`	`brettnem`	`+b`
`16:25.19`	`WildPikachu[BAR]`	`i don't quite understand how sip works`
`16:25.32`	`tzanger`	`brettnem: ahhh that might explain it`
`16:25.42`	`tzanger`	`if I am hearing echo the far end needs a can`
`16:25.45`	`brettnem`	`Have you tried reading the RFC? it's pretty readable if I remember right`
`16:25.48`	`shido6`	`glock u told it to answer`
`16:25.51`	`brettnem`	`tzanger: right`
`16:25.52`	`shido6`	`but didnt tell it where to go`
`16:25.53`	`tzanger`	`if they hear echo I need the can, and my ecan seems to work then`
`16:26.02`	`brettnem`	`right`
`16:26.02`	`tzanger`	`brettnem: or you need a nice hardware echo can that does it in both directions`
`16:26.11`	`tzanger`	`whcih I was under the impression that the zap software echo cans could do as well`
`16:26.15`	`brettnem`	`yeah, and that's pretty compilcated actually`
`16:26.22`	`tzanger`	`brettnem: oh I realize that :-)`
`16:26.27`	`brettnem`	`:)`
`16:26.31`	`anthm`	`WildPikachu[BAR], it's simple you configure it, it doesn't work as expected, then you swear at the PC and rip out all your hair.`
`16:26.40`	`brettnem`	`really, the right way is to have the other end have the cans..`
`16:26.57`	`brettnem`	`damn hybrid coupling`
`16:27.01`	`tzanger`	`brettnem: well that ain't gonna happen since my zap span is on a Bell Canada PRI`
`16:27.06`	`tzanger`	`and they claim they never put echo cans on PRIs`
`16:27.15`	`brettnem`	`how far is the switch from you?`
`16:27.25`	`WildPikachu[BAR]`	`ok.... my linphone wants a sip address, it looks like asterisk is bitching about a username... /me goes to read the wiki about adding users`
`16:27.25`	`tzanger`	`brettnem: the switch? about 60km or so`
`16:27.26`	`brettnem`	`hmm..what's the echo rule? 700 miles?`
`16:27.42`	`anthm`	`hmm cans, that gives me an idea , how bout chan_cans_and_string`
`16:27.45`	`tzanger`	`the CO is only about 500m`
`16:28.04`	`brettnem`	`WildPikachu: perhaps you should learn what SIP does instead of learning asterisk first`
`16:28.07`	`brettnem`	`hmm`
`16:28.18`	`brettnem`	`heh`
`16:28.42`	`brettnem`	`and are you making a local or LD call?`
`16:28.51`	`bkw_`	`brb`
`16:29.02`	`brettnem`	`pri to a switch shouldn't have any impedence matching problems.`
`16:29.11`	`tzanger`	`brettnem: of course not`
`16:29.11`	`brettnem`	`especially at 500m`
`16:29.24`	`brettnem`	`so ld or local?`
`16:29.28`	`tzanger`	`brettnem: anything really`
`16:29.38`	`brettnem`	`anything? what about when you call yourself?`
`16:29.42`	`tzanger`	`brettnem: it depends heavily on the destination number (or source number)`
`16:29.56`	`tzanger`	`i.e. if they call aDID on my PRI it depends on where they're coming from`
`16:30.01`	`tzanger`	`or if I call a # on my PRI it depends on the destination`
`16:30.02`	`brettnem`	`well it should`
`16:30.05`	`brettnem`	`right`
`16:30.12`	`tzanger`	`some are completely clean, and some are horrible`
`16:30.18`	`brettnem`	`so I'm talking about the nature of the other end of the call.. ld or local? or both?`
`16:30.23`	`tzanger`	`I can call sivana's house through the pri and it's shitty`
`16:30.29`	`tzanger`	`brettnem: local and ld`
`16:30.36`	`brettnem`	`haha then it's sivana's fault.. :P`
`16:30.48`	`tzanger`	`brettnem: nah, it's his LEC's switch`
`16:30.52`	`coppice`	`why would ld or local make much difference?`
`16:30.55`	`brettnem`	`so even if you call yourself?`
`16:30.55`	`tzanger`	`or rather CO`
`16:31.01`	`tzanger`	`brettnem: not so much if I call myself`
`16:31.12`	`brettnem`	`coppice: well it's who we are pointing the finger at.. the ld provider or the lec`
`16:31.13`	`tzanger`	`but if I call myself there are echo cans on both sides :-)`
`16:31.22`	`brettnem`	`LEC's dont usually use cans on anything.`
`16:31.58`	`brettnem`	`so it's most likely an IXC issue.. there are lots of crappy IXCs out there that just don't do the echo can thing`
`16:32.10`	`Essobi`	`anyond used soxmix to retain originating and destications in an audio source by mixing on the left and the right?`
`16:32.12`	`brettnem`	`coppice, what do you think. You know more about this than I do`
`16:32.19`	`coppice`	`its just the hybrid at the other end`
`16:32.20`	`Essobi`	`The manual for soxmix reads like a tv repair handbook.`
`16:32.42`	`stevekstevek`	`hey Steve..`
`16:32.59`	`brettnem`	`but digital end-to-end (no hybrid other than the handset) will still experience echo sometimes, right?`
`16:33.00`	`tzanger`	`brettnem: this is all in Bell's netowrk I think`
`16:33.02`	`WildPikachu[BAR]`	`hrmmmmm`
`16:33.16`	`brettnem`	`you in canada?`
`16:33.17`	`WildPikachu[BAR]`	`i setup my linphone in sip.conf`
`16:33.24`	`WildPikachu[BAR]`	`although i think thats how one adds users?`
`16:33.33`	`brettnem`	`what is a "user"?`
`16:33.46`	`brettnem`	`WildPikachu: looks like you are jumping in a little too quick`
`16:33.54`	`WildPikachu[BAR]`	`sip user..... sip:user@ip ?`
`16:33.56`	`WildPikachu[BAR]`	`:(`
`16:33.56`	`tzanger`	`coppice: the question though is if it's the far-end hybrid, why do I get no echo at all when I call them with a POTS line as opposed to my PRI`
`16:34.03`	`brettnem`	`WildPikachu: I think you'll enjoy your asterisk experience more if you learn how SIP works`
`16:34.08`	`coppice`	`you only need one hybrid to get echo`
`16:34.18`	`WildPikachu[BAR]`	`i searched wiki but i can't find what i'm after`
`16:34.24`	`WildPikachu[BAR]`	`:(`
`16:34.32`	`tzanger`	`coppice: right... but again... if it is their end causing the echo should I not technically get it with a POTS line as well as PRI?`
`16:34.32`	`coppice`	`tzanger: you get echo. you just don't hear echo`
`16:34.32`	`brettnem`	`~google SIP RFC`
`16:34.59`	`*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)`
`16:34.59`	`*** topic/#asterisk is Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released \|\| development conf call today 2pm CST IAX2/66.250.68.194/996 #astersk-dev`
`16:35.05`	`Essobi`	`Long time coppice`
`16:35.10`	`Essobi`	`How's it going?`
`16:35.17`	`brettnem`	`well interestingly enough, you don't know what that PRI is doing (network infrastructure wise)`
`16:35.20`	`coppice`	`echo is only a problem with VoIP or cellular. the same thing with POTS calls is called reverberation`
`16:35.29`	`tzanger`	`coppice: hmm`
`16:35.48`	`tzanger`	`so basically just suck it up and get a hardware 2-way 64ms echo can wtih T.8 (I think?) tone detection`
`16:35.51`	`bkw_`	`hey guys check this kewl stuff out http://bugs.digium.com/bug_view_page.php?bug_id=3550`
`16:35.56`	`brettnem`	`coppice: what is the root cause in VoIP networks?`
`16:35.57`	`Essobi`	`reverb is an electromechanical problem ain't it?`
`16:36.04`	`jaiger`	`does anyone have a couple polycom phones that you could run a simple test case for me?`
`16:36.10`	`coppice`	`you always get the echo. if the delay is really short you perceive it as reverb and it sounds nice`
`16:36.14`	`tzanger`	`Essobi: it's an effect on my guitar amp :-)`
`16:36.18`	`Essobi`	`Tehe.`
`16:36.23`	`brettnem`	`bkw: do you have a description of how to use it?`
`16:36.24`	`j_f00b3r`	`turn up the reverb knob!`
`16:36.32`	`coppice`	`VoIP and cellular cause big delays`
`16:36.38`	`tzanger`	`coppice: yes I understand that -- but what I'm saying is that I didn't think that there'd be that much additional delay in the PRI and Zap system to show the echo`
`16:36.45`	`*** join/#asterisk sneak (~sneak@198.22.65.197)`
`16:36.50`	`tzanger`	`but I suppose even 25ms would be very noticeable`
`16:36.54`	`sneak`	`hi`
`16:36.55`	`bkw_`	`brettnem, it has and example`
`16:37.06`	`coppice`	`zap works in 20ms chunks. it always adds considerable delay`
`16:37.10`	`WildPikachu[BAR]`	`brettnem, so i would define my linphone as an extension?`
`16:37.24`	`brettnem`	`bkw_: I think I see.. that's pretty cool.. what's the rational as why this is better than just using a dial plan? more effecient?`
`16:37.41`	`tzanger`	`ahh okay`
`16:37.51`	`tzanger`	`as always, coppice, speaking with you is enlightening :-)`
`16:37.52`	`*** join/#asterisk zno (~zeno@ip-160-79-174-100.autorev.intellispace.net)`
`16:38.25`	`tzanger`	`bkw_: wtf does 3550 have to do with 3487??`
`16:38.26`	`brettnem`	`bkw_: So how's that IMAP storage for VM going??? ;)`
`16:38.36`	`coppice`	`cellular systems work in 20ms chunks, too. If the echo canceller on a GSM call screws up it sounds awful`
`16:38.48`	`tzanger`	`coppice: we use gsm codec exclusively`
`16:38.54`	`tzanger`	`ilbc always sounds like ass for some reason`
`16:39.03`	`tzanger`	`Outstanding frames: 2167 (2165 ingress, 2 outgress)`
`16:39.05`	`tzanger`	`whoohoo`
`16:39.07`	`bjohnson`	`WildPikachu: just make a guest entry in sip.conf for now (no secret)`
`16:39.10`	`j_f00b3r`	`because ilbc always sounds like ass ;)`
`16:39.19`	`brettnem`	`"sounds like ass"`
`16:39.41`	`coppice`	`well, the GSM codec adds 20ms on top of the 20ms due to the zap channels. GSM networks tend to avoid that, and have only one 20ms delay`
`16:40.15`	`bjohnson`	`WildPikachu: define a context for incoming calls from guest to go to. make sure there is a s,1 extension in that context. use your linphone to dial SIP/500@ipofasterisk and see what you get`
`16:40.31`	`ManxPower`	`~doc`
`16:40.34`	`ManxPower`	`~docs`
`16:40.35`	`jbot`	`Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org`
`16:40.39`	`jaiger`	`tzanger: you get the echo canceller yet?`
`16:41.10`	`tzanger`	`jaiger: not yet`
`16:41.51`	`WildPikachu[BAR]`	`bjohnson, thanks man... i changed the sip config in configs/ directory not my asterisk config ... feels stupid`
`16:42.28`	`coppice`	`even GSM handset to GSM handset needs EC. enough voice spills from the earpiece to the mic to cause trouble without it.`
`16:43.35`	`brettnem`	`hey anthm. Can you think of anyway to match up the CDR records generated with app_intercept. It looks like a totally new call and the original call actually ends with a BUSY`
`16:44.13`	`tzanger`	`coppice: does g729 have the same kind of "problems" (not really problems, i know, just the nature of thebeast)`
`16:44.26`	`anthm`	`yah its the curse of the masqueraded channel`
`16:44.40`	`brettnem`	`WildPikachu[BAR]: Try /me feels stupid instead.. ;) hehe hey don't worry we all start pretty green.. someday you'll be helping people out`
`16:44.50`	`WildPikachu[BAR]`	`:-)`
`16:44.57`	`*** join/#asterisk outsidefactor (barf@203-206-254-16.dyn.iinet.net.au)`
`16:44.58`	`brettnem`	`Doh!`
`16:45.04`	`WildPikachu[BAR]`	`i would love to help ppl out with asterisk... just must get into it in my spare time`
`16:45.17`	`adjacent`	`ok. ive almost got my system up and running. still getting a fast busy when i dial. but after watching the SIP traffic between the sip server and the pstn gateway im seeing.`
`16:45.18`	`anthm`	`I could probably make it clone the cdr from the real channel`
`16:45.35`	`brettnem`	`WildPikachu: really tho.. reading the SIP RFC would probably help`
`16:45.36`	`coppice`	`any complex codec works in blocks. Any VoIP channel works in blocks. Many radio channels work in blocks. As soon as you leave the traditional PSTN echo is a problem on even the shortest range calls.`
`16:45.51`	`adjacent`	`INVITE.sip:number Trying... 404.Not.Found ACK`
`16:45.58`	`WildPikachu[BAR]`	`asterisk now sees the call comming in on context "test" ... but then i get this... Stopping retransmission on '3121626364@192.168.3.242' of Response 20: Found`
`16:46.07`	`brettnem`	`anthm: yeah, I'm not sure how it should look. there is a part of me that thinks it's ok.. should it look like an inbound or outbound call?? heh`
`16:46.29`	`adjacent`	`however the number is correct but the VG cant seem to find it =/`
`16:46.38`	`tzanger`	`coppice: well no I meant gsm codec + TDM gives you an immedate 40ms lag if I'm understanding what you're saying`
`16:46.43`	`Luke-Jr`	`hmm fun`
`16:47.00`	`Luke-Jr`	`Unlocked PAP2-NA for $50 :)`
`16:47.08`	`tzanger`	`ok this new packet loss concealment jitter buffer is definitely making asterisk unstable`
`16:47.32`	`eKo1`	`If I want to receive calls from my sip provider, I need to add a register => ... statement in sip.conf right?`
`16:48.10`	`Luke-Jr`	`eKo1: yep`
`16:48.38`	`eKo1`	`What about authorization? How do I stop * from challenging invites from my sip provider?`
`16:48.38`	`brettnem`	`eKo1: If you don't they'll never know where you are`
`16:48.59`	`eKo1`	`Because it keeps challenging them I never get the call.`
`16:49.03`	`brettnem`	`eKo1: you have to have a login for them. maybe it just matches their IP`
`16:49.17`	`Sauron	`Dum di dum.`
`16:49.26`	`Sauron	`I got the new and improved chan_sccp working.`
`16:49.31`	`Sauron	`Bwahaha. Evil people.`
`16:49.36`	`WildPikachu[BAR]`	`bjohnson, well... asterisk sees it but i get the error/message above`
`16:49.38`	`eKo1`	`I contacted them and they said to turn of authorization.`
`16:49.46`	`brettnem`	`Is it worth using if you could be using cisco+SIP?`
`16:50.02`	`brettnem`	`eKo1: SIP or IAX?`
`16:50.03`	`eKo1`	`sip`
`16:50.18`	`brettnem`	`do you have something in sip.conf for them? host=provider.ip.address.com ?`
`16:50.51`	`brettnem`	`there must be some kind of confirmation of who they are.. user/pw/ip... something.. otherwise anyone could access your PBX`
`16:51.05`	`eKo1`	`Well, I have the register => username:password@sip-provider.com under the [general] section.`
`16:51.06`	`brettnem`	`and if THAT is what you want to do.. just put your DID into your default context`
`16:51.12`	`brettnem`	`that's not enough`
`16:51.17`	`*** join/#asterisk objective (~objective@dialup-4.239.222.201.Dial1.Philadelphia1.Level3.net)`
`16:51.32`	`eKo1`	`And I have a [sip-provider] entry in sip.conf set to friend with insecure=yes,`
`16:51.39`	`objective`	`What's the most economical way to ship 20 Sipura SPAs (~$1400) to Canada?`
`16:51.44`	`brettnem`	`try doing a [sip.provider] type=friend host=sip.provider.com context=sip-in or something like that`
`16:51.49`	`Luke-Jr`	`eKo1: seperate user and peer; remove 'secret' from the incoming one`
`16:51.50`	`eKo1`	`objective: Teletransportation.`
`16:52.03`	`eKo1`	`brettnem: That's exactly what I have.`
`16:52.07`	`jarrod`	`anyone here runnin SER`
`16:52.12`	`brettnem`	`so you specify host?`
`16:52.15`	`tzanger`	`odd`
`16:52.16`	`brettnem`	`and is the call coming form that?`
`16:52.17`	`eKo1`	`brettnem: Yep.`
`16:52.25`	`brettnem`	`jarrod: I've run some SER.. but not a lot`
`16:52.27`	`eKo1`	`Luke-Jr: I'll try that.`
`16:52.34`	`tzanger`	`slackware 9.1, asterisk shows only ONE process in ps ax. slackware 10.0, I have like 12`
`16:52.36`	`eKo1`	`brettnem: Yep.`
`16:52.36`	`Luke-Jr`	`eKo1: that fixed SIPPhone for me :)`
`16:52.41`	`brettnem`	`and you have dialplan entries in the context specified?`
`16:52.52`	`Luke-Jr`	`tzafrir: NPTL?`
`16:52.53`	`eKo1`	`brettnem: Yep.`
`16:53.02`	`eKo1`	`I did a debug and I always get 407`
`16:53.02`	`brettnem`	`tzanger: that has to do with what kernel you are running`
`16:53.03`	`jarrod`	`im wondering if there is a way around asterisk seeing all calls from ser as coming from a single context`
`16:53.03`	`tzanger`	`Luke-Jr: you mean me?`
`16:53.04`	`brettnem`	`2.4 or 2.6`
`16:53.06`	`brettnem`	`I think`
`16:53.08`	`tzanger`	`brettnem: both 2.4`
`16:53.12`	`tzanger`	`I think`
`16:53.12`	`brettnem`	`hmm`
`16:53.12`	`Luke-Jr`	`tzanger: yea`
`16:53.16`	`tzanger`	`NTPL?>`
`16:53.23`	`Luke-Jr`	`tzanger: Native POSIX Threading Lib`
`16:53.35`	`tzanger`	`Luke-Jr: yeah but it's both slackware... should be very similar, no?`
`16:53.42`	`Luke-Jr`	`tzanger: It's new in 2.6 kernels`
`16:53.48`	`tzanger`	`brettnem: 2.4.25 shows about 12 processes and 2.4.26 shows only 1`
`16:53.52`	`Luke-Jr`	`tzanger: ps shows diff info w/ it tho`
`16:53.58`	`brettnem`	`jarrod: perhaps use sip domain names... and check the ${SIPDOMAIN} var`
`16:54.06`	`eKo1`	`Luke-Jr: So you have seperate user/peer entries but do you have register => statement as well?`
`16:54.08`	`WildPikachu[BAR]`	`ok.... i created a context in extensions.conf which answers, and does musiconhold for 10s then hangs up..... when i try connect using SIP it doesn't complain about not finding a context, but i get an error/message :(`
`16:54.09`	`Luke-Jr`	`Not sure if NPTL can be backported to 2.4`
`16:54.14`	`Luke-Jr`	`eKo1: yes`
`16:54.19`	`brettnem`	`I don't know when they changed all that threading stuff.. I thought that was a 2.6 thing`
`16:54.20`	`Luke-Jr`	`eKo1: register is needed to tell them your IP`
`16:54.23`	`brettnem`	`it's ok tho`
`16:54.32`	`Luke-Jr`	`brettnem: it is... maybe backported`
`16:54.41`	`BoRiS`	`I thought NPTL was a Glibc thing...`
`16:54.57`	`brettnem`	`Luke-Jr: you really think that would get turned on by mistake?`
`16:55.00`	`Luke-Jr`	`BoRiS: It's a different 'protocol' between glibc and the kernel`
`16:55.09`	`WildPikachu[BAR]`	`BoRiS, kernel + glibc`
`16:55.14`	`Luke-Jr`	`brettnem: I don't know it's an option`
`16:55.34`	`brettnem`	`ok.. I need to get back to work.. heh.. chat with you guys later.. anth, thanks for the app_intercept patch.. it really should get added to core!!`
`16:56.56`	`jarrod`	`brett`
`16:56.59`	`jarrod`	`thats a thought`
`16:57.14`	`brettnem`	`it's something.. ;)`
`16:57.19`	`Luke-Jr`	`What's the best way to dial the dev conf call?`
`16:57.24`	`brettnem`	`lots of ways to do it tho`
`16:57.31`	`brettnem`	`add some digits at the begining`
`16:57.48`	`Luke-Jr`	`Well, isn't there a standard way? o.o`
`16:58.02`	`AgiNamu`	`fuck, those retarded flag macros are really annoying when reading code`
`16:58.09`	`brettnem`	`Luke-Jr: that was for jarrod.. sorry`
`17:03.26`	`adjacent`	`how do i determine the reason my call INVITE gets a 404 return?`
`17:03.54`	`Zeeek`	`AgiNamu what IAX phone do you have, pls? and hopw do you like it?`
`17:04.09`	`Luke-Jr`	`adjacent: see what extention the call is for`
`17:04.18`	`ManxPower`	`adjacent, 404 usually means the same it does on the web, i.e. the request is not valid.`
`17:04.31`	`adjacent`	`ManxPower: the number i passed is valid...`
`17:04.35`	`ManxPower`	`usually means "I have no clue how to dial the number you just gave me"`
`17:04.50`	`ManxPower`	`adjacent, not according the the remote system, it isn't.`
`17:05.12`	`adjacent`	`ManxPower: right. how do i debug the reason it 404's?`
`17:05.33`	`Luke-Jr`	`The reason it 404s is that you gave it the wrong #`
`17:05.42`	`Luke-Jr`	`there's not much reason besides that`
`17:05.52`	`Zeeek`	`for that matter, what IAX hardphone does anyone have and how do they like it?`
`17:06.10`	`ManxPower`	`adjacent, If the remote system is Asterisk then perhaps you have a CONTEXT ISSUE`
`17:06.30`	`hardwire`	`much better`
`17:06.31`	`*** join/#asterisk moonwick (~moonwick@core.dump.net)`
`17:06.50`	`*** join/#asterisk tessier_ (~treed@146.82.146.22)`
`17:06.55`	`adjacent`	`Luke-Jr: no its not. the number is valid. it works from other phones. even other VOIP phones. ive read that things like a codec mismatch can cause this too`
`17:07.16`	`adjacent`	`ManxPower: its a VOCAL provisioning/marshalling server connectting to a cisco voice gateway`
`17:07.53`	`*** join/#asterisk Hmmhesays (~Hmmmhesay@66.173.103.108)`
`17:08.10`	`Hmmhesays`	`anyone in here using spandsp successfully?`
`17:08.16`	`shido6`	`VOCAL`
`17:08.18`	`shido6`	`nooooooo`
`17:08.24`	`shido6`	`how has vocal come along these days`
`17:08.25`	`shido6`	`?`
`17:08.47`	`Hmmhesays`	`i think i might be having a problem with libtiff`
`17:08.47`	`Zeeek`	`someone said turing off echocancle made spandsp work for him`
`17:08.48`	`adjacent`	`shido6: this is my first attempt`
`17:09.04`	`Zeeek`	`or was that you, Hmmm?`
`17:09.12`	`adjacent`	`shido6: it was recommended, but im open to othe reccommendations.`
`17:09.14`	`eKo1`	`Dang it. Didn't work with seperate [sip-provider-in]/[sip-provider-out] peer/user.`
`17:09.21`	`Hmmhesays`	`no, that's not me i just got it running`
`17:09.37`	`Zeeek`	`I never got it to work`
`17:09.45`	`Hmmhesays`	`i have it 90% working`
`17:09.49`	`Zeeek`	`but someone recently said echocancel off was the trick`
`17:09.58`	`Zeeek`	`90% isn't good enough for us`
`17:10.30`	`Hmmhesays`	`i know it, i'm getting small parts of the fax that are garbled`
`17:10.38`	`Hmmhesays`	`but i'll try that`
`17:10.48`	`Zeeek`	`I really don't want to call a cust and ask them to resend a fax, esp when resending won't change anything`
`17:11.08`	`Luke-Jr`	`adjacent: where do you buy Vocal stuff??`
`17:11.28`	`adjacent`	`Luke-Jr: downloaded from www.vovida.org`
`17:11.43`	`Luke-Jr`	`oh... you mean the VOCAL software, I guess`
`17:11.58`	`Hmmhesays`	`did that guy turn off echocancel in asterisk or in his gateway`
`17:11.59`	`adjacent`	`yeh`
`17:12.17`	`Luke-Jr`	`adjacent: There's a company called Vocal that sells cheap ATAs`
`17:12.28`	`Luke-Jr`	`can't figure out where they are sold tho`
`17:12.39`	`*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)`
`17:12.40`	`adjacent`	`oh. i have like 20 cisco ata 186's if someone wants to buy them`
`17:12.49`	`Luke-Jr`	`Cheap as in $8-$30`
`17:12.51`	`Hmmhesays`	`are they all locked?`
`17:13.02`	`Sedorox`	`adjacent: how much? and unlocked?`
`17:13.09`	`adjacent`	`they are all NIB, besides 4-5 of them`
`17:13.46`	`jaiger`	`I don't know about the zap echo canceller but my tellabs echo cancellers hear the fax tone and auto-disable on the fly`
`17:14.00`	`Hmmhesays`	`i'm sending this to a sip endpoint`
`17:14.09`	`Sedorox`	`adjacent:..... how much?`
`17:14.23`	`adjacent`	`$50`
`17:14.29`	`Sedorox`	`hmmm`
`17:14.43`	`adjacent`	`hold on. i need to check our bill first`
`17:15.00`	`Luke-Jr`	`Are Cisco ATA 186s better than Linksys PAP2s? o.o`
`17:15.25`	`Sedorox`	`dunno.. thats what I would like to know.. but I can't find any unlocked pap2's`
`17:15.33`	`*** join/#asterisk Liquide (~liquide@liquide.user)`
`17:15.49`	`Liquide`	`"Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 96." whats wrong there?`
`17:15.57`	`adjacent`	`sorry, those were the cheapo ata's. i can sell the ciscos for $80-100. i have 40+ in inventroy`
`17:16.10`	`Hmmhesays`	`hmm, this has to be a libtiff problem`
`17:16.13`	`jaiger`	`Hmmhesays: unless you can guarantee quality of service to the sip endpoint, I wouldn't expect fax to work reliably - even then there are time-critical sections to T.30`
`17:16.16`	`Luke-Jr`	`Sedorox: Teliax sells them w/ their service`
`17:16.29`	`Hmmhesays`	`jaiger, i'm using a crossover cable between this endpoint and asterisk`
`17:17.01`	`Sedorox`	`locked?`
`17:17.06`	`*** join/#asterisk Nix (~Nix@81.213.125.220)`
`17:17.09`	`Luke-Jr`	`Sedorox: unlocked`
`17:17.29`	`*** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net)`
`17:17.33`	`Sedorox`	`nice`
`17:17.33`	`Luke-Jr`	`"The pap2-na is shipped to you configured for Teliax service so that you can plug it in and start making calls. However, it is NOT locked to Teliax. You can log into the device and make changes as you wish."`
`17:17.43`	`Sedorox`	`nice`
`17:19.03`	`pointer-gaim`	`nice`
`17:19.23`	`Luke-Jr`	`pointer-gaim: careful, their webpage/email is pretty broken`
`17:19.34`	`adjacent`	`yeh. all mine are unlocked also`
`17:19.44`	`pointer-gaim`	`Luke-Jr: well...I can't even dial the UK with my BV account`
`17:19.46`	`Sedorox`	`I'm looking at it now...`
`17:19.52`	`coppice`	`Hmmhesays: if you FAX across a VoIP channel expect nothing but pain and misery`
`17:19.56`	`pointer-gaim`	`a working webpage is useless if it service doesn't work`
