irclog2html for #asterisk on 20060814

`00:00.00`	`mcreedjr`	`delmar: thats the way i'm leaning now, but we'll see`
`00:00.02`	`delmar`	`get the latest zaptel and asterisk.. just so Digium cant tell you .. to get it later...`
`00:00.13`	`delmar`	`mcreedjr, and try the KB1 deal`
`00:00.23`	`mcreedjr`	`delmar: i will certainly try that and the new echo can`
`00:00.26`	`delmar`	`mcreedjr, also.. remember that the other party doesn't hear any echo....`
`00:00.39`	`delmar`	`mcreedjr, but the asterisk end.. may do.. for a couple of seconds...`
`00:01.17`	`delmar`	`mcreedjr, most people have a front door.... so while the wench is saying.. " Hi and welcome to Alisons porn shop... " the echo-can just took care of that...`
`00:01.26`	`mcreedjr`	`delmar: so i may never be able to get rid of that? the asterisk end echo is what my customer is complaining about`
`00:01.27`	`delmar`	`mcreedjr, by the time it hits an extension.. it should be taken care of.`
`00:02.04`	`delmar`	`mcreedjr, I occasionally.. randomly.. get a little at the start of a call... but since putting the front door up... no more issues anyway...`
`00:02.06`	`evilbit`	`is there a way to get app_cepstral to read a file?`
`00:02.21`	`delmar`	`mcreedjr, but i did have it cured almost completly.`
`00:02.29`	`mcreedjr`	`delmar: i could do that i suppose, a quick AA greeting or something`
`00:04.19`	`delmar`	`mcreedjr, do try the KB1 echo-can. it almost completly cured it for me, and this other install I have jut put in.. polycom501's => Asterisk qTDM400 => FXO etc... i had no front-door on that to begin with and was even able to ramp up the TX/RX gains to like.. TX=8.0 RX=4.0 and no issues.`
`00:04.47`	`delmar`	`mcreedjr, I jsut put a front-door on that one now... because they wanted one.. no complaints about echo at the start of calls`
`00:05.08`	`delmar`	`mcreedjr, the odd thing is.. incoming calls was always the worst for echo at the start of a call.. if ever it happened.`
`00:05.41`	`mcreedjr`	`delmar: yeah, at this install, there is no echo on outbound calls whatsoever, only on inbound calls`
`00:05.49`	`delmar`	`mcreedjr, this was the case on the X100p as well... always lots of echo on incoming and less when a call goes out the FXO`
`00:06.05`	`delmar`	`mcreedjr, yeah. pain in the ass.`
`00:06.25`	`mcreedjr`	`delmar: i hate analog`
`00:06.29`	`delmar`	`mcreedjr, so surely the "ringing" on the line is doing something lame.`
`00:06.44`	`delmar`	`mcreedjr, yep. and digital is just too expensive to do.. ie.. Bri.`
`00:07.02`	`delmar`	`mcreedjr, and pointless now.. since you can get DID's over IP`
`00:07.34`	`mcreedjr`	`delmar: i just have the internet gateway equipment to do proper QoS yet though`
`00:07.41`	`mcreedjr`	`delmar: might have to do taht route eventually`
`00:08.00`	`delmar`	`mcreedjr, i would drop this damn pots line if i could but in this pathetic country we have bloody hopeless Telco that requires you pay for the voice/pots line first.. with or without DSL.`
`00:08.31`	`delmar`	`mcreedjr, ah.. well.. once the echo is sorted.. thats your next really important thing.. the QoS.`
`00:08.46`	`mcreedjr`	`delmar: they just started offering naked DSL here in the US. I almost fell out of my chair when the AT&T rep asked if I wanted dialtone with my DSL`
`00:08.46`	`delmar`	`mcreedjr, yet another thing I have worked hard on and know what needs to be done.`
`00:09.11`	`delmar`	`mcreedjr, yeah thats great. we are waiting for it here. be another 12-18months.`
`00:09.19`	`mcreedjr`	`delmar: I've got locally QoS setup, but just not QoS through the internet gateway. It's only used intra office anyway, no offsite extensions`
`00:09.45`	`delmar`	`mcreedjr, our government here had to actually force this... they will be passing legislation to force the Telco to unbundle.`
`00:10.04`	`delmar`	`mcreedjr, heh. when u say.. local QoS.. explain :P`
`00:10.49`	`mcreedjr`	`delmar: QoS setup in the switch, so traffic is prioritized when traversing the switch`
`00:10.52`	`delmar`	`mcreedjr, gonna grab a coffee while you run some more details past me.. then ill explain a QoS solution ive just put in.`
`00:11.06`	`mcreedjr`	`delmar: but not QoS through the gateway`
`00:15.21`	`mcreedjr`	`delmar: i'm justing CoS now, because I don't need L3 priorities on the packets as it never leaves my local switched domain`
`00:15.40`	`mcreedjr`	`delmar: no sense in over complicating matters`
`00:18.21`	`delmar`	`ok back`
`00:18.31`	`delmar`	`mcreedjr, so there is just the one switch involved ?`
`00:18.40`	`mcreedjr`	`delmar: aye`
`00:18.47`	`mcreedjr`	`delmar: it's a small setup`
`00:18.50`	`delmar`	`mcreedjr, yeah thats easy.`
`00:19.04`	`mcreedjr`	`delmar: agreed, hence why i took the easy road :)`
`00:19.11`	`delmar`	`mcreedjr, there are a couple of things.. firstly.. the router on the head end of the network...`
`00:19.36`	`delmar`	`mcreedjr, a key feature of the QoS that it must support.. is being able to set the MAX burst rates for up/down`
`00:19.43`	`delmar`	`mcreedjr, most good routers do this now`
`00:20.07`	`mcreedjr`	`delmar: not to stuff the pipe you mean?`
`00:20.18`	`delmar`	`exactly`
`00:20.22`	`delmar`	`mcreedjr, the reason is because most of the time.. the upstream sender of say.. web traffic.. will be able to max out your pipe...`
`00:20.33`	`delmar`	`packets will then queue on the telco side.. and cause latency`
`00:20.36`	`mcreedjr`	`delmar: not a whole lot of good for UDP sip/rtp traffic :) keeps other tcp stuff out of the way though`
`00:20.49`	`delmar`	`our Telco here.. fuckwits that they are.. have actually made it so HTTP has priority.`
`00:21.06`	`delmar`	`they have no interest in how it effects real-time sensitive applications...`
`00:21.22`	`delmar`	`they want to protect their lucrative Tolls/Voice business... so they don't care if they screw up our VoIP`
`00:21.23`	`mcreedjr`	`delmar: at another one of my clients, i have tcp rate-limiting setup on the cisco edge router`
`00:22.25`	`delmar`	`mcreedjr, yep. so its really important to have burst rate caps to ensure packets don't queue up at either end. better that you slow it down below the cap, and dont let that happen`
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`00:22.40`	`delmar`	`mcreedjr, the other really important thing at any site.. is Human QoS`
`00:23.22`	`delmar`	`mcreedjr, a bit of education goes a long way. you cant have users burning up torrents and all kinds of crap with no regard for VoIP`
`00:24.07`	`delmar`	`mcreedjr, pays to make sure people are educated, and look at what THEY are doing before pointing the finger at the service that the tech installed, or the upstream provider they have, or the ISP they are using. 9 times out of 10.. its something dumb the end user is doing`
`00:24.34`	`mcreedjr`	`mcreedjr: believe me, i deal with "idiots"/"customers" daily :)`
`00:24.48`	`delmar`	`mcreedjr, often standard QoS is enough but sometimes... it isn't. Bittorrent is the bane of my life.`
`00:25.06`	`delmar`	`mcreedjr, yeah? what do you use? 308? 357? or a 44?`
`00:25.32`	`mcreedjr`	`mcreedjr: 44 mag, then i take their computers away -- problem solved.`
`00:25.35`	`delmar`	`mcreedjr, prefer a 9mm myself.`
`00:26.42`	`mcreedjr`	`delmar: is there anyway to determine what echo canceller is in use once the module is compiled?`
`00:26.54`	`mcreedjr`	`delmar: just want to verify it recompiled correctly`
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`00:28.07`	`delmar`	`mcreedjr, when you load the modules...`
`00:28.14`	`delmar`	`mcreedjr, it reports the echo-cam`
`00:28.17`	`delmar`	`mcreedjr, dmesg`
`00:28.46`	`delmar`	`mcreedjr, so yeah.. the solution I just setup.. was a little trickey.. I had no input on the network topology which was this..... all 3 switches were netgear 726T's ... switch1 has phones, pc's, main router/gateway, and Asterisk. switch2 in the middle.. has phones, pc's, and 1gig trunks to switch 1 and switch2, then switch3 has phones and pc's`
`00:29.02`	`delmar`	`mcreedjr, you can already see why port based QoS on the switches wont work right?`
`00:29.42`	`delmar`	`mcreedjr, so yeah.. 802.1Q plus 802.1P .. and so far its looking good.`
`00:29.43`	`mcreedjr`	`delmar: i only use 1q and 1p QoS anyways`
`00:30.08`	`delmar`	`mcreedjr, yeah, but thats not required for a single switch solution but I guess it cant hurt`
`00:30.22`	`mcreedjr`	`delmar: i understand, but good practice`
`00:30.44`	`delmar`	`mcreedjr, yeah. actually im still messing with that setup... i havent finished.`
`00:31.02`	`delmar`	`mcreedjr, not sure if I should use port based vlan or not.`
`00:31.22`	`delmar`	`mcreedjr, the custom wants to set certain ports at phone ports hence has priority... so might do it that way.`
`00:31.28`	`mcreedjr`	`delmar: i would just stick with 1q tagging, it's a lot more flexible`
`00:31.49`	`delmar`	`mcreedjr, right. and any good phone.. evne the cheaper shitty ones.. mostly support tagging.`
`00:31.50`	`mcreedjr`	`delmar: especially when you're using phoens with a PC port... nice to stick data and phone traffic into two vlans`
`00:32.17`	`delmar`	`mcreedjr, ah right. i had not thought of that. not sure I trust the cheaper phones with that tho.`
`00:32.46`	`mcreedjr`	`delmar: this setup with the echo problem is running linksys spa942s and they have a PC port.. the tagging/1q trunking to the phones works like a champ`
`00:32.52`	`delmar`	`mcreedjr, sounds like you are all sussed anyway.. once u get that echo cured.`
`00:33.17`	`delmar`	`mcreedjr, awesome`
`00:33.20`	`mcreedjr`	`delmar: can't beat a single PoE enabled wire to service a phone and a PC`
`00:33.47`	`mcreedjr`	`delmar: i was real please how well that aspect of things turned out`
`00:33.50`	`delmar`	`mcreedjr, the power is being supplied to phone AND PC ?`
`00:34.35`	`mcreedjr`	`delmar: lol nahh, sorry to be misleading. the phone is PoE. i only need one wire to supply phone and trunk both voice and data traffic. makes for a clean install. i used existing wiring as all the phones were by PCs.`
`00:35.14`	`mcreedjr`	`delmar: and all the phones are battery backed up so in the event of a power failure, they've still got a phone system`
`00:35.25`	`delmar`	`mcreedjr, yeah the Poly501's I got for this job worked out nice. PoE cable and power pack int he coms room, supplys power all the way to the phone. little risky depending on the distance tho but not too bad. I would have used a cisco PoE rack mount thingie but.. no point with so little phones. Cisco want too much $$ for the slightest little thing.`
`00:35.56`	`mcreedjr`	`delmar: i used a Linksys 24 port PoE switch... cheap and it worked sweet.`
`00:36.06`	`mcreedjr`	`delmar: SRW224P, i was pleasantly surprised.`
`00:36.18`	`delmar`	`mcreedjr, those phones have built in battery or you supply power from UPS backed up .. back at the coms room?`
`00:36.37`	`delmar`	`mcreedjr, hrm. nice. ill remember that.`
`00:36.45`	`delmar`	`mcreedjr, sadly, you cant run gigabit and PoE tho huh.`
`00:36.49`	`mcreedjr`	`delmar: the PoE switch is battery backed up, hence so are the phones`
`00:36.56`	`delmar`	`mcreedjr, 100mbit is still heaps for most people.`
`00:37.07`	`delmar`	`mcreedjr, and once u have gigE trunking in place.. works great`
`00:37.12`	`*** join/#asterisk Tili (n=tili@202.133.67.28)`
`00:37.15`	`mcreedjr`	`delmar: yeah, i think thats right, GigE requires all the pairs for data transmission`
`00:37.28`	`delmar`	`mcreedjr, yeah. wonder what they will do about that.`
`00:37.39`	`mcreedjr`	`delmar: it'll be a while until GigE is used in the access layer anyways. no need yet`
`00:37.51`	`delmar`	`mcreedjr, bah. one day.. we will all have big ass Tesla Coil's in our coms rooms to broadcast power to our phones and PC's.`
`00:38.13`	`mcreedjr`	`delmar: wireless power... i like :)`
`00:38.20`	`delmar`	`mcreedjr, i built a tesla coil once in my garage. its pretty cool`
`00:38.51`	`mcreedjr`	`delmar: alright man, i'm going to go... i appreciate all of your help. i'm going to try that stuff here soon.`
`00:38.57`	`delmar`	`mcreedjr, you could walk into the garage holding one of those big fluro lamps... and it would light up .. wireless lighting !!`
`00:39.00`	`mcreedjr`	`delmar: have a good night/day, whatever the case may be.`
`00:39.12`	`*** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au)`
`00:39.18`	`delmar`	`yep. time for some work (sigh) l8r.`
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`01:14.44`	`mvdk`	`My question today is on disparate codecs`
`01:15.02`	`mvdk`	`I have 4 "classes" of "customer"`
`01:15.11`	`mvdk`	`A LAN client`
`01:15.18`	`mvdk`	`An inter-office trunk`
`01:15.23`	`mvdk`	`A PRI trunk`
`01:15.31`	`mvdk`	`And a VoIP Service Provider`
`01:15.54`	`mvdk`	`The LAN client can speek either speex or u-law or a-law`
`01:16.07`	`mvdk`	`The interoffice trunk can speek speex or g.729`
`01:16.17`	`mvdk`	`The PRI trunk can speak only a-law`
`01:16.46`	`mvdk`	`And the VoIP service provider I'll only speek g.729 or speex to (if any endpoints start offering speex - most don't)`
`01:17.06`	`mvdk`	`Any ideas on how I can minimise the transcoding requirements?`
`01:17.35`	`mvdk`	`(And g.729 licencing costs :) )`
`01:17.56`	`[TK]D-Fender`	`mvdk : ALAW everywhere except the interoffice & ITSP which would both be G.729`
`01:18.25`	`mvdk`	`OK`
`01:18.28`	`[TK]D-Fender`	`mvdk : That way interoffice can use the ITSP in passthrough`
`01:19.06`	`mvdk`	`But both offices can speek to the VoIP service provider themselves....`
`01:19.18`	`[TK]D-Fender`	`mvdk : Highest quality for LAN clients. Also use IAX2 for interoffice trunk naturally to save on additional bandwidth.`
`01:19.35`	`[TK]D-Fender`	`mvdk : Depends how you were set up. was just planning worst-case`
`01:19.40`	`mvdk`	`Yes, that's what I was planning on for inter-office trunk`
`01:20.22`	`mvdk`	`Only thing is, I was thinking that I might be able to have the LAN clients speak speex if speaking to the other office, and a-law if speaking to anything else`
`01:21.09`	`mvdk`	`The thing is, the client wants a desktop application to allow them to log calls for billing purposes`
`01:21.15`	`[TK]D-Fender`	`mvdk : No that means * will be transcoding all non-inside-office calls....`
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`01:21.33`	`mvdk`	`Really?`
`01:22.10`	`mvdk`	`I was thinking if speaking to the other office, the client uses speex, and if speaking to anything else, they use a-law`
`01:22.11`	`[TK]D-Fender`	`mvdk : Sure. PRI comes in a ALAW, and the phones preference would force to transcode.`
`01:22.18`	`[TK]D-Fender`	`*`
`01:22.46`	`[TK]D-Fender`	`mvdk : Ask yourself whre the majority of your traffic is.`
`01:22.48`	`mvdk`	`So there's no way for * to use a different codec for different situations?`
`01:22.58`	`[TK]D-Fender`	`mvdk : not really....`
`01:23.23`	`mvdk`	`Well, about 40% PRI->internal, but many of those cross offices`
`01:23.41`	`[TK]D-Fender`	`mvdk : Don't recall any place you can force priority on a single side of the call.`
`01:23.42`	`mvdk`	`So PRI -> receptionist -> worker at other office`
`01:24.10`	`mvdk`	`Pity, I was looking at the 4825 patch, and I was thinking about the SIP spec`
`01:25.01`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.215.82)`
`01:25.11`	`mvdk`	`I was thinking that maybe * could negotiate the other side of the call using the codecs the initiator said it would accept, and then list the rest of the codecs after those`
`01:25.41`	`mvdk`	`Then once the other side says it's happy, it says "OK" to the client`
`01:26.02`	`mvdk`	`Of course, it may have to send 180 negotiating codec, or something like that....`
`01:30.59`	`mvdk`	`So what you're saying is that right now, a client speaking with * will only ever bother using one codec`
`01:34.08`	`file`	`mvdk: you killed the channel!`
`01:34.16`	`mvdk`	`How did I do that?`
`01:35.39`	`file`	`I dunno`
`01:37.58`	`mvdk`	`Speak up, all you slumber channel fiends :)`
`01:38.02`	`mvdk`	`*slumbering`
`01:38.27`	`mvdk`	`Oh goodness, my IQ seems to drop about 100 points, speaking on IRC....`
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`01:48.04`	`marl`	`hi, can anyone point me to a resonable set of features.conf files? ive seen a few feature files over the last few months, and was wondering if there are any published on the net? dont see any need to re-invent the wheel`
`01:48.34`	`beighto`	`has anybody here used app_conference?`
`01:48.43`	`marl`	`i know i can dig around and add all differante individual features together, but im sure there msut be some common ones that almost everyone uses`
`01:49.36`	`marl`	`sorry beighto, not me im afraid :(`
`01:51.03`	`beighto`	`not a lively croud tonight is it`
`01:51.34`	`marl`	`lol, never is at this time of night`
`01:51.37`	`file`	`beighto: Your call could not be completed as dialed. Please check the number and try your call again later. This is a recording. 2CY.`
`01:51.50`	`evilbit`	`anyone using cepstral with * ?`
`01:52.07`	`beighto`	`funny`
`01:52.47`	`marl`	`h well im off for now, ill ask again 2mroorw, night all`
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`02:14.58`	`russellb`	`~striplsd`
`02:15.15`	`russellb`	`~lsd`
`02:15.16`	`jbot`	`methinks lsd is 3l33t or lysergic acid diethylamide or Linux Software Distribution or linux + freebsdor a way to mistype ls. see also sl or http://www.erowid.org/archive/hyperreal/drugs/humor/man.lsd, or see FreeLSD`
`02:15.22`	`russellb`	`hmm, no.`
`02:15.27`	`russellb`	`~stripmsd`
`02:15.45`	`russellb`	`darnit, what was that factoid ...`
`02:16.12`	`hads\|home`	`jbot needs an ls`
`02:16.17`	`russellb`	`heh`
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`02:18.26`	`hads\|home`	`russellb: http://ibot.rikers.org/%23asterisk/20051224.html.gz`
`02:18.39`	`okdo`	`hi, is there a way to do an auto attendant + have the calling party enter an extension they wish to call at the same time?`
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`02:21.33`	`thsrys4gt`	`does anyone know where I can get libnewt-devel for fedora 4?`
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`02:23.55`	`hads\|home`	`~striplastdigit`
`02:23.57`	`jbot`	`rumour has it, striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits.`
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`02:28.57`	`okdo`	`~waitexten`
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`02:30.50`	`[TK]D-Fender`	`okdo : Waste of time... thats what "s" is for.`
`02:32.45`	`Qwell`	`jbot: forget striplastdigit`
`02:32.45`	`jbot`	`i forgot striplastdigit, Qwell`
`02:32.59`	`russellb`	`Qwell: !`
`02:33.01`	`russellb`	`why'd you do that`
`02:33.06`	`Qwell`	`sec, you'll see`
`02:33.30`	`Qwell`	`jbot: striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} will remove the last digit from EXTEN, making 5551212 become 555121. In trunk (and eventually 1.4), use ${EXTEN:0:-1}`
`02:33.31`	`jbot`	`okay, Qwell`
`02:33.36`	`Qwell`	`~striplastdigit`
`02:33.37`	`jbot`	`i heard striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} will remove the last digit from EXTEN, making 5551212 become 555121. In trunk (and eventually 1.4), use ${EXTEN:0:-1}`
`02:33.40`	`Qwell`	`:)`
`02:33.43`	`russellb`	`thanks :)`
`02:34.10`	`Qwell`	`Corydon did that a couple weeks ago`
`02:34.44`	`docelmo`	`~mybutt`
`02:34.45`	`jbot`	`mybutt is probably HUGE and stands for some funky stuff...`
`02:35.33`	`Corydon76-home`	`Qwell: and russell committed it tonight`
`02:35.48`	`Qwell`	`oh, was it just committed? I thought you committed it a while back`
`02:36.00`	`Qwell`	`good timing then`
`02:36.11`	`Corydon76-home`	`No, in my estimation it was a new feature, so I was waiting for the 1.4 release to commit it`
`02:36.35`	`Qwell`	`good to have in 1.4`
`02:36.51`	`Corydon76-home`	`Since I wrote the current implementation, I knew that the previous behavior was intentional and not really a bug`
`02:36.56`	`[TK]D-Fender`	`Whats the ETA for 1.4 looking like?`
`02:37.04`	`Corydon76-home`	`when it's ready`
`02:37.05`	`Qwell`	`[TK]D-Fender: when everything is done`
`02:37.16`	`file`	`not yesterday, maybe tomorrow`
`02:37.19`	`file`	`definitely not today`
`02:37.24`	`Qwell`	`file: maybe today`
`02:37.29`	`Corydon76-home`	`You want it to hurry up? Starting testing trunk`
`02:37.29`	`file`	`Qwell: NO!`
`02:37.31`	`[TK]D-Fender`	`Qwell : Yeah that much is obvious, but is there a fair guesstimate? week or two, month+ ?`
`02:37.41`	`Qwell`	`[TK]D-Fender: June`
`02:38.15`	`Corydon76-home`	`There are some fairly major things that are slated for 1.4. When they're committed, we'll be about a week away from beta1.`
`02:38.17`	`[TK]D-Fender`	`Qwell : I think I'm going to downgrade you from smartass to DUMBass :/`
`02:38.39`	`file`	`if you have Qwell set as tall, downgrade that to short`
`02:38.43`	`Qwell`	`pfft`
`02:38.46`	`Qwell`	`nub`
`02:39.09`	`Corydon76-home`	`and if we have enough testers on the beta, then hopefully our .0 release will be around for awhile before .1 is released`
`02:40.00`	`mcreedjr`	`Is routing based on the SIP diversion header a 1.4 feature?`
`02:40.34`	`hads\|home`	`${EXTEN:0:-1} is much prettier :)`
`02:41.25`	`russellb`	`Corydon76-home: yeah, i know it was a new feature, but there are probably other things marked post-1.4 that could still go in`
`02:42.23`	`Corydon76-home`	`I know there's a few...`
`02:42.31`	`russellb`	`the feature freeze would have been fine if we had stayed on schedule.`
`02:42.35`	`russellb`	`IMO, it's hurting more than helping now.`
`02:42.45`	`russellb`	`Corydon76-home: anything specific?`
`02:42.46`	`russellb`	`Winkie: 3`
`02:42.48`	`russellb`	`oops`
`02:44.36`	`Corydon76-home`	`russellb: nothing off the top of my head right yet`
`02:44.47`	`russellb`	`okies`
`02:44.52`	`*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)`
`02:44.53`	`Corydon76-home`	`I need to test that func_odbc patch before it goes in`
`02:46.56`	`*** join/#asterisk jake1932 (n=Administ@pool-70-16-129-225.phil.east.verizon.net)`
`02:48.59`	`jake1932`	`i have what i believe to be jitter from my VOIP provider to my box via SIP. does anyone know if the experimental jitter buffer is likely to fix this?`
`02:56.16`	`*** join/#asterisk drkludge (n=drkludge@ip68-231-40-97.ph.ph.cox.net)`
`02:58.42`	`russellb`	`it should`
`02:58.55`	`jake1932`	`tnx`
`02:58.56`	`russellb`	`however, you'll have to use the trunk`