`17:19.56`	`Luke-Jr`	`pointer-gaim: Voipjet?`
`17:20.03`	`pointer-gaim`	`Luke-Jr: broadvoice`
`17:20.11`	`Sedorox`	`for biz. especially.. looking for a toll-free provider for US and Canada.. and that doesn't have a limit on in and out connections...`
`17:20.16`	`Luke-Jr`	`pointer-gaim: So fallback to VoipJet...`
`17:20.26`	`pointer-gaim`	`Luke-Jr: the tech wouldn't even make a test call to the UK for me....he just escalated it to their engineering dept`
`17:20.37`	`pointer-gaim`	`Luke-Jr: then i actually have to pay for the call`
`17:20.44`	`pointer-gaim`	`Luke-Jr: with BV it's free`
`17:20.51`	`Luke-Jr`	`pointer-gaim: free?`
`17:20.54`	`Luke-Jr`	`pointer-gaim: or included?`
`17:21.24`	`pointer-gaim`	`Luke-Jr: I'm on the 19.95 plan.....unlimited calling to the us/canada and 19 other countries (including the UK)`
`17:21.34`	`Luke-Jr`	`ah`
`17:21.39`	`pointer-gaim`	`Luke-Jr: and calls to italy are terminating fine via BV`
`17:21.48`	`pointer-gaim`	`Luke-Jr: so I think it's strictly a UK term problem`
`17:21.52`	`Luke-Jr`	`I'm using a mix of different providers for my stuff`
`17:21.58`	`pointer-gaim`	`I've tried 4 different UK #s`
`17:22.04`	`*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)`
`17:22.09`	`pointer-gaim`	`Luke-Jr: nod same here... I have BV/nufone/voipjet`
`17:22.21`	`Luke-Jr`	`$5/mo DID from Teliax; Voipjet for outgoing; SIPPhone & FWD for toll free`
`17:22.37`	`Sedorox`	`hmmm`
`17:22.56`	`pointer-gaim`	`Luke-Jr: hmm....I had a 40 minute call to Italy earlier this week...that was free with my BV plan`
`17:23.08`	`pointer-gaim`	`that kind of thing would get expensive quick with other providers`
`17:23.18`	`Luke-Jr`	`ah`
`17:23.26`	`Luke-Jr`	`for me, I'm expecting about $11/mo total`
`17:24.22`	`*** join/#asterisk JohnWayne (jeff@doublewide.hitechredneck.net)`
`17:24.53`	`JohnWayne`	`is voicepulse down`
`17:24.54`	`JohnWayne`	`?`
`17:25.08`	`Luke-Jr`	`Zeeek: how?`
`17:25.08`	`Sedorox`	`Zeeek: how?`
`17:25.13`	`Sedorox`	`lol`
`17:25.23`	`Zeeek`	`voipjet, voicepulse and nufone`
`17:25.36`	`Zeeek`	`no local DID but I don't need one`
`17:25.41`	`Luke-Jr`	`Zeeek: voipjet isn't $0.00...`
`17:25.45`	`*** join/#asterisk Marouani (~ltostain@i01v-30-28.d4.club-internet.fr)`
`17:25.47`	`Sedorox`	`but how can you call locak for free?`
`17:25.50`	`Zeeek`	`per month? yes it is`
`17:25.55`	`JohnWayne`	`all my IAX calls stall at -- Called user:pass@gwiaxt01.voicepulse.com/18067944118`
`17:25.58`	`Luke-Jr`	`Zeeek: I'm including expected usage`
`17:26.18`	`Zeeek`	`don't call expected usage "monthly"`
`17:26.28`	`Zeeek`	`call it expected usage`
`17:26.32`	`Luke-Jr`	`I called it total`
`17:27.01`	`Zeeek`	`but you meant average :)`
`17:27.02`	`Sedorox`	`hmmm`
`17:27.03`	`WildPikachu[BAR]`	`aha.... ok`
`17:27.25`	`WildPikachu[BAR]`	`will musiconhold play if i dial an extension with sip and i have musiconhold(10) ?`
`17:27.38`	`WildPikachu[BAR]`	`asterisk debug says its playing... but i don't hear nothing`
`17:28.34`	`Zeeek`	`not on cellphone by any chance?`
`17:28.45`	`Zeeek`	`I've noticed moh doesn't work well on cells`
`17:30.40`	`WildPikachu[BAR]`	`nope.... sip from linphone to asterisk`
`17:31.10`	`WildPikachu[BAR]`	`exten => 100,1,WaitMusicOnHold(10) <= i got that for extension 100`
`17:31.22`	`WildPikachu[BAR]`	`if i dial it... asterisk says its playing but i don't hear anything :(`
`17:31.26`	`Sedorox`	`anyone know what would cause a "machine gun type sound" when you call in through a x100p?`
`17:31.54`	`Sedorox`	`WildPikachu: how long did you let it play? I noticed when I first start.. since it was quiet.. the default music is quiet to start`
`17:32.34`	`JerJer`	`until the enemy is killed`
`17:32.37`	`Zeeek`	`Sedorox - what kind of people do you hang out with? Maybe the worst has happened?`
`17:32.45`	`Sauron	`Woooo.`
`17:32.46`	`jaiger`	`Sedorox: I'd guess either codecs screwed up or interrupt latency`
`17:33.07`	`Sedorox`	`Zeeek: lol... naaa...`
`17:33.25`	`Sedorox`	`thats what I'm thinking.. what codec does * use to work with the fxo's?`
`17:33.34`	`Zeeek`	`I have a joke answering machine tape with an elliot ness scene and machine guns in it`
`17:33.36`	`jaiger`	`Sedorox: ulaw I believe`
`17:33.37`	`*** join/#asterisk abbas_ (nidobas@203.81.198.6)`
`17:33.38`	`nirs`	`hey everybody`
`17:33.41`	`nirs`	`wasssup ?`
`17:33.42`	`Sedorox`	`hmmmm`
`17:33.47`	`nirs`	`any of th Digium wizards around ?`
`17:34.04`	`WildPikachu[BAR]`	`Sedorox, it works if i dial in on the phone line.... let it play 10s`
`17:35.02`	`Sedorox`	`so you want it to start as soon as you dial the exten?`
`17:35.03`	`jaiger`	`Sedorox: I'd guess codec messup is unlikely. also network latency could cause audio issues`
`17:35.07`	`JerJer`	`only us trolls`
`17:35.23`	`Zeeek`	`Sedorox the machine gun sound happens when you answer the X100? On what?`
`17:35.24`	`JerJer`	`who never return phone calls`
`17:35.25`	`jaiger`	`Sedorox: just a guess tho`
`17:35.26`	`nirs`	`anyone has an idea what the following log entry means: [3 lines will now follow]`
`17:35.31`	`nirs`	`Feb 9 16:08:22 WARNING[11937]: Unable to allocate channel structure`
`17:35.31`	`nirs`	`Feb 9 16:08:22 WARNING[11937]: Unable to start PBX on channel 0/2, span 3`
`17:35.31`	`nirs`	`Feb 9 16:08:22 WARNING[11937]: Call specified, but not found?`
`17:35.43`	`Zeeek`	`Sedorox it sounds more like an IRQ share issue to me`
`17:35.55`	`Sedorox`	`jaiger: thats what I'm thinking because we had the problem befor with x-lite when it would call a BT100....`
`17:36.03`	`Sedorox`	`Zeeek: that would cause that effect?`
`17:36.13`	`Zeeek`	`it has been know to happen ya`
`17:36.20`	`Zeeek`	`did you check the IRQ table?`
`17:36.26`	`Sedorox`	`how do you check in fbsd?`
`17:36.41`	`Zeeek`	`cat /proc/interrupt or something close ?`
`17:36.52`	`Zeeek`	`lemmie see here`
`17:36.54`	`Nugget`	`no, in freebsd proc is proc.`
`17:37.01`	`Nugget`	`not that weird linux proc-and-the-kitchen-sink`
`17:37.02`	`WildPikachu[BAR]`	`Sedorox, yea... i'm just testing :)`
`17:37.11`	`Sedorox`	`hehe`
`17:37.13`	`Zeeek`	`if you know tell him else I can ssh to a FBSD box`
`17:37.17`	`Sedorox`	`dunno how to make it start asap...`
`17:37.20`	`mikegrb`	`Nugget: heh well /sys is supposed to fix thaat`
`17:37.32`	`Nugget`	`it needs fixing?`
`17:37.53`	`Sedorox`	`so how do I do it in fbsd then? lol`
`17:37.58`	`Nugget`	`check dmesg.`
`17:38.12`	`Zeeek`	`you guys know anythiong about http://us.shuttle.com/ for asterisk PC use?`
`17:38.22`	`Hmmhesays`	`heh, forgot libxml2-dev`
`17:38.34`	`Nugget`	`grep irq /var/run/dmesg.boot`
`17:39.19`	`Sedorox`	`I don't even see the fxo in dmesg.boot... hmmm`
`17:39.59`	`mikegrb`	`Nugget: fix linux, not fbsd`
`17:40.04`	`Nugget`	`ah, ok. :)`
`17:40.55`	`Zeeek`	`Sedorox I don't have a local FreeBSD box anymore but I'd check that out - at worst, go to the freebsd Usenet group and ask - they've always been receptive to questions`
`17:41.14`	`Sedorox`	`ok.. from what I see.. no shared irq's`
`17:41.17`	`Sedorox`	`the card is using 11`
`17:41.25`	`Sedorox`	`from the dmesg.boot.. nothing else is using 11`
`17:41.41`	`Zeeek`	`ok then that's one less thing to srew around with`
`17:41.45`	`Hmmhesays`	`now the moment of truth`
`17:42.01`	`Sedorox`	`probably codec`
`17:42.03`	`Sedorox`	`I have to look`
`17:42.34`	`Zeeek`	`I've never had codec troubles between ZAP and SIP or IAX`
`17:42.47`	`Sedorox`	`hmmmm`
`17:42.52`	`abbas_`	`whats "payload'`
`17:42.55`	`Sedorox`	`dunno what else it could be`
`17:43.08`	`Essobi`	`Can I get licences g729 for FreeBSD?`
`17:43.13`	`Zeeek`	`wait a sec, I had that sound on moh with freebsd`
`17:43.26`	`Zeeek`	`I just decided not to use moh (don't need it)`
`17:45.40`	`Zeeek`	`anyone using shuttle products?`
`17:45.47`	`Zeeek`	`http://sys.us.shuttle.com/WhyXPC.aspx`
`17:46.01`	`Hmmhesays`	`moment of truth failed`
`17:46.07`	`Zeeek`	`shit!`
`17:46.08`	`Sedorox`	`grrrrr freaking phone`
`17:47.51`	`Essobi`	`BAAAH!`
`17:47.54`	`Essobi`	`That's super weak!`
`17:47.57`	`Sedorox`	`anyone know a voip provider that does canada DID's?`
`17:48.13`	`Zeeek`	`that question comes up almost daily!`
`17:48.14`	`Essobi`	`Asterisk sales just said there will be no G729 for FreeBSD. GRRRR/`
`17:48.29`	`Sedorox`	`lol`
`17:48.36`	`Hmmhesays`	`Essobi, compile it yourself`
`17:48.48`	`Essobi`	`Umm. Yea, that'll work.`
`17:48.54`	`Hmmhesays`	`why not?`
`17:48.57`	`Essobi`	`With what source code? :P`
`17:48.58`	`Hmmhesays`	`I compiled my own`
`17:49.12`	`Hmmhesays`	`g.723 also`
`17:49.18`	`Hmmhesays`	`not on freebsd, but it's worth a shot`
`17:49.27`	`Essobi`	`G729 is still propietary enforced isn't it?`
`17:49.46`	`Hmmhesays`	`yes`
`17:49.48`	`Essobi`	`Where'd you get the code?`
`17:49.49`	`Essobi`	`:P`
`17:49.53`	`RaYmAn-Bx`	`Essobi: there's plenty of sourcecode available. It's just not legal to use in most countries perhaps unless you have licenses`
`17:49.54`	`Hmmhesays`	`intel`
`17:50.00`	`Essobi`	`lol`
`17:50.07`	`Hmmhesays`	`what's so funny?`
`17:51.55`	`Hmmhesays`	`ok`
`17:52.03`	`Hmmhesays`	`I guess I missed the joke`
`17:52.39`	`*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)`
`17:52.39`	`*** mode/#asterisk [+o anthm] by ChanServ`
`17:53.23`	`*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543956pcs.mainf01.in.comcast.net)`
`17:53.39`	`*** join/#asterisk bowman (~rsp@195.46.47.202.static.cablesurf.de)`
`17:54.01`	`bowman`	`does anyone in here have experience with quadBRI boards?`
`17:55.34`	`Logan`	`I have a wav file I generated by using ztmonitor during a call. How can I use this information to figure out how I should adjust rxgain and txgain in order to get good echo cancellation?`
`17:57.13`	`bjohnson`	`Sedorox: broadvoxdirect and iax.cc`
`17:57.28`	`bjohnson`	`Sedorox: few others with limited areas`
`17:57.37`	`*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released \|\| development conf call today 2pm CST IAX2/66.250.68.194/996 #astersk-dev (msg bkw with agenda ideas)`
`17:57.43`	`bjohnson`	`Sedorox: nufone in a month or so`
`17:57.44`	`*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released \|\| development conf call today 2pm CST IAX2/66.250.68.194/996 #astersk-dev (msg bkw_ with agenda ideas)`
`17:57.58`	`Sedorox`	`kk thanks`
`17:58.22`	`bjohnson`	`Sedorox: let me know what you find. I've sent some question to iax.cc before I sign with them`
`17:58.22`	`AgiNamu`	`How can I register code for a relaod?!`
`17:58.38`	`Sedorox`	`ok`
`17:59.11`	`bowman`	`AgiNamu: eh?`
`17:59.46`	`bkw_`	`AgiNamu, in your module?`
`18:02.34`	`bkw_`	`Ok guys help shape the direction of asterisk ... give ideas and comments that we can go over in the dev meeting today.. ;)`
`18:02.42`	`bkw_`	`and don't say we didn't ask ya ;)`
`18:03.08`	`Hmmhesays`	`the endpoint i was faxing with is a pile of shit`
`18:03.15`	`*** join/#asterisk Mw3 (~mw3@internetes.szabadhalo.hu)`
`18:03.18`	`mikegrb`	`bkw_: meetme2`
`18:03.39`	`bkw_`	`its funny you mention that one. ;)`
`18:03.40`	`bowman`	`bkw_: customizable voicemail prompts`
`18:03.43`	`bjohnson`	`bkw_: ldap as a storage system for auth. of voicemil`
`18:03.56`	`mikegrb`	`bkw_: figured it was on the list ;)`
`18:04.01`	`bkw_`	`how about we just kill user/peers`
`18:04.06`	`bkw_`	`and come up with accounts`
`18:04.07`	`mikegrb`	`k`
`18:04.24`	`bjohnson`	`bkw_: ldap as a storage system for user to extension mapping (including numbers to try concurrently or in sequence)`
`18:04.34`	`bkw_`	`bjohnson, you can do that with res_config ya know ;)`
`18:04.51`	`bkw_`	`the framework is there`
`18:04.55`	`bjohnson`	`bkw_: and write ability to ldap for user's changing info`
`18:05.00`	`bjohnson`	`res_config?`
`18:05.02`	`bkw_`	`realtime can do that`
`18:05.12`	`bkw_`	`you just need to write a realtime ldap driver`
`18:05.16`	`bjohnson`	`ahh`
`18:05.30`	`bkw_`	`like we have res_config_odbc`
`18:05.36`	`bkw_`	`and that cheesy mysql one too`
`18:05.44`	`bjohnson`	`bkw_: proposal for future direction .. someone write a read/write realtime ldap driver`
`18:05.46`	`Kirk_22`	`anyone know why my SPA3000 keeps sending 'MSG Waiting Off' to my cordless phone?`
`18:05.50`	`FuRR_`	`bjohnson: you might actually have something there, an LDAP backend would be nice`
`18:06.03`	`mikegrb`	`bkw_: is it made of chedar or swiss?`
`18:06.10`	`FuRR_`	`bjohnson: But I dont think it would work for CDR`
`18:06.38`	`greg_work`	`bowman bkw_: customizable voicemail prompts`
`18:06.39`	`greg_work`	`bjohnson bkw_: ldap as a storage system for auth. of voicemil`
`18:06.55`	`greg_work`	`in case either of you are interested, i finished my agi-based voicemail last night (i think0`
`18:07.20`	`bkw_`	`again res_config/realtime`
`18:07.20`	`bowman`	`bkw_: to clarify this: configurable menu structure for voicemail and configurable sequence of files to be played`
`18:07.34`	`*** join/#asterisk criptos (~root@dsl-200-78-109-139.prod-infinitum.com.mx)`
`18:07.36`	`bkw_`	`ok I was thinking on that yesterday too`
`18:07.39`	`greg_work`	`it uses the * database to store the user password (though reads the default password from voicemail.conf) .. but it's specifically designed to allow any backend`
`18:07.40`	`bkw_`	`voicemail personalities...`
`18:07.43`	`*** part/#asterisk criptos (~root@dsl-200-78-109-139.prod-infinitum.com.mx)`
`18:07.45`	`bkw_`	`so you can have voicemail be like X or Y system`
`18:07.45`	`bjohnson`	`FuRR_: no .. ldap (like db) isn't a solution for everything .. just some things`
`18:07.48`	`bkw_`	`on a per mailbox basis`
`18:07.58`	`bowman`	`right`
`18:07.59`	`greg_work`	`hm, thats an interesting one too`
`18:08.08`	`greg_work`	`though, i still don't like app_voicemail.c`
`18:08.23`	`bkw_`	`check the ivr api anthm posted on bugs`
`18:08.26`	`bjohnson`	`Kirk_22: you have set a mailbox in the spa config and it thinks there is a message waiting?`
`18:08.30`	`bkw_`	`you'll melt when you see it`
`18:08.34`	`FuRR_`	`bjohnson: true, but it definately has its place in handling extensions and peers`
`18:08.46`	`greg_work`	`having something like that is built in. the whole reason i wrote an agi was because i wanted "new features" that are only in HEAD now, but in my 1.0.5 install`
`18:09.05`	`bkw_`	`be a man run head`
`18:09.06`	`bkw_`	`:P`
`18:09.28`	`*** join/#asterisk asher (ben@62.121.2.101)`
`18:09.45`	`asher`	`can anyone tell me how to re-wire a normal telephone headset to work on a cisco phone?`
`18:09.46`	`greg_work`	`yeah well, i'm a wimp, i don't like running cvs version in production ;)`
`18:09.52`	`asher`	`or even point me where some wiring diagrams for both are?`
`18:09.57`	`bjohnson`	`bkw_: realtime is just in HEAD right now correct?`
`18:09.58`	`*** join/#asterisk oej (~oej@apollo.webway.se)`
`18:10.03`	`bkw_`	`yep`
`18:10.15`	`Kirk_22`	`bjohnson-no I haven't set a mailbox in the SPA- but with broadvoice when I get a voice mail, it chances to just MSG waiting - as it should`
`18:10.18`	`bkw_`	`greg_work, why not?`
`18:10.34`	`greg_work`	`bkw_: are there any bugs in HEAD, right now?`
`18:10.51`	`Sedorox`	`ok`
`18:10.54`	`oej`	`greg_work: Possibly a lot of them, so help us find them :-)`
`18:10.56`	`Sedorox`	`something on iax.cc seesms weird`
`18:10.58`	`greg_work`	`exactly`
`18:11.01`	`BoRiS`	`bkw: get rid of zapbarge and creating virtual zap channels and have one function (like ChanSpy) to listen into channels no matter what the protocol used`
`18:11.06`	`bkw_`	`greg_work, possibly alot in stable too`
`18:11.07`	`Sedorox`	`<PROTECTED>`
`18:11.12`	`greg_work`	`finding them in a development or non-critical environment, thats fine`
`18:11.16`	`bkw_`	`BoRiS, channel shims`
`18:11.17`	`Sedorox`	`<PROTECTED>`
`18:11.17`	`Sedorox`	`Pay only 1.7 cents per minute to the US and Canada. Toll free numbers are billed at 2? per minute.`
`18:11.18`	`Sedorox`
`18:11.31`	`greg_work`	`having an office full of people without phones because you found a bug, thats bad :p`
`18:11.38`	`bkw_`	`greg_work, have two systems`
`18:11.47`	`bkw_`	`swing it into play when you find it.. minimal downtime`
`18:12.07`	`Hmmhesays`	`hmm apparently it takes a long time to send an 8x11 photo at 300dpi over fax`
`18:12.21`	`greg_work`	`bkw_: yeah, true enough -- but there are less likely to be bugs in stable. plus cvs is in flux .. maybe no bugs today, maybe tons tomorrow. when do you decide to use the snapshot? :p`
`18:12.23`	`Zaw`	`Sedorox: they charge a one-time $10.00 fee and .30/month for toll free numbers`
`18:12.32`	`bkw_`	`well right now cvs is is pretty stable`
`18:12.36`	`greg_work`	`i probably will run HEAD, once i have my production system up`
`18:12.38`	`bkw_`	`the flux has yet to start`
`18:12.58`	`bkw_`	`when the flux really hits.. thats when all hell might break loose and kill jor boxen`
`18:13.01`	`greg_work`	`fair enough. but its not always like that . i assume you guys are planning a release somewhat soon then?`
`18:13.03`	`bjohnson`	`Sedorox: incoming voip toll free for numbers that work in canada are more expensive. Outgoing toll free through US voip providers do not work for Canadians. ie ignore toll free`
`18:13.13`	`Sedorox`	`to have a toll-free number? or to dial.. because the way it looks.. it makes it out to be that your getting billed for maing toll-free calls...`
`18:13.15`	`bkw_`	`greg_work, well if you want cvs-head/realtime .. now is the time to grab it`
`18:13.31`	`greg_work`	`bkw_: i'll probably do that`
`18:13.32`	`bjohnson`	`Sedorox: you don't want to use toll free through iax.cc`
`18:13.33`	`greg_work`	`thanks`
`18:13.40`	`greg_work`	`but .. still probably wont run it in production :p`
`18:13.51`	`greg_work`	`bkw_: how are you using it?`
`18:14.04`	`bkw_`	`you name it.. we use it`
`18:14.08`	`bjohnson`	`Sedorox: however it is common to be billed for outgoing toll free numbers in voip unless you are on a package plan`
`18:14.34`	`greg_work`	`bkw_: well, i mean .. if i had a * system at home (which i probably will, soon) then i'd run head for sure`
`18:14.37`	`Sedorox`	`but I can call toll-free numbers through fwd and iaxtel for free...`
`18:14.38`	`bkw_`	`bjohnson, ya common for crack headed voip providers`
`18:14.47`	`bkw_`	`we dont even bill for that`
`18:14.50`	`greg_work`	`but for work, i'm less inclined to run something that is likely to break :p`
`18:15.00`	`bkw_`	`greg_work, setup a second dev box and help out man`
`18:15.07`	`bkw_`	`help shape the future of asterisk :P`
`18:15.25`	`bjohnson`	`Sedorox: try calling a toll free number .. the ones that work in canada do not work through a US voip provider. understand`
`18:15.27`	`bjohnson`	`?`
`18:15.38`	`greg_work`	`bkw_: oh, i will for sure . but now my goal is to get my first one up, and in use.`
`18:15.57`	`bjohnson`	`Sedorox: so you can call SOME toll free numbers through FWD but you have no way of knowing which ones`
`18:15.58`	`Sedorox`	`yeaaa yea.. that i understand`
`18:16.03`	`Sedorox`	`ok`
`18:16.05`	`bjohnson`	`Sedorox: and good luck using iaxtel for anything`
`18:16.18`	`*** join/#asterisk Inv_arp (junya@adsl-3-255-228.mia.bellsouth.net)`
`18:16.25`	`Sedorox`	`lol`
`18:16.33`	`Sedorox`	`i know.. i don't`
`18:16.37`	`Sedorox`	`mainly fwd`
`18:16.39`	`Zeeek`	`great for texting... when it's up`
`18:16.43`	`bjohnson`	`Sedorox: so forget using a US based voip provider for toll free .. unless it's incoming toll free on a US/48 only line`
`18:16.56`	`*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released \|\| development conf call today 2pm CST IAX2/guest@66.250.68.194/996 #astersk-dev (msg bkw_ with agenda ideas)`
`18:17.10`	`Sedorox`	`yea`
`18:20.17`	`Zeeek`	`bkw if only these conf were a little earlier`
`18:20.54`	`Zeeek`	`come to think that's only 9PM here`
`18:21.00`	`*** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org)`
`18:22.12`	`Essobi`	`debian.org down for anyone else?`
`18:22.26`	`postel`	`bkw_: would guest@ work with SIP too?`
`18:23.11`	`bkw_`	`let me see`
`18:23.16`	`bkw_`	`no`
`18:23.20`	`bkw_`	`but I can make SIP/IP/996 work`
`18:23.22`	`Essobi`	`Hi Brian! How you been?`
`18:23.37`	`*** join/#asterisk _Brian (brian@unix01.voicenet.com)`
`18:23.48`	`Essobi`	`haha`
`18:23.51`	`bkw_`	`busy as hell`
`18:23.54`	`Zeeek`	`bkw_ is it possible to test the number right now?`
`18:23.56`	`Essobi`	`Nice vhost man.`
`18:23.58`	`Essobi`	`that's funny.`
`18:24.01`	`Hmmhesays`	`well that was a successful fax`
`18:24.04`	`Hmmhesays`	`bout damn time`
`18:24.15`	`postel`	`bkw_: please do then, not all ppl use IAX`
`18:24.29`	`_Brian`	`good day all`
`18:24.36`	`greg_work`	`bkw_: what do you think about decoupling a lot of the app_* stuff from * though?`
`18:25.13`	`_Brian`	`postel: -6`
`18:25.20`	`postel`	`thank you`
`18:25.27`	`_Brian`	`postel: http://www.svetlozar.com/learningcenter/reference/0000000.html`
`18:25.34`	`bkw_`	`greg_work, ok why? can you give me an example of why?`
`18:25.53`	`greg_work`	`bkw_: applications being non-dependant on the version of * actually running`
`18:26.20`	`bkw_`	`well that could work once the API is stable`
`18:26.22`	`*** join/#asterisk izo (~izo@195.69.228.27)`
`18:26.32`	`Liquide`	`"Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 96." whats wrong there? and yes the voicemail.con is where it says it aint`
`18:26.37`	`greg_work`	`yeah, that would be good`
`18:27.20`	`bkw_`	`Liquid open that up in an editor`
`18:27.25`	`bkw_`	`and look before you ask such simple questions`
`18:27.38`	`bkw_`	`I think /etc/asterisk/voicemail.conf is hard coded in the .cgi`
`18:28.07`	`greg_work`	`the way it is now, its kind of like making vi dependent on the kernel .. in the same source repository, so development happens on both at the same time .. if you want to upgrade to a newer version of vi, you have to ugprade the kernel ..`
`18:28.13`	`*** join/#asterisk VoIPMasta (~John@201.133.111.242)`
`18:28.15`	`VoIPMasta`	`Hi`
`18:28.16`	`Liquide`	`yeah, but the thing is that my voicemail.cof is in /etc/asterisk/.. so its saying it cant find it even though its thete`
`18:28.20`	`BoRiS`	`Pay up!`
`18:28.28`	`bkw_`	`when someone asks such a simple questions.. it reminds me "What if an airline pilot come into the cabin and asked 'does anyone know what this light means?'"`
`18:28.29`	`VoIPMasta`	`One quick question: Would it be useful to set up a STUN server if we are using Asterisk?`
`18:28.40`	`bkw_`	`Liquid check perms`
`18:29.24`	`Liquide`	`bkw_ owned by root`
`18:29.30`	`postel`	`bkw_: Any question is simple only if you know the answer ;)`
`18:29.34`	`WildPikachu[BAR]`	`aha`
`18:29.43`	`WildPikachu[BAR]`	`i get a machine gun sound aswell connecting via sip`
`18:30.11`	`bkw_`	`postel, simple questions on require the skillz to find the answer before asking`
`18:30.20`	`WildPikachu[BAR]`	`seems sip gets changed like 20 times between ulaw and gsm`
`18:30.27`	`bkw_`	`daniel san`
`18:30.46`	`BoRiS`	`wax on..... wax off`
`18:30.53`	`bkw_`	`no comment`
`18:30.59`	`_Brian`	`wax on....wax off!`
`18:31.02`	`bjohnson`	`VoIPMasta: no`
`18:31.05`	`bkw_`	`again no comment`
`18:31.06`	`_Brian`	`damn..beat me to it Boris`
`18:31.12`	`BoRiS`	`:-p`
`18:31.26`	`VoIPMasta`	`bjohnson: Can you explain why?`
`18:31.42`	`bjohnson`	`* doesn't need it`
`18:31.54`	`*** join/#asterisk PCadach (~paul@www.east.telecom.kz)`
`18:32.21`	`VoIPMasta`	`bjohnson: wouldn't it help with SIP clients behind a NAT?`
`18:32.36`	`bjohnson`	`yes .. but * doesn't need it`