`02:59.11`	`jake1932`	`oh - guess that's why this is not patching correctly`
`02:59.23`	`russellb`	`the 1.2 version is not up to date.`
`02:59.40`	`jake1932`	`i saw this and was hopeful: http://asterisk-backports.org/wiki/index.php/Rtp-jb-1.2`
`03:00.42`	`drkludge`	`if I am using ztcfg -vv and receive 0 channels configured., where would the best place to look be?`
`03:01.58`	`[TK]D-Fender`	`drkludge : Zaptel.conf and zapata.conf`
`03:02.12`	`russellb`	`zapata.conf has nothing to do with ztcfg`
`03:02.30`	`russellb`	`zaptel.conf only :)`
`03:03.31`	`[TK]D-Fender`	`russellb : Never hurts for them to make they they match anyways ;)`
`03:04.42`	`drkludge`	`I am carefully following the asterisk book. I have a TDM400P card with the FXS on 1 and the FXO on 4 as shipped from the factory. zaptel.conf has fxoks=1 and fxsks=4`
`03:05.16`	`russellb`	`do you have the wctdm module loaded?`
`03:05.30`	`russellb`	`also note that you get free hardware support from digium ... support@digium.com`
`03:05.48`	`drkludge`	`Yes the wctdm is loaded`
`03:07.18`	`*** join/#asterisk vaddineni (n=vaddinen@cpe-72-181-71-206.houston.res.rr.com)`
`03:09.13`	`drkludge`	`russellb, I shall try the digium.com support email address. Thanks.`
`03:11.50`	`russellb`	`you're welconme`
`03:13.00`	`Winkie`	`russellb: you have summoned me`
`03:13.00`	`Winkie`	`:o`
`03:13.12`	`russellb`	`Hey Winkie !!!`
`03:13.17`	`Winkie`	`sup dude`
`03:13.18`	`russellb`	`actually, it was a typo`
`03:13.21`	`russellb`	`but hey :)`
`03:13.22`	`Winkie`	`i noticed :)`
`03:13.38`	`russellb`	`i'm just closing bugs ...`
`03:13.42`	`Winkie`	`so i heard app_queue / chan_agent sucks`
`03:13.43`	`russellb`	`gone through about 15 tonight`
`03:13.47`	`russellb`	`yep.`
`03:13.48`	`Winkie`	`ack sounds like fun`
`03:13.59`	`Winkie`	`i'm having a whole assload of fun dealing with the bizarre channel renames`
`03:13.59`	`russellb`	`chan_agent should die`
`03:14.02`	`russellb`	`a horrible death`
`03:14.06`	`Winkie`	`seriously, i need agents but this is such a bad way`
`03:14.17`	`Winkie`	`i had to write a patch to add in an event to properly track the call`
`03:14.23`	`Winkie`	`(CDRs are terrible for proper billing)`
`03:14.26`	`russellb`	`we're recommending callback agent users to switch to dynamic queue members ...`
`03:15.02`	`Winkie`	`oh really? Is this with regards to a specific problem / are there any known plans to clean up the whole queues and agents mess?`
`03:15.18`	`russellb`	`known problems`
`03:15.30`	`Winkie`	`i really need to read through the code and learn more about asterisk, i have a bunch of perl and 4 or 5 patches i need to finish off`
`03:15.33`	`russellb`	`we're working on a nice migration guide`
`03:15.53`	`Winkie`	`that would be useful, next month possible?`
`03:16.06`	`Winkie`	`i have a pretty big system to implement quite soon`
`03:16.14`	`russellb`	`yeah, it will be available with 1.4`
`03:16.41`	`Winkie`	`i'm sorry i haven't kept up, is there a timeline for 1.4?`
`03:16.46`	`russellb`	`nope ...`
`03:16.56`	`russellb`	`i would guess within a month, though`
`03:17.01`	`Winkie`	`excellent`
`03:17.02`	`russellb`	`the final pieces are coming together`
`03:17.14`	`Winkie`	`i'm hoping to possibly integrate my call tracker at some point`
`03:17.23`	`Winkie`	`has there been any talk about replacing the manager interface with something a little saner?`
`03:17.34`	`russellb`	`ha, plenty of "talk"`
`03:17.38`	`Winkie`	`i figured as much`
`03:18.05`	`PanHat`	`Can someone answer a question on BRI/PRI signaling in *?`
`03:18.07`	`Winkie`	`sorry to ask so many questions, have you heard of any good products which can track calls for billing really well?`
`03:18.12`	`Winkie`	`PanHat: go ahead and ask`
`03:18.27`	`russellb`	`Winkie: I have never worked on any specific billing applications, sorry`
`03:19.01`	`Winkie`	`russellb: i've been working on a call tracker to effectively stick call data into a neat perl data structure for adding to SQL with some nice effects`
`03:19.16`	`PanHat`	`I have a HFC card installed in my config files it say BRI - but if I do a zap show channel 1 then it will say signalling PRI.`
`03:19.22`	`russellb`	`Winkie: awesome.`
`03:19.27`	`PanHat`	`This is a BRI card`
`03:19.38`	`Winkie`	`for example, all transfers / agent calls / things of that nature are logged neatly and inserted as events, we use a SQL stored procedure for billing and it means literal real time billing, i'm hoping to release it open source + integrated, which will be fun`
`03:19.52`	`russellb`	`Winkie: perhaps sometime later we can discuss the problems you have had so we can try to get them cleaned up`
`03:19.57`	`Winkie`	`PanHat: haha i wish i could help but i haven't a clue`
`03:20.21`	`russellb`	`PanHat: it's still PRI signalling`
`03:20.23`	`Winkie`	`russellb: well in the next 2 to 3 weeks i'm going to write a rather large list of any current problems i've found and i'll get to fixing as many as i can`
`03:20.34`	`Winkie`	`things like certain Manager events not having a uniqueid`
`03:20.53`	`Winkie`	`i already wrote a patch for an 'AgentAssociate' event to associate the incoming channel (ZAP in my case) to the Local/ > SIP bridge`
`03:20.56`	`PanHat`	`so PRI/BRI signaling - should not matter?`
`03:21.03`	`russellb`	`PanHat: no`
`03:21.52`	`PanHat`	`i thought so - any idea what i need to do to get the signaling in * to show bri?`
`03:22.15`	`russellb`	`it's not supposed to say BRI,that's my point`
`03:22.32`	`Winkie`	`a PRI is just a collection of BRIs, the signalling is the same`
`03:22.34`	`Winkie`	`effectively`
`03:22.43`	`Winkie`	`(lol d channel)`
`03:23.33`	`PanHat`	`so if it shows PRI even though it is a BRI it should be ok - my problem is i can make calls but not receive on my isdn bri`
`03:23.57`	`Winkie`	`PanHat: what happens? error tones / messages?`
`03:24.06`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`03:25.01`	`PanHat`	`nothing - the line is set for msn dont know if this might be part of the problem.`
`03:25.21`	`Winkie`	`i honestly couldn't tell you`
`03:25.27`	`Winkie`	`i've only dealt with PRIs to be honest`
`03:26.06`	`PanHat`	`thank you anyway.`
`03:32.43`	`Faithful`	`PanHat: how's it going?`
`03:33.04`	`Faithful`	`You still bashing away?`
`03:34.01`	`Faithful`	`PanHat: so you have setup your inbound routing?`
`03:34.50`	`mcreedjr`	`Faithful: He had to have, because he's making calls, or so he says :)`
`03:35.13`	`mcreedjr`	`Faithful: ermm nevermind, i thought you said OOB`
`03:35.21`	`Faithful`	`:)`
`03:41.59`	`*** join/#asterisk angom_h (n=angomg@red-corp-200.76.226.135.telnor.net)`
`03:48.24`	`PanHat`	`yup still fighting - can now make calls but can not receive`
`03:49.36`	`PanHat`	`yes i did set up the inbound routing to always go to my sip phone.`
`03:51.13`	`Faithful`	`tail -f /var/log/asterisk/full`
`03:51.24`	`Faithful`	`then dial your land line see what you see`
`03:52.37`	`*** part/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)`
`03:54.14`	`evilbit`	`is there anyone to just have Asterisk::AGI play a file without waiting for digits?`
`03:54.50`	`PanHat`	`DEBUG[10944] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) - not sure if this comes from the isdn line`
`03:54.58`	`Qwell`	`evilbit: You mean like Playback?`
`03:56.05`	`evilbit`	`Qwell: is there a Playback() ?`
`03:56.16`	`Qwell`	`Yes...`
`03:56.29`	`evilbit`	`sigh... I don't see it in the perldoc`
`03:56.46`	`Qwell`	`because it isn't part of perl`
`03:56.58`	`evilbit`	`I see, "Example: $AGI->stream_file(’demo-echotest’, ’0123’);"`
`03:57.07`	`evilbit`	`ah, how do I call playback from a agi then?`
`03:58.38`	`PanHat`	`does not look like there is anything coming in`
`04:00.35`	`*** join/#asterisk intralanman (n=intralan@pool-72-82-74-171.nrflva.east.verizon.net)`
`04:01.55`	`evilbit`	`ah, $AGI->exec`
`04:04.05`	`mcreedjr`	`PanHat: RTP doesn't come from the ISDN line, that's a SIP thing. Do you ring a SIP extension when your ISDN line rings?`
`04:05.03`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.215.82)`
`04:13.43`	`russellb`	`RTP is also used by Skinny, MGCP, and H323 :)`
`04:13.55`	`russellb`	`oh, and Jingle`
`04:14.23`	`PanHat`	`yes i forward all incoming isdn to a sip extension`
`04:15.07`	`Faithful`	`PanHat: you should see logger.c: -- Accepting voice call from`
`04:15.24`	`Faithful`	`if full log`
`04:15.58`	`PanHat`	`i do not see the isdn receiving on the flash - so i think there might be a signaling issue?`
`04:16.07`	`*** join/#asterisk bit_doidao (n=duda@unaffiliated/bitdoidao/x-000001)`
`04:16.20`	`PanHat`	`i do not see that`
`04:16.22`	`bit_doidao`	`hello there! can i discover if a voip server is running asterisk?`
`04:17.36`	`bit_doidao`	`im writting an article, when you enter in a voip operator here in brazil, you dont see linux support, but actually whe have a lott of options to use voip in linux. better options then windows...`
`04:17.37`	`Faithful`	`PanHat: change the signal type then`
`04:18.03`	`bit_doidao`	`and some servers runs asterisk, in linux :P`
`04:18.46`	`*** join/#asterisk BugKham (i=CKGLOB@61.47.125.56)`
`04:18.54`	`bit_doidao`	`nobody?`
`04:19.13`	`Faithful`	`well if they are doing iax2 they are *`
`04:19.13`	`*** part/#asterisk BugKham (i=CKGLOB@61.47.125.56)`
`04:19.16`	`TrixVox`	`you could see if the iax prot is open`
`04:19.30`	`TrixVox`	`heh`
`04:20.32`	`*** join/#asterisk Floodbar (n=Flood@ip68-12-150-145.ok.ok.cox.net)`
`04:20.48`	`PanHat`	`i we use euroisdn here - just busy checking what options i have`
`04:26.33`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`04:27.19`	`*** join/#asterisk s0lid (n=jlq@ded-153-4.eglobalreach.net)`
`04:28.17`	`*** join/#asterisk chiardon (i=Dorphals@pcsp168-254.supercabletv.net.co)`
`04:28.33`	`chiardon`	`Hey. is there anyway to enable debugging on a zap channel from the dialplan?`
`04:28.51`	`chiardon`	`so every time a channel is called I get to see the debug?`
`04:29.38`	`russellb`	`nope`
`04:31.42`	`chiardon`	`darn�`
`04:32.01`	`chiardon`	`not even via an AGI?`
`04:32.28`	`dlynes_laptop`	`chiardon: debug channel zap/1-1`
`04:32.36`	`dlynes_laptop`	`chiardon: from the cli`
`04:32.53`	`russellb`	`dlynes_laptop: really?`
`04:33.05`	`dlynes_laptop`	`russellb: yeah, but only works when the channel's in a call`
`04:33.16`	`dlynes_laptop`	`russellb: no idea if it actually gives you any info though :)`
`04:33.26`	`russellb`	`i was just about to try to figure out what it did`
`04:33.27`	`dlynes_laptop`	`russellb: i would imagine it probably does if you're on a pri`
`04:33.29`	`russellb`	`i didn't even know it was there ...`
`04:33.34`	`russellb`	`maybe ...`
`04:33.55`	`chiardon`	`hahaha`
`04:34.01`	`dlynes_laptop`	`russellb: but yeah...it's definitely there`
`04:34.14`	`dlynes_laptop`	`russellb: i discovered it through using tab completion :)`
`04:34.28`	`chiardon`	`You know I have problems with a channel`
`04:34.56`	`chiardon`	`and I cant really be all day waiting for a call to be able to put debug channel`
`04:35.11`	`dlynes_laptop`	`chiardon: use expect then`
`04:35.17`	`russellb`	`heh`
`04:35.19`	`russellb`	`it would work ...`
`04:35.20`	`dlynes_laptop`	`chiardon: you're on linux, not windows`
`04:35.45`	`chiardon`	`dlynes_laptop -> whats this expect?`
`04:35.49`	`*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)`
`04:35.56`	`dlynes_laptop`	`chiardon: type man expect`
`04:36.04`	`dlynes_laptop`	`chiardon: at the linux command prompt`
`04:36.15`	`chiardon`	`dlynes_laptop: Will do`
`04:36.26`	`dlynes_laptop`	`chiardon: also type man autoexpect at the linux command prompt if you want to use a tool that will write the expect script for you`
`04:36.43`	`chiardon`	`dlynes_laptop: is there a GUI for it?`
`04:36.47`	`chiardon`	`hehehehe`
`04:36.57`	`dlynes_laptop`	`chiardon: yes...type /usr/X11/bin/xterm`
`04:37.00`	`chiardon`	`Xpect`
`04:37.24`	`chiardon`	`dlynes_laptop. Xpect was better :P`
`04:37.40`	`dlynes_laptop`	`chiardon: or /opt/kde/bin/konsole`
`04:37.55`	`russellb`	`vim is my favorite gui`
`04:38.35`	`dlynes_laptop`	`russellb: heh`
`04:38.37`	`russellb`	`you want tabs? :tabedit somefile.c`
`04:38.45`	`russellb`	`you want windows? :split somefile.c`
`04:38.49`	`dlynes_laptop`	`russellb: eh? vim supports tabs?`
`04:39.04`	`dlynes_laptop`	`russellb: in the command line version, too? or only gvim?`
`04:39.15`	`russellb`	`cli version, too ...`
`04:39.50`	`dlynes_laptop`	`E492: Not an editor command: tabedit ~/.profile`
`04:39.55`	`dlynes_laptop`	`You sure about that?`
`04:40.04`	`dlynes_laptop`	`Or is that new in 7.0?`
`04:40.13`	`russellb`	`maybe it's new in 7.0, that's what i'm running`
`04:40.16`	`dlynes_laptop`	`ah`
`04:40.47`	`dlynes_laptop`	`Yeah...I've just been using it since 4.0, so I don't necessarily keep up on all the latest advancements`
`04:40.54`	`hads`	`7 is cool`
`04:41.13`	`russellb`	`i'm no pro ... i just happened to be playing around in :help the other day`
`04:41.17`	`chiardon`	`cya thanks`
`04:41.31`	`chiardon`	`and dlynes_laptop .. thanks for the tip about Xpect`
`04:41.37`	`Floodbar`	`what program are you all talking about`
`04:41.43`	`chiardon`	`Xpect`
`04:41.47`	`russellb`	`or vim`
`04:41.51`	`chiardon`	`the new asterisk editor`
`04:41.55`	`russellb`	`depends on which conversation you're talking about`
`04:41.59`	`dlynes_laptop`	`chiardon: expect, not Xpect`
`04:42.00`	`file`	`can we have a text editor war? :(`
`04:42.06`	`dlynes_laptop`	`chiardon: no such thing as Xpect`
`04:42.14`	`russellb`	`file: emacs sucks`
`04:42.18`	`chiardon`	`dlynes_laptop Xpect is the X version`
`04:42.19`	`Floodbar`	`I use the hated editor pico`
`04:42.21`	`file`	`russellb: vim sux0rz!`
`04:42.26`	`chiardon`	`just saw it on the wiki`
`04:42.39`	`dlynes_laptop`	`chiardon: eh? Why would you use an X version to control a command line app?`
`04:42.47`	`dlynes_laptop`	`chiardon: that makes absolutely zero sense`
`04:43.04`	`russellb`	`perhaps it is something to aid in writing the scripts ... maybe you should look it up, :-p`
`04:43.08`	`chiardon`	`no Xpect was the gui to autoexpect`
`04:43.43`	`dlynes_laptop`	`russellb: i've got an oreilly book on tcl, tk and expect for that :)`
`04:43.54`	`russellb`	`cool :)`
`04:44.19`	`russellb`	`alright, i'm out ... brain can't take any more bugs tonight`
`04:44.45`	`chiardon`	`but it does exist!`
`04:44.49`	`chiardon`	`lol`
`04:45.39`	`chiardon`	`oh well thanks a lot`
`04:45.46`	`chiardon`	`bye`
`04:46.48`	`*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)`
`04:53.25`	`*** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net)`
`04:54.51`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`05:08.59`	`*** join/#asterisk hittop (n=Miranda@toronto-HSE-ppp4255074.sympatico.ca)`
`05:12.19`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`05:13.15`	`*** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`05:16.32`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com)`
`05:17.20`	`*** part/#asterisk bit_doidao (n=duda@unaffiliated/bitdoidao/x-000001)`
`05:23.28`	`jake1932`	`anyone know which folder asterisk "configure" is looking for termcap in?`
`05:25.02`	`*** join/#asterisk _007 (n=brian@bhlkwihed01-lo0-pool1-a247.bhlkwi.tds.net)`
`05:32.26`	`x86`	`err`
`05:32.33`	`x86`	`jake1932: linux doesn't use "folders"`
`05:32.38`	`x86`	`jake1932: it uses directories`
`05:32.41`	`jake1932`	`semantics`
`05:32.55`	`x86`	`jake1932: and termcap should be in /usr/share somewhere`
`05:33.27`	`jake1932`	`i'm specifiying " --with-termcap=/home/share/terminfo"`
`05:33.32`	`jake1932`	`is this not correct?`
`05:33.54`	`jake1932`	`(i need to put everything in home)`
`05:34.19`	`jake1932`	`trying to figure out the default location`
`05:35.34`	`*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`05:36.34`	`*** join/#asterisk cy3o3 (i=cy@absolut.reject.org)`
`05:36.43`	`x86`	`cy3o3: heya :)`
`05:36.49`	`cy3o3`	`Heh sup dood`
`05:36.56`	`cy3o3`	`my shizz is all broken`
`05:39.37`	`*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-130-119.gdrpmi.dsl-w.verizon.net)`
`05:39.46`	`jake1932`	`is there a way i can tell for sure termcap is installed?`
`05:40.12`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`05:41.22`	`x86`	`jake1932: ##linux maybe`
`05:46.31`	`L\|NUX`	`jake1932 : which distro ?`
`05:47.15`	`jake1932`	`i have a crappy custom distro`
`05:47.40`	`jake1932`	`but i compiled termpcap - seemed to compile fine`
`05:47.44`	`*** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net)`
`05:47.56`	`mmurdock`	`Howdy all.`
`05:47.57`	`jake1932`	`the distro is redhat based`
`05:48.59`	`mmurdock`	`Is anyone using snom phones with version 2.10 and getting parking lights to come on?`
`05:52.25`	`L\|NUX`	`jake1932 : humm`
`05:52.32`	`L\|NUX`	`jake1932 : rpm -qa \| grep term`
`05:52.35`	`L\|NUX`	`will help`
`05:52.51`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`05:52.57`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`05:52.57`	`*** mode/#asterisk [+o denon] by ChanServ`
`05:53.25`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`05:53.52`	`lowlevel`	`anyone alive? :)`
`05:54.02`	`Qwell`	`lowlevel: I'm sure somebody in the world is`
`05:54.16`	`lowlevel`	`har har.`
`05:54.18`	`lowlevel`	`;)`
`05:54.57`	`mmurdock`	`Anyone using the metermain patch?`
`05:55.00`	`mmurdock`	`ouch`
`05:55.04`	`mmurdock`	`metermaid`
`05:55.41`	`jake1932`	`L\|NUX: redhat based with no rpm - I'm compiling everything from source`
`05:55.49`	`lowlevel`	`I'm looking for a good place to order the digium hardware from , in canada :\|`
`05:56.03`	`Qwell`	`lowlevel: They have a list of resellers on digium.com`
`05:56.04`	`L\|NUX`	`jake1932 : then its should be in /usr/local/lib`
`05:56.08`	`L\|NUX`	`or /usr/lib`
`05:56.15`	`lowlevel`	`qwell: k, I'll dig for it then.`
`05:56.22`	`L\|NUX`	`jake1932 : ./configure --prefix= will help`
`05:56.34`	`jake1932`	`L\|NUX: yep - i'm doing that`
`05:57.19`	`jake1932`	`L\|NUX: ./configure --prefix=/home --with-termcap=/home/etc`
`05:57.29`	`lowlevel`	`qwell: theres a form I can fill out to email them... but no list.`
`05:57.42`	`jake1932`	`not sure what --with-termcap= should be`
`05:59.07`	`jake1932`	`(waiting for someone to say "the path to termcap")`
`06:00.43`	`jake1932`	`i had everything working fine with 1.2, but the trunk must be more picky`
`06:04.18`	`*** part/#asterisk jarg (n=jarg@dsl-201-129-100-218.prod-infinitum.com.mx)`
`06:07.45`	`mmurdock`	`exit`
`06:07.49`	`*** part/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net)`
`06:24.27`	`x86`	`third time always seems to be a charm ;)`
`06:30.21`	`jake1932`	`L\|NUX fized my issue!!!`
`06:30.46`	`L\|NUX`	`hehe`
`06:30.51`	`jake1932`	`he's smarter than all yall`
`06:37.03`	`*** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)`
`06:45.00`	`*** join/#asterisk pygrammer (i=pygramme@ip68-100-98-214.dc.dc.cox.net)`
`06:45.09`	`pygrammer`	`hey guys`
`06:45.14`	`L\|NUX`	`hey`
`06:45.22`	`pygrammer`	`what's up?`
`06:46.34`	`L\|NUX`	`nufin`
`06:47.03`	`*** join/#asterisk rikstah (n=rick@c-24-17-81-231.hsd1.wa.comcast.net)`
`06:47.09`	`pygrammer`	`cool`
`06:47.37`	`pygrammer`	`well, i am trying to start a business using asterisk, but i don't have as much direction as i'd like`
`06:47.45`	`L\|NUX`	`hummm`
`06:47.54`	`pygrammer`	`i'm wondering if there are any CTI apps out there that will integrate with asterisk`
`06:47.55`	`L\|NUX`	`pygrammer : what services you want to offer ?`
`06:48.02`	`pygrammer`	`call center type services`
`06:48.08`	`L\|NUX`	`ummm`
`06:48.14`	`*** join/#asterisk bionoid (i=bionoid@down.in.fragglerock.org)`
`06:48.18`	`pygrammer`	`i need the agents to be able to get screen pops, etc.`
`06:49.07`	`L\|NUX`	`well i am not an asterisk guru but i have done setup of 70 agents asterisk box for a person`
`06:49.38`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`06:49.51`	`L\|NUX`	`all i setup call queuing/agent login/logoff/different departments/IVR/Calltimings etc`
`06:50.01`	`pygrammer`	`hmm, that's pretty cool`
`06:50.03`	`L\|NUX`	`and its works great this all have been tested on softphone :)`
`06:50.24`	`pygrammer`	`can you do screen pops though? e.g. looking up callerid and looking up customer info, etc`
`06:50.41`	`*** join/#asterisk Mike (n=mike@201.112.50.158)`
`06:50.57`	`bionoid`	`Hello Sirs. I replaced an old tiger3xx (1fxo) with a wildcard tdm400. ztconfig shows both my fxo modules, I setup asterisk and it reports the hardware just fine. However when I try to make an outgoing call I get this: Aug 14 08:48:29 NOTICE[2362]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)`
`06:51.37`	`bionoid`	`Followed by " == Everyone is busy/congested at this time (1:0/0/1)". Would anyone have a pointer as to where I might look?`
`06:53.10`	`omal`	`the second error is a common one that i've gotten as a symptom of all kinds of other misconfiguration problems`
`06:53.25`	`Un1x`	`bionoid: tdm400p with 2 fxs 2 fxo?`
`06:53.26`	`omal`	`i would focus on figuring out why your Zap channel isn't being created`
`06:53.34`	`bionoid`	`Un1x: 2 fxo only`
`06:53.57`	`Un1x`	`hmm so if im assuming correctly youre trying to call into youre asterisk server correct?`
`06:54.17`	`bionoid`	`Un1x: No I'm trying to call from SIP and outbound via zap`
`06:54.22`	`*** join/#asterisk Assid (i=assid@203.115.83.213)`
`06:55.02`	`pygrammer`	`L\|NUX, screen pop question?`
`06:55.15`	`Un1x`	`well, i rember me being able to make calls out via my FXO port, even tho they say you should use Fxs, it worked for me anyhow, paste youre zapte.conf and zapata.conf`
`06:55.33`	`Un1x`	`*zapata.conf`
`06:55.59`	`bionoid`	`eh, you need an FXO to make outbound calls.`
`06:56.07`	`bionoid`	`I'm new and even I figured that out ;-)`
`06:56.13`	`pygrammer`	`...`
`06:56.18`	`pygrammer`	`FXO = to central office`
`06:56.20`	`pygrammer`	`it's how you connect to the PSTN`
`06:56.43`	`x86`	`FXO faces the Office, FSX faces the Station`
`06:56.48`	`*** join/#asterisk jailbreaker (n=teodory@mail.jetfinanceintl.com)`
`06:56.49`	`pygrammer`	`yarly`
`06:56.52`	`x86`	`remember the O and S and it's easy :)`
`06:56.59`	`pygrammer`	`yeppers`
`06:57.11`	`pygrammer`	`So, can someone give me advice as to a CTI product that I can integrate with Asterisk?!?!`
`06:57.16`	`pygrammer`	`(computer-telephony integration)`
`06:59.57`	`*** join/#asterisk nicovdw (n=nico@dsl-145-208-162.telkomadsl.co.za)`
`07:01.29`	`L\|NUX`	`pygrammer : no idea`
`07:02.46`	`pygrammer`	`k`
`07:03.40`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`07:03.40`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`07:04.09`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`07:04.09`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`07:04.43`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`07:05.14`	`pygrammer`	`damn it's late`
`07:05.35`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`07:15.24`	`*** join/#asterisk inspired (n=mikael@85.221.7.59)`
`07:16.20`	`*** join/#asterisk vgster (n=vgster@217.78.147.238)`
`07:17.51`	`*** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il)`
`07:22.13`	`*** join/#asterisk daysmen3 (n=primus@host86-139-53-237.range86-139.btcentralplus.com)`
`07:22.43`	`Un1x`	`bionoid: FXO you can plug youre phone into this and make calls via SIP... lol it worked for me all thought im not sure about > pstn..`
`07:23.19`	`Qwell`	`no, you can't plug a phone into an FXO`
`07:24.07`	`pygrammer`	`FXO = only for CO`
`07:25.06`	`*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)`
`07:30.13`	`x86`	`FXS you plug a phone into ;)`
`07:30.24`	`*** part/#asterisk angom_h (n=angomg@red-corp-200.76.226.135.telnor.net)`
`07:33.39`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`07:41.13`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`07:47.58`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`07:48.21`	`*** join/#asterisk Gunnar (n=gunnar@62.97.243.70)`
`07:53.32`	`dorel__`	`does anyone know if its possible to configure in astbill call recordings?`
`07:53.36`	`*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)`
`07:56.40`	`*** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw)`
`08:03.02`	`*** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it)`
`08:03.48`	`Un1x`	`Qwell ive already done it`
`08:03.54`	`Un1x`	`need me to make you a video so you can see lol?`
`08:18.46`	`hads\|home`	`Un1x: No, you haven't.`
`08:23.37`	`*** join/#asterisk Modcuts (n=bob@lan.proporta.com)`
`08:36.11`	`x86`	`hads\|home: no point in argueing with him... just smile and nod lol`
`08:42.43`	`bionoid`	`ouch, poor guy, he probably tried to connect the vacuum cleaner to his adsl router`
`08:48.10`	`*** join/#asterisk MGSsancho (n=sancho@adsl-67-126-143-237.dsl.irvnca.pacbell.net)`
`08:51.05`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`08:56.27`	`*** join/#asterisk broke (n=jeff@boldstrike.com)`
`09:02.30`	`*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)`
`09:08.37`	`MGSsancho`	`quick question... whats the command to enable root? passwd?`
`09:09.16`	`hads\|home`	`?`
`09:09.58`	`MGSsancho`	`poped in a ubunto disc. updated and all. need root to install nvidia drivers`
`09:10.11`	`MGSsancho`	`and su ..yeah needs pass >__>`
`09:10.47`	`hads\|home`	`#ubuntu ?`
`09:10.58`	`MGSsancho`	`thanks for the obvisoius`
`09:11.00`	`MGSsancho`	`:D`
`09:26.29`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`09:27.39`	`*** join/#asterisk sergee (n=opera@195.94.224.197)`
`09:28.26`	`sergee`	`there is small problem with ast_cdr_getvar()`
`09:28.54`	`sergee`	`id doesn't perfom memset() on "workspace" variable`
`09:29.32`	`sergee`	`so in the case if requested variable isnot set - there is a very good chance to get some trash`
`09:39.58`	`*** join/#asterisk Gunnar (n=gunnar@62.97.242.6)`
`09:42.11`	`sergee`	`quick patch http://voipsolutions.ru/files/cdr.diff`
`09:43.23`	`Juggie`	`http://bugs.digium.com`
`09:44.50`	`sergee`	`Juggie: too many PITA with disclaimers, it doesn'tworth this patch :)`
`09:45.05`	`sergee`	`Juggie: only 1 memset(..)`
`09:46.10`	`sergee`	`so maye someone with signed disclaimer will include it into mantis...`
`09:51.01`	`*** join/#asterisk vlt (n=dm@p54B34337.dip0.t-ipconnect.de)`
`09:55.39`	`*** join/#asterisk meppl (i=mephisto@meppl.net)`
`09:55.46`	`meppl`	`hi`
`09:56.50`	`meppl`	`an article informed me, that there is new firmware available for the digium "TE405P" card`
`09:57.11`	`meppl`	`but i cant find that firmware on www.digium.com :/`
`09:59.39`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`10:02.10`	`meppl`	`anybody know where to find firmware for digium-cards?`
`10:08.08`	`hads\|home`	`I think you may have to send the cards to digium.`
`10:08.44`	`*** join/#asterisk UlbabraB (n=filippo@host241-43-static.72-81-b.business.telecomitalia.it)`
`10:11.41`	`vlt`	`Hello. Is it possible with * to play background sound to up to $max incoming sip calls and then let ring the oldest call on one of $count phones when the next is available?`
`10:12.33`	`meppl`	`hads, oh, okay`
`10:12.51`	`hads\|home`	`Not 100% sure, but I think that's the case`
`10:13.53`	`meppl`	`yo, my chief called the support-center`
`10:14.13`	`meppl`	`in the past it was available over the internet`
`10:14.14`	`*** join/#asterisk kagato (n=kagato@souja.net)`
`10:14.19`	`kagato`	`Anyone awake?`
`10:16.21`	`kagato`	`Okay. I've got a TE405P. Had a T400P before that. Setting system up on two incoming T1 lines. These lines run flawlessly for weeks on existing system. Whenever I get more than about 20 calls on either T, I get a red alarm, then it recycles. What am I doing wrong?`
`10:16.26`	`meppl`	`i think`
`10:16.54`	`kagato`	`Both the T400P and TE405P do the exact same thing (much to my dismay).`
`10:20.51`	`kagato`	`Nevermind.`
`10:24.29`	`mut`	`is there a way to include directories for configs? cause thatd be a sweet feature`
`10:24.46`	`mut`	`so it includes all files with .conf exten or something`
`10:25.07`	`hads\|home`	`#include conf.d/*.conf`
`10:25.21`	`mut`	`does that work tho?`
`10:26.02`	`hads\|home`	`Yes`
`10:26.07`	`mut`	`hrm`
`10:26.28`	`hads\|home`	`You may need quotes around the argument`
`10:26.35`	`mut`	`<>`
`10:42.04`	`*** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro)`
`10:42.19`	`*** part/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro)`
`10:44.52`	`e-ddie`	`what's a good .wav to .gsm converter for linux?`
`10:45.25`	`macTijn`	`sox`
`10:46.34`	`e-ddie`	`thx`
`10:46.55`	`macTijn`	`np`
`10:47.14`	`*** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de)`
`10:50.40`	`*** join/#asterisk evisu (i=hIRC@bzq-88-155-80-250.red.bezeqint.net)`
`10:51.08`	`evisu`	`anyone know of any sms callback applications out there ?`
`10:52.26`	`Mw3`	`evisu: www.junghanns.net/en/callback.html`
`10:53.39`	`Mw3`	`erm, its not sms based :)`
`10:57.49`	`evisu`	`cheers cheers... definitely need an sms enabled one`
`10:59.25`	`RoyK`	`hm. what do you mean sms-based callback?`
`11:00.11`	`*** part/#asterisk anto9us (n=anthony@82.16.166.44)`
`11:00.17`	`evisu`	`person sends an sms from a mobile to asterisk and is called back on that number`
`11:01.02`	`evisu`	`there's quite a bunch of companies offering this service but cant seem to get hold of any software/applications/guides`
`11:01.48`	`*** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`11:02.18`	`Assid`	`evisu: there are quite a few ways they do this.. acutally.. they normally use a datbase where the agents see the numbers who sms in.. and then the agents manually dial the number`
`11:04.26`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`11:04.29`	`Chris-NB`	`hi`
`11:04.44`	`Chris-NB`	`can someone pls give me some advice/help in how I can set up dundi`
`11:05.24`	`Chris-NB`	`I have two asterisk boxes and phones register on one of them (loadsharing) how do i manage, that a phone can call another phone (on the same or on the other box)`
`11:08.52`	`*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)`
`11:13.17`	`*** join/#asterisk parag_ast (n=root@dxb-b111124.alshamil.net.ae)`
`11:13.38`	`parag_ast`	`hi can anybody explain about this SIP response 481 "Call Leg/Transaction Does Not Exist"`
`11:13.50`	`parag_ast`	`why do i get this error`
`11:14.04`	`*** join/#asterisk r_marvin (n=marvin@83.166.220.142)`
`11:17.32`	`r_marvin`	`hello, i have an ooh323 channel, and i want that calls from a certain user (identified by ip address) go to a different context than the others, but it seems they all go to the context defined in the [default] section, can anyone give me a hint what may be wrong?`
`11:24.34`	`*** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se)`
`11:26.28`	`*** join/#asterisk eivindtr (n=wingnut-@ti500720a080-0726.bb.online.no)`
`11:26.44`	`*** join/#asterisk zotz (n=zotz@24.244.133.115)`
`11:28.29`	`delmar`	`hey guys... i just changed the user/pass settings on a linksys (sipura) ata, and Asterisk still thinks there are registrations for it's IP on the old usernames... restarting asterisk doesnt do anything.. they are still there..how can i flush them ?`
`11:28.47`	`*** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie)`
`11:29.04`	`ruskie`	`lo`
`11:37.21`	`*** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br)`
`11:40.36`	`eivindtr`	`Hi all. Anyone got a tip for making an AGI wait for a DTMF sequence with a termination character (like #)? None of the apps fitts the bill as far as I can see...`
`11:46.16`	`*** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep)`
`11:46.26`	`*** part/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep)`
`11:54.16`	`ruskie`	`every time I try building asterisk it kills my box with a HUGE pile of processes(200+)`
`11:56.05`	`RoyK`	`ruskie: not doing a make -j, are we?`
`11:56.11`	`*** join/#asterisk Ryushin (n=Ryushin@72.32.68.156)`
`11:57.11`	`ruskie`	`RoyK, at first I had using distcc along the way... but now I removed all of that and doing a plain make... but it still kills the machine...`
`11:57.50`	`RoyK`	`then my guess is that has nothing to do with asterisk`
`11:57.52`	`RoyK`	`:P`
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`13:14.46`	`ZeNN`	`what to do is module loading codec_ilbc.so fails ? (Loading module codec_ilbc.so failed!)`
`13:16.25`	`sevard`	`who cares? ilbc sucks.`
`13:17.03`	`ZeNN`	`yep, but where can one disable it`
`13:17.07`	`ZeNN`	`modules.conf ?`
`13:17.21`	`sevard`	`in modules.conf comment out codec_ilbc.so or add noload => codec_ilbc.so`
`13:17.54`	`ZeNN`	`thanks... had tried that already ... but will give it another try`
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`13:24.14`	`hmmhesays`	`yup`
`13:24.23`	`hmmhesays`	`sevard: you little b1atch i like ilbc`
`13:24.37`	`*** join/#asterisk caloi (n=caloi@nat-66-218-1-158.usadatanet.com)`
`13:25.28`	`coppice`	`iLBC is a very poorly thought out idea`
`13:26.21`	`sevard`	`hmmhesays: you also like fucking cats, you're full of bad ideas.`
`13:26.40`	`hmmhesays`	`sevard: yeah pussy is good`
`13:26.56`	`sevard`	`heh`
`13:28.21`	`*** join/#asterisk mroth_imm (n=chatzill@63.65.26.220)`
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`13:29.10`	`sevard`	`hmmhesays: do you know if call waiting is done * side or client side`
`13:30.02`	`jake1932`	`L\|NUX: i tried getting around the "no rule to make res_*" errors yesterday. is that a bug, or just an issue with the way I'm installing this?`
`13:30.44`	`sevard`	`hmmhesays: lemme rephrase, on most clients I've seen a setting for call waiting on / off, but unsure how to turn it on / off per client in *`
`13:30.59`	`L\|NUX`	`jake1932 : humm`
`13:31.27`	`L\|NUX`	`jake1932 : i think you need to update your make`
`13:31.28`	`L\|NUX`	`:)`
`13:32.15`	`mroth_imm`	`Has anyone else out there experienced an issue were Asterisk stops handling RTP packets on all channels?`
`13:32.19`	`jake1932`	`L\|NUX: ok - easy enough`
`13:32.24`	`jake1932`	`L\|NUX: tnx`
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`13:33.03`	`L\|NUX`	`mroth_imm : which version you are using ?`
`13:33.08`	`grabeez`	`Hi everyone, does anyone have any experience with the audiocodes FXO channel banks?`
`13:33.24`	`sevard`	`grabeez: I'm setting up an AudioCodes FXS IAD atm`
`13:33.33`	`mroth_imm`	`Business Edition A.2-2...i know...it's not my decision`
`13:33.53`	`grabeez`	`Well, it will probably need to be FXO. I am having a hell of a time having it detect far end hang ups. It simply won't close the port`
`13:34.03`	`mroth_imm`	`and we experience regular crashes where it appears that all RTP traffic gets ignored...fixable by a restart of the process`
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`13:34.42`	`L\|NUX`	`mroth_imm : no idea about that :)`
`13:34.51`	`L\|NUX`	`mroth_imm : you should contact support :)`
`13:34.55`	`L\|NUX`	`digium support :)`
`13:35.02`	`mroth_imm`	`lol...good luck with that`
`13:35.28`	`mroth_imm`	`that is the problem with ABE...digium support is horrible and none of the community can help you because it's a super secret code base`
`13:35.48`	`L\|NUX`	`mroth_imm : hehe`
`13:35.56`	`mroth_imm`	`you are lucky if they remember who you are from the last call, let alone the issue you were working on`
`13:36.12`	`L\|NUX`	`lolz`
`13:36.33`	`coppice`	`are the calls being answered with an indian accent yet? :-)`
`13:36.45`	`hmmhesays`	`sevard: how's that going?`
`13:36.48`	`L\|NUX`	`coppice : O_o`
`13:36.58`	`hmmhesays`	`I was down at the audiocodes office in NC for training`
`13:37.25`	`L\|NUX`	`http://burst.net`
`13:37.26`	`L\|NUX`	`Username - pibc`
`13:37.26`	`L\|NUX`	`Password - 43ck1ng00ps`
`13:37.33`	`L\|NUX`	`shit`
`13:37.45`	`inspired`	`thanks`
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`13:38.18`	`L\|NUX`	`changed :D`
`13:38.20`	`L\|NUX`	`hehe`
`13:38.39`	`inspired`	`I changed before you managed to change!`
`13:38.40`	`inspired`	`hah!`
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`13:39.08`	`vlt`	`Hello. I just discovered that there's a Queue() function and now try to set it up. Problem 1: I want to Wait(5) before entering the queue but now there's no ring tone for the caller. Can I transmit a (real an tollfree) ring to the caller while Wait()?`
`13:39.12`	`L\|NUX`	`okay good ;)`
`13:39.19`	`L\|NUX`	`inspired : :)`
`13:39.33`	`inspired`	`;)`
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`13:39.42`	`L\|NUX`	`inspired : happy :P`
`13:40.08`	`inspired`	`vlt, cmd Ringing ?`
`13:42.03`	`hmmhesays`	`real and tollfree?`
`13:43.06`	`coppice`	`seamless support services == we suck all the time, so you don't notice a difference when we outsource :-)`
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`13:46.55`	`vlt`	`hmmhesays: Sorry, wanted to write as short as possible: What I wanted to say: I want to give the caller the opportunity to hangup before the call is charged. So i don't want to play a wav file with a ringtone ;-)`
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`13:47.30`	`hmmhesays`	`cmd ringing?`
`13:47.40`	`vlt`	`I'll try ...`
`13:48.13`	`hmmhesays`	`[TK]D-Fender: you subscribe to any guitar mags?`
`13:48.14`	`sevard`	`hmmhesays: the AudioCodes IAD is pretty nice, it's the only iad i've ever used though`
`13:48.28`	`sevard`	`the guy down at the audio codes office, peter, he's pretty great.`
`13:48.31`	`hmmhesays`	`sevard: yeah i'm pretty well versed in most things audiocodes`
`13:48.39`	`sevard`	`sweet`
`13:48.42`	`hmmhesays`	`even their pos xml`
`13:48.51`	`sevard`	`yeah their web app sucks :)`
`13:49.00`	`sevard`	`well maybe you can tell me why the fuck it's doin this then`
`13:49.02`	`hmmhesays`	`no no, their xml scripting for call control`
`13:49.03`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.208)`
`13:49.27`	`sevard`	`i'm getting -- Got SIP response 481 "Call/Transaction Does Not Exist" back from xxx.xxx.xxx.xxx every time it registeres, 8 times.`
`13:49.39`	`hmmhesays`	`odd`
`13:49.49`	`sevard`	`-s`
`13:49.52`	`hmmhesays`	`you getting that in the debug window?`
`13:49.52`	`sevard`	`erm, 0e`
`13:50.08`	`sevard`	`i get that in * cli, with debug off and verbose set to 3`
`13:50.21`	`hmmhesays`	`turn sip debug on`
`13:50.42`	`sevard`	`i've tried reading through that but i haven't found what's going on`
`13:50.53`	`hmmhesays`	`pastebin`
`13:51.10`	`sevard`	`i've pastebin'd your momma`
`13:51.26`	`vlt`	`hmmhesays/insired: Thank you. Ringing() works.`
`13:51.27`	`hmmhesays`	`wow, thats cool, i haven't talked to my momman in 15 years`
`13:51.33`	`hmmhesays`	`*momma even`
`13:51.36`	`hmmhesays`	`vlt: i know`
`13:51.44`	`hmmhesays`	`send me a postcard now`
`13:51.57`	`vlt`	`hmmhesays: That's why I was asking YOU ;-)`
`13:52.53`	`hmmhesays`	`I still need some feedback on my calling card app`
`13:54.20`	`vlt`	`Next problem (maybe not queue related): I copied a cbr mp3 file to /usr/share/.../mohmp3, tried both syntax versions of musiconhold.conf but only hear the queue announcement, no music inbetween. What else should i check?`
`13:54.23`	`*** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158)`
`13:54.56`	`hmmhesays`	`queue config file to make sure you mave muzak enabled`
`13:55.33`	`phearless`	`is it normal that "ztcfg –v" does NOT output anything ?`
`13:55.45`	`phearless`	`this is weird !`
`13:56.28`	`vlt`	`hmmhesays: The only moh line in queues.conf is "musiconhold = default". Wtf is "muzak"?`
`13:58.31`	`phearless`	`?`
`14:02.00`	`phearless`	`oh now it works ! more and more weird`
`14:02.03`	`hmmhesays`	`music`
`14:02.36`	`[TK]D-Fender`	`vit : Did you install the asterisk-addons package which provides support for MP3?`
`14:02.39`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.215.82)`
`14:02.51`	`vlt`	`When I edit musiconhold.conf to match the new syntax then "started/stopped music on hold" does't even appear on CLI. Using the old "default => ..." syntax shows the line on CLI but I can't hear anything.`
`14:03.14`	`vlt`	`[TK]D-Fender: I'm on ubuntu and there's no pkg *-addons`
`14:03.23`	`vlt`	`[TK]D-Fender: Where can I get that?`
`14:03.26`	`GerbilWrk`	`hmmhesays, what calling card app?`
`14:03.56`	`[TK]D-Fender`	`vit : you should always be compiling from source, and I don't know the details of whre you'd find a compatible packaged form...`
`14:04.17`	`hmmhesays`	`GerbilWrk: I whipped up a basic calling card app using no AGI`
`14:04.31`	`[TK]D-Fender`	`vit :You can get it off Digium's website and try compiling it in.`
`14:04.40`	`GerbilWrk`	`hmmhesays: nice, would love to take a look at that`
`14:04.44`	`[TK]D-Fender`	`vit : Or FTP rather.`
`14:04.48`	`hmmhesays`	`http://www.thelostpacket.org/tricks.php`
`14:04.54`	`hmmhesays`	`GerbilWrk: there you go`
`14:05.09`	`*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)`
`14:05.15`	`vlt`	`[TK]D-Fender: What if I converted the .mp3 to .gsm?`
`14:05.21`	`*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)`
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`14:07.56`	`vlt`	`I have converted .mp3 to .gsm with sox. How must I edit "quietmp3:/"?`
`14:08.07`	`vlt`	`... in musiconhold.conf`
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`14:12.40`	`*** mode/#asterisk [+o anthm] by ChanServ`
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`14:21.01`	`Greek-Boy`	`hello`
`14:21.08`	`Greek-Boy`	`ls /dev/zap`
`14:21.27`	`Greek-Boy`	`Notice: Configuration file is /etc/zaptel.conf`
`14:21.28`	`Greek-Boy`	`line 142: Unable to open master device '/dev/zap/ctl'`
`14:21.35`	`Greek-Boy`	`can someone help me with that?`
`14:21.46`	`RoyK`	`~docs`
`14:21.47`	`jbot`	`docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`14:21.48`	`RoyK`	`~book`
`14:21.50`	`jbot`	`i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`14:21.50`	`Greek-Boy`	`I just installed a TDM2404E card`
`14:21.50`	`RoyK`	`~rtfm`
`14:21.52`	`jbot`	`somebody said rtfm was Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM.`
`14:23.29`	`hmmhesays`	`bwhaha`
`14:25.16`	sp0n9e`	`what would cause libpri not to be detected even though it's installed?`
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`14:29.44`	`*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)`
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`14:31.24`	`[TK]D-Fender`	`vlt: If you swith to GSM you need to use Native MoH. Quietmp3 uses MPG123 which you don't want anyways.`
`14:32.29`	`hmmhesays`	`ok how do I prompt for user input from a bash script?`
`14:32.46`	`*** join/#asterisk parag_ast (n=root@dxb-b111124.alshamil.net.ae)`
`14:33.17`	`parag_ast`	`Can anybody let me know I want that asterisk should not send Registration again and again`
`14:33.25`	`sevard`	`hmmhesays: hahaha, really?`
`14:33.32`	`sevard`	`hmmhesays: man bash`
`14:33.33`	`sevard`	`:)`
`14:33.36`	`sevard`	`RTFM NUB`
`14:33.36`	`parag_ast`	`it sends registration msgs again and again`
`14:33.37`	`hmmhesays`	`sevard: just tell me`
`14:33.45`	`sevard`	`hmmhesays: THE TABLES HAVE TURNE.`
`14:33.47`	`sevard`	`+D`
`14:33.49`	`hmmhesays`	`ha`
`14:33.51`	`sevard`	`heh`
`14:34.01`	`hmmhesays`	`so don't be a little bItch about it and tell me`
`14:34.24`	`coppice`	`ok how do I bash a user to prompt for script input? :-)`
`14:34.25`	`sevard`	`b ahwhahahahah this is awesome`
`14:34.31`	`sevard`	`coppice: no problem`
`14:34.32`	`hmmhesays`	`fine`
`14:34.34`	`hmmhesays`	`fine`
`14:34.40`	`hmmhesays`	`I see how ya'll are`
`14:34.48`	`sevard`	`say you'll love me to the end of time`
`14:34.52`	`parag_ast`	`why asterisk sends registration info again and again`
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`14:35.27`	`hmmhesays`	`naw I figured it out`
`14:35.36`	`sevard`	`hmmhesays: 'read' :)`
`14:36.08`	`*** join/#asterisk Avalone (n=Avalone_@83.239.191.33)`
`14:36.36`	`sevard`	`echo -n "How many times is hmmhesays a nub? "`
`14:36.38`	`sevard`	`read nub`
`14:36.58`	`sevard`	`echo $nub`
`14:37.02`	`sevard`	`400000000000`
`14:38.24`	`parag_ast`	`when i try to register asterisk sends registration info again and again`
`14:38.41`	`sevard`	`a neat tidbit about read is you can read only the digits that matter to you, like echo -n "gender: " ; read -n 1 gender`
`14:40.33`	`hmmhesays`	`yeah sevard too late`
`14:40.50`	`hmmhesays`	`I'm writing an install script for vicidial`
`14:40.53`	`hmmhesays`	`because it is a pain in the ass`
`14:40.53`	`sevard`	`maybe if you RTFM tardface`
`14:41.13`	`hmmhesays`	`stuff it`
`14:41.27`	`RoyK`	`echo -n "hmmhesays? " ; read hmmhesays ; echo $hmmhesays`
`14:41.27`	`RoyK`	`hmmhesays? rtfm`
`14:41.27`	`RoyK`	`rtfm`
`14:41.42`	`hmmhesays`	`you know for all the advice I give out in here, you guys suck`
`14:41.47`	`RoyK`	`rotfl`
`14:42.24`	`sevard`	`bawhhahaha`
`14:42.57`	`sevard`	`hmmhesays: do you need help with any more 'basic bash commands' ?`
`14:43.05`	`hmmhesays`	`no sevard thanks`
`14:43.24`	`sevard`	`no problem, just though i'd ask`