`18:32.38`	`sezuan`	`Does anybody know if the PA168 DSP will have ilbc support in a future firmware relase?`
`18:33.20`	`Zeeek`	`does anybody know how well the PA168 based phones work for SIP and IAX?`
`18:33.33`	`VoIPMasta`	`bjohnson: so nat=yes will take care of everything? even if it's a "restricted cone" NAT?`
`18:33.35`	`bjohnson`	`Zeeek: agniamu says they're good`
`18:33.42`	`Zeeek`	`I asked but he was gone`
`18:33.49`	`bjohnson`	`VoIPMasta: I don't know about your cones`
`18:33.58`	`*** join/#asterisk low_rad (~hibbert@cblfw1.newnova.com)`
`18:34.05`	`bjohnson`	`VoIPMasta: nat=yes works for some`
`18:34.08`	`Zeeek`	`VoIPMasta port forwarding`
`18:34.15`	`postel`	`bkw_: you're a professional brian, you cant expect people to have either the time, or the knowledge or the resources you do. And people are not good with technical docs, which is good and bad, the bad side is you get questions like "Do i need to call voicemail exten to retrieve my voicemail?" , the good side is people like yourself get jobs deploying and supporting systems for those people, dont complain :-)`
`18:34.28`	`sezuan`	`Zeeek: I have no problems with my PA168 base phone and *`
`18:34.28`	`bjohnson`	`VoIPMasta: Zeeek has double nat working with port forwarding and setting externip in sip.conf`
`18:34.36`	`sezuan`	`Zeeek: SIP only.`
`18:34.44`	`bkw_`	`postel, all self taught too ;)`
`18:34.52`	`bjohnson`	`VoIPMasta: the simple answer is no .. it might help in certain cases`
`18:35.02`	`bkw_`	`postel, I just think more problem solving skillz need to be taught in schools`
`18:35.25`	`VoIPMasta`	`bjohnson: The asterisk box is outside the NAT, on the Internet, however the SIP clients are inside a NAT. All clients are on dynamic IPs so it's not possible to set a fixed IP address on them`
`18:35.32`	`Sedorox`	`nice...`
`18:35.49`	`greg_work`	`bkw_: oh, i thought of another benefit of agi vs built-in (even though we weren't exactly talking about agi specifically .. a frozen api would work too .. anyhow) .. the barrier to entry in the api is a lot higher than AGI`
`18:36.05`	`bjohnson`	`VoIPMasta: refer back to your original question and decide why I couldn't guess your configuration`
`18:36.11`	`sezuan`	`Zeeek: there seem to be a pitfall with * and PA168 in the same subnet, see http://www.voip-info.org/wiki-Giptel+IP+phones`
`18:36.13`	`bkw_`	`agi blah`
`18:36.24`	`greg_work`	`having never done programming for asterisk, i was able to understand and use AGI very quickly .. maybe spent a few minutes reading the docs before coding.`
`18:36.31`	`*** join/#asterisk loick (~loick@APuteaux-151-1-51-175.w82-124.abo.wanadoo.fr)`
`18:36.40`	`bkw_`	`true`
`18:36.42`	`bkw_`	`agi does have its place`
`18:36.46`	`VoIPMasta`	`bjohnson: Ok the scheme would be: [sip client]==[NAT]===[Dynamic IP link to the internet]====[Asterisk server]`
`18:36.52`	`bkw_`	`I personally do not like agi as much as other stuff/options`
`18:36.54`	`*** join/#asterisk chfn (~adolfo@c90618ac.virtua.com.br)`
`18:37.07`	`bkw_`	`res_perl is 1000% more powerful than standard AGI`
`18:37.08`	`greg_work`	`bkw_: i've heard the speed argument .,. but are there other technical benefits?`
`18:37.17`	`bkw_`	`agi is easy`
`18:37.29`	`greg_work`	`i mean of the api over agi`
`18:37.42`	`bjohnson`	`VoIPMasta: I haven't done that. I am told that * on a public IP allows sip devices to connect to it directly without knowing the devices IP address .. just nat=yes in sip.conf`
`18:37.49`	`bkw_`	`well thats what AGI is`
`18:37.50`	`chfn`	`I heard right now the sponsor adv in my pvt. THANKSSSSSSS FOR YOUR PROJECT and i will try this project and i will help whatever is possible in source code`
`18:38.17`	`Zeeek`	`sezuan thx for that`
`18:38.38`	`bjohnson`	`VoIPMasta: I am also told that some routers don't deal with SIP nicely nd you may then have to find an alternate method`
`18:38.46`	`greg_work`	`yeah, res_perl is kinda neat`
`18:39.10`	`bjohnson`	`VoIPMasta: this is why everyone anxiously awaits a good supply and selection of iax devices .. these issues go away`
`18:39.44`	`greg_work`	`If you stare at the source code from about 2 feet back for approx 30 seconds you will start to see that the entire thing is in itself 1 large bug. I am not so much a C programmer as I am a Perl programmer so that was my motivation for this idea but it probably is sucky to some arrogant linux zelot somewhere out there. If that is you, ha ha I made you download it.....`
`18:39.45`	`greg_work`	`lol`
`18:39.55`	`*** join/#asterisk schwagner (~andrew@68.143.92.248.nw.nuvox.net)`
`18:40.26`	`*** join/#asterisk HostingGeek (~uname@200.48.233.220.exetel.com.au)`
`18:40.48`	`low_rad`	`how do I use 2 modems with asterisk? everytime I get a "Unable to autodetect modem" and "Unable to register channel" errors`
`18:41.03`	`HostingGeek`	`VoIP sucks it will kill telstra`
`18:41.09`	`HostingGeek`	`and teltsra rules`
`18:41.17`	`HostingGeek`	`:rolleyes:`
`18:41.24`	`HostingGeek`	`GOOO VoIP`
`18:41.33`	`bjohnson`	`GOOO AWAY`
`18:41.34`	`HostingGeek`	`kill hell$$$ra`
`18:41.50`	`schwagner`	`does anyone else get wierd beeps when using an X100P card?`
`18:41.50`	`HostingGeek`	`OOOOKAY`
`18:41.56`	`*** part/#asterisk HostingGeek (~uname@200.48.233.220.exetel.com.au)`
`18:42.06`	`Hmmhesays`	`when you press buttons on the phone?`
`18:42.09`	`FuRR_`	`anyone used the ArtDio SIP phones?`
`18:42.17`	`bjohnson`	`Hmmhesays: hahahaha`
`18:42.19`	`schwagner`	`heh, no, not then`
`18:42.31`	`schwagner`	`just randomly during the conversation`
`18:42.43`	`bkw_`	`ok guys now lay down the specifics of what you wanna see in the dev-conf`
`18:42.51`	`schwagner`	`i can hear them, but the other end can't`
`18:42.53`	`*** join/#asterisk pointer-gaim (~pointer@router.cathey.us)`
`18:42.53`	`Inv_arp`	`are the HT486's able to be locked?`
`18:43.00`	`mikegrb`	`bkw_: meetme2`
`18:43.03`	`Hmmhesays`	`that reminds me of that one commerical where they are saying the IVR menu's`
`18:43.03`	`mikegrb`	`bkw_: ;)`
`18:43.04`	`low_rad`	`anyone?`
`18:43.05`	`Hmmhesays`	`lol`
`18:43.10`	`greg_work`	`bkw_: stable api ;)`
`18:43.10`	`bjohnson`	`schwagner: I get that sometimes but not with a X100P .. I'm wondering if it is * transcoding codecs`
`18:43.21`	`bkw_`	`show about a standard app api`
`18:43.21`	`bjohnson`	`bkw_: that ldap driver thingy you said`
`18:43.23`	`Hmmhesays`	`are you running asterisk on a 486?`
`18:43.25`	`bkw_`	`tht should "never" change`
`18:43.35`	`low_rad`	`p4`
`18:43.40`	`bkw_`	`bjohnson, you willing to pay someone to dev that?`
`18:43.56`	`bkw_`	`we also wanna talk about res_storage`
`18:43.58`	`greg_work`	`bkw_: yeah. to the point of being able to take msot of the apps/ directory and put it in a seperate package, like asterisk-apps`
`18:44.08`	`schwagner`	`bjohnson: ip to ip, or ip to pstn?`
`18:44.08`	`bkw_`	`check astsx`
`18:44.17`	`bkw_`	`you can compile apps outside the asterisk src tree with it`
`18:44.18`	`bjohnson`	`beware .. bkw_'s questions are a trick. He will just answer with "you willing to pay someone to dev that?"`
`18:44.26`	`bjohnson`	`hehe`
`18:44.28`	`bkw_`	`bjohnson, haha well it takes time`
`18:44.34`	`bkw_`	`and i'm sure someone can do it that knows the api`
`18:44.40`	`nestAr`	`master is tricksey`
`18:44.41`	`bkw_`	`but $$$ does speed that up ya knwo`
`18:44.56`	`BoRiS`	`bkw: if only that worked for T.38 passthru :-p`
`18:44.58`	`bkw_`	`can't blame a girl for trying`
`18:45.06`	`BoRiS`	`that gurl!`
`18:45.17`	`bkw_`	`how about some sort of data/t.38 transport system`
`18:45.32`	`bkw_`	`wonder if we can be more reliable ... t.38 is a can of worms`
`18:45.33`	`greg_work`	`bkw_: uh, where is astsx? can't find any references on google/voip-info..`
`18:45.35`	`bkw_`	`HUGE can of works`
`18:45.39`	`bkw_`	`its in the contrib/scripts`
`18:45.46`	`bkw_`	`it should be installed by default`
`18:46.01`	`greg_work`	`bkw_: astxs you mean? :)`
`18:46.02`	`BoRiS`	`more ideas :)`
`18:46.05`	`bkw_`	`ya that`
`18:46.08`	`greg_work`	`heh`
`18:46.08`	`bjohnson`	`schwagner: ip tp pstn locally and also ip to voip provider to pstn`
`18:46.09`	`bkw_`	`damn it.. dyslexic`
`18:46.10`	`VoIPMasta`	`bjohnson: I would be more than glad to use IAX devices, however in this case I'm using SIP based softphones, I know there are some IAX based softphones but I need g.729 support`
`18:46.25`	`bjohnson`	`schwagner: we don't do much ip to local ip`
`18:46.27`	`*** join/#asterisk LarsAC (~LarsAC@pD95005F4.dip0.t-ipconnect.de)`
`18:46.33`	`bkw_`	`Video Meetme?`
`18:46.34`	`bkw_`	`anyone`
`18:46.42`	`bkw_`	`pipe dream`
`18:46.49`	`BoRiS`	`cuseeme?`
`18:46.54`	`bowman`	`moin LarsAC`
`18:46.58`	`greg_work`	`i still think the idea of t.38 is dumb. have asterisk rxfax() it, then txfax() from the other machine`
`18:47.05`	`bkw_`	`how about g.722 wide band codec support?`
`18:47.10`	`bkw_`	`hifi calling`
`18:47.11`	`greg_work`	`its not like it matters that the fax is getting passed through in realtime`
`18:47.17`	`LarsAC`	`hi bowman`
`18:47.20`	`bjohnson`	`bkw_: ability to handle modem calls would be nice but probably of limited interest`
`18:47.24`	`schwagner`	`bjohnson: randomly during the call, for about 1-5 seconds?`
`18:47.35`	`bkw_`	`zapras`
`18:47.43`	`bjohnson`	`schwagner: usually just short beeps`
`18:47.48`	`BoRiS`	`t38+modem support would be amazing`
`18:47.52`	`bkw_`	`ok`
`18:47.54`	`bkw_`	`here is a job`
`18:47.55`	`greg_work`	`that would actually also let * receive multiple faxes simultaenously, then transmit them all one by one to a remote fax`
`18:48.01`	`bkw_`	`find a c t.38 ref code`
`18:48.07`	`bkw_`	`and we'll see what we can do`
`18:48.08`	`*** join/#asterisk criptos (~criptos@dsl-200-78-109-139.prod-infinitum.com.mx)`
`18:48.34`	`greg_work`	`and still no one has told me what benefits there are of t.38 over rxfax()ing then txfax()ing .. .`
`18:48.41`	`schwagner`	`bjohnson: well, either way, i have no idea why it keeps happening`
`18:48.41`	`BoRiS`	`how about C++ code? :)`
`18:48.50`	`*** join/#asterisk jonp (~jon@81-179-250-111.dsl.pipex.com)`
`18:48.52`	`bkw_`	`greg_work, you can pass it over an ip network out an ata to a real fax machine`
`18:49.09`	`bjohnson`	`schwagner: I haven't figured it out yet either .. but I have users submitted trouble tickets each time it happens`
`18:49.11`	`bkw_`	`so to the end user they fax just like they have for the past 10 years`
`18:49.20`	`*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)`
`18:49.22`	`greg_work`	`bkw_: T.38 sends between two * machines though, right?`
`18:49.25`	`jonp`	`Anyone here knowledgable about ast_channel_masquerade()?`
`18:49.30`	`greg_work`	`or does the ATA have support for t.38?`
`18:49.33`	`bkw_`	`yes`
`18:49.34`	`bkw_`	`most do`
`18:49.37`	`greg_work`	`ah ok`
`18:49.45`	`bjohnson`	`greg_work: is that approach outlined somewhere? it's just for outgoing right?`
`18:49.49`	`bkw_`	`even t.38 pass thru on sip would b great`
`18:49.55`	`*** join/#asterisk Slainte (~mirc@66.55.112.217.ppp.northrock.bm)`
`18:49.55`	`bkw_`	`but that would be a bitch to do`
`18:49.58`	`bkw_`	`now`
`18:50.02`	`greg_work`	`bjohnson: which?`
`18:50.02`	`bkw_`	`if we had something like say codec_t28`
`18:50.05`	`bkw_`	`er t38`
`18:50.09`	`bkw_`	`that would be neat`
`18:50.12`	`JerJer`	`what's wrong with app_rx and tx fax ?`
`18:50.15`	`bjohnson`	`greg_work: rxfax and txfax`
`18:50.17`	`bkw_`	`just think accept a fax and put it in a voicemail box`
`18:50.23`	`bkw_`	`then call voicemail`
`18:50.27`	`bkw_`	`retrieve yoru faxes`
`18:50.39`	`greg_work`	`oh i dunno. you just need a bit of glue to connect rxfax() on one * system to txfax() on another.`
`18:50.39`	`JerJer`	`so its not T.38, but it most righteously works`
`18:50.50`	`greg_work`	`would be a simple little agi, right bkw_ ? :)`
`18:50.54`	`low_rad`	`is there a how-to to configure 2 modems for asterisk ?`
`18:51.05`	`bkw_`	`JerJer, ya its not t.38 but if we wanna play with the big boys we need to have support for it`
`18:51.25`	`bjohnson`	`low_rad: you need voip hardware .. not modems`
`18:51.38`	`greg_work`	`i take it none of the digium ATA's have support for T38`
`18:51.41`	`*** join/#asterisk adjacent (~scott@office.bftwave.com)`
`18:51.45`	`low_rad`	`bjohnson: so modems will never work with asterisk ?`
`18:52.01`	`bkw_`	`I think we need a make.conf`
`18:52.05`	`bkw_`	`so we can have diffrent make targets`
`18:52.08`	`bkw_`	`like make pbx`
`18:52.09`	`bkw_`	`make switch`
`18:52.12`	`bjohnson`	`low_rad: long debate .. end answer is maybe .. if you write the support for it and start with the right hardware`
`18:52.12`	`bkw_`	`make minimal`
`18:52.19`	`drumkilla`	`make menuconfig ? :)`
`18:52.24`	`bkw_`	`now thats an idea`
`18:52.25`	`bkw_`	`too`
`18:52.42`	`drumkilla`	`to enable disable certain modules`
`18:52.43`	`low_rad`	`bjohnson: I see, have an url on where I might be able to find that?`
`18:52.47`	`postel`	`drumkilla: sweeeet`
`18:52.47`	`bjohnson`	`no`
`18:52.48`	`bkw_`	`drumkilla, yep`
`18:52.49`	`Logan`	`Is there a way of modifyi8ng txgain or rxgain while asterisk si running?`
`18:52.55`	`low_rad`	`bjohnson: ok`
`18:52.56`	`bjohnson`	`~docs`
`18:52.57`	`jbot`	`Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org`
`18:52.57`	`bkw_`	`Logan, twisted says you can`
`18:52.58`	`bjohnson`	`low_rad: ^^`
`18:52.59`	`iCEBrkr`	`drumkilla: So you'll have that work by tomorrow night??`
`18:53.03`	`iCEBrkr`	`err working`
`18:53.13`	`drumkilla`	`iCEBrkr: ha.`
`18:53.17`	`iCEBrkr`	`:P`
`18:53.17`	`greg_work`	`drumkilla: as opposed to modules.conf, or for controlling what's built?`
`18:53.27`	`bkw_`	`greg_work, no need to build whats not needed`
`18:53.28`	`drumkilla`	`for controlling what's built`
`18:53.29`	`iCEBrkr`	`bkw_: Dude, keep it in your pants.`
`18:53.29`	`low_rad`	`bjohnson: but does that have dev-docs to integrate modems with asterisk ?`
`18:53.56`	`*** join/#asterisk Mike (~mike@201.135.48.217)`
`18:54.01`	`bjohnson`	`low_rad: I have no idea.. I took the easy way and bought voip hardware`
`18:54.13`	`low_rad`	`bjohnson: hmm, I see, thanks`
`18:54.15`	`low_rad`	`bye`
`18:54.24`	`bjohnson`	`low_rad: btw`
`18:54.30`	`bjohnson`	`no hardware is needed`
`18:54.38`	`FuRR_`	`why would anyone want to run a modem with asterisk`
`18:54.49`	`low_rad`	`ok`
`18:54.58`	`iCEBrkr`	`FuRR_: What do you mean?`
`18:54.58`	`bjohnson`	`low_rad: you could use softphones to connect to * and a voip provider for voip<->pstn`
`18:55.14`	`low_rad`	`bjohnson: I see`
`18:55.17`	`twisted[work]`	`bkw_, ONLY in HEAD - reload chan_zap.so will adjust the tx/rx gain settings`
`18:55.23`	`bkw_`	`ya`
`18:55.26`	`bkw_`	`should have said that`
`18:55.27`	`bkw_`	`:P`
`18:55.37`	`FuRR_`	`iCEBrkr: low_rad was asking if modems would work, and I am just questioning why he would need one`
`18:55.40`	`BoRiS`	`twisted!!!!!!`
`18:55.42`	`postel`	`bkw_: a menuconfig and an apt-like backend to go out there and pick up things that are missing (kernel sources packages and everything) on system that support remote package fetches`
`18:56.06`	`bkw_`	`ok that came out wrong`
`18:56.07`	`bkw_`	`so wrong`
`18:56.08`	`greg_work`	`postel: why not just use apt?`
`18:56.24`	`drumkilla`	`bkw_: do you want to touch my cvs head?`
`18:56.28`	`FuRR_`	`bkw_: i was thinkin the same thing, but being polite`
`18:56.57`	`postel`	`greg_work: you are, but you need something to check for dependencies, and thats gonn ahappen only if you pick up a distro package, if you use the cvs that mechanism aint triggered`
`18:56.58`	`bjohnson`	`drumkilla: are there bugs?`
`18:57.05`	`drumkilla`	`no bugs :)`
`18:57.55`	`greg_work`	`postel: but if you're running cvs you should have the skills to do that (and you only have to do it once...)`
`18:58.03`	`drumkilla`	`we could have an autoconf for checking dependencies`
`18:58.08`	`drumkilla`	`but I don't know a thing about that`
`18:58.21`	`greg_work`	`yeah, autoconf is fine`
`18:58.29`	`FuRR_`	`drumkilla: i just use some scripts i lifted out of Asterisk@Home`
`18:58.53`	`FuRR_`	`it pulls down everything you need and then cvs and then makes it all`
`18:58.54`	`bkw_`	`drumkilla, I dont know what to think .. you saying that..`
`18:58.59`	`bkw_`	`I didn't corrupt you.. I'm sure of it`
`18:59.08`	`drumkilla`	`haha`
`18:59.14`	`greg_work`	`but its a lot of work to write a whole package fetching system that'll work across different distros (which may also involve setting up a repository).. there are better things to work on ;P`
`18:59.24`	`*** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no)`
`18:59.49`	`greg_work`	`oh yeah, isn't there some asterisk-get.sh script? is that what you're talking about FuRR_ ?`
`19:00.10`	`postel`	`greg_work: well, yes, but we're lazy bastards remember? someone from here coded ADM to drop the volume of your music on the system when a call hits your 79xx and be able to callout by middle button clicking on a gtk widget in the tray ;)`
`19:00.27`	`adjacent`	`anyone have a sip proxy i can use without auth? this is just to test`
`19:00.51`	`*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01v-27-233.d4.club-internet.fr)`
`19:01.12`	`*** join/#asterisk Ahewes (~rsb@209.81.2.58)`
`19:01.39`	`greg_work`	`ah, here it is: http://www.voip-info.org/wiki-Asterisk+Download "asterisk-update.sh"`
`19:02.01`	`FuRR_`	`greg_work: yes`
`19:02.47`	`JerJer`	`adjacent: install asterisk, configure sip.conf`
`19:02.50`	`JerJer`	`problem solved`
`19:03.15`	`modulus_`	`jbot babelfish de en est ist zeit fur rachte!`
`19:03.44`	`*** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net)`
`19:03.53`	`modulus_`	`jbot babelfish de en du wirdest eine krankenschwester brauchen`
`19:04.01`	`modulus_`	`jbot babelfish de en du werdest eine krankenschwester brauchen`
`19:04.19`	`_Brian`	`~lart modulus_`
`19:05.16`	`modulus_`	`jbot babelfish de en _Brian bist eine kleine dummer schweinehund`
`19:05.26`	`*** join/#asterisk zotz (~zotz@24.231.32.191)`
`19:05.26`	`bkw_`	`HEY`
`19:05.30`	`bkw_`	`be nice`
`19:05.35`	`Luke-Jr`	`What is a competitive price for converting an existing analogue phone system to use VoIP? (~1250 users)`
`19:05.43`	`bkw_`	`Luke-Jr, do the research`
`19:05.50`	`bkw_`	`we aren't your research team`
`19:05.53`	`modulus_`	`luke-jr, ask mister jeeves he cares`
`19:05.55`	`Luke-Jr`	`bkw_: I don't know how/where`
`19:06.01`	`bkw_`	`www.voip-info.org`
`19:06.03`	`bkw_`	`google`
`19:06.07`	`*** join/#asterisk PakiPenguin (~info@202.176.254.1)`
`19:06.14`	`PakiPenguin`	`hello everyone`
`19:07.15`	`jarrod`	`it seems like it would be hard to keep track of calls on asterisk coming from ser on where it was coming from`
`19:07.18`	`*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)`
`19:07.30`	`jarrod`	`is there a better way to do accounting on ser -> asterisk from asterisk?`
`19:07.49`	`Slainte`	`every so often when someone uses voicemail (outside caller) I get these errors. WARNING[26415]: Failed to write frame`
`19:08.07`	`*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)`
`19:09.59`	`bjohnson`	`<PROTECTED>`
`19:10.17`	`paulc`	`replacing a small CO with asterisk? ;-)`
`19:10.20`	`paulc`	`Or maybe it's a networked system`
`19:10.26`	`bjohnson`	`sounds like it`
`19:10.41`	`paulc`	`or.. it could really be a large analog system.. large Meridian or something, possibly networked? (think: hospital, university, business park)`
`19:10.49`	`Logan`	`When I dial an invalid extension with my SIP phone, the statemenst for the i extension are never executed. Instead, I just get a busy signal.`
`19:10.50`	`Luke-Jr`	`paulc: a college`
`19:10.59`	`bjohnson`	`I think they are usually digital .. not analogue`
`19:11.02`	`paulc`	`there we go - we have a winner :)`
`19:11.14`	`paulc`	`digital back end, sure.. but I'll bet there's a lot of POTS/2500 type sets on desks..`
`19:11.23`	`paulc`	`no digital feature phones etc.. blinky lights = $$`
`19:11.51`	`FuRR_`	`paulc: what university`
`19:11.54`	`paulc`	`Luke-Jr: You could have a nice networked asterisk setup..`
`19:12.03`	`paulc`	`FuRR_: not me - it's Luke-Jr`
`19:12.09`	`Hmmhesays`	`ahh you can't go wrong with blinky lights`
`19:12.18`	`bjohnson`	`cheapest would be to resuse existing wiring`
`19:12.18`	`FuRR_`	`<~ still waking up`
`19:12.23`	`Hmmhesays`	`the more blinky, the better`
`19:12.31`	`*** join/#asterisk Inv_arp (~junya@adsl-3-255-228.mia.bellsouth.net)`
`19:12.36`	`bjohnson`	`channel banks out the wazoo`
`19:12.37`	`Delvar`	`nn all`
`19:12.41`	`FuRR_`	`i wonder if anyone makes an ATA for Nortel Meridian Sets`
`19:12.52`	`bjohnson`	`FuRR_: nortel`
`19:12.58`	`modulus_`	`furr, i have a meridian fxo<->fxs adapter`
`19:13.24`	`FuRR_`	`bjohnson, modulus_ : do they actually work with the digital handsets with the blinky lights`
`19:13.27`	`paulc`	`There's another company too.. datapulse rings a bell`
`19:13.47`	`Luke-Jr`	`oh well... no useful info on voip-info about such things :(`
`19:13.53`	`bjohnson`	`FuRR_: kind of`
`19:14.11`	`FuRR_`	`i just wish my Nortel IP phone on my desk supported SIP`
`19:14.20`	`tzanger`	`show channels`
`19:14.23`	`tzanger`	`ha`
`19:14.26`	`bjohnson`	`FuRR_: it's usually not worth the headache`
`19:14.37`	`greg_work`	`hey, somewhat OT, but anyone interested in a config tool for spa-841's?`
`19:14.44`	`paulc`	`Luke-Jr: What's your current setup like?`
`19:14.51`	`paulc`	`greg_work: tell me more ;)`
`19:14.52`	`FuRR_`	`bjohnson: true, but i have a crapload of meridian gear here and i want to devlop a test lab and not spend any money`
`19:14.52`	`Slainte`	`logan pastebin your extensions.conf section`
`19:14.59`	`*** part/#asterisk criptos (~criptos@dsl-200-78-109-139.prod-infinitum.com.mx)`
`19:15.23`	`modulus_`	`who here likes to look at perl code?`
`19:15.35`	`*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)`
`19:15.42`	`Inv_arp`	`modulus_: perl code gasp`
`19:15.44`	`modulus_`	`i wrote something for agi in perl`
`19:15.47`	`bjohnson`	`FuRR_: just set up the nortel brain and put * in front of it`
`19:16.01`	`bjohnson`	`FuRR_: and send me some M7310s`
`19:16.04`	`greg_work`	`paulc: i'm thinking, create .ini file with settings, [global] call_return_code=*69 .... [04-32-d4-2d-24-2e] username=119 .... where [04-... is the MAC address of the specific unit`
`19:16.05`	`bjohnson`	`I need about 8`
`19:16.20`	`modulus_`	`junk-y, imma pastebin`
`19:16.25`	`modulus_`	`do i need login for pastebin`
`19:16.26`	`modulus_`	`?`
`19:16.31`	`JunK-Y`	`nah`
`19:16.35`	`Slainte`	`pastebin.ca na`
`19:16.38`	`JunK-Y`	`i wont login to pb to see that.`
`19:16.40`	`paulc`	`greg_work then calling the spc tool and creating a bunch of compiled config files?`
`19:16.40`	`tzanger`	`yup`
`19:16.43`	`greg_work`	`then you run config-spa myconfigfile.conf or config-spa --phone=04-d3-35... myconfigfile.conf`
`19:16.45`	`JunK-Y`	`email me that.`
`19:16.47`	`tzanger`	`our PRI receives echo like mad`
`19:16.52`	`tzanger`	`zap echo can does NOSING for it`
`19:16.59`	`tzanger`	`NO-SING!`
`19:17.10`	`greg_work`	`paulc: no, then it would lookup the IP address based on mac (using rarp) and just submit it using http`
`19:17.11`	`Slainte`	`tzanger, is it all calls on your PRI or only some?`
`19:17.20`	`tzanger`	`Slainte: it seems like 90% of them`
`19:17.30`	`FuRR_`	`bjohnson: nah`
`19:17.38`	`paulc`	`greg_work ah, ok, gotcha.. using the web based config thing.. pretending to be a user/browser..`
`19:17.39`	`bjohnson`	`FuRR_: bastard`
`19:17.42`	`Slainte`	`Hmm, try this.`