`14:44.30`	`sevard`	`hmmhesays!`
`14:44.36`	`sevard`	`guess what comes tomorrow!`
`14:44.54`	`hmmhesays`	`your real doll?`
`14:45.22`	`sevard`	`EVEN BETTER`
`14:45.23`	`Sonderblade`	`in the permit variable in the manager.conf conf file, can you set the net mask so that all from 192.168.. can access AMI but noone else?`
`14:45.36`	`sevard`	`(btw if i had a real doll i'd model it after you)`
`14:46.03`	`sevard`	`i'm getting my 37" LCD HDTV Widescreen ViewSonic`
`14:46.43`	`sevard`	`there's always been a howto, 'man bash'`
`14:47.33`	`Nugget`	`or 'man tcsh' if you're one of those freaky bsd people. :)`
`14:47.42`	`sevard`	`icky.`
`14:47.56`	`hmmhesays`	`RoyK: it is over now, you are not fashionably late`
`14:47.57`	`Nugget`	`you say to-may-toe.`
`14:48.07`	`hmmhesays`	`you are my retarded 1/2 cousin late`
`14:48.29`	`sevard`	`I see where the hmmhesays genepool comes into play`
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`14:49.07`	`RoyK`	`~hmmhesays`
`14:49.08`	`jbot`	`methinks hmmhesays is not really here...`
`14:49.15`	`RoyK`	`good`
`14:49.34`	`RoyK`	~nickometer sp0n9e`
`14:50.10`	`sevard`	`hmmhesays: did you see this? http://www.asteriskvoipnews.com/asterisk_development/twincities_asterisk_users_group_saturday_august_12th_1130am.html`
`14:50.44`	`hmmhesays`	`no`
`14:50.51`	`sevard`	`how strange is that`
`14:50.52`	`*** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com)`
`14:51.53`	`sevard`	`apparently been going on a while`
`14:52.40`	`hmmhesays`	`cool`
`14:52.44`	`hmmhesays`	`I wish I could go`
`14:52.44`	`rowter`	`~nickometer rowter`
`14:52.56`	`hmmhesays`	`~nickometer hmmhesays`
`14:53.02`	`hmmhesays`	`coo`
`14:53.24`	`sevard`	`hmmhesays: want to go ?`
`14:53.36`	`hmmhesays`	`I don't drive`
`14:53.36`	`*** join/#asterisk chorlick (n=chorlick@gateway.digium.com)`
`14:53.45`	`sevard`	`disability?`
`14:53.56`	`hmmhesays`	`legalality`
`14:54.07`	`sevard`	`whiskey plates?`
`14:54.10`	`*** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu)`
`14:54.21`	`hmmhesays`	`for the next 9 months`
`14:54.31`	`sevard`	`wtf did you do`
`14:54.38`	`jake1932`	`2 much jd`
`14:54.39`	`*** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu)`
`14:54.46`	`hmmhesays`	`gf wrecked my car`
`14:54.52`	`*** part/#asterisk parag_ast (n=root@dxb-b111124.alshamil.net.ae)`
`14:55.04`	`hmmhesays`	`grabbed the fscking steering wheel while she was plastered`
`14:55.05`	`sevard`	`is she still your gf then`
`14:55.10`	`hmmhesays`	`heh, no`
`14:55.14`	`sevard`	`very good.`
`14:55.18`	`sevard`	`i could pick your ass up`
`14:55.22`	`quid246`	`what kind of ride?`
`14:55.27`	`quid246`	`did you have`
`14:55.34`	`sevard`	`heh, did`
`14:56.14`	`hmmhesays`	`it was an old pos`
`14:56.31`	`quid246`	`still it was your wheels`
`14:56.34`	`hmmhesays`	`ye`
`14:56.47`	`hmmhesays`	`unfortunately I was slightly over the limit also`
`14:56.57`	`quid246`	`sucks`
`14:57.27`	`quid246`	`I dont touch the stuff anymore if I drive... I'm a good driver, but fall asleep easy`
`14:57.30`	`sevard`	`so you ruined your car and now you have a whiskey license`
`14:57.47`	`hmmhesays`	`no`
`14:57.54`	`sevard`	`quid246: same, even if I had just one beer i'll wait.`
`14:58.22`	`hmmhesays`	`sevard: you are young`
`14:58.53`	`sevard`	`i've always been a heavyweight but it's not worth the risk a cop smelling your breath.`
`14:59.03`	`hmmhesays`	`not if you're under 21`
`14:59.07`	`sevard`	`hmmhesays: you are retarded`
`14:59.20`	`sevard`	`1/2 cousin`
`14:59.22`	`sevard`	`abwhaha`
`14:59.36`	`sevard`	`so, want to go?`
`14:59.43`	`hmmhesays`	`when is it?`
`15:00.01`	`sevard`	`closed the page, can't recall`
`15:00.07`	`hmmhesays`	`yeah so did I`
`15:00.14`	`quid246`	`the biggest contradiction I've seen is at the local police association's social club... parking lot littered with beer bottles`
`15:00.27`	`sevard`	`quid246: take pictures`
`15:00.32`	`quid246`	`and we're talking a police forc for 1 million people`
`15:00.47`	`quid246`	`notsome small town`
`15:01.06`	`mut`	`i'de rather have cops that drink a bit than some uppity people out to prove to the world they follow the law to the core`
`15:01.17`	`*** join/#asterisk bXi (i=bluepunk@irssi.co.uk)`
`15:01.23`	`bXi`	`hi`
`15:01.26`	`sevard`	`HI`
`15:01.30`	`sevard`	`HOWRU`
`15:01.34`	`bXi`	`if i change something in my sip.conf do i need to restart?`
`15:01.39`	`*** join/#asterisk marv[work] (n=timr@64.89.118.139)`
`15:01.41`	`quid246`	`no just reload`
`15:01.41`	`sevard`	`sip reload`
`15:01.58`	`bXi`	`sip reload?`
`15:02.02`	`bXi`	`where do i do that?`
`15:02.04`	`sevard`	`cli`
`15:02.25`	`coppice`	`mut: sounds like a good plot for a play. i'll call it "measure for measure". oh, damn, its been taken :-)`
`15:02.39`	`bXi`	`is there a way to do that externally`
`15:02.46`	`bXi`	`i'm trying to automate asterisk usage`
`15:02.50`	`bXi`	`trough a webinterface`
`15:02.53`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`15:02.59`	`mut`	`me no likey plays`
`15:03.00`	`sevard`	`hmmhesays: want to write an ncurses front end to cli with me?`
`15:03.15`	`mut`	`turn it into a cool scifi with special fx`
`15:03.22`	`sevard`	`bXi: probably by the manager interface`
`15:03.45`	`vlt`	`Hello. I downloaded the latest src from asterisk.org. When I try to compile it on Debian Sarge an error "termcap support not found" appears. What to do now?`
`15:04.33`	`hmmhesays`	`sevard no`
`15:04.40`	`sevard`	`hmmhesays: why not?`
`15:04.57`	`*** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it)`
`15:05.58`	`quid246`	`vlt: sounds like you are missing a package`
`15:06.43`	`quid246`	`bxi: you could run it as a cron job`
`15:07.11`	`jake1932`	`vlt: what distro?`
`15:07.33`	`jake1932`	`vlt: apt-get libtermcap`
`15:07.46`	`jake1932`	`vlt: there are a few packages that are needed`
`15:08.19`	`quid246`	`anybody here ran Asterisk in 64-bit mode?`
`15:08.37`	`file`	`quid246: I run it on a 64-bit CPU, does that count? ;)`
`15:08.47`	`coppice`	`it runs fine on an AMD64 machine`
`15:08.53`	`vlt`	`jake1932: Thank you, I'll try that.`
`15:08.57`	`neo`	`but compiled in 32 bits :)`
`15:09.39`	`coppice`	`it runs fine on an AMD64 machine, running FC5 64 bit version, and compiled as 64 bit code`
`15:10.33`	`quid246`	`I dunno..I'm a newb at 32vs64 bit... but from what I read, 64 Bit is heck of alot more "iffy"`
`15:10.49`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`15:11.37`	`coppice`	`where did you read that?`
`15:11.52`	`sevard`	`<PROTECTED>`
`15:12.39`	`quid246`	`coppice: haven't read in one sport particular, but general consensus`
`15:12.50`	`vlt`	`jake1932: There's no pkg called libtermcap here (Debian Sarge).`
`15:13.00`	`quid246`	`like packages aren't all that available`
`15:13.46`	`coppice`	`the general consensus of many heavy users is its just fine. don'e try it with pentiums earlier than core2, though. their 64 bit performance is terrible`
`15:13.52`	`[TK]D-Fender`	`vlt: there is, could be thats not the perfect spelling of it. I think its something like lib-termcap.... look it up.`
`15:14.31`	`quid246`	`yeah I am trying to decide on a dual-xeon or dual-opteron setup and 32 vs 64 bit for *.`
`15:15.27`	`*** join/#asterisk Mike (n=mike@201.112.50.158)`
`15:15.28`	`*** join/#asterisk dotbit (n=dotbit@home-020108.b.astral.ro)`
`15:15.30`	`ZeNN`	`compiling zaptel 1.2.7 with make linux26 gives me the following error:`
`15:15.30`	`ZeNN`	`collect2: ld returned 1 exit status`
`15:15.31`	`ZeNN`	`make: *** [ztcfg] Error 1`
`15:15.33`	`ZeNN`	`what to do ?`
`15:15.33`	`vlt`	`jake1932, [TK]D-Fender: apt-cache search termcap only returns one termcap pkg: termcap-compat. Do I need this?`
`15:15.42`	`sevard`	`I wish gnome-terminal worked more like putty`
`15:15.48`	`sevard`	`I haven't found a terminal for X that I actually like`
`15:15.54`	`coppice`	`well, on an opteron * runs faster in 64 bit. on a pentium before core 2 it was a lot slower. dunno about core2, but it looks like they improved the 64 bit performance somewhat`
`15:16.27`	`sevard`	`I like putty's resize text when fullscreen'd option, gnome-terminal will go full screen but it will simply increase the size of the terminal without decorations`
`15:16.38`	`neo`	`sevard: xterm ?`
`15:16.40`	`neo`	`aterm ?`
`15:16.51`	`*** join/#asterisk FBJimmy (n=jamest@131.202.90.23)`
`15:16.59`	`neo`	`(aterm allows you to make windows transparent)`
`15:17.00`	`sevard`	`xterm won't go fullscreen without a patch and regardless won't resize the text to retain the same aspect ratio`
`15:17.11`	`*** join/#asterisk funxion (n=nunya@63.214.236.169)`
`15:17.14`	`sevard`	`i'm not sure about aterm, but i don't care about window transparancy, it's distracting`
`15:17.32`	`sevard`	`i want white on black and resize text when fullscreend`
`15:17.50`	`quid246`	`coppice: yeah, benchmarks on * are dodgy at best...`
`15:17.51`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`15:18.15`	`sevard`	`putty is a damned good terminal, just not for linux :(`
`15:18.25`	`quid246`	`surprised no one has standardized a test to benchmark diff. processor performance, perhaps there isn't a big enough audience yet`
`15:18.39`	`vlt`	`I just installe dthe pkg "termcap-compat" but the error "termcap support not found" still stops compiling asterisk. Does anyone know the correct pkg name on Debian?`
`15:18.40`	`sevard`	`quid246: sipsak?`
`15:19.11`	`tzafrir`	`vlt: you probably miss a -dev or -dev package`
`15:19.26`	`tzafrir`	`vlt, which distro?`
`15:19.35`	`quid246`	`sevard: Or SIPP... but nobody has attempted to really put together a good central resouce of that data`
`15:19.36`	`vlt`	`tzafrir: Debian Sarge`
`15:19.38`	`tzafrir`	`vlt: apt-get build-dep asterisk`
`15:20.09`	`vlt`	`tzafrir: What does it?`
`15:20.42`	`tzafrir`	`asterisk is an existing debian package. If you have a deb-src source of standard debian, this will give you its build dependencies`
`15:21.04`	`tzafrir`	`Which are basically what you need to build the asterisk tarball`
`15:21.12`	`tzafrir`	`not to mention:`
`15:21.22`	`tzafrir`	`apt-get install build-essential`
`15:21.47`	`vlt`	`tzafrir: Ok, I'll try. Thank you.`
`15:22.06`	`*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)`
`15:22.13`	`tzafrir`	`sevard, putty for linux is also a pretty nice terminal. But xterm/ssh is better`
`15:22.30`	`variable_office`	`i am looking for a decent speakerphone + multiline phone(at least 4 lines) for $200 or less, whats the good word on some phones that would fit this description?`
`15:22.55`	`*** join/#asterisk jarg (n=jarg@200.56.225.61)`
`15:23.56`	`tzafrir`	`sevard, what's the problem with full-screen xterm? Any x program would go full screen if the window manager would tell it so`
`15:24.00`	`vlt`	`tzafrir: apt-get build-dep asterisk --> E: You have to put some "source"-URIs to sources.list ...`
`15:24.14`	`vlt`	`tzafrir: Mmh.`
`15:25.21`	`tzafrir`	`vlt, you probably have one remmed-out. take a look at your sources.list`
`15:25.45`	`*** join/#asterisk cbrake (n=cbrake@cpe-65-25-51-117.neo.res.rr.com)`
`15:25.46`	`vlt`	`tzafrir: And look for ...?`
`15:26.01`	`tzafrir`	`grep deb-src /etc/apt/sources.list`
`15:26.49`	`vlt`	`tzafrir: There's no deb-src line in sources.list`
`15:27.49`	`variable_office`	`or i guess i should say, what all phones do you use?`
`15:28.06`	`*** join/#asterisk eVoke (n=vircuser@62.96.103.66)`
`15:29.25`	`vlt`	`tzafrir: Ok, I added the same line as for deb again for deb-src in sources.list ...`
`15:29.38`	`tzafrir`	`should work`
`15:29.44`	`tzafrir`	`run apt-get update`
`15:30.02`	`vlt`	`tzafrir: Yes, downloading now ... ;-)`
`15:30.22`	`vlt`	`tzafrir: (apt-get build-dep asterisk)`
`15:30.28`	`*** join/#asterisk ajohnson (i=ajohnson@ip68-104-215-193.ph.ph.cox.net)`
`15:30.43`	`ajohnson`	`where is the asterisk database file kept?`
`15:30.54`	`ajohnson`	`On a typical install`
`15:31.02`	`tzafrir`	`<PROTECTED>`
`15:31.06`	`ajohnson`	`thanks`
`15:31.26`	`ajohnson`	`and I can remove that file if it is corrupted and Asterisk will rebuild it?`
`15:31.42`	`vlt`	`tzafrir: Now compiling ...! :-)`
`15:31.56`	`tzafrir`	`what version?`
`15:32.00`	`ajohnson`	`1.2`
`15:32.31`	`*** join/#asterisk momelod (n=momelod@Toronto-HSE-ppp3892982.sympatico.ca)`
`15:32.41`	`momelod`	`greetings channel`
`15:33.36`	`eVoke`	`hi, short question : Does it gives a possibility to access asterisk by visual basic?`
`15:33.49`	`*** join/#asterisk toerkeium (i=oo@201.216.206.221)`
`15:33.55`	`*** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)`
`15:34.42`	`tzafrir`	`eVoke, read about the asterisk manager interface`
`15:35.45`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`15:35.45`	`*** mode/#asterisk [+o mog] by ChanServ`
`15:36.05`	`momelod`	`i want to record all incomming calls, but i would like to seperate the recorded calls in directories so that my agents can be given access to the dir holding thier calls.. so in my extensions.conf i tried something like Monitor(wav,${EXTEN}/{$EXTEN}=${TIMESTAMP}) ..but since i use macros for all the extensions, the ${EXTEN} var hold 's' instead of say '242' ..how can i preserve the dialed extention, or is there another variable i sh`
`15:36.16`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co)`
`15:36.21`	`eVoke`	`i search it by google, without success, do you have a good link for me?`
`15:36.41`	`vlt`	`I started compiling asterisk but get error messages: chan_zap.c: union has no member named digit ...`
`15:37.19`	`vlt`	`It's the current 1.2.10 src from asterisk.org`
`15:37.34`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`15:40.26`	`hmmhesays`	`bah vicidial has a lot of dependencies`
`15:40.46`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`15:42.10`	`sevard`	`is it even worth it?`
`15:42.17`	`vlt`	I did `apt-get build-dep asterisk`before.
`15:42.31`	`*** join/#asterisk DeVilSoulBlacK (n=devilsou@gea-gye-internet.telconet.net)`
`15:45.55`	`*** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18)`
`15:46.58`	`funxion`	`I'm getting WARNING[23470]: file.c:587 ast_readaudio_callback: Failed to write frame could this have something to do with not having zaptel drivers laoded?`
`15:52.44`	`*** join/#asterisk _007 (n=brian@bhlkwihed01-pool0-a187.bhlkwi.tds.net)`
`15:52.44`	`*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)`
`15:53.12`	`*** join/#asterisk luckasrg (n=root@204.97.196.20)`
`15:53.13`	sp0n9e`	`what does the smdi module do?`
`15:53.13`	`mog`	`allows asterisk to talk to old voicemail servers`
`15:53.13`	`mog`	`over serial`
`15:53.13`	sp0n9e`	`doesn't sound important for me`
`15:53.14`	`mog`	`it is if you own one`
`15:53.14`	`mog`	`most modules though if you dont use them only take up memory`
`15:53.14`	sp0n9e`	`hence the qualifying statement "for me"`
`15:53.14`	`mog`	`so it usually doesnt matter`
`15:53.14`	`mog`	`if its loaded or unloaded`
`15:53.14`	sp0n9e`	`well, i just want the console to start up clean`
`15:53.41`	`mog`	`then do asterisk -c`
`15:53.48`	`mog`	`its pretty clean ^_^`
`15:53.51`	`mog`	`or do asterisk`
`15:53.53`	`mog`	`then connect`
`15:53.59`	`funxion`	`I'm getting WARNING[23470]: file.c:587 ast_readaudio_callback: Failed to write frame could this have something to do with not having zaptel drivers laoded??`
`15:54.33`	sp0n9e`	`yeah, noload'ing smdi freaks out chan_zap.so`
`15:54.36`	`luckasrg`	`is there anything i can read that can help me whit this`
`15:57.16`	`luckasrg`	`q`
`15:57.34`	`*** join/#asterisk SplasPood (n=jwb@206.252.198.101)`
`15:58.45`	`luckasrg`	`i have been installing a asterisk box and everithing works fine but the tranlation of sip to h323 is not working for me what can i do do to get it to work`
`15:59.12`	`*** part/#asterisk eVoke (n=vircuser@62.96.103.66)`
`15:59.56`	`luckasrg`	`or if there is anythig i can read that can help me to get this to work`
`16:02.33`	`*** join/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com)`
`16:05.04`	`*** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com)`
`16:06.26`	`*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)`
`16:06.37`	`*** join/#asterisk Greek-B0y (n=Greek-Bo@196.46.109.164)`
`16:08.23`	`phearless`	`hello folks ! :-)`
`16:09.34`	`intralanman`	`hey phearless`
`16:11.07`	`*** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it)`
`16:12.43`	`*** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66)`
`16:15.27`	`dotbit`	`hello ^^`
`16:15.49`	`*** join/#asterisk easydone (n=notdone@eksel.demon.nl)`
`16:17.03`	`phearless`	`anybody know how to "debug" my problem ? when I do an outbound call, I got nothing but a weird sound, like crrr crrr crrr crrr ... I do now know what I should check !`
`16:18.21`	`jbalcomb`	`phearless: you should check your logs`
`16:18.33`	`dotbit`	`i am new with asterisk but have you tried "debug 10"?`
`16:18.43`	`dotbit`	`solves a ton of problems for me ^^`
`16:20.01`	`phearless`	`http://rafb.net/paste/results/EsoQLE64.html`
`16:20.04`	`phearless`	`ok so here is the log`
`16:20.32`	`phearless`	`i do not really know what all that stuff means`
`16:20.42`	`jbalcomb`	`"phearless: /you/ should check your logs"`
`16:21.26`	`*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`16:21.42`	`phearless`	`..`
`16:21.50`	`jtexter3`	`Anybody here have a setup where automon recordings are sent to the persons voicemail box?`
`16:22.23`	`syzygyBSD`	`people still use voicemail boxes?`
`16:22.48`	`jtexter3`	`voicemail folder`
`16:23.09`	`syzygyBSD`	`ya, I (and all my clients) just have it all emailed`
`16:23.47`	`jtexter3`	`The recordings made when pressing *1 during a call? I assume you modifed app_monitor to accomplish that?`
`16:23.53`	`jbalcomb`	`syzygyBSD: are you running them through text to speach? i have 150 users who don't have speakers.`
`16:24.21`	`syzygyBSD`	`hmm, ya, all of them have speakers, I dont' trust text to speech`
`16:24.49`	`sevard`	`Alright, here's a brain teaser`
`16:24.55`	`*** join/#asterisk Kylun (i=StarHawk@adsl-068-157-090-228.sip.bct.bellsouth.net)`
`16:25.05`	`syzygyBSD`	`ya, I know a lot of people still use call in voicemail, Any that have tried it prefer email though`
`16:25.07`	`sevard`	`I have a PRI card in my * box, when somebody calls out I get a 'double ring'`
`16:25.30`	`syzygyBSD`	`sevard: ya, i get that too, just haven't worried about fixing it yet`
`16:25.38`	`jbalcomb`	`syzygyBSD: i'm not sure i see the benefit of adding 150 pieces of equipment at $1500+ os people dont have to press <msg>`
`16:26.01`	`syzygyBSD`	`the problem is your Dial command sends back the ring, as well as your provider`
`16:26.02`	`Kylun`	`Can anyone suggest a good phone (2-4 line preferred) for a brand new asterisk install, in a business environment of about 4 people`
`16:26.10`	`syzygyBSD`	`there should be an option to dial to turn it off`
`16:26.32`	`Greek-B0y`	`Kylun go for Cisco 7916G`
`16:26.33`	`jbalcomb`	`Kylun: polycom ip 501 is good`
`16:26.38`	`momelod`	`Kylun: i like the polycom ip501`
`16:26.53`	`momelod`	`i want to record all incomming calls, but i would like to seperate the recorded calls in directories so that my agents can be given access to the dir holding thier calls.. so in my extensions.conf i tried something like Monitor(wav,${EXTEN}/{$EXTEN}=${TIMESTAMP}) ..but since i use macros for all the extensions, the ${EXTEN} var hold 's' instead of say '242' ..how can i preserve the dialed extention, or is there another variable i sh`
`16:26.55`	`*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)`
`16:27.03`	`Kylun`	`how much those phones run, anyone have a link? Also, do they do intercom? that would be a very useful feature.`
`16:27.08`	`jbalcomb`	`Kyler: do NOT get Grandstream gxp-2000s`
`16:27.27`	`momelod`	`Kylun: about $250CND`
`16:27.33`	`momelod`	`yes they have intercome`
`16:27.34`	`vlt`	`Now I'm testing queuing calls on another machine v1.2.7. Music on hold works with native .gsm files but now no announcment about the queue position is made. Where to set that?`
`16:27.35`	`jbalcomb`	`kylun: $240ish`
`16:27.58`	`jbalcomb`	`vlt: queues.conf`
`16:28.25`	`jbalcomb`	`vlt: announce-holdtime = no`
`16:28.45`	`jbalcomb`	`vlt: periodic-announce-frequency = 0`
`16:29.11`	`*** join/#asterisk Givemelove (n=non@208.57.229.162)`
`16:29.29`	`*** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw)`
`16:30.34`	`*** join/#asterisk fiber0pti (n=John@207.114.199.107)`
`16:30.46`	`fiber0pti`	`what is a "." perform in an extension? i.e _1X.`
`16:30.51`	`eKo1`	`I'm have 2 E1 PRIs. The first E1 works like a charm. On the second E1, only the channels lower than the signaling channel (47) work. If I make a call with a channel number higher than 47, the call goes through but I get no audio.`
`16:30.59`	`Greek-B0y`	`whats the best way to install zaptel on debian? from source or apt-get?`
`16:31.03`	`eKo1`	`fiber0pti: matches anything thereafter`
`16:31.13`	`jbalcomb`	`fiber0pti: it may match any amount of any character`
`16:31.17`	`vlt`	`fiber0pti: every digit after`
`16:31.23`	`eKo1`	`Greek-B0y: I always go with source.`
`16:31.32`	`jbalcomb`	`Greek-B0y: ditto`
`16:31.35`	`momelod`	`Greek-B0y: go with source, cause then u can recompile if u want to make changes`
`16:31.45`	`fiber0pti`	`so it's expecting ATLEAST a "1" and then any number PLUS is will allow for any numbers thereafter?`
`16:32.02`	`[TK]D-Fender`	`Kylun: Avoid Cisco.... Polycom is considerably cheaper and has better SIP support.`
`16:32.35`	`eKo1`	`fiber0pti: no, if you dial just 1, it won't match.`
`16:32.36`	`Kylun`	`I'll check into it, does it have good intercom support?`
`16:32.37`	`Qwell`	`[TK]D-Fender: But, polycom doesn't do skinny ;)`
`16:32.52`	`eKo1`	`fiber0pti: it will match any number greater than 1 that starts with 1 though`
`16:32.58`	`[TK]D-Fender`	`Qwell: You say that as though it were a BAD thing;)`
`16:33.06`	`file`	`ooooooooh`
`16:33.10`	`jbalcomb`	`fiber0ptil: something like that yes, there may be something there that it /requires/ atleast the 1 and another character`
`16:33.15`	`syzygyBSD`	`jbalcomb: you are missing the added benifit of loss of productivity for them having a computer in front of them too`
`16:33.20`	`[TK]D-Fender`	`Kylun: No different than theother options really.`
`16:33.26`	`[TK]D-Fender`	`jbalcomb: PM`
`16:33.30`	`Qwell`	`[TK]D-Fender: SIP is going to be deprecated in 1.4`
`16:33.32`	`syzygyBSD`	`1500+ starts looking cheap`
`16:33.34`	`Qwell`	`</troll>`
`16:33.38`	`[TK]D-Fender`	`Qwell: LOL!`
`16:33.38`	`momelod`	`k, i use the standard macro for my internal extensions.. i want to record incomming calls to /var/spool/asterisk/monitor/${EXTEN}/blah-${TIMESTAMP}.wav. how can i do that?`
`16:33.45`	`file`	`yeah, no more SIP`
`16:34.03`	`jbalcomb`	`syzygyBSD: ummm.. it ain't like they have to go down to the basement to get to thier phone fella`
`16:34.08`	`[TK]D-Fender`	`file: Replaced by the new and vastly superior chan_gulp? :)`
`16:34.16`	`Qwell`	`momelod: just set the right variable.. MONITOR_FILE or whatever it is`
`16:34.18`	`file`	`[TK]D-Fender: :D`
`16:34.22`	`Qwell`	`(That isn't actually it)`
`16:34.28`	`Qwell`	`chan_biggulp?`
`16:34.42`	`syzygyBSD`	`jbalcomb: I was saying if you got them all a computer (hence the 1500 price tag right?)`
`16:35.34`	`momelod`	`Qwell: yeah, thats my question.. because im using a macro for extension the ${EXTEN} var is 's' but i would like it to be the exten dialed, eg 222. how do i preserve the dialed entension number?`
`16:35.45`	`Qwell`	`${MACRO_EXTEN{`
`16:35.46`	`Qwell`	`}`
`16:35.53`	`momelod`	`sweet!!`
`16:35.54`	`momelod`	`ty`
`16:36.01`	`*** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net)`
`16:36.04`	`Qwell`	`${MACRO_CONTEXT} too`
`16:36.20`	`[TK]D-Fender`	`momelod: Pass the exten as a parameter to your macro`
`16:37.08`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`16:37.13`	`momelod`	`k, now if the call goes into a queue, how do i extract the final extension?`
`16:37.37`	`momelod`	`..where the call was answered`
`16:37.37`	`Qwell`	`So..`
`16:37.48`	`*** join/#asterisk Assid (i=assid@203.115.83.213)`
`16:37.53`	`*** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`16:38.27`	`[TK]D-Fender`	`momelod: I'd avoid using MACROEXTEN and opt for passing it as a parm. Much more flixble and you may want to call your macro OTHER than just from what my a user perspective nuber.`
`16:38.29`	`Qwell`	`Just so you all know...`
`16:38.46`	`file`	`yes`
`16:38.57`	`file`	`Qwell is now on the Digium team everyone`
`16:39.00`	`Qwell[Digium]`	`:D`
`16:39.06`	`Assid`	`he is???`
`16:39.11`	`Qwell[Digium]`	`He is`
`16:39.14`	`Assid`	`when did you join bro?`
`16:39.18`	`Qwell[Digium]`	`like...an hour ago`
`16:39.37`	`Assid`	`nice`
`16:41.03`	`syzygyBSD`	`is there a silence MoH option?`
`16:41.20`	`[TK]D-Fender`	`syzygyBSD: Change the class to one that won't have music to play`
`16:41.42`	`FBJimmy`	`Anyone have any experience with filing bug reports with Digium? I filed one on Thursday (#7706), but it doesn't look like any action has been taken on it. Is there anything I can do to get it moved along?`
`16:41.46`	`Assid`	`brb`
`16:42.04`	`syzygyBSD`	`ok, that is a way to hack what I want, but is there another way to make dial not ring to the callee?`
`16:42.28`	`file`	`FBJimmy: you have to wait :)`
`16:42.38`	`rpm`	`syzygyBSD: you mean page?`
`16:43.46`	`syzygyBSD`	`page? dont' think so, no. I don't want the ring sound from the dial application to be passed to the callee. There is already one sent from my provider and I don't want a duplicate sound passed to the callee`
`16:43.59`	`*** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18)`
`16:44.48`	`hmmhesays`	`so my vicidial installer kicks ass so far`
`16:44.49`	`eKo1`	`Will the telco. send me calls on zap channels that I've destroyed?`
`16:44.52`	`hmmhesays`	`booyeah`
`16:44.55`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`16:45.08`	`TripleFFFF`	`whats a cheap but good ata with 1 pstn port ?`
`16:45.16`	`Qwell[Digium]`	`eKo1: define "destroyed"`
`16:45.22`	`hmmhesays`	`mediatrix 2102`
`16:45.26`	`Qwell[Digium]`	`I imagine that you you ripped the cable out, it probably won't`
`16:45.30`	`Qwell[Digium]`	`s/you/if/`
`16:45.31`	`hmmhesays`	`midrange price, fantastic functionality`
`16:45.42`	`cytrak`	`anyone here has a lagacy PBX attached to asterisk ? I have PSTN -> asterisk -> LPBX`
`16:45.49`	`hmmhesays`	`yep`
`16:45.53`	`TripleFFFF`	`basicalyy trying to patch a ata to nortel meridian`
`16:46.00`	`hmmhesays`	`do it every day to do least cost routing cytrak`
`16:46.46`	`TripleFFFF`	`any way ?`
`16:47.16`	`*** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek)`
`16:47.18`	`cytrak`	`hmmhesays: I have a siemens pbx with siemens phones attached to it .... one thing that I still haven't been able to figure out is how to make asterisk voicemail work for those internal phones`
`16:47.52`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co)`
`16:47.53`	`*** join/#asterisk sugardave (n=sugardav@cpe-70-112-127-158.austin.res.rr.com)`
`16:47.55`	`TripleFFFF`	`hmmhesays mediatrix is like.. not too promising for me.. any other ?`
`16:47.57`	`TripleFFFF`	`spa3000 ?`
`16:48.34`	`cytrak`	`hmmhesays: I checked the pbx manager and I know that a PBX extension can be setup as a voicemail extension which passes the caller ID to the voicemail app , but asterisk can't read that`
`16:48.52`	`TripleFFFF`	`or any idea ona sterisk => nortel merdiains ?`
`16:49.03`	`cytrak`	`the other voicemail app (which of course is comparable with the seimens box) could`
`16:49.54`	`*** part/#asterisk vlt (n=dm@p54B34337.dip0.t-ipconnect.de)`
`16:50.59`	sp0n9e`	`i used d to turn on debugging, but /var/log/asterisk/debug doesn't exist.`
`16:51.25`	`syzygyBSD`	`cytrak: asterisk can't read the callerid from your pbx system?`
`16:51.41`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`16:52.12`	`cytrak`	`syzygyBSD: well it's not like you are thinking`
`16:53.27`	`cytrak`	`syzygyBSD: I used routing on the PBX to do the following`
`16:55.53`	`*** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net)`
`16:56.08`	`cytrak`	`person on phone 1444 calls phone 1555, so PBX calls the number if no one answers it asterisk voicemail system should pick it up but when it does it sees the PBX calling in and not the 1444 user`
`16:56.30`	`cytrak`	`its wierd to explain`
`16:56.41`	`Qwell[Digium]`	`cytrak: How are you calling Voicemail()?`
`16:56.57`	`syzygyBSD`	`ok, I see what I think the order is...`
`16:57.07`	`CunningPike`	`TripleFFFF: We have a Meridian - what are you trying to do?`
`16:57.09`	`*** join/#asterisk SplasPood (n=jwb@206.252.198.101)`
`16:57.30`	`syzygyBSD`	`when the PBX dials asterisk to go to voicemail, how is callerid set`
`16:57.53`	`cytrak`	`I guess I need to check on the seimens manager again to explain better to you guys , I haven't checked it in a while`
`16:58.16`	`TripleFFFF`	`CunningPike trying to terminate nortel phones onto voip`
`16:58.21`	`syzygyBSD`	`lol, that is where I think the problem would be`
`16:58.33`	`CunningPike`	`TripleFFFF: Got a PRI card in your nortel>`
`16:58.37`	`CunningPike`	`?`
`16:58.49`	`TripleFFFF`	`no idea`
`16:58.53`	`syzygyBSD`	`there is probably an option in there of whether to send the callerid from the PBX system or the phone (complete guess that this is happening)`
`16:59.13`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`16:59.53`	`*** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com)`
`16:59.56`	sp0n9e`	`how do i get debug information into /var/log/asterisk/debug ? that file doesn't exist on my system, i've tried asterisk -cd`
`17:00.14`	`*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)`
`17:00.39`	`Qwell[Digium]`	sp0n9e`: check logger.conf - make sure debug => is uncommented
`17:00.51`	`CunningPike`	`TripleFFFF: Well, find out - it will make life a _lot_ easier`
`17:01.43`	`TripleFFFF`	`if no card is there an option`
`17:01.46`	`TripleFFFF`	`its just for 1 line`
`17:01.47`	`*** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)`
`17:01.48`	`TripleFFFF`	`one phone`
`17:01.53`	`TripleFFFF`	`for now`
`17:02.10`	`cytrak`	`ok on the siemens there is an extension which is 5050 and its setup as a voicemail extension which tells the PBX that the last 4 digits of the caller should be passed to the voicemail application. So 5050 is part of a routing group which dials 350 (voicemail extension). previously 350 would go out through an analog line to a voicemail box, but now I route it through the PRI which connects to *`
`17:02.25`	`cytrak`	`asterisk sees 5050 as the caller`
`17:02.28`	`variable_office`	`is the Aastra 9133i a good phone/ what experience doo people have with it?`
`17:02.39`	`variable_office`	`*`
`17:02.40`	`variable_office`	`do`
`17:02.47`	`TripleFFFF`	`and with a cisco 2620 can we do ?`
`17:03.18`	`cytrak`	`the previous voicemail app also saw the 5050 but then would wait until it received the mailbox 4 digit extension passed by the PBX`
`17:05.19`	`syzygyBSD`	`ok... so let me see if I got this; your pbx send a callerid of 5050, dials 350, then sends dtmf tones of the extension that is trying to get voicemail?`
`17:05.40`	`*** join/#asterisk luckasrg (n=lucasrg@204.97.196.20)`
`17:05.52`	`*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)`
`17:05.53`	`luckasrg`	`hi all`
`17:06.04`	`syzygyBSD`	`high luckasrg`
`17:06.12`	`cytrak`	`syzygyBSD: yes that's how it looked like`
`17:06.48`	`luckasrg`	`hi i am trying to get sip to H323 to work but some tranlation thing is not working`
`17:06.48`	`cytrak`	`syzygyBSD: In the past I tried to tell asterisk to read any inputs but that didn't work`
`17:07.13`	`[TK]D-Fender`	`variable_office: Its decent, but in Nort America, not a phone I'd recommend`
`17:07.27`	`cytrak`	`syzygyBSD: I also remeber that during the installation of the old voicemail app on another machine that I had to choose the siemens box protocol I think`
`17:07.31`	`variable_office`	`[TK]D-Fender what is a good phone for multiline and speakerphone?`
`17:07.40`	`*** join/#asterisk _deg_ (n=deg@200.163.193.247)`
`17:07.41`	`TripleFFFF`	`ok so i guess not`
`17:07.42`	`[TK]D-Fender`	`variable_office: Any Polycom.`
`17:07.46`	`luckasrg`	`does any one know if there is any documentation that i can read to get this to work`
`17:08.04`	`syzygyBSD`	`cytrak: give me a sec to create a voicemail app for you real quick`
`17:08.12`	`[TK]D-Fender`	`variable_office: Actually, with speakerphone in mind any but the IP 301 which isn't a real speakerphone.`
`17:08.21`	`[TK]D-Fender`	`variable_office: Where are you located?`
`17:08.25`	`*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)`
`17:08.26`	`variable_office`	`[TK]D-Fender IL`
`17:09.00`	`[TK]D-Fender`	`variable_office: Here's the best pricing I've seen on them : http://www.telephonydepot.com/Polycom_s/25.htm`
`17:09.03`	`variable_office`	`whats the least expensive 4 line?`
`17:09.22`	`*** join/#asterisk dalfz (n=dman@179.80-203-19.nextgentel.com)`
`17:09.23`	`cytrak`	`syzygyBSD: you mean like a voicemail * extension`
`17:09.33`	`syzygyBSD`	`yes`
`17:09.41`	`[TK]D-Fender`	`variable_office: Just to clarify the concept of "lines" is that any polycom can shuffle at least 24 calls at a time, its a quesiton of IDENTITIES.`
`17:09.44`	`dalfz`	`for maximum performance of the TE412P card, what hardware do i need on the server? cpu/mem`
`17:10.20`	`[TK]D-Fender`	`variable_office: Most users would be well served by an IP 430 or IP 501. Only a receptionist should need the 601 (suggested with attendant modules)`
`17:10.28`	`*** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net)`
`17:10.36`	`variable_office`	`[TK]D-Fender ah, so how would you switch between the different lines?`
`17:10.53`	`[TK]D-Fender`	`dalfz: Decent normal P4 w/ 1 gig should be more than comfortable if you're not transcoding like nuts`
`17:12.01`	`dalfz`	`[TK]D-Fender, no transcoding.. so an active channel doens't consume much mem or cpu, i understand`
`17:12.01`	`[TK]D-Fender`	`variable_office: Well you can treat each "line-key" as supporting 1 call each and subsequent calls will cascade to the next free line key, or a single line key can be made to support MULTIPLE calls and you'd use the cursor keys to navigate them.`
`17:12.01`	`*** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome)`
`17:12.11`	`[TK]D-Fender`	`dalfz: You could get by with 512mb and any decept P4 then really....`
`17:12.28`	`dalfz`	`greatness!`
`17:12.30`	`dalfz`	`is asterisk multithreaded for dual core cpu environment btw?`
`17:12.30`	`syzygyBSD`	`pastebin is down, any suggestions?`
`17:12.43`	`Supaplex`	`nopaste`
`17:12.47`	`[TK]D-Fender`	`syzygyBSD: tried .ca?`
`17:12.51`	`cytrak`	`pastebin.ca`
`17:12.57`	`syzygyBSD`	`ya, that is the one i tried`
`17:13.02`	`[TK]D-Fender`	`ew... yeah its dead`
`17:13.05`	`syzygyBSD`	`problems with their DB from the looks of it`
`17:13.08`	`variable_office`	`[TK]D-Fender ah cool, so what is the difference between the 430 and the 501? just the additional "line"?`
`17:13.08`	`cytrak`	`nervermind`
`17:13.13`	`Supaplex`	`http://rafb.net/paste/`
`17:13.13`	`[TK]D-Fender`	`.com seems functional again`
`17:13.17`	`syzygyBSD`	`thanks`
`17:13.25`	`hmmhesays`	`lots of steps in this damn installation`
`17:14.06`	`*** join/#asterisk pbx1 (i=pbx1@netblock-66-245-213-61.dslextreme.com)`
`17:14.13`	`Supaplex`	`dams are usually labor intensive to build.`
`17:14.34`	`[TK]D-Fender`	`variable_office: IP 430 has built-in PoE support, lighted indicators, a smaller screen, 2 line-keys (instead of 3). Still a great general purpose phone. Your wiriing possibilities would determine my recommendation`
`17:15.15`	`variable_office`	`[TK]D-Fender i can wire anything id like`
`17:15.23`	`[TK]D-Fender`	`variable_office: If you were interested in deploying PoE and wanted to save some $ I'd say go with the IP430. If PoE doesn't factor in I'd say go for the %01 for the bigger screen and more line keys (just looks a bit nicer.`
`17:15.25`	`*** part/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net)`
`17:16.02`	`variable_office`	`ya`
`17:16.42`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co)`
`17:16.50`	`[TK]D-Fender`	`variable_office: In that case I'd suggest you go PoE on these and examine your call flow. I own each of Polycom's desktop phone models and they are all great choices depending on usage. IP 430 is your baseline full-feature phone at a bargain. IP 301 is SUPER budget that in all practicality only really sacrifices the speakerphone.`
`17:16.55`	`syzygyBSD`	`cytrak: http://rafb.net/paste/results/srPGFi75.html`
`17:16.58`	`variable_office`	`ill go for the 501 i think, is it massive, or of reasonable physical size?`
`17:17.06`	`syzygyBSD`	`try that, let me know if you have questions`
`17:17.06`	`[TK]D-Fender`	`variable_office: very reasonable.`
`17:17.12`	`*** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net)`
`17:17.17`	`ionix`	`hmm, I think polycom's PoE is bullshit`
`17:17.23`	`*** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net)`
`17:17.26`	`ionix`	`do they still force you to use an adapter for each phone at 50$?`
`17:18.16`	`[TK]D-Fender`	`variable_office: Measurements should be on their site`
`17:18.21`	`variable_office`	`[TK]D-Fender what phone do you personally use?`
`17:18.21`	`*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`17:18.29`	`syzygyBSD`	`variable_office: I thought it was massive, at least compared to the 841 I have next to it`
`17:18.41`	`syzygyBSD`	`only about 2-3" wider though`
`17:18.52`	`[TK]D-Fender`	`ionix: The only bad part is the ones using the PoE adapter cable instead of built-in support (IP 430 / 601)`
`17:19.14`	`variable_office`	`syzygyBSD who makes the "841"?`
`17:19.43`	`[TK]D-Fender`	`syzygyBSD: Yes, bigger than the 841, but the 841 has a super shitty LCD.... polycom's (501/601) have a nice large readble LCD.`
`17:19.52`	`cytrak`	`syzygyBSD: I have also tried that before ... I waited up to 40 seconds but I did not use the other _xxxx line so I'll give it a shot it might work.... thanks`
`17:19.56`	`syzygyBSD`	`linksys now, I have a sipura here, boss has a linksys in his office, right next to my polycom, got a couple grandstreams around here somewhere too`
`17:20.05`	`cytrak`	`never thought about a _xxxx line`
`17:20.13`	`[TK]D-Fender`	`variable_office: Discontinued POS phone..... replaced by Linksys' new line (which isn't worth it.`
`17:20.19`	`ionix`	`[TK]D-Fender So a IP430/601 will connect to a real POE switch right out of the box, right?`
`17:20.35`	`variable_office`	`ionix looks like it`
`17:20.37`	`[TK]D-Fender`	`ionix: Yup`
`17:20.41`	`syzygyBSD`	`[TK]D-Fender: I have never had a problem reading the LCD, but it is less then half the size`
`17:21.23`	`[TK]D-Fender`	`syzygyBSD: Yes, and it also convey's a fraction of the information as well with poorer fonts. then again the 841 costs a lot less. You do (for the mostpart) get what you pay for...`
`17:22.14`	`[TK]D-Fender`	`ionix: I run pretty much all IP 600's at the office (1 x IP 301 w/ PoE cable for reception) off D-Link DES-1526 PoE switches. great setup.`
`17:22.23`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`17:22.33`	`syzygyBSD`	`[TK]D-Fender: fonts isn't something I really care about on a phone (the resolution of any of those LCD's sucks) Configuring the phones through the web is much easier for me on the 841 though`
`17:22.34`	`ionix`	`great`
`17:22.44`	`[TK]D-Fender`	`ionix: At home I have an IP 301, 430, and 501 all on power bricks because I don't have PoE there "yet"`
`17:22.48`	`syzygyBSD`	`and I don't have to wait 5 minutes between changing 2 things on a 841`
`17:23.15`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`17:23.22`	`variable_office`	`i dont have enough poe stuff here to justify a poe switch`
`17:23.30`	`luckasrg`	`i get this two messages.`
`17:23.31`	`[TK]D-Fender`	`syzygyBSD: You won't hear me argue there. If you are a very basic user the 841 is a fair bit easier to set up, but nowhere near as flexible.`
`17:23.33`	`luckasrg`	`channel.c:2706 ast_channel_make_compatible: No path to translate from SIP/5541-09a22190(4) to H323/5743251731-1(256)`
`17:23.33`	`luckasrg`	`Aug 14 12:22:06 WARNING[7519]: app_dial.c:1595 dial_exec_full: Had to drop call because I couldn't make SIP/5541-09a22190 compatible with H323/5743251731-1`
`17:23.38`	`javar`	`what hardware recommend me, for 200 concurrents calls?`
`17:23.51`	`[TK]D-Fender`	`variable_office: Well... how many phones are you looking at getting?`
`17:23.51`	`FuriousGeorge`	`200 phones :)`
`17:23.56`	`javar`	`yeah`
`17:24.03`	`[TK]D-Fender`	`javar: Depends on how they come in and transcoding.`
`17:24.03`	`variable_office`	`[TK]D-Fender lol 1`
`17:24.03`	`ionix`	`I am looking to enable our employees with ip phone at home. I work for a major English teaching business in Japan and they spend money like crazy calling back home and making calls between them. I would need a 1 line cheap but not shitty phone. What phone do you suggest? Should be very simple and if possible, remote provisioned enabled.`
`17:24.20`	`ionix`	`Sound quality over features of course, it would be used as a traditional phone`
`17:24.23`	`[TK]D-Fender`	`variable_office: .... so you just want a single IP phone for what... home?`
`17:24.26`	`syzygyBSD`	`[TK]D-Fender: very true, however agreeing with you I will give new clients the 501 for the options they want most of the time, but for small offices I would recommend a 941 I think`
`17:24.27`	`*** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl)`
`17:24.31`	`javar`	`i need receive 200 calls.. from many proxies.. what hardware i need on my asterisk server`
`17:24.41`	`variable_office`	`[TK]D-Fender no my office, but ill be the only one using it for now`
`17:25.22`	`[TK]D-Fender`	`ionix: If you don't need speakerphone, IP 301, if you do IP 430. If you want to be really cheap just get an SPA-2002 ATA and use a normal phone.`
`17:25.26`	`syzygyBSD`	`[TK]D-Fender: what OS does the polycom run off of?`
`17:25.29`	`ZeNN`	`while compiling zaptel, is exits with an error concerning ztcfg.o .. what could be the probem ?`
`17:25.29`	`ZeNN`	`libtonezone.a(tonezone.lo)(.gnu.linkonce.t.__i686.get_pc_thunk.bx+0x0):/root/zaptel-1.2.7/tonezone.c:41: first defined here`
`17:25.29`	`ZeNN`	`collect2: ld returned 1 exit status`
`17:25.34`	`[TK]D-Fender`	`javar: I jsut asked you some specific questions....`
`17:25.57`	`javar`	`which ?`
`17:25.59`	`syzygyBSD`	`if it has a base of linux it might be worth it to hack the firmware to fix a lot of the problems I have with it`
`17:25.59`	`ionix`	`How's the call quality on a SPA-2002?`
`17:26.09`	`ManxPower`	`syzygyBSD, lime most phones it runs it's own OS`
`17:26.17`	`Qwell[Digium]`	`limeOS`
`17:26.19`	`variable_office`	`[TK]D-Fender are grandstream phones of good quality?`
`17:26.30`	`ManxPower`	`variable_office, no they are not.`
`17:26.33`	`eKo1`	`Grandstream sucks.`
`17:26.40`	`syzygyBSD`	`thanks Qwell[Digium]`
`17:26.43`	`[TK]D-Fender`	`variable_office: Grandstream is GARBAGE`
`17:26.53`	`variable_office`	`what about for ATAs ?`
`17:27.12`	`Qwell[Digium]`	`I would just like to say - for the record, that Sprint sucks.`
`17:27.16`	`ManxPower`	`variable_office, SIPura`
`17:27.23`	`ionix`	`GrandStream Budgetone is crap, I already tried`
`17:27.38`	`Qwell[Digium]`	`and that (again, for the record) tmobile does not suck`
`17:27.40`	`variable_office`	`i had one, but it died, i figured it was a power thing`
`17:27.42`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`17:27.50`	`variable_office`	`so for ATAs sipura is preferred?`
`17:27.50`	`[TK]D-Fender`	`variable_office: Even then wouldn't touch them. Go with Linksys for ATA's, or Mediatrix if you want high-end`
`17:27.52`	`javar`	`[TK]D-Fender: tell me`
`17:28.20`	`[TK]D-Fender`	`javar: (1:24:02 PM) [TK]D-Fender: javar: Depends on how they come in and transcoding.`
`17:28.38`	`variable_office`	`which sipura atas are routers as well so that sip doesnt have to go through nat?`
`17:28.46`	`[TK]D-Fender`	`javar: Explain where the calls would come in from and how you would process them.`
`17:28.54`	`Qwell[Digium]`	`variable_office: well, if it's a router, it's doing nat...`
`17:29.06`	`syzygyBSD`	`Qwell[Digium]: ya, but tmobile won't let me change my outgoing callerid :(`
`17:29.25`	`Qwell[Digium]`	`syzygyBSD: Asterisk + DISA + ITSP`
`17:29.41`	`[TK]D-Fender`	`variable_office: BGelieve me when I tell you you don't want the router-based ones unless you are SURE you aren't going to expand. changing it mean replacing the router role it server. Their normal models work just fine behind NAT`
`17:29.44`	`variable_office`	`Qwell[Digium] not necessarily, but i want the ata to be the router so that sip can be on the public, not behind the nat`