`19:17.44`	`greg_work`	`paulc: yes`
`19:18.01`	`tzanger`	`DEEZ GOGGLES... DEY DO NOSING`
`19:18.11`	`Slainte`	`in your zapata.conf`
`19:18.17`	`tzanger`	`er NOSSING is a better way to type it, heh`
`19:18.26`	`Slainte`	`echocancel=128`
`19:18.32`	`modulus_`	`http://pastebin.ca/5597`
`19:18.36`	`modulus_`	`there's my agi script`
`19:18.37`	`Slainte`	`echotraining=800`
`19:18.40`	`modulus_`	`that does billing`
`19:18.57`	`modulus_`	`sorta`
`19:19.07`	`paulc`	`greg_work sounds cool.. I've been working on a web based Sipura provisioning tool.. management of multiple devices etc, it just manages the config files, then you get the SPA devices to provision from a given URL etc`
`19:19.08`	`*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)`
`19:19.12`	`Slainte`	`play with the training value. It will cause a minor pause as the system tries to determine what level of echo there is`
`19:19.24`	`Slainte`	`I only have it for about 20% of my calls and those values helped a fair bit`
`19:19.31`	`Slainte`	`I still need to play with them some more.`
`19:19.33`	`tzanger`	`Slainte: I don't have echo training but I have echocancel=128`
`19:19.33`	`HitTop`	`can TDM400p run on pci 3.3v?`
`19:19.43`	`Slainte`	`the echotraining is the smart one`
`19:19.47`	`tzanger`	`Slainte: yes I know what echotraining does and I am not using it becuase I don't want the pause`
`19:19.50`	`postel`	`ok, little trouble, i have _6,. and _7., and _8., to go over voip providers, common scenario, now a friend passed me his SIP url which is something 75@host, i can URL dial from the ciscos 79xx but it matches the dialplan and goes thru the provider, any nice way to avoid the trouble of building the dialplan from scratch?`
`19:20.00`	`Slainte`	`tzanger, then deal with the echo :)`
`19:20.03`	`tzanger`	`all the echotraining value does is send an impluse and measure the response to more quickly train the echo canceller... that's it`
`19:20.04`	`modulus_`	`who likes my perl code?`
`19:20.11`	`*** join/#asterisk Jas_Williams (~jas_willi@213.94.168.254.wlan-ce1.redcow-hotel-dublin.cust.eircom.net)`
`19:20.31`	`Slainte`	`but it will help train the echo cancellation`
`19:20.38`	`Slainte`	`you can also try the echocancel=256`
`19:21.06`	`paulc`	`postel: Build a rule cisco dialplan that rewrites.. like.. "if I dial *, strip it, then pass whatever after the timeout delay"`
`19:21.11`	`JerJer`	`postel: those are not valid regexp's`
`19:21.11`	`tzanger`	`Slainte: no 256 is too much... I typically used 64 and 128`
`19:21.22`	`Luke-Jr`	`paulc: I don't know much technical details of the current setup, ATM. It's some old computer system doing voicemail, mainly.`
`19:21.24`	`tzanger`	`Slainte: but as I said, echotraining just helps the echocanceller settle faster -- this isn't settling at all..`
`19:21.48`	`JerJer`	`_6X. is valid`
`19:21.53`	`*** join/#asterisk HD (~Henk@82-136-197-93-mx.xdsl.tiscali.nl)`
`19:21.53`	`Slainte`	`tzanger, is it a new PRI?`
`19:22.15`	`Luke-Jr`	`paulc: I'm guessing there's PRIs from the main switch to all the dorm switches`
`19:22.21`	`paulc`	`Luke-Jr: You could probably replace the voicemail server with Asterisk.. then have some SIP extensions hanging off it.. play, learn, etc.. then look at rolling out more asterisk boxes to serve more users..`
`19:22.38`	`postel`	`paulc: sure, i can exclude the bastard, and the next one, and the next one, but i cant just keep on doing it, is it a better way to ignore the dialplan when URL dialling?`
`19:22.39`	`tzanger`	`Slainte: no, coming up on 9 months now`
`19:22.43`	`Luke-Jr`	`paulc: That's what I was thinking-- gradually move it to VoIP and such`
`19:22.47`	`paulc`	`Luke-Jr: Yeah, that'd make sense.. see if you can find out if there's one honking great big switch with a shit load of PRIs.. or if it's a networked system of smaller PBXs`
`19:22.55`	`modulus_`	`there's a 1.5 second pause when my agi script runs`
`19:22.57`	`postel`	`well that "ignore the dialplan" doesnt make sense, hope you got what im saying`
`19:22.59`	`modulus_`	`the question is, where?`
`19:23.03`	`tzanger`	`wtf`
`19:23.06`	`tzanger`	`hang on I think I'm just stupid`
`19:23.09`	`Luke-Jr`	`paulc: Well, the main phone switch is like 4 huge towers ;)`
`19:23.20`	`tzanger`	`need to wait for inactivity to test that theory though`
`19:23.23`	`Luke-Jr`	`paulc: and mostly electronic-less at the dorms`
`19:23.27`	`paulc`	`postel: I had the same situation with a client of mine who wanted to dial INOC-DBA but had a clash with the local dialplan.xml on the Ciscos..`
`19:23.29`	`JerJer`	`Luke-Jr: evil`
`19:23.31`	`Slainte`	`you fat finger a config option?`
`19:23.34`	`JerJer`	`Luke-Jr: our switch is 8U`
`19:23.40`	`Luke-Jr`	`JerJer: 8U?`
`19:23.44`	`Slainte`	`I would love to hear your theory, as I have a handful of calls still echo'ed`
`19:23.49`	`JerJer`	`8 rack units`
`19:23.50`	`Slainte`	`always to the same places as well`
`19:24.02`	`postel`	`paulc: dont you leave me hanging here.. and....? ;)`
`19:24.02`	`tzanger`	`8U? jesus what are you planning on doing, sleeping in that thing?`
`19:24.12`	`tzanger`	`Slainte: yes`
`19:24.14`	`tzanger`	`Slainte: badly`
`19:24.17`	`paulc`	`postel: And.. I did what I said.. we created a dialplan rule to override it.. let me see if I can find it`
`19:24.20`	`JerJer`	`32 DS-3s`
`19:24.36`	`tzanger`	`JerJer: that's a lot of fucking channels`
`19:24.45`	`Slainte`	`wowo that is`
`19:24.50`	`tzanger`	`twenty one thousand channels`
`19:24.52`	`tzanger`	`holy fuck`
`19:25.06`	`JerJer`	`we don't have that many lit up`
`19:25.08`	`tzanger`	`21504`
`19:25.10`	`*** join/#asterisk Astinus_ (~abba@213.167.111.138)`
`19:25.21`	`Slainte`	`JerJer: Well that makes it ok then.`
`19:25.23`	`Slainte`	`hehe`
`19:25.23`	`JerJer`	`but the capability is there`
`19:25.31`	`*** join/#asterisk flosch (~flosch@my-247-014.dsl.ligado.de)`
`19:25.39`	`tzanger`	`well`
`19:25.45`	`paulc`	`Postel: Hmm.. cant' find it.. I think we replaced it with a catchall.. what's in your dialplan.xml?`
`19:25.57`	`Luke-Jr`	`paulc: Basically, I'm considering offering to convert their stuff, but I don't know how much I should quote o.o`
`19:25.58`	`JerJer`	`man what a waist of good cocaine`
`19:26.02`	`JerJer`	`waste ?`
`19:26.05`	`tzanger`	`21376 with four 7-DS1 NFAS groups with no backup Ds`
`19:26.12`	`slePP`	`JerJer: i could inject it into your veins, but dry coke could be bad`
`19:26.13`	`postel`	`paulc: wait, i think i xml-ed my way out of it`
`19:26.13`	`bkw_`	`slePP, really?`
`19:26.20`	`tzanger`	`fix-my-udp?`
`19:26.26`	`Luke-Jr`	`slePP: you're the pastebin.ca person! =p`
`19:26.26`	`paulc`	`Luke-Jr: Pick a number, double it, add 30%..`
`19:26.32`	`slePP`	`bkw_: sorta really :>`
`19:26.34`	`paulc`	`HE IS HE IS!`
`19:26.46`	`Luke-Jr`	`paulc: O.o;;`
`19:26.52`	`*** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net)`
`19:27.00`	`Logan`	`If I use ast_waitstream to allow playback of something to be interrupted by a digit, what happens to that digit?`
`19:27.33`	`JerJer`	`vi apps/app_background.c`
`19:27.35`	`slePP`	`Luke-Jr: i am, yeh`
`19:27.46`	`bjohnson`	`Luke-Jr: even at $50 per port your at $62,500 .. and that's assuming no wiring and no time included`
`19:27.49`	`slePP`	`thanks to a pleasant script kiddie`
`19:27.49`	`twisted[work]`	`the digit is returned as the result of waitstream`
`19:28.10`	`lilo`	`I've never managed to find a rationale in that sort of behavior`
`19:28.19`	`Luke-Jr`	`bjohnson: you don't think the existing hardware would work?`
`19:28.26`	`bjohnson`	`applying paulc's rule ( double it, add 30%) you're at $162,500`
`19:28.27`	`Logan`	`I looked at pbx_builtin_background, and I'm trying to make some custom code of mine behave the same way.`
`19:28.31`	`lilo`	`it's hard to believe people can really equate removing useful resources with constructive activity`
`19:28.34`	`slePP`	`Luke-Jr: what about the pbin?`
`19:28.41`	`twisted[work]`	`lilo, LOL`
`19:28.45`	`bjohnson`	`Luke-Jr: what existing hardware. We have no idea what is there`
`19:28.58`	`Logan`	`A pressed digit does terminate the playback, but then the next statement (which is also a Background) starts running, instead of jumping to the context of the digit that was entered.`
`19:29.01`	`lilo`	`twisted[work]: well, I know there are people who do equate them, but I think it's more in the nature of a rationalization than anything else`
`19:29.02`	`paulc`	`bjohnson: sounds like a bargain to me!`
`19:29.10`	`Luke-Jr`	`slePP: nothing... just never got a reply from you when I IM'd`
`19:29.13`	`lilo`	`twisted[work]: "we demonstrated a problem with your security" 8)`
`19:29.22`	`Logan`	`Whereas if I press the same digit during the Background statement that precedes or follows my custom application, it jumps directly to the extension entered.`
`19:29.23`	`slePP`	`Luke-Jr: uhm. when'd you im? :P`
`19:29.24`	`twisted[work]`	`lilo, yea. I'm not one of them, I just find that sort of thing funny`
`19:29.28`	`Luke-Jr`	`bjohnson: don't know the details, but it currently interfaces to some kind of computer`
`19:29.31`	`lilo`	`twisted[work]: some people are really sincere about it, but maybe they're not thinking through the implications`
`19:29.31`	`slePP`	`i removed yer post for you`
`19:29.39`	`Luke-Jr`	`slePP: right`
`19:29.46`	`twisted[work]`	`or thinking in general :)`
`19:29.54`	`lilo`	`twisted[work]: I would tend to agree`
`19:30.04`	`twisted[work]`	`ah well.. lunch over - back to work ;)`
`19:30.07`	`JerJer`	`slePP: btw, the never expire option, set by default, is pretty evil`
`19:30.16`	`JerJer`	`on your pb`
`19:30.27`	`slePP`	`i like never expires.`
`19:30.31`	`slePP`	`otherwise the archives are just silly`
`19:30.35`	`JerJer`	`Google like indexing them`
`19:30.51`	`JerJer`	`and morons don't bother to censor passwords`
`19:30.59`	`JerJer`	`search for nufone and pastebin :)`
`19:31.17`	`JerJer`	`or nufone and secret`
`19:31.49`	`slePP`	`bad, bad people`
`19:31.57`	`paulc`	`twisted[work]: Hyatt Saint Claire eh? ;-)`
`19:32.40`	`*** part/#asterisk bowman (~rsp@195.46.47.202.static.cablesurf.de)`
`19:32.55`	`slePP`	`yeh. i figure that if they make a 'user', then the post can be deleted by that user if they're logged in. and if they aren't logged in, then it'll check the IP and if remote IP == poster IP, it'll allow deletion/title changes`
`19:33.12`	`JerJer`	`perfect`
`19:33.20`	`FuRR_`	`anyone ever use these`
`19:33.21`	`FuRR_`	`http://www.voipsupply.com/product_info.php?cPath=95_105&products_id=262&desc=ArtDio%20IPF-2000`
`19:34.03`	`*** join/#asterisk mbranca_home (~matteo@adsl-84-222-11-154.tiscali.it)`
`19:35.33`	`HitTop`	`I would like to ask about Sipura SPA-3000. I provides 1 fxs and 1 fxo, I wonder how it connects to asterisk`
`19:35.48`	`modulus_`	`who here is good with perl?`
`19:35.49`	`bjohnson`	`HitTop: sip`
`19:35.55`	`paulc`	`HitTop: The answer is "very well"`
`19:36.11`	`glock`	`Anyone please help me with fax detection? In my extensions.conf i have exten => fax,1,Dial,Zap/4 exten => s,1,Dial,Zap/1. Although it rings on zap1 even if it is a fax. I've tried exten => s,1,Answer - But then it doesnt ring anywhere. any idea?`
`19:36.35`	`HitTop`	`so there's an ethernet jack to connect to the network as well?`
`19:36.40`	`bjohnson`	`yes`
`19:36.51`	`paulc`	`HitTop: Yup.. a port for a phone line, a port for a phone, ethernet jack + power connector`
`19:36.56`	`HitTop`	`and asterisk no need to install zaptel drivers?`
`19:36.56`	`modulus_`	`furr_, agi`
`19:37.02`	`tzanger`	`glock: do you have faxdetect in your zapata.conf?`
`19:37.50`	`modulus_`	`furr_, my agi script causes 1.5 second pause`
`19:37.54`	`modulus_`	`i want to eliminate that`
`19:37.54`	`godsmoke`	`who has voicepulse connect ....`
`19:37.58`	`godsmoke`	`I need questions answered :)`
`19:38.01`	`modulus_`	`godsmoke, i do`
`19:38.03`	`FuRR_`	`pastebin it`
`19:38.16`	`modulus_`	`http://pastebin.ca/5598`
`19:38.19`	`modulus_`	`there's my agi script`
`19:38.20`	`HitTop`	`thank you!!!^^`
`19:38.22`	`modulus_`	`it's all in perl`
`19:38.39`	`Sauron	`Hum.`
`19:38.41`	`modulus_`	`godsmoke, yes i've played with ani and caller-id a little with vp`
`19:38.51`	`godsmoke`	`well, that wasn't really my question`
`19:39.36`	`modulus_`	`godsmoke, yes i still get callerid via my ANI on blocked calls`
`19:39.44`	`godsmoke`	`really?`
`19:39.54`	`godsmoke`	`VP Connect said no`
`19:39.55`	`godsmoke`	`:)`
`19:40.01`	`modulus_`	`oh`
`19:40.08`	`modulus_`	`i don't have my toll-free with them`
`19:40.16`	`godsmoke`	`they don't offer toll-free`
`19:40.24`	`jarrod`	`why is it when i login as an agent it keeps my phone on`
`19:40.40`	`jarrod`	`and when i hang up it logs me out`
`19:40.44`	`godsmoke`	`modulus_: so you're sure ... that ANI is valid from the PSTN on CID-blocked calls?`
`19:40.48`	`modulus_`	`godsmoke, i only use them for about 6 numbers`
`19:41.08`	`modulus_`	`godsmoke, i have rec'd callerid info when it was blocked`
`19:41.17`	`godsmoke`	`hrm ...`
`19:41.21`	`godsmoke`	`on which string?`
`19:41.24`	`godsmoke`	`CPN or ANI?`
`19:41.43`	`*** join/#asterisk wwhome (~andreas@woffi.planix.com)`
`19:43.44`	`FuRR_`	`modulus_: dont create and recreate the database connection`
`19:44.04`	`modulus_`	`furr, that's an obvious`
`19:44.13`	`modulus_`	`but that wouldn't cause 1.5 seconds of lag`
`19:44.23`	`FuRR_`	`it could`
`19:44.37`	`godsmoke`	`modulus_: have you seen, does the Caller ID come in on the CPN and/or ANI strings on blocked calls?`
`19:44.38`	`FuRR_`	`if your database machine was on a 56k frame halfway across the world`
`19:44.41`	`modulus_`	`i'm thinking about just writing a daemon and having agi talk to the deamon via ipc`
`19:44.43`	`*** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)`
`19:44.57`	`modulus_`	`the daemon would always keep query pool open`
`19:45.31`	`modulus_`	`godsmoke, good question i'm not sure.`
`19:45.36`	`godsmoke`	`alright`
`19:45.46`	`FuRR_`	`modulus_: you die if the select statement fails -- but shouldnt you die up top where you set SIPUSER to ""`
`19:45.49`	`godsmoke`	`but, you answered the important question, which is that it does come in :)`
`19:45.53`	`godsmoke`	`very nice, thanks`
`19:46.06`	`modulus_`	`godsmoke, ani doesn't use CPN afaik`
`19:46.07`	`FuRR_`	`err undef`
`19:46.15`	`godsmoke`	`use?`
`19:46.20`	`modulus_`	`errr`
`19:46.25`	`godsmoke`	`heh -- no, ANI and CPN are separate ...`
`19:46.27`	`modulus_`	`ani isn't sent by calling party`
`19:46.31`	`modulus_`	`yes`
`19:46.36`	`FuRR_`	`modulus_: line 26`
`19:46.38`	`godsmoke`	`correct -- but it's sent by the originating CLEC switch`
`19:46.42`	`modulus_`	`furr, lemme see`
`19:47.14`	`FuRR_`	`you die at line 35 if the select statement fails, and since SIP user is in the select statement, then you prolly want to bow out at 26`
`19:47.15`	`modulus_`	`furr, yeah exit should occur if no sipuser matched`
`19:47.26`	`modulus_`	`furr, anything more useful?`
`19:47.31`	`godsmoke`	`modulus_: well, thanks for the info -- I'll be setting up VP Connect in a few days`
`19:47.38`	`modulus_`	`furr, thanks for your insight but those are all obviouses`
`19:48.10`	`FuRR_`	`modulus_: sometimes the forrest is best seen by looking at the trees`
`19:48.13`	`modulus_`	`i need a solution to eliminate the 1.5 seconds of pause`
`19:48.31`	`modulus_`	`furr, my daemon idea would probably be best`
`19:48.39`	`modulus_`	`unless agi just sucks, then i'm fucked`
`19:48.50`	`Sauron	`I got * working with 7940/sccp. Nyah Nyah. :)`
`19:48.56`	`FuRR_`	`ive got a couple of php and perl agi's and they dont sucl`
`19:48.57`	`modulus_`	`actually i could re-write in C, that'll work faster`
`19:49.03`	`modulus_`	`furr, may i?`
`19:49.16`	`glock`	`tzafrir: I have faxdetect=incoming in zapata.conf`
`19:50.14`	`*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)`
`19:50.59`	`bprice20`	`did the format that realtime inserts records into mysql database, recently change`
`19:51.13`	`bprice20`	`the duration column in particular`
`19:51.27`	`bprice20`	`it was in seconds b4 now its minutes:seconds`
`19:51.29`	`*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)`
`19:51.51`	`FuRR_`	`modulus_: you can do whatever you want....`
`19:52.28`	`modulus_`	`furr, perl code?`
`19:53.45`	`*** join/#asterisk Grooby (~Grooby@12.22.232.212)`
`19:54.25`	`jarrod`	`i dislike having to remain with my phone off hook to receive calls from a queue`
`19:55.01`	`Sauron	`You could have the queue call your phone`
`19:55.11`	`Grooby`	`hola`
`19:55.47`	`Grooby`	`anyone here following the SIP jitter thread on the mailing list?`
`19:57.46`	`greg_work`	`paulc: hm, what would work better.. [1.2.3.4] mac=3cd32f23 ... so you can sipuraconfig -u 1.2.3.4 (to just config that unit), or call the phone section something arbitrary, like [119] and allow to specify either an IP or a MAC, then you could do sipuraconfig --unit=119 ?`
`19:57.59`	`greg_work`	`i kinda like the last way i think`
`19:58.07`	`jarrod`	`sauron like member => SIP/yadiyadi ?`
`19:58.18`	`Sauron	`jarrod: Look at queues.conf`
`19:58.40`	`paulc`	`yeah I kinda like that too.. I don't care about IP addresses (which may be dynamic) or MAC addresses.. this phone is extension 123, now go configure it already!`
`19:58.41`	`Sauron	`; A strategy may be specified. Valid strategies include:`
`19:58.48`	`paulc`	`I guess dynamic IP addresses get "interesting"`
`19:59.36`	`*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)`
`19:59.56`	`eKo1`	`What codec are you guys using for the conference call?`
`20:01.07`	`*** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca)`
`20:01.36`	`jarrod`	`oh werd it works :)`
`20:02.28`	`modulus_`	`furr, what do your perl agi scripts do?`
`20:03.54`	`FuRR_`	`modulus_: convience stuff, queue registration, call center logging`
`20:04.16`	`modulus_`	`furr, any database queries?`
`20:04.17`	`FuRR_`	`modulus_: time/date, wheather, directions`
`20:04.18`	`MrClean`	`Can someone suggest a strategy for using the queue position/hold time announcement facility for announcing a generic "all agents are busy" message sans queue position? I know I can override some of the recordings, but there doesn't appear to be a way of overriding the position number itself.`
`20:04.23`	`FuRR_`	`modulus_: yes`
`20:04.56`	`FuRR_`	`modulus_: why do you loop through each row returned from your select statement, can a given SIP extentsion have more than one record in your database`
`20:05.49`	`modulus_`	`i don't know`
`20:06.55`	`bjohnson`	`eKo1: ulaw`
`20:07.57`	`schwagner`	`bjohnson: have you noticed if the beeps you hear are related to pci bus traffic, such as a network card?`
`20:08.22`	`Jas_Williams`	`.`
`20:08.35`	`modulus_`	`furr, that while loop wouldn't cause that much pause`
`20:08.41`	`bjohnson`	`schwagner: haven't found the cause yet`
`20:08.53`	`dan2`	`kram: you might want to turn up the volume on the phone`
`20:08.53`	`*** join/#asterisk randu (~randu@dsl-pppoe-pool1-37.d02.ncshlt.infoave.net)`
`20:09.07`	`eKo1`	`bjohnson: Bullshit because I'm using ulaw.`
`20:09.18`	`schwagner`	`bjohnson: well, i've pretty much nailed mine down to network traffic, give that a shot`
`20:09.21`	`FuRR_`	`modulus_: no, but its unnessecary`
`20:09.36`	`FuRR_`	`modulus_: and would screw your app up if for some reaosn someone did get a 2nd entry`
`20:09.58`	`modulus_`	`no, sipuser is unique key`
`20:10.18`	`*** join/#asterisk jets (jetsn@omega.pmt.org)`
`20:10.30`	`*** join/#asterisk Vico (vico@c-666470d5.1011-1-64736c12.cust.bredbandsbolaget.se)`
`20:10.36`	`modulus_`	`furr_, can you pastebin some of your perl?`
`20:10.57`	`FuRR_`	`not off hand, dont have any on me`
`20:11.05`	`modulus_`	`just login to your * box`
`20:11.06`	`modulus_`	`duh`
`20:11.52`	`FuRR_`	`modulus_: that would require my * to be connected to an outbound internet source -- currently its in a lab, that i am not near, other wise i would`
`20:12.06`	`jets`	`If you have agents login, but there extension is an agi script-- will app_queue assume an agi working with a call was the agent answering?`
`20:12.06`	`modulus_`	`you suck`
`20:12.08`	`FuRR_`	`when do you get the pause`
`20:12.17`	`modulus_`	`as soon as agi executes`
`20:12.33`	`FuRR_`	`so the extension picks up .. pause .. and then proceeds`
`20:13.10`	`FuRR_`	`take away -T and -w from your perl statement, you dont want warnings going to console, and you dont need taint`
`20:13.12`	`modulus_`	`yeah it executes when a sip user dials out`
`20:13.27`	`modulus_`	`furr, you take away taint checking?`
`20:13.45`	`modulus_`	`i'm shocked and appalled`
`20:14.15`	`FuRR_`	`what are you going to do with error messages?`
`20:14.22`	`FuRR_`	`its not like you can actually see them`
`20:14.49`	`jets`	`Maybe I can do what I want with an extension plan -- I want to queue calls, but instead of sending directly to a device/phone, i want both parties to be put in a conference room -- but only if the agent first answer's there phone`
`20:15.14`	`bjohnson`	`eKo1: worked for me (tm)`
`20:15.32`	`modulus_`	`furr, future bugs can be found quicker`
`20:16.33`	`FuRR_`	`modulus_: at what expense...lets think about this`
`20:16.38`	`FuRR_`	`what inputs are you receiving?`
`20:16.55`	`machinehd`	`I have an d* box with a pri card. I can call in and out just fine. However there is no dialtone when the sip handset is picked up. Is there a setting I've messed up?`
`20:17.02`	`FuRR_`	`are you modifying ENV{PATH} ?`
`20:17.04`	`modulus_`	`furr, $ARGV[0] for one`
`20:17.16`	`FuRR_`	`modulus_: yes, but that data comes from *`
`20:17.25`	`FuRR_`	`so you know what the data is, or what it can be`
`20:17.28`	`modulus_`	`also, $AGI->ReadParse()`
`20:17.29`	`eKo1`	`heh, got a 403 Forbidden`
`20:17.36`	`modulus_`	`i don't trust data from *`
`20:17.44`	`modulus_`	`i never trust external data`
`20:17.52`	`FuRR_`	`wait`
`20:18.00`	`FuRR_`	`this application is for SIP users right?`
`20:18.03`	`modulus_`	`yup`
`20:18.11`	`jets`	`machinehd: hard phones generate the 'dialtone'`
`20:18.12`	`FuRR_`	`and YOU define SIP users in sip.conf right?`
`20:18.15`	`jets`	`the pbx doesn't--`
`20:18.18`	`*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)`
`20:18.22`	`modulus_`	`furr, no`
`20:18.26`	`modulus_`	`mysql`
`20:18.55`	`FuRR_`	`so your reading your sip users out of mysql as well?`
`20:19.05`	`modulus_`	`yes`
`20:19.27`	`jets`	`Can anyone explain Asterisks 'dial' (M) command option? An example of a dial() macro would be helpful`
`20:21.52`	`jarrod`	`dcd`
`20:22.26`	`*** join/#asterisk fixx_ (~null@tbnb-165-58-16.telkomadsl.co.za)`
`20:22.50`	`fixx_`	`helo`
`20:23.09`	`bprice20`	`who isn't using realtime nowadays`
`20:23.09`	`*** join/#asterisk jayden (~ircatjerr@65.170.43.34)`
`20:23.16`	`fixx_`	`could you guys please assist me with the sip config of asterisk?`
`20:23.16`	`bprice20`	`realtime = the norm`
`20:23.28`	`bprice20`	`fixx_ sup`
`20:24.04`	`FuRR_`	`modulus_: so why not add a column to the existing mysql sip config table and just use * db variables ?`
`20:24.13`	`fixx_`	`i want to just configure a basic setup that will allow me to connect two pc's on my lan to my asterisk pbx and then let the one pc phone the other's extension`
`20:24.29`	`machinehd`	`jets, thanks. So if there's no dialtone then it's a problem with the phone?`
`20:24.40`	`eKo1`	`Anybody using Kayote for call termination?`