`17:30.13`	`javar`	`i receive the calls in G711u and i need to transfer in G711...`
`17:30.20`	`variable_office`	`[TK]D-Fender i was under the impression that sip is not usable behind a nat?`
`17:30.22`	`[TK]D-Fender`	`Qwell[Digium]: On the ones that are a router the SIP part isn't NAT'd, its on the WAN interface.`
`17:30.22`	`ManxPower`	`variable_office, there is no real reason to do that.`
`17:30.26`	`Qwell[Digium]`	`oh`
`17:30.39`	`ManxPower`	`variable_office, SIP works just fine behind NAT`
`17:30.44`	`[TK]D-Fender`	`variable_office: Sure it is. I run * in double-NAT'd scenarios just fine all the time including my own home.`
`17:30.52`	`javar`	`i receive the calls from many proxies via SIP`
`17:31.00`	`variable_office`	`without any special configs on the nating router?`
`17:31.05`	`javar`	`no hardware connect on my server.`
`17:31.18`	`ManxPower`	`variable_office, the only time you need special configs on the NAT router is of ASTERISK is behind NAT.`
`17:31.29`	`ManxPower`	`of == if`
`17:31.41`	`variable_office`	`oh, so in asterisk just say nat=yes eh?`
`17:31.51`	`syzygyBSD`	`Qwell[Digium]: ya, I was looking at that but would love to use all the stored phonebook entries in my cell phone vs having to redial them after calling myself`
`17:31.58`	`ManxPower`	`variable_office, if the SIP clients are behind NAT and Asterisk is on a public IP, yes.`
`17:32.05`	`[TK]D-Fender`	`variable_office: not that much... 2-3 settings depending what side is NAT'd`
`17:32.10`	`Qwell[Digium]`	`syzygyBSD: most phones let you send your phonebook entry as dtmf`
`17:32.24`	`Qwell[Digium]`	`so, dial your asterisk box, hit disa, then hit a speeddial entry`
`17:32.26`	`variable_office`	`ok, i guess its worth a try`
`17:32.35`	`variable_office`	`why dont you like the router+ata ones?`
`17:32.46`	`syzygyBSD`	`ya, just a pain vs them fixing my callerid at their switch level, it is possible, just a pain for all my calls`
`17:33.27`	`Qwell[Digium]`	`8005551212p12345p5551212`
`17:33.28`	`syzygyBSD`	`I love tmobiles callerid authenticated voicemail though`
`17:33.30`	`javar`	`[TK]D-Fender: the band width is no problem with me`
`17:33.48`	`syzygyBSD`	`Qwell[Digium]: :( cell phones don't have a way to pause in them`
`17:33.53`	`[TK]D-Fender`	`variable_office: Because SIP is only active on the WAN interface which adds complications. Also if there is a router in place you'd be replacing it with this ATA one with potentially less features you wan'tneed, and if you do get on and decide you want to move it elsewhere you'll need to buy a new router to replace it.`
`17:33.54`	`Qwell[Digium]`	`they do too`
`17:34.02`	`Qwell[Digium]`	`syzygyBSD: hold * or # for a second..it'll give you a p`
`17:34.09`	`syzygyBSD`	`oh.. how? I looked for that`
`17:34.11`	`syzygyBSD`	`thanks!`
`17:34.25`	`[TK]D-Fender`	`variable_office: And the fact its only configured on the WAN port blows. Trust me... really not worth it unless you are never going to think about the site you install it for again.`
`17:34.42`	`Qwell[Digium]`	`on mine, I have to hold 0 to get a +, then * gives me p`
`17:35.17`	`variable_office`	`[TK]D-Fender ah, well if sip does indeed work behind existing routers without problems then i guess i dont need the router ata anyways`
`17:35.33`	`syzygyBSD`	`ya, verified that p does work on my cell phone, I may have to have some fun later with that`
`17:35.52`	`[TK]D-Fender`	`variable_office: Yeah, and they're cheaper too....`
`17:36.12`	`[TK]D-Fender`	`variable_office: You can afford to take it out of service to reboot etc without disrupting your network.`
`17:36.52`	`variable_office`	`well, the atas werent for my network`
`17:37.10`	`[TK]D-Fender`	`variable_office: That of wherever you are planning to install them`
`17:37.21`	`funxion`	`does anyone know of a way to accept an ooh323c call playa prompt read digits and dialout without supplying answer supervision until it is passed though from the far end?`
`17:38.05`	`variable_office`	`[TK]D-Fender whered telephony depot ship out oh?`
`17:38.08`	`variable_office`	`*of`
`17:38.18`	`ManxPower`	`funxion, no. audio is only one way on unanswered calls.`
`17:39.02`	`[TK]D-Fender`	`variable_office: Somewhere in thye US but most places it dirt cheap to ship US48`
`17:39.07`	`syzygyBSD`	`ManxPower: which way?`
`17:39.23`	`Juggie`	`syzygyBSD, guess.`
`17:39.24`	`ManxPower`	`from the server to the caller`
`17:39.27`	`variable_office`	`[TK]D-Fender i was just wondering if it was in the midwest so it got here quicker`
`17:39.33`	`[TK]D-Fender`	`variable_office: Ask them about that part and look at these 2 other places : www.atacomm.com and www.voipsupply.com`
`17:39.45`	`[TK]D-Fender`	`variable_office: I just showed you the cheapest place for them first.`
`17:39.46`	`ManxPower`	`So the server can play things like "your call cannot be completed as dialed" messages and stuff like that without answering the line.`
`17:39.58`	`variable_office`	`[TK]D-Fender already ordered`
`17:40.19`	`[TK]D-Fender`	`variable_office: Look them all up and see if its worth a day or two more tops to get it at abetter price (or not even longer..)`
`17:40.26`	`[TK]D-Fender`	`variable_office: :O`
`17:40.33`	`javar`	`[TK]D-Fender: so?`
`17:40.40`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`17:41.24`	`variable_office`	`im going to go eat some lunch, thanks for the help, ill bbl`
`17:41.30`	`syzygyBSD`	`ManxPower: ok, so new dtmf tones arn't allowed from the caller untill the call is answered? what about the option to dial that accepts those?`
`17:42.37`	`ManxPower`	`syzygyBSD, what about the option? if the far end doesn't answer, the telco won't pass any audio to the destination`
`17:42.39`	`luckasrg`	`is this error a codec error app_dial.c:1595 dial_exec_full: Had to drop call because I couldn't make SIP/5541-096bb210 compatible with H323/100500543251731-2`
`17:42.53`	`ManxPower`	`luckasrg, sounds like a codec problem`
`17:43.24`	`ManxPower`	`luckasrg, but we can't tell you WHAT specific codec problem until we see the lines above it that includes the codec info`
`17:43.35`	`syzygyBSD`	`ManxPower: thought it might have been frequency ranges that wern't passed, maybe dtmf were passed`
`17:43.40`	`luckasrg`	`is it the codec or the translation betwen sip and h323`
`17:43.52`	`ManxPower`	`luckasrg, SIP and H323 are NOT codecs`
`17:44.03`	`luckasrg`	`Aug 14 12:41:28 WARNING[7574]: chan_h323.c:614 oh323_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 8/8)`
`17:44.06`	`ManxPower`	`syzygyBSD, The telco does this to prevent fraud.`
`17:44.24`	`syzygyBSD`	`no... nobody would EVER abuse that....`
`17:44.41`	`ManxPower`	`luckasrg, 8 is ulaw and 256 is G729.`
`17:45.01`	`ManxPower`	`so your problem is that you have not purchased a G729 license.`
`17:45.13`	`luckasrg`	`ok i get it`
`17:45.26`	`ManxPower`	`luckasrg, "show codecs" tells you the codec numbers`
`17:45.47`	`luckasrg`	`i need it installed on the asterisk or on the client i am using to make tha calls?`
`17:46.23`	`syzygyBSD`	`I won't ever use g729 again I think, caused me too many problems`
`17:46.36`	`ManxPower`	`luckasrg, if the client supports G729 then the client should already have a G279 license.`
`17:46.51`	`ManxPower`	`free softphones don't have G729`
`17:47.15`	`ManxPower`	`syzygyBSD, G729 saves my clients about $400/month`
`17:47.50`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`17:47.59`	`coppice`	`G.729 should be one of the least troublesome codecs. its so widely supported`
`17:48.07`	`[TK]D-Fender`	`G729's value depends on its application.`
`17:48.15`	`syzygyBSD`	`what kind of load does that server have? It may have just been a bad codec on the servers I was working on (I didn't install it) but it couldn't hadle over 20 calls at the same time`
`17:48.27`	`[TK]D-Fender`	`If you can affor ULAW hey, don't touch anything else...`
`17:49.11`	`syzygyBSD`	`s/hadle/handle/`
`17:49.14`	`javar`	`[TK]D-Fender: you can help me pls...`
`17:49.15`	`coppice`	`G.729 has a fairly heavy compute requirement`
`17:49.15`	`javar`	`=[`
`17:49.22`	`javar`	`:-$`
`17:49.33`	`*** join/#asterisk aw22 (n=aw@201.203.98.10)`
`17:49.46`	`Qwell[Digium]`	`javar: Does ":-$" mean you're putting your money where your mouth is, and are willing to pay a consultant?`
`17:49.49`	`syzygyBSD`	`ya, after switching to ulaw I went from 100% load at 50 calls to 15% at 90`
`17:49.53`	`aw22`	`can someone help me with an asterisk setup..`
`17:50.01`	`eKo1`	`Man, this E1 issue is voodoo. I think I'll be bringing in a shaman to sing some spell to my equipment and hopefully it will start working normally.`
`17:50.30`	`mog`	`eKo1, im a shaman of sorts`
`17:50.34`	`mog`	`whats the prob`
`17:50.35`	`jbroome`	`~zaptel`
`17:50.41`	`jbot`	`i guess zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access.`
`17:50.53`	`jbroome`	`bot-lag!`
`17:51.12`	`luckasrg`	`what free codec should i use for testing porpuces with a cisco gateway`
`17:51.18`	`Qwell[Digium]`	`ulaw`
`17:51.21`	`mog`	`ulaw forever`
`17:51.21`	`javar`	`Qwell[Digium]: sorry, i try to put emoticon`
`17:51.24`	`luckasrg`	`that you recomend me`
`17:51.38`	`Qwell[Digium]`	`or lpc10?`
`17:51.52`	`jbroome`	`you can pass marine mammals through a cisco gateway? :)`
`17:51.58`	`[TK]D-Fender`	`Qwell[Digium]: Domo arigato!`
`17:52.08`	`mut`	`yess`
`17:52.08`	`mut`	`these people had to see that..`
`17:52.12`	`mut`	`http://independentsources.com/2006/07/12/worst-company-urls/`
`17:52.15`	`*** join/#asterisk Ast001 (n=uros@212.200.219.228)`
`17:52.15`	`eKo1`	`mog: well, channels 48 - 62 of my second E1 result in no audio.`
`17:52.36`	`mog`	`what does ztmonitor say`
`17:52.56`	`mog`	`and can you cat /dev/zap/1 , cat /dev/zap/2 , cat /dev/zap/3 to pastebin`
`17:53.09`	`[TK]D-Fender`	`javar: Have you answered the questions I asked you?`
`17:53.12`	`aw22`	`sorry i am newbie on asterisk, can someone help a little with a office pbx i am programming`
`17:53.30`	`[TK]D-Fender`	`aw22: Ask some specific questions and maybe we can.`
`17:53.40`	`CunningPike`	`aw22: Just ask specific questions - people will answer if they know the answer`
`17:53.55`	`fiber0pti`	`how can I list all meetme conferences through the CLI? meetme list conferences doesn't seem to work`
`17:54.02`	`CunningPike`	`aw22: In the meantime:`
`17:54.04`	`javar`	`[TK]D-Fender: yes i said you... i receive the calls in G711u and i need to transfer in G711...i receive the calls from many proxies via SIP no hardware connect on my server.`
`17:54.05`	`CunningPike`	`~thebook`
`17:54.06`	`jbot`	`hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`17:54.20`	`mog`	`aw22, the answer is yes and you need exten => 1234,1,playback(tt-weasels)`
`17:54.49`	`CunningPike`	`mog: You need Answer() first ;)`
`17:55.01`	`mog`	`no you dont`
`17:55.06`	`aw22`	`ok`
`17:55.14`	`aw22`	`i have a debian pc`
`17:55.40`	`aw22`	`kernel 2.6, just apt the asterisk install, then configure 2 sip user and 2 extesions`
`17:55.53`	`syzygyBSD`	`fiber0pti: 'meetme'`
`17:55.57`	`aw22`	`i try to run asterisk /etc/init.d/asterisk start`
`17:56.09`	`aw22`	`then nmap localhost and see no 5060 running listening`
`17:56.58`	`aw22`	`the logs show me the following errors Aug 14 13:32:45 WARNING[19755]: Unable to open pseudo channel for timing... Sound may be choppy.`
`17:57.01`	`syzygyBSD`	`5060 is UDP`
`17:57.05`	`aw22`	`Aug 14 13:32:45 WARNING[19755]: Found no files in '/usr/share/asterisk/mohmp3'`
`17:57.10`	`aw22`	`Aug 14 13:32:45 WARNING[19755]: unable to spawn mp3player`
`17:57.12`	`syzygyBSD`	`do a netstat -na\|grep 5060`
`17:57.15`	`aw22`	`Aug 14 13:32:45 NOTICE[19755]: registered database handle 'mysql1' dsn->[MySQL-asterisk]`
`17:57.15`	`aw22`	`Aug 14 13:32:45 NOTICE[19755]: registered database handle 'mysql2' dsn->[MySQL-asterisk]`
`17:57.39`	`aw22`	`Aug 14 13:32:45 WARNING[19755]: Unable to open IAX timing interface: No such file or directory`
`17:57.40`	`aw22`	`Aug 14 13:32:45 WARNING[19755]: Unable to get our IP address, Skinny disabled`
`17:57.44`	`aw22`	`this are the logs`
`17:57.51`	`syzygyBSD`	`aw22: use pastebin`
`17:58.00`	`aw22`	`ok , sorry`
`17:58.00`	`aw22`	`netstat -na\|grep 5060`
`17:58.01`	`aw22`	`udp 0 0 200.122.132.181:5060 0.0.0.0:*`
`17:58.08`	`syzygyBSD`	`so it is running`
`17:58.28`	`aw22`	`i guess`
`17:58.34`	`syzygyBSD`	`i know`
`17:58.39`	`aw22`	`but if I telnet localhost 5060 nothing answers`
`17:59.06`	`[TK]D-Fender`	`aw22: telnet = TCP. SIP = UDP`
`17:59.06`	`syzygyBSD`	`once again, you are using TCP, sip is UDP`
`17:59.28`	`aw22`	`ooookkkkk get it... :S`
`17:59.31`	`ManxPower`	`THE NEWBIES! OH GOD THE NEWBIES!!!`
`17:59.33`	`*** part/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com)`
`17:59.42`	`*** join/#asterisk Assid (i=assid@203.115.83.213)`
`17:59.53`	`jbroome`	`those logs look like you started * w/o configuring anything`
`17:59.58`	`[TK]D-Fender`	`How very "Armageddon Now"....`
`18:00.07`	`aw22`	`so the daemon is running and should be no problem`
`18:00.29`	`syzygyBSD`	`ManxPower, that a recording from allison?`
`18:00.41`	`[TK]D-Fender`	`aw22: well.. runing yes... but doesn't mean its configured right....`
`18:00.44`	`CunningPike`	`[TK]D-Fender: ;)`
`18:01.13`	`aw22`	`ok, so i am gonna configure 2 pcs with the softphone and test it, then tell you how was it.. thanks.. sorry for bothering`
`18:01.41`	`[TK]D-Fender`	`aw22: Its no trouble.`
`18:01.53`	`syzygyBSD`	`how much work does allison get?`
`18:01.55`	`[TK]D-Fender`	`aw22: You got to asking your questions rather directly`
`18:02.13`	`CunningPike`	`~suggestions`
`18:02.15`	`jbot`	`i guess suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and ...`
`18:03.10`	`[TK]D-Fender`	`aw22: Yeah, follow the ablove, use some common sense and all is good.`
`18:03.18`	`aw22`	`ok`
`18:03.38`	`*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)`
`18:03.38`	`*** mode/#asterisk [+o Cresl1n] by ChanServ`
`18:03.50`	`FuriousGeorge`	`anyone using asterlink in here?`
`18:03.58`	`dalfz`	`[TK]D-Fender, what approach should i use to set up asterisk serving 1500 softphones? i cannot enter a record for each phone in sip.conf and externsions.conf`
`18:05.24`	`syzygyBSD`	`dalfz: why not?`
`18:05.50`	`syzygyBSD`	`well, extensions is easy, you can do pattern matching, but you want an entry for each in sip.conf`
`18:06.28`	`syzygyBSD`	`s/but you/I think you/`
`18:06.49`	`syzygyBSD`	`mmm, jbot is nice`
`18:07.04`	`dalfz`	`syzygyBSD, suppose i can generate the files by scripts`
`18:07.20`	`*** part/#asterisk jarrod (n=jarrod@juniperyour.net)`
`18:07.42`	`syzygyBSD`	`we have a script pulling out our sip.conf.auto file from our database`
`18:07.58`	`dalfz`	`ah`
`18:08.08`	`syzygyBSD`	`the extensions.conf is easy to do small though`
`18:09.17`	`syzygyBSD`	`as long as each extension doesn't have custom routing... we have that so we have an extensions.conf.auto file as well`
`18:09.48`	`dalfz`	`btw, if my lines go down, can asterisk automatically re-route to another redundant server?`
`18:09.57`	`*** join/#asterisk bofh42 (n=bofh42@p54829E40.dip0.t-ipconnect.de)`
`18:10.08`	`syzygyBSD`	`look at least cost routing`
`18:10.27`	`syzygyBSD`	`the short answer is yes`
`18:10.34`	`dalfz`	`aye, thanks`
`18:11.59`	`Kylun`	`I loaded asterisk with ztdummy, I just got some cheap 100 cards to test with, can I just recompile the zaptel driver, and add trunks?`
`18:12.31`	`intralanman`	`syzygyBSD: what do you use for LCR? lcDial?`
`18:12.32`	`dalfz`	`how many parallell conversations can a TE412P card do? 128*4?`
`18:12.34`	`variable_office`	`speaking of grandstream budgetone, the wan port randomly died, anyone had this happen before? is it something i can reset, or more likely it is hardware and is toast?`
`18:12.48`	`syzygyBSD`	`I dont' use anything`
`18:14.13`	`intralanman`	`ahh, i was looking into 'the calling card applications' and was going to use lcdial with them.... the calling card part didn't work out so well though`
`18:14.52`	`luckasrg`	`hi it worked`
`18:15.16`	`luckasrg`	`i used alaw and the call whent trougth just fine`
`18:15.48`	`mog`	`yay`
`18:17.15`	`*** join/#asterisk jab_work (i=jbroome@martha.stuart.is.gonna.innovateyour.com)`
`18:17.24`	`*** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)`
`18:17.50`	`*** join/#asterisk Ebola (n=Ebola@84.13.200.98)`
`18:20.47`	`jab_work`	`Before i get too involved in this, let me bounce something off the channel to see if it's sane:`
`18:21.34`	`jab_work`	`I have asterisk working a-ok with a voicepulse connect #. The client also wants a FXO card incase they lose their 'net connection they can still receive POTS calls`
`18:22.02`	`tzanger`	`when a SIP phone returns "temporaily moved", is there any dialplan variable set with the forwarding address/extension that the SIP phone provides?`
`18:22.04`	`jab_work`	`can i do a dialplan for ZAP and #include it in my VP [incoming] context?`
`18:22.07`	`*** join/#asterisk n0n0x (n=n0n0x@201.216.229.213)`
`18:22.48`	`n0n0x`	`Hi!`
`18:23.23`	`CunningPike`	`jab_work: How will your incoming calls get redirected to your POTS line?`
`18:23.53`	`jab_work`	`CunningPike: well, they're two different phone #s.`
`18:24.20`	`syzygyBSD`	`jab_work: yes`
`18:24.27`	`hmmhesays`	`bhahahahahahha`
`18:24.29`	`jab_work`	`sweet, thanks`
`18:24.29`	`CunningPike`	`jab_work: OK - simple then - just do what you say and handle the incoming calls the same way`
`18:24.42`	`file`	`hmmhesays: pay your membership fees!`
`18:24.43`	`CunningPike`	`hmmhesays: You always were insane`
`18:24.47`	`hmmhesays`	`haha yeah`
`18:25.09`	`hmmhesays`	`i am paying my membership fee, I'm writing an install script for vicidial so there`
`18:25.30`	`CunningPike`	`Kylun: You may not need to recompile - just load the module appropriate for your card`
`18:25.32`	`file`	`that's not a membership fee, that's torture`
`18:25.46`	`hmmhesays`	`file: with all the dependencies? yeah`
`18:26.01`	`*** join/#asterisk redox- (n=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com)`
`18:26.17`	`hmmhesays`	`I have everything automated from the scratch_install guide up to astguiclient install`
`18:26.25`	`Kylun`	`Thank you. :) I'm just shutting down the box now, to try out the cards.`
`18:27.17`	`n0n0x`	`I'm having problems with DTMF, there are some IVRs that doesn't detect dial tones. I've forced the dial plan to use rfc2833 with SIPDtmfMode command but still doesn't work. Could the outbound gateway be a problem? or it's (phone\|asterisk) side?`
`18:27.51`	`hmmhesays`	`where are your calls coming from`
`18:28.21`	`*** join/#asterisk slinabery (n=slinaber@smooth.worldcycling.com)`
`18:28.24`	`quid246`	`do alot of providers usually set a maximum call length?`
`18:29.15`	`hmmhesays`	`Mr Sparkle!`
`18:29.40`	`quid246`	`haha Mr Sparkle.. you talking about the Homer Japanese dish soap`
`18:29.49`	`n0n0x`	`hmmhesays, your msg was for me? (where are your calls coming from)?`
`18:30.00`	`*** join/#asterisk jake1932 (n=Administ@pool-70-16-129-225.phil.east.verizon.net)`
`18:30.52`	`jake1932`	`How can i ignore this line "Peer audio RTP is at port 192.168.3.100:18112"? I tried nat=yes`
`18:32.00`	`jake1932`	`rtp debug still shows it is trying 192.168.3.100`
`18:32.17`	`*** join/#asterisk Ebola (n=Ebola@89.240.183.206)`
`18:32.41`	`hmmhesays`	`n0n0x: yeah`
`18:32.47`	`hmmhesays`	`quid246: yeah`
`18:33.24`	`*** join/#asterisk sugardave (n=sugardav@cpe-70-112-127-158.austin.res.rr.com)`
`18:33.28`	`quid246`	`quid246: when I Was in Japan a few years back, I saw the Simpsons there... the tone of Homer's voice is exactly like the American one, except speaking Japanese`
`18:34.15`	`sugardave`	`what are some causes for the following error: username mismatch, have <VoicePulse3>, digest has <s>`
`18:35.15`	`sugardave`	`and Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@64.61.93.87>;`
`18:35.27`	`sugardave`	`any time I call my DID, that's what I get`
`18:36.03`	`jab_work`	`sugardave: sounds like you don't have the right auth keys for VP in your config`
`18:36.33`	`n0n0x`	`hmmhesays, the problem isn't incoming calls, the problem are outgoing calls. When i call an IVR that needs a bunch of numbers (let's say i've to dial a 8 digits UserID). The IVR doesn't recognize those digits.`
`18:36.42`	`sugardave`	`jab_work: I thought that, too, so I regenerated my credentials and re-downloaded the key file`
`18:36.55`	`hmmhesays`	`what is your call flow`
`18:37.08`	`n0n0x`	`hmmhesays, just 1`
`18:37.30`	`n0n0x`	`this PBX is in development state`
`18:37.33`	`hmmhesays`	`call flow dude, from origination point to termination`
`18:39.37`	`*** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com)`
`18:39.48`	`*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`18:40.34`	`n0n0x`	`hmmhesays, sry I'm not an expert using asterisk. Could you please explain what "call flow" means for you?`
`18:40.46`	`sugardave`	`it's weird because I can see "Registration successful" messages and my sip registry looks okay`
`18:42.13`	`*** join/#asterisk dasenjo_ (n=dasenjo@208.195.215.23)`
`18:42.41`	`n0n0x`	`with "call flow" do you mean the different nodes that`
`18:43.31`	`*** join/#asterisk MACscr (n=MACscr@66.73.154.70)`
`18:43.51`	`MACscr`	`anyone know of a company that does managed asterisk/freepbx hosting?`
`18:44.10`	`n0n0x`	`hmmhesays, sry, i think i get what do you mean with 'call flow'.. it just goes out from an Linksys PAP.. pass through the asterisk and then it goes out through a gateway provider.`
`18:44.34`	`*** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com)`
`18:44.47`	`dos000`	`hi i am getting res_odbc: Error SQLConnect=-1 errno=2013 [unixODBC][MySQL][ODBC 3.51 Driver]Lost connection to MySQL server during query`
`18:45.10`	`dos000`	`any ideas ? i have checked the driver and all configs should be ok`
`18:45.59`	`intralanman`	`dos000: have you successfully connected to the dsn using isql?`
`18:46.18`	`hmmhesays`	`n0n0x what provider?`
`18:46.26`	`n0n0x`	`Auris`
`18:46.34`	`*** join/#asterisk backblue (n=moo@87-196-67-20.net.novis.pt)`
`18:47.06`	`hmmhesays`	`do you have any problems with the asterisk ivr's and dtmf?`
`18:48.54`	`dos000`	`intralanman, ok .. im trying that now .. thanks for the tip`
`18:49.23`	`intralanman`	`np`
`18:49.43`	`file`	`[TK]D-Fender: they ordered my case from the supplier!`