`20:24.49`	`fixx_`	`i am using x-lite on both machines, but i don't think they are connected to the pabx`
`20:24.57`	`modulus_`	`furr, b/c i have no idea what that all means`
`20:24.58`	`fixx_`	`even though the status says connected`
`20:25.06`	`riksta`	`anyone got any feedback on ADM for me?`
`20:25.19`	`*** join/#asterisk jsolares (~jsolares@200.30.141.85)`
`20:25.21`	`malcolmd`	`riksta: i'm trying to d/l it..server's timing out`
`20:25.35`	`riksta`	`really?`
`20:25.37`	`malcolmd`	`er..scratch that..`
`20:25.37`	`malcolmd`	`heh`
`20:25.44`	`riksta`	`i was gonna say`
`20:25.53`	`malcolmd`	`you switched to 0.7 in between a timeout and my clicking`
`20:25.57`	`riksta`	`no`
`20:26.09`	`riksta`	`it's been 0.7 since 5pm`
`20:26.12`	`riksta`	`GMT`
`20:26.30`	`*** join/#asterisk chaoscon (~ph33r@chaoscon.user)`
`20:26.49`	`malcolmd`	`it's been a while since i click'd :D maybe i hit it at just the wrong time ;)`
`20:26.57`	`malcolmd`	`anywho, i'll fire it up shortly`
`20:27.02`	`fixx_`	`can i only edit extensions.conf and sip.conf for this ?`
`20:27.10`	`riksta`	`great, read the the faq in the readme`
`20:27.17`	`riksta`	`you need to hack a perl module, that has a bug in it`
`20:27.21`	`malcolmd`	`will do`
`20:27.32`	`fixx_`	`this connection i am irc'ng on so lagged :/`
`20:28.10`	`*** join/#asterisk DaPrivateer (Privateer@CRIMSON.OFF-HOURS.COM)`
`20:28.18`	`nestAr`	`whee..`
`20:28.23`	`*** join/#asterisk criptos (~criptos@dsl-200-78-109-139.prod-infinitum.com.mx)`
`20:28.25`	`nestAr`	`i'm almost finished with my dialplan`
`20:28.40`	`criptos`	`Can I use 4 tdm400p with 4 fxo ports at a single machine ?`
`20:28.43`	`adjacent`	`nice. i got it working =)`
`20:28.45`	`*** part/#asterisk jayden (~ircatjerr@65.170.43.34)`
`20:28.47`	`nestAr`	`my extensions.conf is getting long.`
`20:28.56`	`criptos`	`machine: pIV 3gigs 1gig ram fsb 800 mhz`
`20:29.02`	`nestAr`	`criptos: you want 4 incoming lijes? yeah.. it'll work`
`20:29.07`	`FuRR_`	`modulus_: you might wanna try EAGI and put your AGIs on another box for testing, maybe the 1.5sec delay isnt the AGI at all, maybe its ite the box itself`
`20:29.13`	`adjacent`	`thanks everyone for the help. ended up having to prefix with 999 to direct calls through the PSTN becuase of the previous providers dialplan`
`20:29.18`	`criptos`	`16 incoming lines... nestAr`
`20:29.19`	`DaPrivateer`	`heh... i dunno how much n00b's are tolerated here... but im trying to set up asterisk for a PSTN to SIP gateway... if anyone can provide me with some help, or point me to a url, it would be greatly appreciated`
`20:29.29`	`nestAr`	`criptos: o, sorry`
`20:29.46`	`nestAr`	`criptos: does the machine have an irq controller?`
`20:29.53`	`criptos`	`so? I machine can handle 16 incoming analog lines at a 4xtdm400p`
`20:30.12`	`nestAr`	`sounds like a pri would be cheaper than 16 lines`
`20:30.33`	`moonwick`	`and simpler`
`20:30.39`	`nestAr`	`yeah.. what he said`
`20:30.49`	`nestAr`	`(s)`
`20:31.14`	`criptos`	`I�m unable to use a pri, neither a e1... they what to "charge me" for the 16 channles at the e1 different thatn 16 analog lines`
`20:31.14`	`FuRR_`	`isnt e1 30 chan?`
`20:31.28`	`nestAr`	`well, in theory, it should work..`
`20:31.41`	`VoIPMasta`	`32 chans`
`20:31.43`	`nestAr`	`but i've noticed that the tdm400's can be flakey for some.. and rock solid for others.`
`20:31.49`	`FuRR_`	`criptos: you could take your 16 analog lines and put them in a channel bank, and then run a T1/E1 from the chan bank to your *`
`20:32.02`	`criptos`	`yeap, but tmy provider, doesn�t want to give a e1 for only 16 channels. They want to charge me for the FULL e1`
`20:32.02`	`*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)`
`20:32.02`	`*** join/#asterisk PakiPenguin (~meow@202.176.254.1)`
`20:32.02`	`PakiPenguin`	`hello`
`20:32.02`	`VoIPMasta`	`E1 = 32 channels`
`20:32.40`	`*** join/#asterisk DevilFish (~me@staff211.qtm.net)`
`20:32.41`	`*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)`
`20:33.00`	`adjacent`	`you could use the other 16 for internet traffic =)`
`20:33.04`	`criptos`	`Furr: 1 e1 diium single channel card + channel banck is more expensive thatn 4 tdm400p card`
`20:33.07`	`paulc`	`E1 = 30B+D :-)`
`20:33.12`	`DevilFish`	`I'm really struggling with this and need some help, could any PolyCom guru have a quick look`
`20:33.13`	`DevilFish`	`http://www.mail-archive.com/asterisk-users@lists.digium.com/msg70340.html`
`20:33.16`	`modulus_`	`furr, what's the difference between eagi and agi?`
`20:33.17`	`nestAr`	`i guess an E1 is more than 16 lines.. around here a POTS line runs about $80 our PRI is 400`
`20:33.19`	`modulus_`	`jbot eagi?`
`20:33.21`	`PakiPenguin`	`is there a web signup script for * ? like users signup from the web and it adds the enteries in extensions & reload it`
`20:33.21`	`FuRR_`	`criptos: but youll have IRQ issues with 4 cards`
`20:33.27`	`adjacent`	`ManxPower: i got it working. thanks for the direction earlier`
`20:33.47`	`FuRR_`	`modulus_: eagi basically offs the proccessing of the agi to another resource`
`20:33.54`	`criptos`	`FuRR even if I use a good motherboard? and each card has its own irq?`
`20:34.15`	`paulc`	`DevilFish: I've got a Polycom IP600 and have had calls in excess of 2 hours with no trouble, NAT phone to public IP`
`20:34.21`	`FuRR_`	`http://www.voip-info.org/tiki-index.php?page=Asterisk%20AGI`
`20:34.21`	`paulc`	`can you replicate with a softphone/other phone?`
`20:34.23`	`DevilFish`	`I should also add that the Switch I'm using is a metaswitch 3500`
`20:34.39`	`modulus_`	`furr, which resource would that be?`
`20:34.58`	`FuRR_`	`criptos: who knows, i just generally try to stick as few cards in a box as possible, less to go wrong....`
`20:35.16`	`FuRR_`	`modulus_: another linux box ...`
`20:35.21`	`DevilFish`	`paulic, I've held them for as much a three hours actually but it just seems random.. plus know one knows about the "failed to grab lock"`
`20:35.35`	`modulus_`	`furr, that's so messy`
`20:35.39`	`DevilFish`	`sometimes we get a "ignoring too old packet" type notice`
`20:35.40`	`FuRR_`	`modulus_: toaster oven... it doesnt really matter, just whatever resource is running your app. Say like you have a VoiceXML Application server`
`20:36.05`	`Madkiss`	`/usr/asterisk/usr/sbin/asterisk:/usr/asterisk/usr/lib/asterisk/modules/chan_sip.so: undefined symbol 'ast_moh_stop'`
`20:36.08`	`Madkiss`	`lazy binding failed!`
`20:36.10`	`Madkiss`	`Now what the hell is _that?!`
`20:36.12`	`Madkiss`	`zsh: segmentation fault (core dumped) /usr/asterisk/usr/sbin/asterisk -cvvvvvvvvvvvvvvvv`
`20:36.34`	`ManxPower`	`adjacent, What did I tell you to do?`
`20:36.39`	`DevilFish`	`paulc: it really only seems to be calls to the pstn that it happens too`
`20:37.11`	`paulc`	`DevilFish: Hmm.. PSTN via PRI or through a VoIP provider? And which version of asterisk / Polycom firmware?`
`20:37.17`	`DevilFish`	`ManxPower: U use polycoms right?`
`20:37.41`	`DevilFish`	`lastest polycom firmware`
`20:37.46`	`ManxPower`	`DevilFish, Well not personally, but we have some on our VoIP network`
`20:37.53`	`DevilFish`	`asterisk is 1.0.5 I believe`
`20:37.55`	`DevilFish`	`lemme check`
`20:38.12`	`adjacent`	`ManxPower: read the man page =)`
`20:38.20`	`FuRR_`	`modulus_: actually if you ran your app out of apache/mod_perl you might get better performance in that you can cache database calls and such`
`20:38.20`	`adjacent`	`just kidding. just gave some good pointers`
`20:38.30`	`DevilFish`	`I got: Connected to Asterisk CVS-v1-0-01/17/05-11:18:25 currently running on pbx`
`20:38.33`	`FuRR_`	`it would scale alot better and not potentially bring down your pbx`
`20:38.35`	`*** join/#asterisk yebo (private@64.119.201.17)`
`20:38.39`	`yebo`	`'allo!`
`20:39.07`	`FuRR_`	`but then again i tend to overbuild`
`20:39.10`	`DevilFish`	`PSTN via SIP is what we're doing to our own metaswitch 3500`
`20:39.10`	`yebo`	`has anyone here done any research into Verisign's SIP7 service?`
`20:39.29`	`adjacent`	`im thinking about moving the working VOCAL setup to an *, i dont have a digium card, but i have a cisco interfacing my PRI. can i do this with a p2 500MHz, 256 RAM? or do i need to beef it up. looking at around 30 lines.`
`20:40.50`	`*** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com)`
`20:41.08`	`ManxPower`	`adjacent, Beef up the CPU.`
`20:41.27`	`ManxPower`	`adjacent, 500Mhz will prolly work, but why take the chance?`
`20:42.17`	`*** join/#asterisk voipme (~chatzilla@83-70-245-167.b-ras1.prp.dublin.eircom.net)`
`20:42.36`	`*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)`
`20:42.40`	`*** join/#asterisk greendisease (~jack@greendisease.fedora)`
`20:42.54`	`terrapen`	`wheeeee`
`20:43.11`	`criptos`	`I have a 4 fxo ports card. 3 ports 2-4 are Actual HookStaet: offhook and 1 is Actual Hookstate: Onhook`
`20:43.12`	`terrapen`	`1-866-859-7359`
`20:43.14`	`terrapen`	`try it out.`
`20:43.17`	`criptos`	`and this is not true...`
`20:43.21`	`criptos`	`what can i debug?`
`20:43.33`	`jarrod`	`hmm i need a way to determine the source of the inbound call from a SER proxy so I can forward different people to different contexts`
`20:43.35`	`terrapen`	`i need to pay for some real voice recordings`
`20:43.49`	`*** join/#asterisk Wireless (~bad@220.233.77.87)`
`20:44.19`	`Wireless`	`i'm having a problem with one-way audio between two asterisk boxes running sip`
`20:44.36`	`Wireless`	`both boxes are behind individual NAT's`
`20:44.47`	`Wireless`	`and both have 5060 and RTP ports forwarded.`
`20:44.53`	`bjohnson`	`jarrod: can you modify the destination extension at the SER?`
`20:45.18`	`paulc`	`terrapen: that's cool! :-D`
`20:45.19`	`Wireless`	`friend settings on both accounts have nat=yes, localnet and externip's have all been set.`
`20:45.23`	`terrapen`	`paul: thanks!`
`20:45.27`	`Wireless`	`what else could it be?`
`20:45.27`	`jarrod`	`so i can make it say begin with 100 for this customer.. 102 for this customer.. and so forth?`
`20:45.28`	`jarrod`	`hmm`
`20:45.30`	`jarrod`	`not a bad idea`
`20:45.32`	`terrapen`	`i wrote it in about 30min last night`
`20:45.36`	`paulc`	`there were a bunch of weather related prompts added to asterisk-sounds recently - you should check 'em out - might do what you want`
`20:45.43`	`bjohnson`	`Wireless: double nat works for some .. not others`
`20:45.44`	`terrapen`	`paul, none do what this does`
`20:45.48`	`terrapen`	`this is live, real-time weather`
`20:45.51`	`terrapen`	`updated every 5sec`
`20:45.57`	`terrapen`	`from the station on my roof`
`20:46.09`	`Wireless`	`bjohnson: thanks. what sort of problems did you run into and how did you solve them? were you running sip?`
`20:46.10`	`paulc`	`AGI script?`
`20:46.13`	`terrapen`	`i wrote a gathering daemon that pulls the data over the serial port`
`20:46.21`	`bjohnson`	`Wireless: I didn't solve them`
`20:46.27`	`terrapen`	`and then the AGI queries the DB file that the daemon writes to`
`20:46.34`	`Wireless`	`bjohnson: have you had my one way audio problem?`
`20:46.39`	`bjohnson`	`Wireless: got audio from the server to the device but nothing back`
`20:47.02`	`Wireless`	`exactly!`
`20:47.09`	`Wireless`	`the called party does not hear audio.`
`20:47.14`	`bjohnson`	`Wireless: that's the problem ith SIP`
`20:47.21`	`riksta`	`malcolmd: does it work ok?`
`20:47.28`	`*** join/#asterisk aokaze (aokaze@modem92.supermusca.bigsky.net)`
`20:47.33`	`DaPrivateer`	`sigh no one can help me with seting up PSTN -> SIP ?`
`20:47.40`	`terrapen`	`everybody has AGIs that pull from NOAA or weather.com`
`20:47.43`	`bjohnson`	`Wireless: you need a SIP server on a public IP`
`20:47.43`	`terrapen`	`but what fun is that?`
`20:47.53`	`terrapen`	`i wanted to hear the weather right here, right now`
`20:47.58`	`Wireless`	`and when i do sip show channels on both machines, the calling party can see the other's real IP, but the called party could only see the (unreachable) local ip of the caller.`
`20:48.16`	`VoIPMasta`	`Wireless: set nat=yes on sip.conf`
`20:48.20`	`Wireless`	`hence i gather that all rtp packets are being sent to that internal ip, which doesn't work.`
`20:48.30`	`Wireless`	`voipmasta=doesn't help, both that on both boxes.`
`20:48.55`	`Wireless`	`bjohnson - strangely enough, doublenat with my box and a sipura works.`
`20:49.25`	`bjohnson`	`Wireless: I thought of signing all the devices to FWD and using FWD as a go between`
`20:49.49`	`greendisease`	`would anyone be willing to donate a party line for linuxworld for a few hours?`
`20:49.52`	`Wireless`	`bj: i still think i missed something though...`
`20:51.11`	`VoIPMasta`	`which one uses less bandwidth? ilbc or gsm? and is there a significant audio quality loss with ilbc (compared to gsm)?`
`20:51.28`	`tzanger`	`gsm is lower than ilbc`
`20:51.46`	`tzanger`	`manxpower swears ilbc sounds better but every single time I switch over to it I get complaints about audio quality, so I stick with gsm`
`20:51.54`	`VoIPMasta`	`tzanger: which one would you use for dial up connections?`
`20:52.21`	`CoaxD`	`hahaha. We've now got a business whose' phone is solely based upon a 1-800 voip number`
`20:52.27`	`CoaxD`	`(and a local number forwarding to it)`
`20:52.31`	`tzanger`	`I used ilbc over a dialup connection and I was amazed at the quality (this was firefly on win2k --- dialup --- internet provider -- internet -- asterisk`
`20:52.31`	`Wireless`	`bj: can you confirmed that you have tried port forwards and nat=yes and externip/localnet and it still has one way audio?`
`20:52.46`	`PatrickDK`	`coaxd, sounds like fun`
`20:52.47`	`tzanger`	`CoaxD: I have that :-)`
`20:52.52`	`tzanger`	`100% of calls are VOIP`
`20:52.54`	`CoaxD`	`PatrickDK: It is! We've never done it before`
`20:53.01`	`VoIPMasta`	`CoaxD: almost everyone here uses VoIP + DIDs`
`20:53.11`	`CoaxD`	`Eventually i'm going to get her set up with a branch off a local line for outbound local calls`
`20:53.26`	`VoIPMasta`	`tzanger: so should I stick to ilbc for dial up users?`
`20:53.32`	`CoaxD`	`(So she doesn't pay for outgoing usage)`
`20:53.38`	`tzanger`	`VoIPMasta: try it`
`20:53.42`	`CoaxD`	`but beyond that, its all voip! WOOHOO!`
`20:53.56`	`VoIPMasta`	`CoaxD: guess what... the world is round :)`
`20:54.05`	`bjohnson`	`Wireless: no`
`20:54.05`	`CoaxD`	`VoIPMasta: YOU'RE FUCKING KIDDING ME!`
`20:54.17`	`Wireless`	`bj: so what have you tried so far?`
`20:54.25`	`VoIPMasta`	`CoaxD: Don't take me wrong, I'm just having some fun...`
`20:54.31`	`CoaxD`	`VoIP: Hehe. me too`
`20:54.36`	`bjohnson`	`Wireless: zeeek has. I want a solution that does not require remote router config or knowing remote ips`
`20:54.48`	`slePP`	`but i wonder if you're discussing otherstuff`
`20:54.51`	`*** join/#asterisk zeut (~zeutzeut@ool-4350c7ad.dyn.optonline.net)`
`20:54.52`	`PatrickDK`	`coaxd, you ever sell them fxo modules?`
`20:54.53`	`CoaxD`	`VoIP: keep this in mind: I've been doing the voip thing for a while now. 8 months maybe? I use it. it is an integral part of my life. I've just never solely relied on it for a business`
`20:54.53`	`VoIPMasta`	`CoaxD: in here we all use VoIP to handle our calls, it's no "big news" to have a 1-800 DID forwarded through VoIP`
`20:54.57`	`bjohnson`	`Wireless: stun .. got me the one way audio`
`20:55.08`	`slePP`	`stun is annoying`
`20:55.09`	`CoaxD`	`PatrickDK: fxo modules? You must be thinkin' of the wrong guy`
`20:55.17`	`VoIPMasta`	`CoaxD: .ok I see your point`
`20:55.23`	`PatrickDK`	`maybe, just remember it starts with c :)`
`20:55.26`	`Wireless`	`bj: knowing remote real ip is no problem here... who's zeek?`
`20:55.27`	`CoaxD`	`VoipMasta: yeah, same over here. I've had multiple DIDs for months and months`
`20:55.33`	`eKo1`	`VoIPMasta: It isn't "big news" but it is a bitch to get working.`
`20:55.33`	`bjohnson`	`Wireless: thinking of trying fwd to act as go between .. but haven't tried yet (should work though)`
`20:55.36`	`sezuan`	`How can I do a 'factory reset' on PA168 based phones? The new 1.42 firmware behaves a bit funny.`
`20:55.40`	`CoaxD`	`VoipMasta: Its just cool that we're actually using it for business that way now.`
`20:55.46`	`bjohnson`	`Wireless: guy from France`
`20:55.51`	`VoIPMasta`	`eKo1: ??`
`20:55.52`	`tzanger`	`hmm why do DTMF tones need to be twisted?`
`20:55.53`	`CoaxD`	`VoipMasta: (I've been using an X100P and asterisk to handle another of my offices for a while now, too.)`
`20:55.53`	`Wireless`	`bj: and i didn't think * supports stun, and don't want to set up second computer as SER`
`20:55.58`	`Wireless`	`bj: why not use IAX?`
`20:56.05`	`zeut`	`hello`
`20:56.15`	`terrapen`	`now i need some ideas for a new AGI project`
`20:56.24`	`slePP`	`terrapen: finish my cc app`
`20:56.27`	`zeut`	`Anyone care to answer some newbieish questions about converting my company phone system to asterisk?`
`20:56.34`	`slePP`	`terrapen: spandsp -> hylafax backend script`
`20:56.37`	`twisted[work]`	`tzafrir, huh?`
`20:56.37`	`VoIPMasta`	`terrapen: release it under the GPL :)`
`20:56.42`	`twisted[work]`	`er tzanger`
`20:56.48`	`terrapen`	`ugh, hylafax`
`20:56.55`	`slePP`	`hylafax is the best`
`20:56.56`	`terrapen`	`voip: the weatherstation agi?`
`20:57.06`	`terrapen`	`hylaffax is oooooooold`
`20:57.11`	`terrapen`	`IRIX days`
`20:57.11`	`bkw_`	`no its not`
`20:57.13`	`tzanger`	`terrapen: so? It's still very good`
`20:57.14`	`slePP`	`er... no, it's not. :P`
`20:57.15`	`bkw_`	`NO IT IS NOT`
`20:57.16`	`terrapen`	`bullshit, bkw`
`20:57.22`	`slePP`	`it is old in age, but not in development :>`
`20:57.23`	`bkw_`	`it may have origins from way back`
`20:57.24`	`redder86`	`HylaFAX is mature, not old`
`20:57.27`	`terrapen`	`hylafax dates from, i believe the late 80s`
`20:57.28`	`slePP`	`yes, mature :>`
`20:57.36`	`redder86`	`terrapen: 1990`
`20:57.37`	`slePP`	`redder86: can you give me any suggestions on keeping hylafax happy with couriers?`
`20:57.44`	`terrapen`	`ok, close enough :)`
`20:57.44`	`bkw_`	`hylafax is currently still under development`
`20:57.48`	`redder86`	`slePP: throw them away?`
`20:57.49`	`terrapen`	`wow`
`20:58.02`	`slePP`	`redder86: no :P i know that's the _best_ advice.. but i got 3 16 port courier banks for $100 CAD`
`20:58.03`	`redder86`	`Class 1 would be better.`
`20:58.04`	`bkw_`	`value=blah\|blah`
`20:58.09`	`bkw_`	`class1 is da best`
`20:58.16`	`slePP`	`class 1 frightens me :>`
`20:58.18`	`bkw_`	`and fax compatibility is hard to do accross the board anyway`
`20:58.22`	`terrapen`	`ph33r 1496+`
`20:58.27`	`terrapen`	`they have LCDs`
`20:58.29`	`bkw_`	`ph33r faxing`
`20:58.36`	`bkw_`	`rxfax is very stable now it seems`
`20:58.44`	`slePP`	`redder86: can i run class 1 w/ rtscts and 115200? since they're going for distinctive ring answering, i want to be sure the flow control is right when the dialups come in`
`20:58.50`	`terrapen`	`i wonder if i could find an updated firmware ROM for my ZyXEL`
`20:58.51`	`CoaxD`	`bkw: Let me ask you a question. Have you stabilized my ass yet?`
`20:59.03`	`CoaxD`	`bkw: I've been asking you to work on that for MONTHS.`
`20:59.16`	`slePP`	`but rxfax lacks what hylafax gets me... a pretty frontend (if you call WHFC pretty)`
`20:59.26`	`zeut`	`ok, I'll ask anyway :-) I want to convert my company to asterisk, we currently run with 12 POTS lines connected to a maxed out avaya Partner system.`
`20:59.28`	`terrapen`	`voipmasta, did you want the WeatherStation AGI?`
`20:59.47`	`PatrickDK`	`heh, partner system, I never liked that thing`
`20:59.48`	`slePP`	`zeut: have you considered bringing in a PRI? or do you want to pure VoIP it, or do you want to use the existing 12 lines as is?`
`20:59.57`	`PatrickDK`	`but the telcos around here love to dish them out`
`20:59.58`	`eKo1`	`zeut: You haven't asked anything.`
`21:00.09`	`zeut`	`easiest I think would be to use the existing 12 lines.`
`21:00.16`	`slePP`	`and you want to connect your phones to it how?`
`21:00.23`	`slePP`	`use the existing hardware, or replace with SIP/IAX devices?`
`21:00.45`	`zeut`	`I did a little looking and there was someone who created a device so that we could use our existing phones.`
`21:00.46`	`slePP`	`and why are you replacing it?`
`21:01.02`	`slePP`	`"created a device".. wouldn't that just be a big channel bank? :>`
`21:01.08`	`*** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com)`
`21:01.10`	`zeut`	`the main reason is that it only supports 48 mail boxes, and we could use 100 or more`
`21:01.20`	`johnnyb`	`Has anyone here used a Swissvoice IP 10 S phone?`
`21:01.30`	`slePP`	`that's a lot of mailboxes :>`
`21:01.43`	`zeut`	`we are a real estate company and we have lots of people who double up on desks`
`21:01.46`	`slePP`	`basically, zeut, it really boils down to how far you wnt to go with it`
`21:02.00`	`slePP`	`if you want to fully integrate asterisk into the existing pbx, you're probably in for more headaches than if you just replace everything`
`21:02.05`	`zeut`	`how far I want to go, and how to get there is what I'm trying to figure out :-)`
`21:02.32`	`slePP`	`asterisk can do everything you use the avaya for, but to keep the avaya, you have to get that thing happy to not do anything but manage extensions in/out to the 12 lines you'd run into your asterisk server`
`21:02.35`	`zeut`	`by replace everything you mean bring in a PRI (T1?) and use all voip hard phones to connect to asterisk?`
`21:02.49`	`slePP`	`well, don't need to replace the incoming lines, but a T1 would be cheaper i bet.`
`21:03.22`	`terrapen`	`zeut, you should sell them an IVR system where prospective homebuyers can call in and listen to a recorded msg about a particular home`
`21:03.24`	`zeut`	`perhaps, but assuming I keep the 12 lines, I would need at least 3 cards to go in the asterkisk server`
`21:03.42`	`slePP`	`zeut: or you get a T100P or TE4xxP, and use an FXO channel bank`
`21:03.44`	`zeut`	`well I work for them so I'm not going to seem them anything, but I do like the idea.`
`21:03.57`	`slePP`	`what you'll probably want to do, is get a 12 FXS/12 FXO channel bank and a T1 controller for asterisk`
`21:04.06`	`zeut`	`ok, can you explain how I would use an FXO channel bank?`
`21:04.07`	`slePP`	`connect 12 FXS to the avaya, and 12 FXO to the telco`
`21:04.33`	`slePP`	`so, an incoming call will end up coming down the FXO into asterisk, and it can handle it from there. then it can dial back to the avaya using the FXS`
`21:04.58`	`slePP`	`so you'd still be stuck with 12 channels in/out from the internal extensions and pbx`
`21:05.22`	`slePP`	`but it'd work :> again, telling the avaya to behave nicely would probably be the messiest bit`
`21:05.40`	`slePP`	`it's good for other ppl that may be wondering and aren't asking :>`
`21:05.43`	`zeut`	`ok, it sounds simpler to dump the avaya`
`21:05.45`	`PatrickDK`	`heh, hope you have the programming stuff for the system`
`21:05.46`	`zeut`	`I have no problem with that.`
`21:05.51`	`terrapen`	`slepp, who makes these?`
`21:05.53`	`slePP`	`it is definitely easier to get rid of it and replace everything with real phones`
`21:05.56`	`slePP`	`terrapen: makes what?`
`21:05.56`	`terrapen`	`and do they make 24-port versions?`
`21:06.02`	`slePP`	`24 FXS/24 FXO?`
`21:06.03`	`terrapen`	`this device you speak of`
`21:06.07`	`terrapen`	`channel bank`
`21:06.07`	`terrapen`	`yes`
`21:06.14`	`slePP`	`you can get like 16/8 zhone's`
`21:06.15`	`zeut`	`what do you mean by real phones?`
`21:06.24`	`slePP`	`and with things like Adit's, you can get expansion cards to do things in blocks of 4 or 8`
`21:06.28`	`terrapen`	`ah`
`21:06.30`	`slePP`	`zeut: hard voip phones`
`21:06.38`	`terrapen`	`screw it, we're just gonna get VoIP phones`
`21:06.39`	`slePP`	`are the phones you have avaya specific?`
`21:06.51`	`terrapen`	`i think we're going to get a bunch of IP500s for our new store`
`21:06.53`	`zeut`	`not sure... they are AT&T MLS-12`
`21:06.56`	`slePP`	`terrapen: heh. voip phones are good.. channel banks are useful for integration to existing systems. (or modems/faxes)`
`21:06.59`	`terrapen`	`and hopefully convert the other stores over`
`21:07.16`	`slePP`	`so i'm guessing they're pretty pbx specific. they have 'line' buttons and all that?`
`21:07.21`	`terrapen`	`slepp, we're currently using an ancient AT&T Merlin system from the 80s`
`21:07.25`	`terrapen`	`actually, we're using three`
`21:07.27`	`slePP`	`terrapen: nice.`
`21:07.27`	`zeut`	`yes they line buttons`
`21:07.30`	`terrapen`	`everybody has three phones on their desk`
`21:07.31`	`slePP`	`zeut: k.`
`21:07.34`	`slePP`	`terrapen: ew....`
`21:07.35`	`terrapen`	`i'm not kidding, either`
`21:07.45`	`terrapen`	`the funny thing is...`
`21:08.04`	`terrapen`	`my boss (who is also my father) swears by it and is super-reluctant to give it up`
`21:08.04`	`slePP`	`zeut: k. keep the avaya, but program it to just basically pass any digits through to the phone line, and have each line coming out of the avaya have an extension number`
`21:08.09`	`slePP`	`basically, the avaya gets to be a big pass-through switch`
`21:08.09`	`terrapen`	`because its simple and he can understand it`
`21:08.12`	`slePP`	`and asterisk does everything else`
`21:08.18`	`terrapen`	`but i'm going to replace it.`
`21:08.49`	`slePP`	`zeut: you could tell it to handle internal extensions still, but that will limit some call features asterisk can provide (but with 12 FXS, you'd only get 6 internal extension->extension calls, if asterisk had to deal with it)`
`21:08.55`	`slePP`	`terrapen: good :>`
`21:09.15`	`zeut`	`ok, so dump the avaya...`
`21:09.21`	`slePP`	`zeut: what asterisk can do is pickup the line to the avaya and dial an extension..`
`21:09.39`	`slePP`	`well, again, the only way to dump the avaya is to replace everything with VoIP phones, or to get more channel banks and use regular phones`
`21:09.41`	`zeut`	`so what I would need is 12 FXS ports and about 30 FXO ports (we have about 30 phones in the office)`
`21:09.53`	`slePP`	`other way around..`
`21:09.57`	`slePP`	`FXO you'd use to get incoming from telco`
`21:10.01`	`slePP`	`FXS provides dialtone/etc.`
`21:10.02`	`zeut`	`when you say channel bank what do you mean?`
`21:10.07`	`*** join/#asterisk TekLexus (~Mnemonic@167.206.75.24)`
`21:10.10`	`slePP`	`exactly that... search ebay for it`
`21:10.19`	`slePP`	`a channel bank takes a T1 and converts it into FXS/FXO/whatever`
`21:10.24`	`zeut`	`ahh sorry had that backwords, so 12 FXO's and 30 FXS's`
`21:10.44`	`slePP`	`but, i'm thinking pretty much 100%, those phones won't work on a regular FXS port off a channel bank`
`21:11.00`	`*** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net)`
`21:11.07`	`slePP`	`quick test, take one and plug it into a regular phone jack. if you get dialtone and stuff, yay. if not, no go..`
`21:11.09`	`zeut`	`or they might sorta work, but not well....`
`21:11.11`	`slePP`	`(beware of possibly frying a phone)`
`21:11.12`	`riksta`	`malcolmd: does it work ok?`
`21:11.19`	`slePP`	`all the 'line' features and such will not work`
`21:11.30`	`zeut`	`well I don't have a "regulal phone jack" handy.`
`21:11.31`	`slePP`	`it will all have to be dialed to do things like transfers or whatever`
`21:11.32`	`Uajal`	`whether it is possible`
`21:11.32`	`Uajal`	`1. to write real IVR script for autodialer`
`21:11.32`	`Uajal`	`2. point to table in database with phone numbers`
`21:11.32`	`Uajal`	`3. point to table where results should be`
`21:11.35`	`TekLexus`	`Hey, i just installed a TDM400P with 1 FXO card on bank 4... I tried looking up the config settings for asterisk on digium's website but i cannot find the settings for zaptel.conf and zapata.conf`
`21:12.04`	`slePP`	`zeut: my advice is either drop everything including the phones, or keep the avaya nad just reprogram it. it shouldn't be hard, actually`
`21:12.14`	`slePP`	`the avaya acts like a pbx already?`
`21:12.24`	`zeut`	`yes`
`21:12.26`	`slePP`	`so you can just have asterisk dial extension 110 to ring that phone`
`21:12.31`	`slePP`	`but disable voicemail, the pbx, etc. on the avaya`
`21:12.36`	`Uajal`	`I meant using * as autodialer`
`21:12.42`	`slePP`	`so each incoming line expects nothing but an extension to be dialed`
`21:13.07`	`zeut`	`I"m sorry, i'm not sure I follow the question...`
`21:13.11`	`slePP`	`k...`
`21:13.12`	`slePP`	`this is the path:`
`21:13.35`	`slePP`	`Telco/channel bank (FXO) -> asterisk -> channel bank(FXS) -> avaya (FXO) -> phones (whatever/avaya)`
`21:13.41`	`slePP`	`the avaya will handle the phones as they are, all 30 of them`
`21:13.47`	`zeut`	`I'm an experience network admin, but have almost no experience with phones :-) (part of the reason askterisk sounded good :-)`
`21:13.49`	`slePP`	`right now, you have the avaya plugged into the telco`
`21:14.00`	`slePP`	`and when you dial it, what do you get?`
`21:14.05`	`slePP`	`a person or a voice prompt?`
`21:14.14`	`zeut`	`yes right now avaya is connected to thet telco`
`21:14.23`	`zeut`	`a person`
`21:14.26`	`slePP`	`k.`
`21:14.41`	`slePP`	`can the avaya be setup to answer with a voice prompt and say 'please enter the extension youw ish to call' and wait for more digits?`
`21:14.48`	`*** join/#asterisk ZX81 (~matt@222-153-18-2.jetstream.xtra.co.nz)`
`21:15.05`	`zeut`	`yes, you can have the avaya do an auto-attendant answer (if that is what you mean)`
`21:15.09`	`slePP`	`that's what you want to do with the avaya. have it answer, and then wait for someone to dial an extension`
`21:15.12`	`slePP`	`that's what i mean, yes.`
`21:15.18`	`Uajal`	`Is it possible to use asterisk as autodialer so that it will read phone numbers from database or file (which will be dynamically updated by application) and call the numbers according to some IVR logic?`
`21:15.28`	`slePP`	`have it answer in auto-attendant mode, and say 'enter an extension' and wait for digits. in reality, you don't want/need the voice to say anything. i tjust has to wait`
`21:15.40`	`zeut`	`ok, but management here will want a customer to call and not get ahuman voice`
`21:15.42`	`slePP`	`and asterisk will dial the extension`
`21:15.45`	`AgiNamu`	`how can i lower the IAX2 reregister interval?`
`21:15.55`	`TekLexus`	`anyone know the configuration for the TDM400P?`
`21:16.01`	`slePP`	`asterisk can do anything you need it to. it replaces the functions of the avaya, short of managing all those 30 phones`
`21:16.06`	`AgiNamu`	`for a specific peer`
`21:16.18`	`slePP`	`it can answer the call in an auto-attendant mode, or it can directly dial various extensions internally`
`21:16.29`	`zeut`	`interesting, so I would have the 12 lines go into asterisk, and then back out into the avaya....`
`21:16.33`	`slePP`	`yup`
`21:16.45`	`Uajal`	`I read documentation but didn't find how can I write IVR logic?`
`21:16.49`	`slePP`	`and the avaya would handle where each real phone is, but asterisk would tell avaya which realphone it wants by using avaya's 'auto-attendant'`
`21:16.53`	`zeut`	`ok, so I would have to be able to transfer a call being managed by the avaya into voicemail handled by asterisk`
`21:17.02`	`slePP`	`right..`
`21:17.06`	`slePP`	`which can be done a few ways...`
`21:17.22`	`slePP`	`if a real person is transfering it, the avaya would pickup the phone (to asterisk) and dial something to go straight to voicemail`
`21:17.33`	`slePP`	`if the line is ringing, ringing, ringing, never gets answered, asterisk will hangup and you can go on into voicemail`
`21:18.20`	`zeut`	`ok, so a customer calls and askterisk routes the call to the avaya which rings a desk and it's answered`
`21:18.27`	`slePP`	`keep in mind, how users use stuff inside will definitely change for some things, like voicemail transfers, since the avaya is going to imagine it's an outgoing call to the real world.. when it's not. :>`
`21:18.29`	`slePP`	`yes.`
`21:18.42`	`slePP`	`they can then transfer it using the regular avaya transfer features`
`21:18.50`	`*** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl)`
`21:18.59`	`slePP`	`or transfer it to voicemail, but transfering the call to an outgoing line`
`21:19.48`	`zeut`	`one downside of this setup is that it's going to limit then number of outside liens that can be used when someone is checking their voicemail`
`21:19.59`	`slePP`	`yes`
`21:20.12`	`slePP`	`and it'll take 2 lines to leave a voicemail (maybe)`
`21:20.17`	`slePP`	`depending on the how the avaya works`
`21:20.27`	`zeut`	`right two lines for someone from the outside world to leave a voicemail`
`21:20.30`	`slePP`	`yeh`
`21:20.48`	`zeut`	`ok, that sounds interesting.... but option two is dump the avaya`
`21:20.59`	`zeut`	`is there anyway to use our existing phones with asterisk without the avaya`
`21:21.24`	`slePP`	`you're limiting yourself keeping the avaya, but the cost of moving everything to voip phones could be very high`
`21:21.27`	`slePP`	`(ie: $9,000)`
`21:21.45`	`zeut`	`how are you getting that number $300 per phone?`
`21:21.54`	`slePP`	`that's about the price for the IP300's`
`21:21.58`	`*** join/#asterisk DaPrivateer (Privateer@CRIMSON.OFF-HOURS.COM)`
`21:22.00`	`slePP`	`Polycom SoundPoint IP300`
`21:22.04`	`slePP`	`(i'm canadian, btw)`
`21:22.20`	`zeut`	`ok... is that sort of a standard phone, or can you get away with something cheaper?`
`21:22.22`	`slePP`	`and no, i doubt there's a useful way to use the avaya phones with asterisk. you'd have to research the MLS-12 and asterisk or channel banks`
`21:22.26`	`Madkiss`	`Is somebody with FWD here who could help me a bit with testing?`
`21:22.57`	`slePP`	`how many features do you need on a phone?`
`21:22.59`	`DaPrivateer`	`I am trying to running Asterisk on a FreeBSD 5.3 box... got the software installed and running, but my VoIP phone can not connect to it... no clue what i am doing wrong.. can some1 please help?`
`21:23.29`	`zeut`	`not many... really all we do is anser calls, put them on hold, transfer to voicemail and use the intercomm.`
`21:23.54`	`slePP`	`k`
`21:24.02`	`slePP`	`well, there are the Budgetones, which are like $65 USD each`
`21:24.06`	`slePP`	`but they're pretty simple phones`
`21:24.42`	`zeut`	`well that's only 2k for 30 phones, which sounds better :-)`
`21:24.51`	`*** join/#asterisk Dow (~Gaia@213.245.200.90)`
`21:25.06`	`slePP`	`heh`
`21:25.10`	`slePP`	`but they're not impressive at all`
`21:25.12`	`zeut`	`so if I were to dump the avaya and the phones then I would need a good PC, 3 4port cards from Digium and 30 voip phones.`
`21:25.30`	`slePP`	`yeh`
`21:25.37`	`slePP`	`or again, a channel bank with FXO and a T1 controller`
`21:25.56`	`slePP`	`the 4 port cards, 3 of them, should equal the cost of a single T1 controller and channel bank`
`21:25.59`	`slePP`	`(used channel bank)`
`21:26.18`	`zeut`	`so with the channel bank, I could use a T1 card, but keep the 12 POTS lines?`
`21:26.22`	`jaiger`	`yeah, with that many channels you're probably better off with a T1`
`21:26.25`	`slePP`	`yeh`
`21:26.30`	`slePP`	`and in future, move to a PRI`
`21:26.45`	`zeut`	`ok, about how much $$$ is the channel bank?`
`21:26.55`	`slePP`	`i got a full FXS one for $20 to $80 USD`
`21:27.02`	`jaiger`	`zeut: I have a wct100p and channel bank for 4 analog PSTN lines`
`21:27.04`	`slePP`	`but a one with FXO can be had fairly cheap ($150+)`
`21:27.07`	`slePP`	`again, search ebay :>`
`21:27.17`	`slePP`	`there are fancy ones for $4k, and non-fancy for $200`
`21:27.21`	`JunK-Y`	`slePP: adtran?`
`21:27.30`	`slePP`	`JunK-Y: adtran's tend to be more than that`
`21:27.34`	`jaiger`	`and I use some fxs channels for local fax machines`
`21:27.36`	`JunK-Y`	`yep, 5k$`
`21:27.42`	`slePP`	`but the zhone's are cheap, and i know at least the old premisys i have works like a charm with FXS`
`21:27.55`	`zeut`	`zhones cheap, Adtran expensive`
`21:27.59`	`slePP`	`yup`
`21:28.17`	`slePP`	`the one thing people have said about the zhone's with FXO is their caller id is either not there at all, or sometimes doesn't work`
`21:28.18`	`*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)`
`21:28.34`	`slePP`	`if you're not afraid of spending $600 on a nice bank, go that route`
`21:28.40`	`slePP`	`get the T1 controller`
`21:28.49`	`jaiger`	`I got a used adtran ta750 on ebay for ~$350 with 4 fxo and 12 fxs`
`21:28.51`	`zeut`	`ok, but I won't need the bank at all if I get a PRI put in`
`21:28.53`	`slePP`	`and you'd be set to do X number of FXO (limited byt he bank), and could move to PRI later`
`21:28.54`	`slePP`	`right`
`21:29.03`	`slePP`	`if you put a PRI in, and replace all the phones, you're better off in a lot of ways`
`21:29.11`	`slePP`	`you could possibly get a half T1 or such too, and get just 11-12 lines`
`21:29.19`	`slePP`	`jaiger: nice`
`21:29.35`	`zeut`	`ok`
`21:30.04`	`zeut`	`is it possible to keep my existing phones, put them all into a channel bank (or two) and use T1 cards in asterisk to talk to the phoens?`
`21:30.05`	`slePP`	`do you have a real budget for this?`
`21:30.15`	`slePP`	`those phones are gonna give you nightmares`
`21:30.19`	`jaiger`	`zeut: if they're analog phones, yes`
`21:30.29`	`twisted[work]`	`hmm`
`21:30.29`	`jaiger`	`then you get the fxs channels in the channel bank`
`21:30.29`	`slePP`	`again, search around for MLS-12 and channel banks or asterisk or FXO/FXS`
`21:30.33`	`twisted[work]`	`what phones?`
`21:30.37`	`slePP`	`twisted: AT&T MLS-12`
`21:30.40`	`slePP`	`hooked into avaya`
`21:30.41`	`zeut`	`well we are looking at $5k to upgrade the avaya system to something with more voimail boxes (but we get to keep the existing phoenes)`
`21:30.46`	`twisted[work]`	`Slainte, hahahahaha`
`21:30.51`	`nirs`	`any of the Digium experts around ?`
`21:30.52`	`slePP`	`slepp, you jackass`
`21:30.56`	`slePP`	`not slainte :>`
`21:31.05`	`DaPrivateer`	`I am trying to running Asterisk on a FreeBSD 5.3 box... got the software installed and running, but my VoIP phone can not connect to it... no clue what i am doing wrong.. anyone?`
`21:31.21`	`slePP`	`twisted: fixed frame slip errors, not we're getting bit shift :>`
`21:31.50`	`slePP`	`zeut: maybe look at doing some weird hybrid, where instead of asterisk being the frontend, asterisk does all the voicemail`
`21:31.56`	`slePP`	`by taking an 'internal' extension`
`21:32.08`	`twisted[work]`	`slePP, eh?`
`21:32.11`	`twisted[work]`	`bit shift?`
`21:32.11`	`slePP`	`so it would be like another phone..`
`21:32.13`	`slePP`	`twisted: yeh.`
`21:32.15`	`zeut`	`that's an interesting thought....`
`21:32.19`	`twisted[work]`	`wtf`
`21:32.24`	`twisted[work]`	`that's bizzare`
`21:32.29`	`slePP`	`twisted: the incoming clock and outgoing clock are not quite on par, and so the signal is not copying dead on the mark`
`21:32.34`	`slePP`	`and that pisses off high speed modem connections`
`21:32.50`	`redder86`	`slePP: you can run rtscts and 115200 baud in Class 1 - assuming your modems support it`
`21:33.01`	`*** join/#asterisk atmel (~vlad@128.54.50.78)`
`21:33.03`	`slePP`	`twisted: so, we have a firmware upgrade coming along that'll fix that clocks to allow the timing on span2 to use tming from span1. so they'll be dead on with each other`
`21:33.08`	`slePP`	`redder86: courier v.everythings`
`21:33.18`	`terrapen`	`zeut, you have 12 phones?`
`21:33.21`	`terrapen`	`in your current system?`
`21:33.21`	`zeut`	`interesting thought... that would allow me to leave the system in place while I test....`
`21:33.35`	`zeut`	`no we have 12 POTS lines from telco, and 30 phoens.`
`21:33.37`	`terrapen`	`ah`
`21:33.37`	`redder86`	`slePP: I don't know much more about the Couriers except that they have the usual USR quirks.`
`21:33.48`	`zeut`	`and we need approx 100 voicemail boxes which is why we need to upgrade our phone system`
`21:33.53`	`terrapen`	`zeut, ok, so you guys are just a tiny bit smaller than us. we have 23 lines, 40 phones`
`21:33.56`	`redder86`	`slePP: usually the advice is to use xonxoff with USR modems ... in Class 2.0 or Class 1`
`21:33.59`	`terrapen`	`zeut, you know how i'm doing this?`
`21:34.11`	`Madkiss`	`If I see this in my asterisk console with "sip debug" while a client tries to call me, what's wrong?`
`21:34.12`	`zeut`	`nope... how?`
`21:34.13`	`slePP`	`redder86: k. and in doing so, i should be able to flip to hardware flow on a data call`
`21:34.14`	`terrapen`	`i set up a testbed * pbx that i'm using for a side project`
`21:34.15`	`Madkiss`	`Reliably Transmitting (no NAT):`
`21:34.17`	`Madkiss`	`SIP/2.0 404 Not Found`
`21:34.18`	`terrapen`	`our eBay sales`
`21:34.19`	`slePP`	`using &H1&K3, etc.`
`21:34.21`	`slePP`	`prior to answer`
`21:34.26`	`terrapen`	`and i made it really cool, with some neat features`
`21:34.29`	`redder86`	`slePP: probably, yes`
`21:34.30`	`terrapen`	`and demoed it to the bosses`
`21:34.40`	`terrapen`	`and, for the most part, they liked it a lot`
`21:34.50`	`terrapen`	`you just have to show them the benefits of an all-IP system`
`21:34.57`	`terrapen`	`but don't just duplicate the current system`
`21:35.00`	`terrapen`	`that has no selling power`
`21:35.11`	`slePP`	`zeut: k. i have to run out`
`21:35.14`	`terrapen`	`you have to offer them something of value (besides just VM) that they don't have`
`21:35.16`	`redder86`	`slePP: there was pretty good evidence that I saw a few years ago that illustrated that hardware flow control on many USR modems was broken - that would probably apply to both fax and data`
`21:35.20`	`slePP`	`redder86: i'll try and let you know. :> iax/spandsp project is started is a nightmare`
`21:35.21`	`zeut`	`ok, thanks very much for the help.`
`21:35.37`	`slePP`	`the data connections work great except for retrains with hardware`
`21:35.40`	`slePP`	`er..`
`21:35.45`	`slePP`	`with hardware, data works fine`
`21:35.50`	`slePP`	`faxes seem to as well, at 14,400`
`21:35.52`	`terrapen`	`zeut, call 866-859-7359`
`21:35.56`	`terrapen`	`toll-free demo`
`21:35.57`	`redder86`	`that's good`
`21:35.57`	`slePP`	`the only issues we're having deal with the channel bank/t1 timing`
`21:36.05`	`zeut`	`ok`
`21:36.09`	`terrapen`	`it's stupid but they love this stuff`
`21:36.10`	`*** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com)`
`21:36.13`	`terrapen`	`just to know that we can do it`
`21:36.17`	`slePP`	`k. i'm out, folks. client needs to see me`
`21:36.55`	`zeut`	`bikeworld weather :-)`
`21:36.57`	`slePP`	`zeut: selling features are conferencing built in, multi-ring of extensions, call parking, music on hold, follow-me-find-me sort of services, unified messaging, etc.`
`21:37.00`	`terrapen`	`yup`
`21:37.03`	`slePP`	`and now i'm gone :>`
`21:37.15`	`slePP`	`(oh, and the best feature of all in IP phones, REALLY easy to configure and maintain)`
`21:37.15`	`slePP`	`:>`
`21:38.00`	`terrapen`	`ok, lunchtime4me`
`21:38.03`	`terrapen`	`good luck zeut`
`21:38.05`	`zeut`	`terrapen: thanks`
`21:38.45`	`zeut`	`is it practicle for me to think I can manage and asterisk system with out any real phone experience?`
`21:38.51`	`*** join/#asterisk t3t (~t3t@cust018.mke.attron.net)`
`21:38.57`	`slePP`	`do you learn quickly?`
`21:38.58`	`slePP`	`:>`
`21:39.01`	`zeut`	`It seems like it should be easy, but the phone world if very diffent.`
`21:39.02`	`slePP`	`the phone bit is pretty easy`
`21:39.02`	`zeut`	`yes`
`21:39.15`	`slePP`	`you'll spend way more time learning asterisk and how it handles things than you will figuring out how to make a phone work`
`21:39.22`	`zeut`	`I have a good programming and netowrking background, just no phone experienc.e`
`21:39.24`	`slePP`	`the big headaches tend to be dealing with telcos over PRI/T1 lines or anything else like that`
`21:39.28`	`slePP`	`yeh, it should be easy enough`
`21:39.30`	`slePP`	`that's where i came from`
`21:39.40`	`slePP`	`and now we run 7 asterisk boxes and two PRIs locally, with about 100 extensions`
`21:39.46`	`zeut`	`yeah, dealing with telcos is a pain...`
`21:40.24`	`zeut`	`question, one thing that would be nice here is having physical extensions and voicmeail boxes decoupled.`
`21:40.28`	`nirs`	`hey all`
`21:40.33`	`zeut`	`which I think is the norm for an asterisk type system.`
`21:40.36`	`*** join/#asterisk OzJames79 (~James@203.208.64.29)`
`21:40.52`	`nirs`	`has anybody ever integrated Asterisk with a connection to a Veraz softswitch, via E1 PRI connections ?`
`21:40.52`	`zeut`	`I would like to be able to sit at a desk, punch in a code into my phone and have it become my "extension"`
`21:40.56`	`zeut`	`all that sounds normal?`
`21:41.24`	`OzJames79`	`hi all i just signedup for voicepulse i have purchased a few extra numbers is it poissble to tell * to send one number to one exten and one number to another`
`21:41.37`	`*** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com)`
`21:42.14`	`*** part/#asterisk oej (~oej@apollo.webway.se)`
`21:43.03`	`redder86`	`does meetme require the presence of a Zap device?`
`21:43.57`	`OzJames79`	`no it doesnt`
`21:44.08`	`*** join/#asterisk PCadach (~paul@www.east.telecom.kz)`
`21:44.31`	`OzJames79`	`it uses ztdummy`
`21:44.46`	`redder86`	`OzJames79: ah, well, it looks like I need to get ztdummy going, then`
`21:45.25`	`OzJames79`	`edit the makefile`
`21:45.38`	`OzJames79`	`take out he # before ztdummy`
`21:45.44`	`OzJames79`	`then recomplie that will fix it all up :)`
`21:45.51`	`redder86`	`OzJames79: the zaptel Makefile?`
`21:48.13`	`OzJames79`	`yep thats right`
`21:48.17`	`redder86`	`thanks`
`21:48.26`	`OzJames79`	`anyone here using voicepulse?`
`21:48.35`	`redder86`	`then I'll need to rebuild asterisk, I think, to get the module built`
`21:48.51`	`*** join/#asterisk Blackvel (~blackvel@dsl-213-023-032-071.arcor-ip.net)`
`21:49.15`	`OzJames79`	`yes`
`21:49.17`	`Uajal`	`<Uajal> I read documentation but didn't find how can I write IVR logic?`
`21:49.55`	`Uajal`	`can anybody help me with writing IVR script?`
`21:50.12`	`redder86`	`OzJames79: do you have to run the zaptel module still?`
`21:50.22`	`Blackvel`	`hey, who is an agi expert? can it write to db and read e.g voiemail from? can it interconnect a caller with a number which is stored in the db? :)`
`21:50.36`	`redder86`	`Blackvel: yup`
`21:50.38`	`_Brian`	`anyone ever have a problem in which they can not connect to asterisk remotely? When trying, I always get "Unable to connect to remote asterisk" even though i have it running on the machine in a different window`
`21:50.52`	`redder86`	`Blackvel: you can do whatever you can program with AGI`
`21:51.08`	`nirs`	`can someone at Digium tech support please pick up the phone, I've been holding for like 10 minutes now`
`21:51.25`	`Blackvel`	`redder86: cool stuff... there are functions for this I guess? for db connection and reading/storing you just use perl?`
`21:51.40`	`Uajal`	`Iwhat is AGI?`
`21:51.47`	`redder86`	`Blackvel: you use whatever tools your programming language gives to you`
`21:51.49`	`*** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com)`
`21:51.54`	`redder86`	`Blackvel: use perl if you like`
`21:52.11`	`*** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com)`
`21:52.20`	`Uajal`	`can the asterisk commands be executed from perl?`