`18:50.04`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`18:50.16`	`[TK]D-Fender`	`file: Have your peope see their poeple... they'll do lunch :)`
`18:50.32`	`blitzrage`	`lunch is overrated`
`18:50.42`	`file`	`:D`
`18:51.19`	`[TK]D-Fender`	`blitzrage: Tell that to Ethiopia....`
`18:51.27`	`blitzrage`	`hmmhesays: nice moves`
`18:51.29`	`hmmhesays`	`sally struthers`
`18:51.45`	`hmmhesays`	`i always play when I come around as a warm up`
`18:51.57`	`n0n0x`	`hmmhesays, Well.. when i call to an extension, i can't hear very well the tones. When i establish a connection between two extensions and i start dialing numbers from one phone.. the dial tone sound that i should hear in the other phone isn't what i expected to hear.. i hear something like 'clitches'`
`18:52.00`	`hmmhesays`	`speaking of speakers, I could use a new cabinet`
`18:52.20`	`file`	`hmmhesays: are you... having a blast?`
`18:52.20`	`hmmhesays`	`n0n0x send a call into local/1000@demo`
`18:52.29`	`file`	`hmmhesays: maybe the time of your life?`
`18:52.37`	`hmmhesays`	`not right now, when I get home and play some guitar I will be`
`18:52.43`	`hmmhesays`	`until i start in on the new stuff`
`18:52.49`	`[TK]D-Fender`	`hmmhesays: too slow, better to start with "Basket Case"`
`18:53.01`	`hmmhesays`	`my hand is still hurting from the bike accident`
`18:53.33`	`Greek-Boy`	`do I need to setup any arguments when loading zaptel kernel module?`
`18:53.46`	`Greek-Boy`	`that is for TDM24xx cards`
`18:53.57`	`DrkShdw`	`hey Worth, when you get a few minutes, can you type up what you did to install all of this in a jail? I've never had a need for jails before.. so have never messed with them. but I think I'd actually rather have all of this inside a jail Or jails)`
`18:54.54`	`mcreedjr`	`Greek-Boy: when actually modprobing the modules, no.`
`18:54.56`	`mcreedjr`	`Greek-`
`18:54.58`	`mcreedjr`	`emm`
`18:55.03`	`*** join/#asterisk MatsK (i=MatsK@83.233.97.229)`
`18:55.14`	`mcreedjr`	`You do need to configure /etc/zaptel.conf and run ztcfg after modprobing though`
`18:55.49`	`Greek-Boy`	`damn, i dunno why i can't get a response from my TDM2404E`
`18:56.15`	`mcreedjr`	`Greek-Boy: what does dmesg show after loading the modules?`
`18:57.21`	`Greek-Boy`	`Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters.`
`18:57.22`	`Greek-Boy`	`<PROTECTED>`
`18:57.41`	`Greek-Boy`	`/lib/modules/2.4.27-2-386/zaptel/wcfxo.o: insmod /lib/modules/2.4.27-2-386/zaptel/wcfxo.o failed`
`18:58.11`	`mcreedjr`	`what modules are you attempting to load?`
`18:58.22`	`Greek-Boy`	`just wcfxo`
`18:58.29`	`mcreedjr`	`ermm try wctdm`
`18:58.33`	`Greek-Boy`	`zaptel`
`18:58.56`	`mcreedjr`	`i don't know what the correct module is, but i don't think it's wcfxo`
`18:59.18`	`[TK]D-Fender`	`Not WCTDM either....`
`18:59.35`	`[TK]D-Fender`	`time to read the docs... its very clear which module to use`
`18:59.35`	`mcreedjr`	`[TK]D-Fender: enlighten us then :)`
`18:59.43`	`hmmhesays`	`so [TK]D-Fender I picked up a dunlop 3.0mm pick`
`18:59.46`	`hmmhesays`	`my god I love it`
`19:00.39`	`Kylun`	`question, do i NEED to redo zaptel if i was using ztdummy, and now i have some 100 cards to test with? It seems like the only thing different is some timing...`
`19:00.53`	`Greek-Boy`	`[TK]D-Fender which module should i be using?`
`19:01.00`	`Greek-Boy`	`wcfxo?`
`19:01.26`	`redox-`	`Whats the best way to see what calls are currently taking place? Something with more info than "show channels"`
`19:01.34`	`Greek-Boy`	`or ztd-eth`
`19:01.58`	`CunningPike`	`Kylun: Did you try it?`
`19:02.32`	`Kylun`	`CunningPike: I'm still trying to check dmesg to find the modems. I installed them, got a CentOS "configure" message, and I'm just trying to figure out where to go from here.`
`19:02.52`	`*** join/#asterisk UnstableFlame (n=unstable@70.58.167.89)`
`19:02.55`	`CunningPike`	`Kylun: What hardware is it?`
`19:03.03`	`Kylun`	`1sec`
`19:04.23`	`*** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net)`
`19:04.32`	`Kylun`	`CunningPike: just 2 generic X100P cards i got for testing`
`19:04.47`	`Greek-Boy`	`what line do i add for dynamic in zaptel.conf?`
`19:05.01`	`Kylun`	`CunningPike: wanted to make sure I could get it working correctly, and that it would suit my needs before i dropped any serious cash at this.`
`19:05.56`	`UnstableFlame`	`I have a quick question. I am installing asterisk for the first time I cant seem to get the cvs server to work is cvs.digium.com still valid and is it up?`
`19:06.12`	`*** join/#asterisk NDT (n=noone@cpe-24-195-66-214.nycap.res.rr.com)`
`19:06.30`	`CunningPike`	`Kylun: You couldn't have picked a worse card, by all accounts ;) The zaptel module you need is wctfxo`
`19:06.45`	`*** join/#asterisk andymul (n=andymul@67.106.157.244.ptr.us.xo.net)`
`19:06.48`	`[TK]D-Fender`	`Greek-Boy: http://www.digium.com/en/docs/misc/quick_install_zaptel_asterisk.pdf says it all....`
`19:06.55`	`Kylun`	`CunningPike: two cards, $25.00. :) Just want to make sure it works.`
`19:06.56`	`CunningPike`	`UnstableFlame: Negatory. s/cvs/svn/g`
`19:07.01`	`Greek-Boy`	`thanks TK`
`19:07.03`	`[TK]D-Fender`	`Greek-Boy: You really should look a bit harder.....`
`19:07.08`	`*** part/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net)`
`19:07.13`	`Kylun`	`CunningPike: I plan on getting one of those 4-port 400 series cards for production release`
`19:07.16`	`andymul`	`Anyone here know the SIP protocol well? I just need to know if a lineid on a contact is supposed to be unique.`
`19:07.23`	`CunningPike`	`Kylun: It does :)`
`19:07.56`	`CunningPike`	`Kylun: We have a TE410P - works great`
`19:08.26`	`CunningPike`	`Kylun: We started with a TE110P for our experiments`
`19:08.32`	`[TK]D-Fender`	`CunningPike: I'm somewhat sure he's not talking T1 there....`
`19:08.47`	`CunningPike`	`[TK]D-Fender: Yup - my bad`
`19:08.49`	`Kylun`	`No T1 here, no. Just 4 analog lines.`
`19:09.10`	`CunningPike`	`Kylun: Sorry - my fault - I always assume PRI :(`
`19:09.16`	`Kylun`	`:) no big deal`
`19:09.56`	`*** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)`
`19:10.39`	`sugardave`	`where should you start looking if you have IAX2 registered peers, but 'iax2 show registry' has no entries?`
`19:10.41`	`*** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net)`
`19:10.53`	`Dovid`	`afternoon all`
`19:10.58`	`sugardave`	`hello`
`19:11.12`	`Dovid`	`is there anyway to use chanisavail and if the channel is in use to hang up on the call that it is on ?`
`19:11.27`	`sugardave`	`Dovid: I'm not that far yet`
`19:11.34`	`Dovid`	`lol`
`19:11.36`	`Dovid`	`tis ok`
`19:11.52`	`Dovid`	`TK: ?`
`19:11.59`	`quid246`	`anybody ever seen this bug... Going from one * system to another * via SIP... sometimes the RTP connects and the call goes through, but SIP still thinks it hasn't answered?`
`19:12.14`	`Dovid`	`quid246: using NAT ?`
`19:12.33`	`Dovid`	`nm: i read ur message wrong`
`19:12.50`	`intralanman`	`dovid: can you use the manager api to do a hangup?`
`19:12.54`	`file`	`quid246: sip debug is a good thing`
`19:12.57`	`brodiem`	`has anyone used any answering machine detection, like app_machinedetect.c?`
`19:12.58`	`quid246`	`Dovid: Yeah, but strangely... I was testing it on 1-800's.. and American Airlines will do this (their various ines).. but other 1-800's get the ANSWER.. and I've tried different *-run providers.`
`19:13.12`	`[TK]D-Fender`	`Dovid: just do a dump of "show channels" in a script, nd grp them out.`
`19:13.22`	`[TK]D-Fender`	`Dovid: And sof-hanup them all :)`
`19:13.29`	`file`	`OH`
`19:13.34`	`file`	`quid246: they're cheating`
`19:13.41`	`file`	`quid246: they didn't answer the phone call`
`19:14.11`	`quid246`	`But I tried it also through a Cisco gateway.. and that connects, just not Asterisk systems`
`19:14.14`	`luckasrg`	`how hard is it to configure a ambient md3200 to work as fxo?`
`19:14.20`	`quid246`	`I mean connects and shows ANSWERED`
`19:14.28`	`redox-`	`Whats the best way to see what calls are currently taking place? Something with more info than "show channels"?`
`19:14.30`	`Dovid`	`[TK]D-Fender: I am not advanced wit scripts :(`
`19:15.03`	`Dovid`	`[TK]D-Fender: its just for 911, they have e911 so worse comes to worse 911 will go out thru e911`
`19:15.26`	`*** join/#asterisk pdt (n=ptinsley@adsl-070-148-122-203.sip.bna.bellsouth.net)`
`19:15.39`	`file`	`quid246: they send the audio as early media so they don't have to pay for part of the call, I distinctly remember someone mentioning American Airlines on the asterisk-users mailing list I think even`
`19:16.15`	`*** join/#asterisk kfudge (n=hgumva@24-217-137-21.dhcp.stls.mo.charter.com)`
`19:16.28`	`kfudge`	`Can anyone tell me what channel.c: Didn't get a frame from channel means?`
`19:16.51`	`quid246`	`file: Interesting... here's the humber anyway if people want to "see it in action" 1-800-223-5436`
`19:17.45`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`19:17.53`	`quid246`	`file: found it... what a bunch of cheats.`
`19:18.11`	`quid246`	`I was worried that my AGI was messed up`
`19:18.19`	`luckasrg`	`has any one had any success with the MD3200 modem os asterisk`
`19:19.11`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`19:22.34`	`[TK]D-Fender`	`luckasrg: Not until YOU decide to invent the driver for it....`
`19:22.54`	`*** join/#asterisk _DAW (n=bob@adsl-35-240-194.msy.bellsouth.net)`
`19:23.27`	`brodiem`	`has anyone used any answering machine detection, like app_machinedetect.c?`
`19:23.30`	`_DAW`	`hello. Is it possible to see what dtmf is sent on a call via pri debug?`
`19:24.07`	`[TK]D-Fender`	`brodiem: Ask iCEBrkr when he;s around, he's done stuff like that before.`
`19:24.13`	`brodiem`	`cool`
`19:24.25`	`jab_work`	`Ok, my card shows up in dmesg with the FXO module, but not in zap show channels: zapata.conf my first stop?`
`19:24.56`	`mcreedjr`	`jab_work: does ztcfg -vv show that is configured the channel right?`
`19:25.12`	`[TK]D-Fender`	`jab_work: zaptel.conf`
`19:25.27`	`jab_work`	`Channel 04: FXS Kewlstart (Default) (Slaves: 04)`
`19:25.28`	`jab_work`	`1 channels configured`
`19:25.32`	`*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)`
`19:25.45`	`mcreedjr`	`jab_work: definately a zaptel.conf problem`
`19:25.51`	`jab_work`	`I liked * a lot more before i started dealing with zaptel. :)`
`19:25.52`	`jab_work`	`thanks guys`
`19:25.53`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
`19:26.16`	`kfudge`	`My error occured when a person was trying to press 3 in someone elses IVR System`
`19:27.25`	`sugardave`	`if someone has VP Connect: how can I make incoming calls come in on IAX? It appears that I can't register the channel...but if I register a SIP, the calls fail with an authorization error`
`19:27.34`	`*** join/#asterisk ellisgl2 (i=keefejoh@gateway/web/cgi-irc/citmedia.org/unlabeled)`
`19:28.05`	`[TK]D-Fender`	`jab_work: This isn't rocket science... you should be done in minutes...`
`19:28.32`	`mcreedjr`	`[TK]D-Fender: positive reinforcement eh? :)`
`19:28.47`	`jab_work`	`[TK]D-Fender: i do seem to be making this harder than it should be`
`19:28.50`	`[TK]D-Fender`	`mcreedjr: Quite!`
`19:29.06`	`[TK]D-Fender`	`jab_work: Or at least giving yourself that impress :)`
`19:29.08`	`mcreedjr`	`jab_work: first one is always the hardest..`
`19:29.42`	`mcreedjr`	`jab_work: did you look at the zaptel.conf examples on voip-info.org? I find examples to be the most informative.`
`19:30.47`	`luckasrg`	`but is not the md3200 a x100p clone?`
`19:33.18`	`*** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw)`
`19:34.18`	`ellisgl2`	`Newb here dealing with t1's. Currently have a 4 port card - have one t1 port up and all the channels up and it is connected to a Adit 600 - I'm trying to make the ds0 connection dail a sip phone (sip stuff is working fine) and all I get is a busy`
`19:34.57`	`jab_work`	`mcreedjr: double checking`
`19:35.18`	`[TK]D-Fender`	`ellisgl2: Check your dialplan and make sure what your dial matches your zapata.conf`
`19:42.26`	`jab_work`	`dear jab_work: the last module on your card is "3" not "4"`
`19:43.12`	`ellisgl2`	`[TK]D-Fender: I have no Dail plan =P`
`19:43.32`	`*** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk)`
`19:48.42`	`*** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu)`
`19:49.10`	`Dovid`	`is this the correct way if some one uses the FXS it should go out thru the FXO ?`
`19:49.12`	`Dovid`	`exten => _XXXXXXXXXX,3,Dial(ZAP/1/1${EXTEN},10)`
`19:49.57`	`Dovid`	`my problem is that when asterisk bridges the call it thinks the call was answered and then it disregards the 10 at the end thinking that the call was picked up`
`19:50.26`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-200-247.red.bezeqint.net)`
`19:51.16`	`Dovid`	`anyone having issues when they go to pastebin.ca ?`
`19:52.12`	`Dovid`	`seems like thier db is down and pb.com jsut dosent work. u hit submit and it sits`
`19:52.15`	`*** join/#asterisk svenna (n=svenna@p548D3C9E.dip0.t-ipconnect.de)`
`19:52.24`	`*** join/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com)`
`19:52.46`	`kfudge`	`Can anyone tell me what channel.c: Didn't get a frame from channel means?`
`19:53.03`	`*** join/#asterisk wunderkin (n=wunderki@216-19-202-3.getnet.net)`
`19:53.30`	`docelmo`	`just what it says`
`19:57.13`	`*** join/#asterisk ellisgl2 (i=keefejoh@gateway/web/cgi-irc/citmedia.org/unlabeled)`
`19:57.36`	`kfudge`	`ok so it didn't get the frame, is there a way to fix? or is it something that happens at random`
`19:59.43`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`19:59.48`	`file`	`kfudge: happens when someone hangs up`
`20:00.07`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`20:00.35`	`kfudge`	`ah so not an issue then`
`20:01.33`	`*** join/#asterisk AJaymn (n=boiwonde@70.59.126.206)`
`20:01.40`	`*** part/#asterisk AJaymn (n=boiwonde@70.59.126.206)`
`20:02.26`	`kfudge`	`one of my users was having a hard time calling another business's phone system. It would no take her entry of 3`
`20:02.39`	`kfudge`	`And I saw that right next to it`
`20:02.57`	`*** join/#asterisk mrec_ (n=revenger@dslb-084-058-030-227.pools.arcor-ip.net)`
`20:03.22`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`20:03.50`	`mrec_`	`hi! does anyone know what "Callback on CID authentication. " means?`
`20:05.19`	`jab_work`	`there's an approved list of phone numbers. if someone calls in from one of those numbers, the system will call them right back.`
`20:05.51`	`[TK]D-Fender`	`Dovid: Analog cards can't tell if the other side picks up. Its considered answered the moment its able to take the line at all`
`20:06.33`	`Dovid`	`TK: Thanks`
`20:07.05`	`Dovid`	`TK: I thought i saw talks on the list that it could. or maybe this just was for signaling to the mabell that the call has been terminated ?`
`20:07.44`	`tzafrir_laptop`	`[TK]D-Fender, fxo card?`
`20:08.55`	`tzafrir_laptop`	`mrec_, you call back to the number that called you. But only if it's one of a pre-selected list.`
`20:09.05`	`[TK]D-Fender`	`tzafrir_laptop: thats what I'm talking about...`
`20:09.06`	`mrec_`	`okay thanks!`
`20:09.26`	`*** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18)`
`20:09.28`	`jab_work`	`gah, had it backwards`
`20:16.45`	`*** join/#asterisk PanHat (n=pieter@myw-stp-196-34-112-228.sentechsa.net)`
`20:18.34`	`Lyfe`	`anyone know off the top of their head if callerid recognition on an e&m wink t1 is automatically used, or if there's something I need to turn on for it?`
`20:18.50`	`*** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir)`
`20:20.08`	`*** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net)`
`20:20.19`	`sudhir492`	`hi`
`20:20.26`	`sudhir492`	`anyone using spandsp here?`
`20:20.41`	`ellisgl2`	`[TK]D-Fender: Ok - any sugguestions - remember I'm a newb here - I'm a data guy being made into a phone guy aswell`
`20:20.50`	`sudhir492`	`I have downloaded the asterisk ver 1.2.10`
`20:21.58`	`eKo1`	`Lyfe: You need to make sure your telco is sending CID info.`
`20:22.07`	`eKo1`	`sudhir492: Lots of folks here do.`
`20:22.21`	`Lyfe`	`eKo1: they wanted to make sure i'd be able to receive it before they turn the feature on, which is why i ask.`
`20:22.40`	`Greek-Boy`	`i configured my zaptel.conf for fxsks but it still say i have the wrong signalling`
`20:22.43`	`Greek-Boy`	`frustrating...`
`20:23.18`	`eKo1`	`Lyfe: Yes, you are able to receive it.`
`20:23.31`	`ellisgl2`	`Greek-Boy: try fxo_ls?`
`20:23.50`	`eKo1`	`fxo uses fxs signaling and vice-versa`
`20:24.06`	`*** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir)`
`20:24.16`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`20:25.55`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`20:26.34`	`ManxPower`	`fxs_ls means "don't try to detect when the far end hangs up"`
`20:27.04`	`Kylun`	`CunningPike: is this what you meant by crappy cards? ZT_CHANCONFIG failed on channel 1: No such device or address (6)`
`20:27.24`	`Lyfe`	`anyone know if queues are under heavy work between 1.2 and 1.4? (im' curious how much work i'll need to redo).`
`20:27.45`	`ManxPower`	`Kyler, What card do you have?`
`20:27.51`	`Kylun`	`generic x100p`
`20:27.54`	`Greek-Boy`	`fxs_ls or fxsls?`
`20:28.06`	`ManxPower`	`Kyler, does "lsmod" show wcfxo loaded?`
`20:28.30`	`Kylun`	`negative...`
`20:28.32`	`Kylun`	`wtf`
`20:28.47`	`ManxPower`	`Kyler, there is your problem.`
`20:28.53`	`marl`	`hi, can anyone point me to a resonable set of features.conf files? ive seen a few feature files over the last few months, and was wondering if there are any published on the net? dont see any need to re-invent the wheel`
`20:28.55`	`Kylun`	`well, that fixed the problem, but why ain't it auto loading... :-/`
`20:29.02`	`marl`	`i know i can dig around and add all differante individual features together, but im sure there msut be some common ones that almost everyone uses`
`20:29.09`	`ManxPower`	`Kyler, it doesn't get loaded automagically by driver fairies.`
`20:29.23`	`Kylun`	`ManxPower: uhh.yeah.. thanks. :)`
`20:29.31`	`Kylun`	`ManxPower: never suggested it did.`
`20:29.33`	`ManxPower`	`IIRC if you run "make config" in the zaptel source it will install an init script.`
`20:29.45`	`Kylun`	`ManxPower: nope, I did that already.`
`20:29.56`	`Lyfe`	`i have a driver fairy that loads it for me.. it's called the sangoma wanpipe module :P`
`20:30.06`	`ManxPower`	`then you can modify the /etc/rc.d/init.d/zaptel script to suit your needs and run "chkconfig zaptel on" to make it load on boot.`
`20:30.19`	`Kylun`	`yep, already on it. thanks. :)`
`20:30.26`	`[TK]D-Fender`	`Lyfe: They prefer to be called "queer" again now ;)`
`20:30.34`	`Greek-Boy`	`maybe i gotta reboot after installation of libri and zaptel`
`20:30.37`	`Lyfe`	`hehe`
`20:30.52`	`Greek-Boy`	`one things for sure, debian packages from apt-get shouldn't be used for zaptel and libri`
`20:31.31`	`blitzrage`	`one things for sure, debian ... shouldn't be used ...`
`20:31.42`	`jab_work`	`aww`
`20:31.45`	`Greek-Boy`	`lol`
`20:31.49`	`blitzrage`	`:)`
`20:32.01`	`Nugget`	`ont things for sure, linux is poo.`
`20:32.21`	`*** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca)`
`20:33.43`	`blitzrage`	`:)`
`20:33.47`	`*** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)`
`20:34.05`	`sudhir492`	`eKo1: What version of spandsp and rxfax, txfax to use`
`20:34.32`	`Lyfe`	`hmm.. is there a current recommended way to use a queue? I'd swear there's 3-4 different ways to configure them`
`20:34.52`	`tzafrir_laptop`	`Greek-B0y, what's the problem with the debs?`
`20:34.58`	`xai`	`ManxPower: hi.. how's the work going? We've been working on a asterisk performance gig in cali.`
`20:35.03`	`tzafrir_laptop`	`I like them...`
`20:35.19`	`*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)`
`20:35.21`	`tzafrir_laptop`	`As for modules loading: simply add the module names to /etc/modules`
`20:35.27`	`*** join/#asterisk anthonyl (n=anthony@ppp-70-227-136-89.dsl.chcgil.ameritech.net)`
`20:35.45`	`ManxPower`	`xai, which work are you referring to?`
`20:36.11`	`marl`	`anyone willing to let me have a features.conf file with a nice bunch of features for * ?`
`20:37.47`	`sudhir492`	`marl, what features do you want?`
`20:38.13`	`xai`	`ManxPower: client had some real problems with kernel, network, and ast setup. We're tuning the whole thing. He's the same guy i talked about last week.`
`20:38.29`	`ManxPower`	`xai, ah.`
`20:38.31`	`sudhir492`	`can anyone tell me which combination of spandsp, rxfax and txfax work?`
`20:38.41`	`ManxPower`	`network and kernel problems would make things complicated for Asterisk`
`20:38.47`	`ManxPower`	`sudhir492, what version of Asterisk?`
`20:38.51`	`xai`	`ManxPower: we're tuning the network stack, kernel stack, and some * parms too... he was running some funky fedora..`
`20:38.56`	`ido`	`has anyone here replaced their POTS home phone line with an SIP/Asterisk solution?`
`20:39.05`	`*** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net)`
`20:39.08`	`sudhir492`	`I have 1.2.10 on one machine, 1.2.7.1 on another`
`20:39.10`	`tzafrir_laptop`	`sudhir492, latest 0.0.2pre spandsp. The modules from the same dir`
`20:39.23`	`sudhir492`	`can use anyo of them for fax.`
`20:39.25`	`marl`	`lol, as meny as posible? ive been going through the * sites, and have come accross plenty of referances to things like hold/transfer/night mode etc. i know i can sit and spend hours getting the differant features together in one file, but thought id ask if anyone was willing share rather than re-inventing the wheel`
`20:39.42`	`ManxPower`	`sudhir492, then you need the version of spandsp, rx_fax, and tx_fax for Asterisk 1.2.x`
`20:40.07`	`*** join/#asterisk boch (n=root@201.216.241.97)`
`20:40.26`	`tzafrir_laptop`	`sudhir492, anyway, they mostly useful for recieving faxes. For sending you'll have the technical problem of getting the fax number from the user`
`20:40.36`	`DrkShdw`	`ido: most people here will tell you that voip is not ready to replace standard analog lines. It's a handy addition to analog lines. It can bring costs down, and has a few other advantages. however it's nowhere near the quality, or reliability of analog`
`20:40.40`	`boch`	`pals, do you know why i cant see the pri debug? i enabled with: pri debug span 1`
`20:40.57`	`boch`	`also set debug 1000`
`20:40.57`	`nextime`	`ido i think that is a bad idea to do that, voip lines are good to have more lines at better cost that your old pots line, but pots line remain more reliable, expecially in case of black out or down of your home connection`