`21:52.22`	`MasterYoda`	`you have only been on hold for 5 mins :)`
`21:52.30`	`MasterYoda`	`some one should be with you shortly`
`21:52.31`	`*** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com)`
`21:52.37`	`redder86`	`Uajal: AGI is customized asterisk`
`21:52.38`	`Blackvel`	`how complicated is to learn AGI for IVR?`
`21:52.48`	`Blackvel`	`several days?`
`21:52.54`	`*** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com)`
`21:53.05`	`HjemmeRoyK`	`Blackvel: IVR?`
`21:53.10`	`modulus_`	`jbot ivr?`
`21:53.12`	`HjemmeRoyK`	`Blackvel: IVR works nicely without AGI`
`21:53.17`	`Uajal`	`what is customized asterisk? any links, pls.`
`21:53.27`	`redder86`	`How did MasterYoda get that message when he wasn't in the room to get it?`
`21:53.32`	`HjemmeRoyK`	`Uajal: agi is cgi for asterisk`
`21:53.40`	`Blackvel`	`probably I it insane to ask for how long AGI programm will take for a special job (when you dont know the details about the job?)`
`21:54.01`	`HjemmeRoyK`	`Blackvel: what sort of agi?`
`21:54.06`	`Blackvel`	`but 1-2 weeks including learning doesn't sound too far?`
`21:54.09`	`Uajal`	`any main AGI links for beginner?`
`21:54.53`	`HjemmeRoyK`	`Uajal: there's a test.agi script in the asterisk source`
`21:54.55`	`HjemmeRoyK`	`see that`
`21:55.00`	`Blackvel`	`well, some stuff which I had asked before when I joined the channel. leaving voicemail/reading voicemail, connecting db, calling ppl e.g :)`
`21:55.01`	`HjemmeRoyK`	`it's pretty self-explainatory`
`21:55.07`	`HjemmeRoyK`	`given you know a little perl.....`
`21:55.20`	`*** part/#asterisk zeut (~zeutzeut@ool-4350c7ad.dyn.optonline.net)`
`21:55.35`	`HjemmeRoyK`	`Blackvel: take a look at it and then see if you understand`
`21:55.41`	`HjemmeRoyK`	`also, rtfw`
`21:55.43`	`Blackvel`	`I will`
`21:55.50`	`Blackvel`	`whats rtf`
`21:55.55`	`HjemmeRoyK`	`~rtfw`
`21:55.56`	`jbot`	`it has been said that rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php`
`21:55.57`	`Blackvel`	`read the fucking and whats w?`
`21:55.59`	`Blackvel`	`hehhehe`
`21:56.07`	`redder86`	`all of my AGI is pretty simple call routing and Caller*ID stuff. I do it all with php and shell.`
`21:56.11`	`Blackvel`	`something new for asterisk`
`21:56.13`	`Blackvel`	`<g>`
`21:56.41`	`Blackvel`	`hm`
`21:56.53`	`Blackvel`	`who owns a PDA here? who uses it for VOIP?`
`21:57.21`	`Blackvel`	`~PDA`
`21:57.22`	`jbot`	`extra, extra, read all about it, pda is Personal Digital Assistant`
`21:57.33`	`*** join/#asterisk r0d3nt\|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)`
`21:57.34`	`Qwell`	`Blackvel: friend of mine uses his with a softphone`
`21:58.18`	`Hmmhesays`	`it's not a personal digital assistant till it services you under the desk while you work`
`21:58.24`	`Blackvel`	`I tried that with a friend too (not xlite pocketpc but something different) and the quality was sooo bad. he hadn't used a bluetooth headset. do you know something in more detail maybe?`
`21:58.56`	`Hmmhesays`	`sjphone has a windows ce client i believe`
`21:59.35`	`Qwell`	`yeah, he uses sjphone`
`21:59.49`	`Blackvel`	`can you shut down the built in pda mic and speakers and replace by a headset? is this an easy task? i am configuring my pda this week to buy one with car navigation`
`22:00.12`	`Blackvel`	`bought a paper about pdas`
`22:00.12`	`godsmoke`	`I use my pda for voip`
`22:00.18`	`godsmoke`	`gphone`
`22:00.33`	`Blackvel`	`and i am sooo confused about all that pdas, mhz, features and different models ..hehe`
`22:00.41`	`godsmoke`	`confused about?`
`22:00.53`	`mikegrb`	`then have someone else buy it`
`22:00.58`	`Blackvel`	`I simply can not decide what I should buy`
`22:01.00`	`Blackvel`	`hehe`
`22:01.01`	`Blackvel`	`you buy it`
`22:01.04`	`Blackvel`	`pay it and send me`
`22:01.04`	`Blackvel`	`:)`
`22:02.04`	`bjohnson`	`Blackvel: I might eventually`
`22:02.09`	`bjohnson`	`linphone on opie`
`22:02.15`	`godsmoke`	`how cpu intensive is codec-conversion in asterisk?`
`22:02.34`	`godsmoke`	`anyone done some testing?`
`22:02.35`	`Uajal`	`Can the AGI be used to read phone numbers from MySQL database and dial them?`
`22:02.39`	`Blackvel`	`is there any requirement to use the latest for voip? e.g 625MHZ PDA's. or should 325MHZ or 400MHZ be sufficent? hey is there asterisk for pocket pcs yet? :)`
`22:02.40`	`Hmmhesays`	`yes`
`22:02.54`	`godsmoke`	`Blackvel: asterisk for pocketpc?`
`22:02.55`	`Hmmhesays`	`to the agi question`
`22:03.06`	`godsmoke`	`you want to run a PBX on your PocketPC?`
`22:03.14`	`Blackvel`	`haha`
`22:03.23`	`Blackvel`	`well`
`22:03.25`	`Nugget`	`that's crazy talk.`
`22:03.27`	`Hmmhesays`	`I want to run an entire corporate key system on my 486`
`22:03.36`	`Blackvel`	`its insane like running asterisk on linksys wrt54g :)`
`22:03.39`	`godsmoke`	`Hmmhesays: no problem`
`22:03.42`	`Hmmhesays`	`with high availability`
`22:03.48`	`jaiger`	`Hmmhesays: can do, but it won't be cheap!`
`22:03.57`	`Uajal`	`Can I check from AGI the status of the call (answering machine or person or silence detection)?`
`22:04.04`	`techie`	`I still got my Comodore64`
`22:04.05`	`godsmoke`	`sigh`
`22:04.09`	`godsmoke`	`I can't talk this craziness`
`22:04.10`	`*** join/#asterisk Capouch (501@12.176.248.4)`
`22:04.15`	`Hmmhesays`	`the comodore64 rocked`
`22:04.21`	`Damascene`	`what do you mean? it STILL rocks`
`22:04.33`	`Damascene`	`Last ninja 1, 2, and 3! and who could forget strip poker.`
`22:04.35`	`Damascene`	`;)`
`22:04.35`	`Hmmhesays`	`i used past tense because they are not manufactured anymore`
`22:04.36`	`Nugget`	`commodore was trash. atari 800 forever.`
`22:04.46`	`techie`	`they need to build to asterisk support for it`
`22:05.00`	`techie`	`s/to/some`
`22:05.02`	`eKo1`	`Man, you people are old.`
`22:05.07`	`Nivex`	`atari800++`
`22:05.14`	`Uajal`	`Did anybody check the status of the call from AGI?`
`22:05.23`	`Damascene`	`i actually had the commodore 128. played all those gold box ssi games on it.`
`22:05.34`	`Damascene`	`and the bard's tale as well.`
`22:05.41`	`techie`	`yeah, like KungFu Master, and Tetris`
`22:05.45`	`Qwell`	`bards tale was great, heh`
`22:05.55`	`Damascene`	`bard's tale 3 that is. what else hmmmmm maniac mansion, zak mc kraken and the alien mind benders.`
`22:06.03`	`Damascene`	`Qwell: i started playing i and II again on the pc.`
`22:06.11`	`Hmmhesays`	`that re-make of maniac mansion was the shit`
`22:06.25`	`Damascene`	`Hmmhesays: i dind't try it yet. you mean the pc remake that someone did using that adventure toolkit?`
`22:06.26`	`Hmmhesays`	`the win32 version`
`22:07.03`	`Damascene`	`yeah! did you try out the 'new adventures of zak mckraken'? it was based off of the fm towns version it hink. also they released a king's quest 1 re-release VGA (re-released as in, not-sierra made)`
`22:07.40`	`Hmmhesays`	`nope, might have to look on emule for that`
`22:07.48`	`Hmmhesays`	`I got maniac mansion if you want it`
`22:07.48`	`Uajal`	`Any great asterisk consultants for hire?`
`22:08.12`	`techie`	`define great`
`22:08.23`	`Hmmhesays`	`cause there's a lot of us for hire`
`22:08.54`	`*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)`
`22:09.16`	`*** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com)`
`22:09.24`	`Chrisf0rd`	`Hello`
`22:09.35`	`adjacent`	`semantically he must mean asterisk consultants of large girth and stature`
`22:09.40`	`Chrisf0rd`	`guys do I have a question that has stumped even Supura`
`22:09.54`	`Beirdo`	`I dunno, do you?`
`22:10.02`	`adjacent`	`which there are problably lots of, also!`
`22:10.15`	`Uajal`	`great means Who is professional in Autodialing and PBX capabilities of Asterisk programming it from AGI or else`
`22:10.54`	`Chrisf0rd`	`I haev a supura 2001 from a Regular telephone I can call my DIAX and haev a 2 way conversation. but from a softphone to the Telephone they can only hear me I can not hear them`
`22:11.30`	`Chrisf0rd`	`when I call from an outside line and put in the ext number I can not hear them but they can hear me.`
`22:11.36`	`adjacent`	`Chrisf0rd: its the routing to the softphone.`
`22:11.50`	`adjacent`	`likely the comp is behind NAT`
`22:11.57`	`Chrisf0rd`	`you mean in the 2001 box itself`
`22:12.10`	`adjacent`	`no. the softphone`
`22:12.35`	`Chrisf0rd`	`when the phone calls out you can hear them but when you call the phone they cannot hear you`
`22:12.51`	`Chrisf0rd`	`either internal or external`
`22:12.52`	`adjacent`	`wait. which phone?`
`22:12.56`	`Chrisf0rd`	`ok`
`22:13.06`	`adjacent`	`be more specific, we dont all know our setup`
`22:13.12`	`adjacent`	`s/our/your`
`22:13.20`	`Chrisf0rd`	`if you call from a reg telephone put in the ext number and get patched threw`
`22:13.28`	`Chrisf0rd`	`they can hear you just fine`
`22:13.29`	`mikegrb`	`you said that already`
`22:13.52`	`eKo1`	`Chrisf0rd: Are these SIP phones you have problems with?`
`22:13.57`	`mikegrb`	`nat is your prooblrm`
`22:14.01`	`Chrisf0rd`	`when you call from a soft phone to that extention i.e ext to ext they can hear you but you cannot hear them`
`22:14.23`	`Chrisf0rd`	`but if the phone calls your sip its ok`
`22:14.30`	`Chrisf0rd`	`or an outside line its ok`
`22:14.33`	`eKo1`	`Chrisf0rd: 'sip debug peer mypeer' on the CLI and check the URIs.`
`22:14.58`	`eKo1`	`Most like it's a NAT issue.`
`22:15.15`	`eKo1`	`s/like/likely`
`22:15.19`	`machinehd`	`any bc'ers in here with a pri from telus?`
`22:15.20`	`adjacent`	`Chrisf0rd: right. because your NAT device (aka. router) can pass outbound UDP fine, but because UDP is connectionless, without forwarding the ports, or using STUN, you router doesnt know where do send inbound UDP`
`22:15.29`	`Chrisf0rd`	`No such peer 'mypeer'`
`22:15.29`	`Chrisf0rd`	`asterisk1*CLI>`
`22:16.00`	`Chrisf0rd`	`oops`
`22:16.05`	`eKo1`	`eh, mypeer is the peer you're having trouble with you ding dong.`
`22:16.15`	`Chrisf0rd`	`so that would be in the router config right`
`22:16.23`	`Chrisf0rd`	`hehehehe`
`22:16.27`	`Chrisf0rd`	`I am new to this`
`22:16.31`	`Chrisf0rd`	`sorry`
`22:16.58`	`eKo1`	`Chrisf0rd: I suggest you read the SIP RFC before proceeding.`
`22:17.04`	`Chrisf0rd`	`SIP Debugging Enabled for IP: 24.209.236.189:5060`
`22:17.09`	`Chrisf0rd`	`that is what I got`
`22:17.24`	`Chrisf0rd`	`is that the port I need to have upd/tcp`
`22:17.49`	`*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)`
`22:17.59`	`eKo1`	`RTFRFC`
`22:18.05`	`adjacent`	`Chrisf0rd: you need the sip stream to be redirected, and the conversation (wc?) stream to be redirected also`
`22:18.09`	`Damascene`	`Hmmhesays: the re-released king's quest vga version is freeware.`
`22:18.41`	`*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)`
`22:18.44`	`Damascene`	`Hmmhesays: http://www.agdinteractive.com/`
`22:19.05`	`*** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com)`
`22:19.07`	`Chrisf0rd`	`ok`
`22:19.32`	`Damascene`	`Hmmhesays: they even redid king's quest 2 in vga! they plan to do qfg2 in vga too!`
`22:19.48`	`*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)`
`22:20.11`	`HitTop`	`for sipura's product, is spa-3000 the only device that gives fxo ?`
`22:20.26`	`gambolputty`	`so far yes`
`22:23.59`	`*** join/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net)`
`22:25.54`	`Qwell`	`Damascene: qfg?`
`22:26.08`	`Damascene`	`qfg == quest for glory`
`22:26.12`	`Qwell`	`oh`
`22:27.52`	`r0d3nt\|m`	`Who do I talk to, to get voice recordings done ???`
`22:27.56`	`r0d3nt\|m`	`@ Digium /`
`22:27.57`	`r0d3nt\|m`	`?`
`22:28.25`	`file[laptop]`	`http://thevoice.digium.com/`
`22:28.36`	`r0d3nt\|m`	`thank you sir`
`22:31.36`	`*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)`
`22:33.31`	`*** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com)`
`22:33.35`	`Chrisf0rd`	`ok`
`22:33.49`	`Chrisf0rd`	`What ports need to be open 5060 and what else`
`22:36.49`	`TokyoJimu`	`Any foolproof way to do SIP where the client is behind NAT? The Wiki talks about special gateway servers to do this. Anyone know more?`
`22:39.17`	`eKo1`	`Make a tunnel?`
`22:39.23`	`eKo1`	`VPN tunnel.`
`22:39.33`	`eKo1`	`Or use STUN.`
`22:39.54`	`*** join/#asterisk florz (nobody@odnb-d9baa432.pool.mediaWays.net)`
`22:39.56`	`eKo1`	`or use IAX and avoid it altogether.`
`22:39.56`	`TokyoJimu`	`Yes, but... I'm talking about offering a service to the public, so it needs to work without any special config on the part of the end user.`
`22:40.07`	`RaYmAn-Bx`	`there's no foolproof way`
`22:40.13`	`RaYmAn-Bx`	`STUN is the closest though`
`22:40.28`	`RaYmAn-Bx`	`some NAT routers messes up that however`
`22:40.30`	`TokyoJimu`	`What does Vonage (et al) do?`
`22:40.46`	`Qwell`	`They have you put the device in front of your router`
`22:41.00`	`TokyoJimu`	`No they don't.`
`22:41.04`	`eKo1`	`Vonage expects that you aren't behind NAT.`
`22:41.16`	`RaYmAn-Bx`	`Qwell: hard to do if it's a router with builtin ADSL modem (provided by the ISP)`
`22:41.16`	`TokyoJimu`	`Not true at all.`
`22:41.33`	`eKo1`	`Either that or they don't use SIP.`
`22:41.38`	`RaYmAn-Bx`	`presumably they do STUN as well?`
`22:41.45`	`TokyoJimu`	`My Vonage service is completely behind NAT with no ports opened, and it works.`
`22:41.57`	`TokyoJimu`	`And it uses just a standard Cisco ATA-186 box.`
`22:42.10`	`eKo1`	`Are they using SIP?`
`22:42.14`	`RaYmAn-Bx`	`STUN most likely. But it's definitely not foolproof.`
`22:42.18`	`TokyoJimu`	`Good question. Probably not.`
`22:42.33`	`*** join/#asterisk scarolan (~scarolan@209-163-182-185.gen.twtelecom.net)`
`22:42.39`	`TokyoJimu`	`At least on my box. But I believe their newer boxes are SIP.`
`22:42.49`	`eKo1`	`Either they use STUN or H.323 or MGCP or ...`
`22:43.01`	`scarolan`	`Greetings friends. This may be OT, but can someone direct me to where I could find a manual or howto to get my snom 190 phone working?`
`22:44.02`	`*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)`
`22:46.49`	`moonwick`	`is there something non-obvious that I need to do in order to get asterisk to accept SIP calls from the internets?`
`22:47.04`	`moonwick`	`I've got a context set up in [general] but any time I try to call it, the caller gets a 404`
`22:47.12`	`moonwick`	`[general] in sip.conf, that is`
`22:48.16`	`eKo1`	`moonwick: 'sip debug' on the cli and look at the dialog.`
`22:48.19`	`*** join/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca)`
`22:48.39`	`eKo1`	`Make sure there is nothing out-of-the-ordinary happening.`
`22:48.57`	`moonwick`	`ooh, damn. I should've thought to try that.`
`22:49.10`	`*** join/#asterisk NeonLevel (~NeonLevel@200.52.142.184)`
`22:50.32`	`NeonLevel`	`Hi everyone, I was wondering if exist analog telephone adapters like the cisco ATA-186, or any other brand, but with more analog ports. anyone????`
`22:50.51`	`NeonLevel`	`I need 16 ports`
`22:50.58`	`eKo1`	`fxs?`
`22:51.14`	`NeonLevel`	`yes`
`22:51.29`	`eKo1`	`Check out mediatrix.com.`
`22:51.41`	`eKo1`	`I think they have 16 and 24 port ones.`
`22:51.48`	`NeonLevel`	`checking....`
`22:51.53`	`NeonLevel`	`thanks`
`22:51.55`	`*** part/#asterisk Capouch (501@12.176.248.4)`
`22:52.18`	`eKo1`	`Quintum should also have some I think.`
`22:52.35`	`NeonLevel`	`Quintum?`
`22:52.40`	`eKo1`	`quintum.com`
`22:52.42`	`NeonLevel`	`ok...`
`22:52.45`	`NeonLevel`	`checking...`
`22:53.00`	`eKo1`	`net2phone also custom made ones.`
`22:53.09`	`NeonLevel`	`really net2phone?`
`22:53.18`	`eKo1`	`For use with their service only.`
`22:53.23`	`eKo1`	`I have one downstairs.`
`22:53.26`	`NeonLevel`	`oh i see :(`
`22:53.36`	`NeonLevel`	`no use for me here...`
`22:53.49`	`NeonLevel`	`trying to connect all ports to an asterisk pbx`
`22:54.18`	`naturalvoice`	`Hey guys ? I need a good international rates for Portugal and Brazil, anyone ?`
`22:54.34`	`eKo1`	`naturalvoice: Check out the wiki.`
`22:54.49`	`eKo1`	`I mean, check the wiki under voip providers.`
`22:55.02`	`jets`	`Using a macro what is the easiest way to redirect both users to a meetme, an agi?`
`22:55.04`	`Sedorox`	`http://www.mediatrix.com/products_devices.php?prodid=3`
`22:55.05`	`Sedorox`	`?`
`22:55.30`	`flosch`	`night`
`22:55.44`	`JunK-Y`	`Sedorox: FXS ?`
`22:56.06`	`Sedorox`	`well.. just in response to NeonLevel`
`22:56.15`	`Sedorox`	`it is fxs...`
`22:57.51`	`AgiNamu`	`Hey, if anyone wants to try out my codec passthrough, I just posted it http://bugs.digium.com/bug_view_page.php?bug_id=0003553`
`22:57.53`	`AgiNamu`	`for IAX only`
`22:58.01`	`*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)`
`22:58.02`	`AgiNamu`	`it's saving my ass`
`22:58.13`	`ariel_`	`hello everyone`
`22:58.26`	`ariel_`	`bkw_, your back good to see you back.`
`22:58.48`	`eKo1`	`AgiNamu: codec passthrough? I don't get it.`
`22:59.05`	`*** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es)`
`22:59.22`	`*** join/#asterisk bowman (~rsp@195.46.47.202.static.cablesurf.de)`
`22:59.55`	`Sedorox`	`eKo1: to pass the codec being used through the * box`
`22:59.56`	`eKo1`	`You can't avoid transcoding if the clients use different codecs.`
`22:59.59`	`AgiNamu`	`eKo1, read the bug. basically, you can't negotiate codecs with incoming clients`
`23:00.05`	`AgiNamu`	`eko, that's not the issues`
`23:00.13`	`AgiNamu`	`telco provider has allow=all`
`23:00.22`	`AgiNamu`	`they will transcode to whatever. But my clients are ulaw, gsm, g729, and ilbc.`
`23:00.38`	`AgiNamu`	`so when a call comes in, i need to negotiate the final codec with the provider`
`23:00.42`	`AgiNamu`	`instead of transcoding.`
`23:00.45`	`Madkiss`	`What do I have to tell my asterisk if I want it to call 012345@sipgate.de directly?`
`23:00.46`	`eKo1`	`Isn't that a limitation of IAX though.`
`23:00.49`	`AgiNamu`	`no`
`23:01.00`	`AgiNamu`	`this is just because this hasnt been written yet`
`23:01.06`	`Madkiss`	`It actually tries to call 012345@sipgate.de via the freenet-out account, but ..`
`23:01.07`	`AgiNamu`	`you dont know the final codec until you go thru the dialplan`
`23:01.15`	`bowman`	`Madkiss: Dial(SIP/012345@sipgate.de) ? :)`
`23:01.16`	`AgiNamu`	`this allows you to specify the final codec right away`
`23:01.26`	`eKo1`	`IAX doesn't negoitate codecs?`
`23:01.32`	`AgiNamu`	`read the bug :)`
`23:01.33`	`Madkiss`	`bowman: Right, but I have that problem with all SIP-adresses`
`23:01.48`	`AgiNamu`	`for instance, you can say, if userX dials in to this extension, then negotiate codec Y for with userX`
`23:01.59`	`AgiNamu`	`because you know that the extension dials a GSM-only user.`
`23:02.21`	`AgiNamu`	`iax will just negotiate from a static list`
`23:02.29`	`dca[laptop]`	`a little birdie told me that digium is not going to pursue making the g729 registration utility available on FreeBSD...any truth to that?`
`23:02.34`	`AgiNamu`	`it wont take into consideration the extension being dialed`
`23:02.44`	`NeonLevel`	`i guess you can define that AgiNamu in iax.conf in every extension, no?`
`23:02.48`	`AgiNamu`	`no`
`23:02.58`	`eKo1`	`AgiNamu: So what you've done is basically forcing which codecs are negotiated?`
`23:03.01`	`AgiNamu`	`this is impossible with asterisk without this patch (according to bkw_ and he wrote the stuff)`
`23:03.04`	`AgiNamu`	`eKo1, yes`
`23:03.11`	`AgiNamu`	`which isnt needed for everyone`
`23:03.14`	`bowman`	`Madkiss: I have no real idea, sorry :)`
`23:03.17`	`AgiNamu`	`but for service providers, its a huge help.`
`23:03.29`	`eKo1`	`I though you can control that with allow & disallow`
`23:03.33`	`AgiNamu`	`no`
`23:03.40`	`AgiNamu`	`see, when a call comes in`
`23:03.43`	`AgiNamu`	`you have no clue where it will end up`
`23:03.54`	`AgiNamu`	`so you dont know what the final, client, codecs are.`
`23:04.02`	`AgiNamu`	`this allows you to specify that`
`23:04.10`	`NeonLevel`	`ah i see...`
`23:04.26`	`NeonLevel`	`interesting...`
`23:04.42`	`AgiNamu`	`so now, when a call comes in, it checks the DID`
`23:04.45`	`eKo1`	`Well, my clients aren't dialing other voip users around the world so...`
`23:04.54`	`AgiNamu`	`eko, again, nothing to do with that`
`23:04.59`	`AgiNamu`	`it has to do when you have a provider coming in`
`23:05.00`	`terrapen`	`oh WORD`
`23:05.04`	`terrapen`	`i love our new MoH`
`23:05.10`	`eKo1`	`I pretty much have full control what codecs are being used where.`
`23:05.12`	`AgiNamu`	`and sees if that client is ilbc, gsm, etc. then it'll negotiate that`
`23:05.24`	`AgiNamu`	`this is useful if your PSTN->VoIP provider of DIDs allows multiple codecs`
`23:05.29`	`AgiNamu`	`and you want to offload transcoding to them`
`23:05.31`	`terrapen`	`Napoleon Dynamite soundtrack`
`23:05.35`	`terrapen`	`the MIDI stuff`
`23:06.23`	`Sauron	`Is the reason Zap cards are CPU intensive, because they use libpri?`
`23:06.33`	`Sauron	`s/they/* or the driver`
`23:07.16`	`bowman`	`Madkiss: pm?`
`23:07.36`	`nestAr`	`hahaha.. anyone using webalizer? offtopic, i know.`
`23:08.00`	`Madkiss`	`bowman: Well, please.`
`23:08.06`	`bowman`	`does anyone in here use a quadBRI junghanns card?`
`23:08.08`	`Sauron	`awstats > webaliser`
`23:08.13`	`Sauron	`s/s/z`
`23:08.55`	`nestAr`	`awstats is such a pain in the ass to get setup.. i just want a program i run..`
`23:09.00`	`nestAr`	`but anyway...`
`23:09.13`	`nestAr`	`i think i'm rolling over the transfer counter in webalizer..`
`23:09.29`	`Sauron	`nestar: Shrug, wasn't a problem here. I scripted it for a couple-50 vhosts I had`
`23:09.38`	`nestAr`	`it reports in K-Bytes.. and appears to be saying I've moved 2 gigs this month..`
`23:09.43`	`nestAr`	`i did the math by hand...`
`23:09.48`	`nestAr`	`~23 gigs..`
`23:09.57`	`*** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net)`
`23:10.18`	`johnnyb`	`What do you all think of the Grandstream Budgetone phones?`
`23:10.37`	`nestAr`	`johnnyb: they make great paper weights`
`23:10.40`	`eKo1`	`I don't think, I hit them with a bat.`
`23:10.52`	`freat`	`johhnyb: friend of mine has one, when I call him if he uses speakerphone it totally sucks`
`23:11.05`	`freat`	`johhnyb: but that's just the speakerphone that I know of`
`23:11.20`	`eKo1`	`johnnyb: in other words, stay away from them.`
`23:11.44`	`freat`	`johnnyb: you'd probably be better off with an ATA`
`23:12.00`	`*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)`
`23:12.08`	`expressfone1`	`any report about C&W blocking SIP protocol over their network ??`
`23:12.14`	`Primer`	`bjohnson: ever get your bluetooth headset working?`
`23:12.39`	`Primer`	`why would any isp block SIP traffic?`
`23:12.48`	`johnnyb`	`An ATA what?`
`23:12.53`	`Primer`	`especially when you can simply change the port`
`23:13.27`	`johnnyb`	`freat: what do you use?`
`23:13.57`	`johnnyb`	`freat: what kinds of problems do you have w/ speakerphone?`
`23:13.59`	`NeonLevel`	`what ISP is doing that Primer?`
`23:14.01`	`eKo1`	`Not the HandyTone...those are just as bad as the phone.`
`23:14.12`	`Primer`	`NeonLevel: according to expressfone1, V&W`
`23:14.22`	`Primer`	`note what he wrote before my question`
`23:15.06`	`NeonLevel`	`so if the Grandstream phones sucks wich is the good of course no as expensive as cisco ones.`
`23:15.14`	`Sauron	`I wonder if residential TWC has blocked SIP on their network`
`23:15.19`	`expressfone1`	`Primer, have some news from USA SIP protocol, net2phone and dialpad don�t working if C&W is in your network path`
`23:15.24`	`Sauron	`cuz it'd compete with their VoIP offering`
`23:15.39`	`mikegrb`	`cuz?`
`23:15.51`	`terrapen`	`i need some other funny MoH stuff`
`23:15.54`	`terrapen`	`but nothing dirty`
`23:16.07`	`godsmoke`	`Sauron: uh ...
`23:16.19`	`godsmoke`	`they don't do it`
`23:16.22`	`godsmoke`	`and they won't`
`23:16.28`	`Primer`	`expressfone1: that's total bullshit`
`23:16.40`	`*** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de)`
`23:17.10`	`terrapen`	`primer, how do you know this?`
`23:17.35`	`Primer`	`I know nothing!`
`23:17.36`	`terrapen`	`i've heard that several large providers are queing SIP packets`
`23:17.42`	`terrapen`	`queueing even`
`23:17.45`	`Primer`	`all I did was ask why an ISP would do that!`
`23:17.50`	`godsmoke`	`terracon: like whom?`
`23:17.53`	`Primer`	`expressfone1 is the one that brought it up!`
`23:17.58`	`terrapen`	`supposedly TimeWarner's RoadRunner is`
`23:18.02`	`godsmoke`	`no`
`23:18.03`	`Primer`	`I use RR`
`23:18.04`	`godsmoke`	`they're not`
`23:18.05`	`Primer`	`RR is fine`
`23:18.12`	`Primer`	`RR hardly blocks _any_ ports`
`23:18.15`	`terrapen`	`oh wait, maybe it was SBC DSL`
`23:18.22`	`terrapen`	`not port blocking`
`23:18.22`	`godsmoke`	`roadrunner doesn't block any ports`
`23:18.23`	`terrapen`	`queueing`
`23:18.32`	`terrapen`	`no filtering/blocking`
`23:18.35`	`Primer`	`SIP does support using other ports`
`23:18.39`	`terrapen`	`just putting your SIP traffic at a lower priority`
`23:18.41`	`liquidno2`	`RR is great.. now if only they had better peiring`
`23:18.56`	`terrapen`	`RR has awesome peering where i am`
`23:18.57`	`godsmoke`	`terracon: I highly doubt that's true -- and it's nearly impossible to prove/disprove`
`23:18.58`	`Primer`	`I mean, it's not like they can't just make it use a different port`
`23:19.00`	`liquidno2`	`their teir 1 peiring is lack luster`
`23:19.07`	`Primer`	`liquidno2: peering`
`23:19.12`	`godsmoke`	`terrapen*`
`23:19.14`	`liquidno2`	`docking`
`23:19.21`	`Primer`	`I think their peering is fine`
`23:19.29`	`liquidno2`	`I didn't realize the speling police were in force today`
`23:19.30`	`Primer`	`I get great throughput and latencies everywhere`
`23:19.30`	`terrapen`	`why would they not do that?`
`23:19.34`	`terrapen`	`they are greedy bastards`
`23:20.00`	`godsmoke`	`terrapen: stop propagating nonsense -- you have no proof, and there's no way to discredit you with proof, only SBC/Yahoo can say`
`23:20.22`	`*** join/#asterisk techie (gus@asterisk.horizonte.us)`
`23:20.22`	`godsmoke`	`I think it's highly unlikely`
`23:20.41`	`liquidno2`	`Primer: then you don't have any interaction with Teir 1 providers`
`23:21.02`	`liquidno2`	`TWRR is 100ms to pnap/internap and even Savvis...`
`23:21.05`	`expressfone1`	`have a lot of friends at FL using sipuras, dialpad and net2phone.. every one have C&W in their network path can�t register`
`23:21.15`	`Primer`	`what about their tier 1 providers?`
`23:21.22`	`liquidno2`	`uhm`
`23:21.33`	`liquidno2`	`Savvis and internap are teir 1 providers`
`23:21.33`	`Primer`	`all I'm saying is that I trace stuff from home all the time`
`23:21.43`	`Primer`	`not many hops, great ping times`
`23:21.47`	`Primer`	`that's all`
`23:21.51`	`*** join/#asterisk StilexIP (~none@pc-24-151-59-224.newt1.ct.charter.com)`
`23:21.59`	`liquidno2`	`Then the stuff you trace is all hosted at rackshack :)`
`23:22.12`	`*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)`
`23:22.15`	`Primer`	`hah`
`23:22.32`	`terrapen`	`RR is way solid for me`
`23:22.34`	`*** join/#asterisk joebaker (~joebaker@rrcs-67-52-9-110.west.biz.rr.com)`
`23:22.46`	`liquidno2`	`Primer: I run a colo in Orlando with Savvis, Internap and Cogent...`
`23:22.48`	`terrapen`	`i don't really care about the hops as long as latency is low and throughput is high`
`23:22.50`	`terrapen`	`and both are.`
`23:22.50`	`Primer`	`I know of one port RR blocks and that's 445 (windows rcp)`
`23:22.58`	`liquidno2`	`I get better rates through Cogent which is saying something`
`23:23.08`	`Primer`	`liquidno2: I'm west coast, perhaps that explains it`
`23:23.13`	`liquidno2`	`perhaps`
`23:23.15`	`Primer`	`I'm 8 hops from work`
`23:23.21`	`liquidno2`	`lucky you`
`23:23.24`	`liquidno2`	`I 20 at best`
`23:23.28`	`joebaker`	`Does asterisk work with Vonage? Can I use the asterisk software to replace the router box they give out?`
`23:23.44`	`liquidno2`	`* works fine with Vonage`
`23:23.47`	`expressfone1`	`Vonage are reporting OUT`
`23:23.49`	`Primer`	`liquidno2: I think they're west coast stuff is newer`
`23:23.56`	`Primer`	`their, even`
`23:24.03`	`*** join/#asterisk fall0ut (~tim@fluorine.idge.net)`
`23:24.05`	`Sedorox`	`Transmitting (NAT):`
`23:24.05`	`Sedorox`	`SIP/2.0 407 Proxy Authentication Required`
`23:24.10`	`Sedorox`	`that asterisk end.. or the far end?`
`23:24.20`	`liquidno2`	`Primer: perhaps... I am routed through Houston, which for where I am is stupid`
`23:24.22`	`terrapen`	`fortunately, the route from my office to my home RR connection never leaves a single building in downtown san antonio`
`23:24.29`	`terrapen`	`it goes, literally, down the hall`
`23:24.33`	`fall0ut`	`Anybody ever done MGCP on Polycom IPx00's ?`
`23:24.53`	`joebaker`	`Thanks liquidno2. are there docs for that you can point me to or good keywords to narrow down the search?`
`23:25.06`	`liquidno2`	`joebaker: check voip-info.org`
`23:25.11`	`joebaker`	`thanks`
`23:25.15`	`liquidno2`	`I seem to recall seein g something there`
`23:25.18`	`terrapen`	`http://www.pastebin.com/240348`
`23:25.20`	`liquidno2`	`it is basic SIP`
`23:25.23`	`nirs`	`ehy all`
`23:25.32`	`nirs`	`can anyone help with some extensions debugging for a few minutes ?`
`23:25.37`	`liquidno2`	`you might need to sniff the packets to get your SIP info`
`23:25.46`	`liquidno2`	`but my knowledge is out of date`
`23:25.55`	`terrapen`	`that's the route from my work to my home`
`23:26.03`	`terrapen`	`and 1/3 of the route is on my private network`
`23:26.08`	`liquidno2`	`terrapen: I am jelaous`
`23:26.12`	`*** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)`
`23:26.28`	`liquidno2`	`the only thing I ping low to is rackshack`
`23:26.33`	`terrapen`	`but my connection to my work ISP's core router is flaky right now`
`23:26.37`	`liquidno2`	`which is solid 50ms 99% of the time`
`23:26.43`	`terrapen`	`for some reason i see occasional > 200ms pings`
`23:26.45`	`terrapen`	`which makes no sense`
`23:26.46`	`Primer`	`liquidno2: I'm in LA, and I leave RR's network in LA almost every time`
`23:26.51`	`terrapen`	`because most of the time they are <4ms`
`23:26.55`	`terrapen`	`what would cause that?`
`23:26.55`	`Primer`	`if not LA, SJ`
`23:27.16`	`liquidno2`	`terrapen: maybe collisions on an interface or a duplexing issue`
`23:27.33`	`liquidno2`	`do you see any pl with that, or is it just the latency?`
`23:27.38`	`terrapen`	`no loss at all`
`23:27.46`	`terrapen`	`just occasional high latency`
`23:27.48`	`terrapen`	`its odd`
`23:27.48`	`liquidno2`	`install mtr`
`23:27.58`	`liquidno2`	`and run an mtr on it when it happens`
`23:28.05`	`liquidno2`	`matt's traceroute`
`23:28.18`	`liquidno2`	`greatest app for debugging weird network latency`
`23:28.32`	`jets`	`In a dial plan how could i redirect obth legs of the call to a meetme?`
`23:28.32`	`JunK-Y`	`which app?`
`23:28.41`	`eKo1`	`mtr rules.`
`23:28.41`	`liquidno2`	`JunK-Y: mtr`
`23:28.59`	`jets`	`I noticed the manager has a redirect -- but couldn't this be pretty easy in a dialplan too?`
`23:28.59`	`terrapen`	`yep`
`23:29.01`	`liquidno2`	`there is a windows port for it somewhere too`
`23:29.03`	`terrapen`	`i'm running mtr right now`
`23:29.20`	`JunK-Y`	`never heard about mtr before.`
`23:29.24`	`terrapen`	`oh shit`
`23:29.26`	`terrapen`	`mtr is the bomba`
`23:29.27`	`eKo1`	`I run mtr to check my connection all the way to miami.`
`23:29.38`	`*** join/#asterisk Nethab (~Anon8834@adsl-67-113-141-170.dsl.sntc01.pacbell.net)`
`23:29.43`	`liquidno2`	`eKo1: where is your equipment?`
`23:29.49`	`liquidno2`	`the old 911 building?`
`23:29.57`	`eKo1`	`central america`
`23:30.03`	`liquidno2`	`ah...`
`23:30.07`	`terrapen`	`wierd`
`23:30.09`	`terrapen`	`look at this:`
`23:30.09`	`terrapen`	`http://www.pastebin.com/240349`
`23:30.10`	`eKo1`	`I go through nwn`
`23:30.11`	`liquidno2`	`yeah mia is a big hub for that`
`23:30.14`	`terrapen`	`that's my upstream ethernet`
`23:30.21`	`*** join/#asterisk soulz- (~Soulz-@cm51.epsilon168.maxonline.com.sg)`
`23:30.23`	`terrapen`	`very little dropped packets`
`23:30.24`	`soulz-`	`hello all`
`23:30.33`	`Nethab`	`yawn`
`23:30.36`	`terrapen`	`what the hell would cause this latency?`
`23:30.46`	`soulz-`	`any broadvoice gurus here? seem to have trouble after the cvs upgrade`
`23:30.58`	`Nethab`	`i use broadvoice`
`23:31.06`	`liquidno2`	`bunch of outbound broadcast terrapen`
`23:31.14`	`fall0ut`	`Nobody ever used MGCP on Polycoms ?`
`23:31.14`	`soulz-`	`nethab: do u need the patch if u use CVS-HEAD-02/06/05-09:10:59`
`23:31.29`	`Nethab`	`no, the patch is not necessary after december 2004`
`23:31.51`	`terrapen`	`liquid, actually that may be the downlink to my firewall`
`23:31.58`	`terrapen`	`i get the two ports mixed up`
`23:32.03`	`terrapen`	`one goes to my provider's switch`
`23:32.06`	`terrapen`	`the other goes to my fw`
`23:32.06`	`*** part/#asterisk bowman (~rsp@195.46.47.202.static.cablesurf.de)`
`23:32.10`	`soulz-`	`nethab: emm, strange, sip.broadvoice.com:5060 2122020832@s 3584 Registered`
`23:32.21`	`soulz-`	`nethab: and when i make the call`
`23:32.26`	`terrapen`	`liquid, their stupid terminal servers spew lots of bcast`
`23:32.30`	`soulz-`	`it goes through my context`
`23:32.46`	`liquidno2`	`terrapen: we put all windows servers in their own vlan at the colo`
`23:32.59`	`liquidno2`	`we have been burned more than once with a broadcast packet storm`
`23:33.31`	`terrapen`	`Last boot: 22:53:18 Tue Jun 15, 2004 (power cycle)`
`23:33.33`	`liquidno2`	`exploited machiens pumping out arp poison`
`23:33.36`	`soulz-`	`nethab: SIP/2.0 100 Trying and says this Destroying call '6899f68d53f3ed0d4a23645d625dacbb@192.168.10.254'`
`23:33.36`	`soulz-`	`<PROTECTED>`
`23:33.55`	`soulz-`	`nethab: know what could be wrong?`
`23:34.12`	`Nethab`	`my sip show registry looks like this`
`23:34.13`	`Nethab`	`sip.broadvoice.com:5060 4087010023 3584 Registered`
`23:34.18`	`Nethab`	`no @s`
`23:34.48`	`*** join/#asterisk Darwin35 (~Darwin@c-24-3-241-22.client.comcast.net)`
`23:35.54`	`liquidno2`	`way off topic... but worthy: http://www.zefrank.com/punc/`
`23:35.58`	`yebo`	`anyone here using asterisk behind a media gateway like the AS5400 or the Telica Plexus 9000?`
`23:36.02`	`Nethab`	`are having trouble calling out using broadvoice, or receiving calls to your broadvoice number`
`23:36.21`	`soulz-`	`nethab: i am just trying to make calls via broadvoiceat the moment`
`23:36.27`	`terrapen`	`i'd love to find someone with a Nortel account`
`23:36.33`	`Darwin35`	`I have incoming problems with * and broadvoice with incoming`
`23:36.34`	`soulz-`	`i haven't configured the receving calls yet`
`23:36.36`	`terrapen`	`so i can get some new firmware for my switch`
`23:36.45`	`*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)`
`23:36.49`	`*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)`
`23:37.02`	`liquidno2`	`terrapen: what are you running?`
`23:37.02`	`terrapen`	`hey shido`
`23:37.10`	`terrapen`	`liquidno2: Alteon AceDirector 4`
`23:37.24`	`liquidno2`	`not familiar with those`
`23:37.46`	`terrapen`	`they were the shit, back in the day`
`23:37.49`	`terrapen`	`dunno if they still are`
`23:38.04`	`terrapen`	`we used them @ ticketmaster/citysearch`
`23:38.14`	`*** join/#asterisk DrukenHME (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com)`
`23:38.28`	`Primer`	`expressfone1: got a URL for this C&W allegation?`
`23:38.34`	`joebaker`	`I noticed that vonage is said not to allow BYOD unless you buy a second DID according to their terms of service. Can you do it anyway and not get caught? What are the ramifications? I guess they just don't want their system hacked in some way.`
`23:38.54`	`liquidno2`	`wow layers up to 7 eh?`
`23:38.58`	`liquidno2`	`not too shabby`
`23:39.00`	`terrapen`	`uh huh`
`23:39.05`	`terrapen`	`http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11187&item=5748835331&rd=1`
`23:39.46`	`liquidno2`	`So you running that on -48DC? ;)`
`23:39.48`	`eKo1`	`Dang it. Why can't voip providers agree on a format for their stupid call rates.`
`23:40.19`	`eKo1`	`Better yet, why can't they have a web service I can access to get the info. when I need it.`
`23:40.36`	`eKo1`	`That would be so sweet.`
`23:40.41`	`terrapen`	`http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=3706&item=5749290558&rd=1`
`23:40.46`	`liquidno2`	`wdsl based call info for a voip provider`
`23:40.49`	`terrapen`	`liquid, no, not in our co-lo`
`23:41.08`	`terrapen`	`what is wdsl-based call info`
`23:41.21`	`liquidno2`	`soap web service`
`23:41.29`	`liquidno2`	`simple object access protocol`
`23:41.31`	`eKo1`	`wsdl`
`23:41.36`	`DrukenHME`	`terrapen: what do those things do ?`
`23:41.41`	`eKo1`	`web service description language`
`23:41.41`	`terrapen`	`i know what SOAP is`
`23:41.49`	`terrapen`	`druken, its a load-balancing switch`
`23:41.53`	`eKo1`	`not wdsl`
`23:42.02`	`liquidno2`	`I typoed`
`23:42.02`	`terrapen`	`wirespeed layer-7 load balancing switch`
`23:42.07`	`liquidno2`	`sue me :)`
`23:42.16`	`terrapen`	`i'll bet you could balance some SER installations nicely with the Alteon AD3`
`23:42.46`	`StilexIP`	`anyone have ser and asterisk working together that can send me a sample ser.cfg file`
`23:42.48`	`liquidno2`	`there was a long diatrabe on the ml about load balancing last week`
`23:42.56`	`eKo1`	`the ad3 is good for as a session border controller`
`23:43.05`	`terrapen`	`i don't read the ml...maybe i should`
`23:43.15`	`liquidno2`	`that was a worthy discussion`
`23:43.15`	`*** join/#asterisk ZX81 (~matt@222-153-101-201.jetstream.xtra.co.nz)`
`23:43.26`	`liquidno2`	`there aren't many :)`
`23:43.27`	`terrapen`	`url?`
`23:44.12`	`liquidno2`	`terrapen: http://lists.digium.com/pipermail/asterisk-users/`
`23:44.18`	`terrapen`	`i actually had my AceDirector 4 given to me by Alteon`
`23:44.20`	`liquidno2`	`use google to search`
`23:44.22`	`terrapen`	`back when they were Alteon`
`23:44.23`	`terrapen`	`and not Nortel`
`23:44.29`	`*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)`
`23:44.29`	`*** mode/#asterisk [+o bkw_] by ChanServ`
`23:44.53`	`liquidno2`	`the topic was load balancing 20 servers`
`23:45.00`	`liquidno2`	`or some deriviative of that`
`23:45.21`	`terrapen`	`not seeing it....`
`23:45.35`	`terrapen`	`ok`
`23:45.36`	`terrapen`	`got it`
`23:45.49`	`*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)`
`23:45.49`	`*** mode/#asterisk [+o bkw_] by ChanServ`
`23:45.59`	`StilexIP`	`anyone have ser and asterisk working together`
`23:46.06`	`eKo1`	`StilexIP: I do.`
`23:46.37`	`terrapen`	`shit, it seems that all you would need is a SIP-aware load balancer`
`23:46.37`	`eKo1`	`The only reason I have an SER is to accomodate some stupid 4-port FXS gateway that would not work with *.`
`23:46.45`	`terrapen`	`one that can keep track of sessions`
`23:46.49`	`Qwell`	`bkw_: Get away while you still can!`
`23:46.54`	`terrapen`	`i'll bet the Alteon can`
`23:47.00`	`*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX \|\| chan_zap in 1.0.4 had a bug, so 1.0.5 has been released`
`23:47.25`	`terrapen`	`i love it how when people ask about load-balancing 20 servers, somebody pipes up about LVS (Linux Virtual Server)`
`23:47.41`	`terrapen`	`if you are going to load-balance 20 servers, layer 7, at wirespeed, you don't want LVS`
`23:47.42`	`StilexIP`	`eKo1: I'm having trouble getting asterisk to accept the calls from ser i get an error of too many sip headers. would you mind sending me a ser.cfg file or giving me some advice?`
`23:47.44`	`terrapen`	`you want a real switch.`
`23:47.47`	`*** join/#asterisk jgaviria (~jgaviria@63.245.86.116)`
`23:48.06`	`dca[laptop]`	`did they talk about g729 for FreeBSD on the conf call today? i heard it wasn't going to happen`
`23:48.17`	`eKo1`	`terrapen: Well, what are you load-balancing exactly?`
`23:48.19`	`terrapen`	`conf call?`
`23:48.36`	`terrapen`	`eko, im not load balancing any voip stuff now...i was just readding this mailinglist post`
`23:48.42`	`dca[laptop]`	`yeah, there was an asterisk conf call at 3pm CST`
`23:48.46`	`terrapen`	`and this guy recommends LVS`
`23:48.59`	`terrapen`	`dca, where do i find out about these?`
`23:49.09`	`eKo1`	`StilexIP: Hmm. Never heard of that problem.`
`23:49.15`	`dca[laptop]`	`heh, well, it was posted on the chan topic before the call occured`
`23:49.33`	`jgaviria`	`hi, i have a problem, i have some agents answering a pri, but the agents doesnt received any notification for the incoming calls, then they get confused, there is some way to asterisk play a beep or something when a call is received?`
`23:49.59`	`StilexIP`	`eKo1: http://pastebin.ca/5007`
`23:51.20`	`terrapen`	`where are the conference calls held? via IAXtel?`
`23:51.25`	`Moc`	`hi everyone`
`23:51.29`	`eKo1`	`StilexIP: Have you done a 'sip debug' and checked the headers?`
`23:52.26`	`StilexIP`	`eko1: yeah it looks like they're replicated, i think asterisk is sending them back to ser and sers sending them back to asterisk until it core dumps`
`23:52.57`	`eKo1`	`What kind of message? INVITE, ACK, what?`
`23:53.36`	`StilexIP`	`invite`
`23:53.51`	`eKo1`	`Do you see any 200 OKs in there?`
`23:54.11`	`liquidno2`	`terrapen: it was posted in the channel topic... it was an IAX2 trunk to an IP address with an extension`
`23:54.29`	`liquidno2`	`something like IAX2/68.x.x.x/996 or something similar`
`23:54.42`	`liquidno2`	`did anybody record it?`
`23:54.51`	`liquidno2`	`or is that asking way too much? :)`
`23:56.50`	`eKo1`	`I had in my extensions.conf but since I couldn't get in, I erased it.`
`23:57.12`	`liquidno2`	`eKo1: jammed the iax server?`
`23:57.33`	`liquidno2`	`I was out with my daughter at ballet class... priorities ;)`
`23:57.34`	`eKo1`	`I got a 404`
`23:58.00`	`eKo1`	`I guess they don't allow just anybody in the conference.`
`23:58.04`	`liquidno2`	`that is the reason (at least what I tell myself) that I work from home. :)`

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