`20:41.03`	`anthonyl`	`i don't know, i can really see asterisk support t38 internetfax with the right hardware`
`20:41.06`	`eKo1`	`boch: check logger.conf`
`20:41.09`	`syzygyBSD`	`DrkShdw: i disagree`
`20:41.10`	`anthonyl`	`pretty soon`
`20:41.36`	`ido`	`nextime, DrkShdw: i agree with you both. is there, then, a less risky alternative that would permit me to allow two people / two phones in my house to have private lines while only exposing one public line?`
`20:41.36`	`boch`	`eKo1: where's that file ?`
`20:41.44`	`DrkShdw`	`syzygyBSD: you are allowed to disagree. But you'd be the minority. anyone who has worked with voip more than a week or two, knows the truth.`
`20:41.54`	`ido`	`for example, i don't want my little sister to clog up the house phone line :)`
`20:42.00`	`ido`	`when someone calls her`
`20:42.01`	`eKo1`	`boch: normally in /etc/asterisk`
`20:42.12`	`syzygyBSD`	`for a home phone, I would agree, but that statement out of that context I don't`
`20:42.13`	`ido`	`or would i need to purchase an ISDN PRI?`
`20:42.31`	`syzygyBSD`	`and I have used it for many many months`
`20:42.38`	`boch`	`eKo1: this is the only uncommented line full => notice,warning,error,debug,verbose`
`20:42.51`	`DrkShdw`	`syzygyBSD: for any phone system. period. voip should only be an addition to, not a replacement of analog lines.`
`20:43.02`	*** join/#asterisk Trystan` (i=tslee@c-67-186-230-12.hsd1.ut.comcast.net)
`20:43.24`	`syzygyBSD`	`hmm, you must have some crappy networking hardware`
`20:43.24`	`nextime`	`ido : a ISDN BRI line is enought to have 2 lines`
`20:43.38`	`dalfz`	`oh my. my server has v1.0.7`
`20:43.39`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`20:43.39`	`sudhir492`	`I tried 0.0.2pre26 of spandsp and app_rxfax, txfax from the same version, but it core dumps`
`20:43.47`	`DrkShdw`	`No. I just understand how the internet works.`
`20:43.47`	`nextime`	`2 lines/2 channels`
`20:43.50`	`sudhir492`	`in 1.2.10`
`20:43.50`	`eKo1`	`boch: read the contents of that file`
`20:43.59`	`shmaltz`	`does the Polycom 601 have POE as is? or does it need the specail $40 cable?`
`20:44.00`	`eKo1`	`you need to have messages => ... in there`
`20:44.09`	`ManxPower`	`nextime, Asterisk does not support BRI in USA/Canada`
`20:44.13`	`marl`	`nextime, your coment about blackouts, doesnt realy hold nowadays, as most people have cordless phones, how meny people actually think that they should have a fixed line phone as a backup in case of powercut`
`20:44.14`	`marl`	`?`
`20:44.15`	`ido`	`nextime: suppose i have 5 people in the house, and i want to route calls to cellphones depending on the caller...do i need to buy a public line for every concurrent call i want?`
`20:44.23`	`ManxPower`	`shmaltz, Have you checked the polycom web site?`
`20:44.33`	`nextime`	`ManxPower : really?`
`20:44.35`	`shmaltz`	`ManxPower, I'm doing that right now`
`20:44.36`	`eKo1`	`ido: yes`
`20:44.40`	`ido`	`damn`
`20:44.41`	`Cresl1n`	`yeah you can`
`20:44.45`	`ido`	`:-/`
`20:44.49`	`syzygyBSD`	`DrkShdw: as do I, which is why I can get 5 9's reliability with Voip`
`20:44.57`	`Cresl1n`	`you just get a card that supports it or you get a converter box from euroBRI to US-BRI`
`20:45.01`	`Cresl1n`	`:-D`
`20:45.09`	`nextime`	`ido : maybe you can use 1 old pots line and add to it 4 voip lines`
`20:45.38`	`ManxPower`	`nextime, Almost nobody uses ISDN BRI in the USA/Canada. It is usually twice as expensive as two analog lines AND the customer must provide the ISDN TA, the telco will not.`
`20:45.39`	`ido`	`nextime: yeah, i was thinking of using incoming calls via voip, and 1 pots line for emergencies`
`20:45.59`	`ido`	`ManxPower is right, ISDN in general is expensive nowadays.`
`20:46.04`	`DrkShdw`	`syzygyBSD: I sincerely hope you don't sell pure voip systems to clients. but if you do, I sincerely hope they sue you for everything you are worth, when there is an outage :)`
`20:46.06`	`ido`	`because nobody uses it for internet service`
`20:46.11`	`ManxPower`	`ido, not in Europe`
`20:46.19`	`nextime`	`ManxPower : this is fucking limiting`
`20:46.52`	`syzygyBSD`	`DrkShdw: have you ever seen companies like vonage, teliax?`
`20:46.56`	`ManxPower`	`ido, in Europe MANY people use it for vlice.`
`20:47.08`	`ManxPower`	`voice, that is.`
`20:47.10`	`ido`	`ManxPower: i meant in the us`
`20:47.12`	`marl`	`DrkShdw, if voip is so bad for calls, why does almost every advertising/telesales company use it for the call centres nowadays?`
`20:47.22`	`ido`	`"where god is watching" ManxPower ;)`
`20:47.35`	`DrkShdw`	`marl: I didn't say it was so bad. I said it's not a replacement.`
`20:47.46`	`ManxPower`	`marl, VoIPoInternet cannot be relied on. VoIPoPrivateNetwork is as reliable as you want it to be.`
`20:47.58`	`*** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net)`
`20:48.02`	`nextime`	`in europe ISDN BRI is the standard for new home lines, noone, almost in italy where i am, install a pots line from many years`
`20:48.27`	`ManxPower`	`I would say that in the best case, VoIPoInternet is as reliable as a cell phone. If that is acceptable to you then you might consider VoIPoInternet`
`20:48.34`	`syzygyBSD`	`DrkShdw says it is never a replacement...`
`20:48.37`	`boch`	`eKo1: the line full => ... contains "debug" level, but i dont see any pri debug`
`20:48.55`	`ManxPower`	`boch, pri debug is a different debug.`
`20:49.05`	`ManxPower`	`"pri debug span X" where X is the span you want to debug.`
`20:49.15`	`marl`	`ive got an isp here thats just about to luanch a voip broadband service, that should get rid of most of the voip-analog problems, allowing you to install a pure voip system in an office, and relying on the isp to do the voip-pstn terminations`
`20:49.44`	`ManxPower`	`marl, THAT is VoIPoPrivate Network`
`20:49.45`	`boch`	`ManxPower: already did that but there is no output`
`20:49.56`	`marl`	`ah, ok got you`
`20:49.59`	`ManxPower`	`boch, then you don't have an ISDN PRI`
`20:50.15`	`DrkShdw`	`syzygyBSD: Of course I have. I also have heard the complaints about vonage. teliax is another story altogether. they don't market themselves as an analog replacement.`
`20:50.19`	`boch`	`asterisk1*CLI> pri show spans`
`20:50.19`	`boch`	`PRI span 1/0: Provisioned, Up, Active`
`20:50.26`	`ManxPower`	`marl, since the part where voice is going over IP is ALL owned and managed by the same company.`
`20:50.40`	`*** join/#asterisk turth (n=MAGiC@64.20.38.74)`
`20:50.42`	`turth`	`hey`
`20:50.52`	`turth`	`what are the best settings for telasip and asterisk`
`20:50.56`	`ManxPower`	`boch, call your telco say "I'm not seeing anything on the D Channel"`
`20:50.57`	`turth`	`so i get a clear signal`
`20:50.59`	`turth`	`it break up alot`
`20:51.07`	`marl`	`instead of it being owned and managed by the same company for analog?`
`20:51.33`	`turth`	`WHat are the best settings for asterisk using telasip as my service. I'm using the GSM codec but it seems to breakup alot. Any Suggestions?`
`20:51.38`	`marl`	`ive delt with BT in the uk for years now, and there service ISN'T that reliable :)`
`20:52.11`	`syzygyBSD`	`turth: try ulaw`
`20:52.16`	`*** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net)`
`20:52.19`	`boch`	`ManxPower: sure is a telco problem? maybe i missconfigured something, shouldnt i see my signaling ?`
`20:52.55`	`ManxPower`	`boch PRI span 1/0: Provisioned, Up, Active shows that there's a line connected, there isn't a red alarm`
`20:53.05`	`aydiosmio`	`how do I get * to route inbound calls directly out another trunk, so it acts as a gateway?`
`20:53.16`	`ManxPower`	`boch, you're not going to get any signaling as a pri_cpe until the pri_net side starts talking`
`20:53.23`	`turth`	`�syzygyBSD� Ulaw uses to much bandwidth... how do i install the g729 codec?`
`20:53.27`	`marl`	`sudhir492, did u have in mind any sites with a feature.conf file for *?`
`20:53.32`	`turth`	`g729 is better than gsm right?`
`20:53.36`	`boch`	`ManxPower: awesome, good explain, thanks`
`20:53.36`	`ManxPower`	`turth, you contact digium and buy a G729 license`
`20:53.40`	`aydiosmio`	`"better"`
`20:53.51`	`turth`	`there are no free g729 codecs?`
`20:53.53`	`syzygyBSD`	`I will never recommend the g729 codec after the problems I have had, however many people use it without a problem`
`20:53.54`	`aydiosmio`	`iLRC is a good codec`
`20:54.03`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`20:54.04`	`ManxPower`	`aydiosmio, there is nothing special about that. Just dial out as you normally would.`
`20:54.39`	`ManxPower`	`turth, because the G729 is copyrighted, patented, trade secreted, and international treaty restricted the answer is NO!`
`20:54.59`	`ido`	`MP3 is a good codec. 8)`
`20:55.04`	`turth`	`what exacly is pass thru mode?`
`20:55.06`	`ido`	`any idea when asterisk will have it? ;)`
`20:55.37`	`syzygyBSD`	`mp3 is patented no?`
`20:55.38`	`ManxPower`	`ido, MP3 is a HORRIBLE codec for real time voice.`
`20:56.10`	`*** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com)`
`20:56.23`	`ManxPower`	`turth, pass thru means that Asterisk just copies the raw audio from one device to another, since it is not encoding/decoding the audio you do not need a license.`
`20:56.23`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`20:56.25`	`turth`	`WHat is pass thru mode... It says I can use the g729 codec free with pass thru mode?`
`20:56.25`	`Greek-Boy`	`Aug 14 23:54:30 NOTICE[2231]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented)`
`20:56.27`	`turth`	`oh`
`20:56.36`	`Greek-Boy`	`why would it do that after my zaptel.conf and zapata.conf are configured?`
`20:56.39`	`ManxPower`	`Greek-B0y, install zaptel BEFORE you install Asterisk`
`20:56.58`	`blitzrage`	`yah... too much overhead and CPU consumption -- translate your MP3s to the codecs you use so you don't have to transcode`
`20:57.03`	`ido`	`ManxPower: for playing hold music it is a great codec :) especially if everything i have is encoded in it...`
`20:57.07`	`turth`	`so therefor its the same as ulaw in that matter right or will it be using g729? For my sip service to get what im saying using g729 do i need a license with asterisk@home?`
`20:57.26`	`Greek-Boy`	`ManxPower what if asterisk is already installed, do i just recompile?`
`20:57.28`	`turth`	`or can I use pass thru?`
`20:57.31`	`blitzrage`	`g729 doesn't need a license to do passthrough, but you need it to decode/encode`
`20:57.31`	`ManxPower`	`turth, If your SIP provider REQUIRES only G729 then you need a license.`
`20:57.34`	`ManxPower`	`Greek-B0y, yes`
`20:57.53`	`turth`	`no my isp allows ulaw and g729`
`20:58.02`	`*** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110)`
`20:58.22`	`turth`	`but if i use pass thru with g729 then its the same thing as using ulaw correct?`
`20:58.27`	`ManxPower`	`turth, ulaw uses alot of bandwidth.`
`20:58.33`	`ManxPower`	`turth, no.`
`20:58.34`	`turth`	`exaclu`
`20:58.34`	`Greek-Boy`	`thanks Manx`
`20:58.39`	`turth`	`thats why i wnat g729`
`20:58.48`	`turth`	`so what exacly would g729 with pas sthru do?`
`20:58.49`	`ManxPower`	`turth, if you use G729 passthru then Asterisk cannot monitor, play, or record audio.`
`20:59.04`	`turth`	`so it just cant record it`
`20:59.06`	`blitzrage`	`EXISTS() is used to check if a variable exists or not, correct?`
`20:59.19`	`turth`	`but will i be able to talk to the person im calling and they can speak to me and we hear each other?`
`20:59.31`	`ManxPower`	`turth, you cannot use Playback(), Background() or Voicemail`
`21:00.03`	`blitzrage`	`and only to the other end if it has a G.729 license as well`
`21:00.03`	`ellisgl2`	`When I make a Zap extention - is the channel number 1 or is it Zap/1-1 ?`
`21:00.16`	`blitzrage`	`Zap/1`
`21:00.20`	`turth`	`yes my isp has g729 so they must have the license right?`
`21:00.29`	`blitzrage`	`turth: if they do, then you just need it and it'll work`
`21:00.37`	`turth`	`perfect`
`21:00.46`	`ellisgl2`	`this is on a quad t1 card`
`21:00.52`	`ManxPower`	`ellisgl2, the -1 is the "call instance on that channel" so if you have two calls on a FXS channel (call waiting) you would have a -1 and a -2`
`21:00.52`	`blitzrage`	`just remember that to decode/encode into G729 you need a license (both sides)`
`21:00.54`	`ellisgl2`	`I'm trying to put extentions on the ds2`
`21:01.05`	`ellisgl2`	`er ds0`
`21:01.31`	`blitzrage`	`ellisgl2: yes, you woudl probably have zomething like Zap/1 -> Zap/23 on a North American PRI, where the 24th channel is the D channel`
`21:01.40`	`turth`	`But as long as my provider has the g729 codec (telasip) I can just use it in pass thru?`
`21:02.04`	`[TK]D-Fender`	`turth : only if your phones support it as well.`
`21:02.09`	`blitzrage`	`turth: no........ your provider doesn't need a license for passthrough -- but you do to encode/decode the audio as an endpoint`
`21:02.16`	`turth`	`my soft phones support it`
`21:02.32`	`[TK]D-Fender`	`turth : Also you'd have to replace all of your prompts with G729 versions, and change all your recording optiosn to use it as well`
`21:02.42`	`[TK]D-Fender`	`turth : Which one are you using?`
`21:02.49`	`turth`	`eyebeam`
`21:02.53`	`turth`	`and asterisk@home`
`21:03.05`	`blitzrage`	`phone 1 (g729) <---> Asterisk <---> phone 2 (g729) ---- Asterisk would "pass through" the audio between the two phones even if it doesn't have a license`
`21:03.41`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.23)`
`21:03.45`	`blitzrage`	`now if Asterisk needs to do anything with the audio (playback, background, voicemail, etc...) then it needs a license between it is not passing through -- it is the end-point`
`21:03.56`	`blitzrage`	`s/between/because`
`21:04.33`	`[TK]D-Fender`	`turth : then its doable, you jus need to make sure that * NEVER has to trnslate from anytihng else to G.729. That includes MoH`
`21:04.37`	`ellisgl2`	`woo hah!`
`21:04.49`	`eKo1`	`Anyone have any suggestion as to why calls through zap channels, 48 - 62, experience no audio.`
`21:05.03`	`[TK]D-Fender`	`blitzrage : not if the recording being sent is IN G729 in the first place.`
`21:05.27`	`blitzrage`	`ok... but if you got the audio into g729, you can probably assume a license was involved somewhere in making that audio`
`21:05.36`	`ellisgl2`	`channel and dial are Zap/1-1 or Zap/1-2 (haven't tried the second ds0 - time to pull out the soldering iron)`
`21:05.52`	`[TK]D-Fender`	`blitzrage : True, but there are native codec pack straight from Digium already...`
`21:05.55`	`*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`21:06.06`	`ManxPower`	`eKo1, the telco has the provisioning screwed up. happens all the time.`
`21:06.08`	`blitzrage`	`[TK]D-Fender: symantics!`
`21:06.33`	`[TK]D-Fender`	`blitzrage : Thats an Anti-virus company! SEMANTICS!`
`21:06.35`	`ellisgl2`	`ok i can call sip to ds0 but not the other way..`
`21:07.03`	`[TK]D-Fender`	`ellisgl2 : Check your dialplan and zap setup`
`21:07.14`	`blitzrage`	`[TK]D-Fender: picky picky :)`
`21:07.23`	`[TK]D-Fender`	`ellisgl2 : could have sworn I said that hours ago....`
`21:07.26`	`blitzrage`	`I hate not having a spellcheck on irssi :)`
`21:07.35`	`ManxPower`	`eKo1, Once we have confirmed that EVERYTHING works except for audio, we make the telco come out with a PRI tester. It's usually fixed withint an hour of the tech arriving.`
`21:07.43`	`ellisgl2`	`[TK]D-Fender: you did - i'm just typing to myself`
`21:07.45`	`ido`	`i'm just glad i have irssi on this old piece of crud 8)`
`21:07.47`	`[TK]D-Fender`	`blitzrage : You need an upgrade. Blitzrage 2.0!`
`21:07.52`	`blitzrage`	`it's true!`
`21:08.28`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co)`
`21:09.11`	`*** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu)`
`21:09.17`	`blitzrage`	`EXISTS(${MY_VAR}) will return 0 if MY_VAR is NULL or does not exist, correct?`
`21:09.29`	`ido`	`what is the cheapest technology i can use with asterisk for POTS integration of one phone line?`
`21:09.44`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`21:09.53`	`blitzrage`	`show function EXISTS() isn't terribly useful :) I thought it wanted the variable name, but now I remember looking at the code a while ago and I think it's just the inverse of ISNULL()`
`21:09.57`	`[TK]D-Fender`	`blitzrage : GotoIf($["${myvar}"=""]?100)`
`21:10.09`	`blitzrage`	`which expects data, not a variable`
`21:10.23`	`eKo1`	`ManxPower: What is a PRI tester?`
`21:10.31`	`blitzrage`	`GotoIf($[${EXISTS(${MY_VAR})}]?my_label) is more fun :)`
`21:10.45`	`[TK]D-Fender`	`blitzrage : And curiously not getting you anywhere!`
`21:11.01`	`ManxPower`	`eKo1, a VERY expensive device the telco plugs into the customer line at the customer site`
`21:11.14`	`ido`	`sorry to reiterate, but is there a cheap way of interfacing my single home phone line (POTS) with my asterisk/sip/voip/ultra-cool-voicy-thingy box?`
`21:11.28`	`blitzrage`	`[TK]D-Fender: it will in a moment when I fix it from passing it the variable name as opposed to the ${variable}`
`21:11.28`	`ManxPower`	`ido, no there is not.'`
`21:11.38`	`eKo1`	`ManxPower: The problem that I'm hooked up to the telco through SS7 and I'm using a SS7<->PRI gateway.`
`21:11.39`	`blitzrage`	`err... vice-versa`
`21:11.44`	`ido`	`ManxPower: well, what's the cheapest way? :/`
`21:11.49`	`blitzrage`	`my brain hurts from beerfest yesterday... :)`
`21:11.51`	`ido`	`i mean, would a voice modem work?`
`21:12.00`	`ManxPower`	`eKo1, there is no way in hell anyone here can help you with that.;`
`21:12.03`	`bXi`	`yo`
`21:12.08`	`ManxPower`	`ido, not with Asterisk`
`21:12.11`	`eKo1`	`ManxPower: hehe, figured.`
`21:12.17`	`ido`	`crap :(`
`21:12.23`	`bXi`	`i'm checking what the bare minimum is that need to define in sip.conf per user`
`21:12.43`	`ManxPower`	`eKo1, T-Berd is what most people call them.`
`21:12.44`	`ido`	`how much do 8-way doohickeys cost?`
`21:12.45`	`justinu\|laptop`	`eko1: are you running NFAS PRIs?`
`21:12.54`	`bXi`	`i currently have type, username, secret, host`
`21:13.02`	`bXi`	`any others which i REALLY need ?`
`21:13.23`	`ManxPower`	`bXi, disallow=all and allow=theonecodecyouwantforthatdevice`
`21:13.36`	`[TK]D-Fender`	`ido : Cheap ass X100P (or knock-off) for $15 or less`
`21:14.09`	`[TK]D-Fender`	`ido : they suck, but can get the job done.`
`21:14.11`	`bXi`	`ManxPower: what will happen if i dont have a disallow in my user bit /`
`21:14.29`	`eKo1`	`justinu\|laptop: the PRI side is not NFAS but the SS7 side is.`
`21:14.29`	`Greek-Boy`	`ok so i got to the point where i am able to dial zap channels but now i can't hear anything`
`21:14.30`	`Greek-Boy`	`lol`
`21:14.34`	`ido`	`[TK]D-Fender: thanks. $15 is cheap enough for me :)`
`21:14.37`	`ManxPower`	`bXi, unless you mean "bare minimum" to mean the same as the "minimum requirements for Windows XP", which is something like "Pentium 200Mhz w/256k of ram".`
`21:14.48`	`ManxPower`	`We all know how helpful THAT bare minimum is.`
`21:15.07`	`bXi`	`ManxPower: i'm making a webinterface for asterisk`
`21:15.16`	`bXi`	`and i want to make it very easy to use`
`21:15.23`	`ManxPower`	`bXi, you poor, misguided creature.`
`21:15.43`	`bXi`	`eh?`
`21:15.43`	`[TK]D-Fender`	`ido : You only need to take in 1 line?`
`21:15.44`	`MACscr`	`bxi: whats wrong with freepbx and such?`
`21:16.04`	`[TK]D-Fender`	`MACscr : besides the obvious? ;)`
`21:16.11`	`bXi`	`it doesnt fit in with the rest of the configuration panel`
`21:16.26`	`justinu\|laptop`	`eko1: how confident are you your SS7->PRI bridge is configured correctly?`
`21:16.42`	`MACscr`	`bxi: what configuration panel?`
`21:16.53`	`blitzrage`	`freepbx and the such is a bitch for consultants when you want something it doesn't do out of the box, but don't want to use straight asterisk`
`21:17.20`	`bXi`	`MACscr: i work for a company which makes linux based server products with a very userfriendly interface`
`21:17.22`	`ido`	`[TK]D-Fender: well, two lines would be better, and i have about 8 telephones to connect to one line, and one telephone to the other line`
`21:17.29`	`ido`	`[TK]D-Fender: "telephone" = person`
`21:17.35`	`eKo1`	`justinu\|laptop: well, it was working before`
`21:17.44`	`justinu\|laptop`	`before what?`
`21:17.45`	`MACscr`	`i thought freepbx was just a frontend to asterisk, didnt know there wasn things that asterisk can do and freepbx didnt`
`21:17.56`	`MACscr`	`figured you just had to get direty with the config files`
`21:17.58`	`eKo1`	`justinu\|laptop: before it stopped working`
`21:18.07`	`justinu\|laptop`	`lol alrighty then`
`21:18.38`	`eKo1`	`Well, the signaling gateway that I'm using is not marvelous either.`
`21:18.49`	`eKo1`	`Actually, it's a POS.`
`21:19.05`	`bXi`	`i'm making a script which will autodetect all lan>phone boxes on the lan and auto configure them`
`21:19.12`	`xai`	`ManxPower: looks like they are still havingproblems with NAT. Something is wierd in their networking.`
`21:19.28`	`justinu\|laptop`	`if signalling on a PRI works, but you get no audio, the usual culprit is timeslot mismatches`
`21:20.37`	`[TK]D-Fender`	`ido : So 8 people sharing 1 line, and 1 guy getting one all themself?`
`21:20.44`	`ido`	`[TK]D-Fender: yes`
`21:21.16`	`dos000`	`hos do i tell registered users when using asterisk and realtime ?`
`21:21.56`	`ido`	`[TK]D-Fender: the one getting a line to himself can just plug the telephone right into the line for all i care.`
`21:22.08`	`dos000`	`basicaly anything to debug realtime ?`
`21:23.21`	`*** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk)`
`21:23.46`	`ido`	`[TK]D-Fender: i am just wanting to do cool stuff with my home phone network, basically...`
`21:24.53`	`eKo1`	`justinu\|laptop: good point. I will recheck the CIC mappings`
`21:25.12`	`*** join/#asterisk aw22 (n=aw@201.203.98.10)`
`21:25.18`	`aw22`	`good evening`
`21:25.46`	`aw22`	`can someone help with this log Unable to get our IP address, Skinny disabled`
`21:26.31`	`ManxPower`	`aw22, are you using Cisco phones?`
`21:27.01`	`aw22`	`no i have asterisk running on debian and trying to test it with diax`
`21:27.44`	`[TK]D-Fender`	`ido : Well if its basically for personal use sure, whatever. I might suggest getting an SPA-3102 for $90 though. That'd give you 1 line and let you plung in an analog phone as well.`
`21:27.59`	`ManxPower`	`aw22, then you can ignore it`
`21:28.11`	`aw22`	`ok`
`21:28.37`	`aw22`	`the matter is that i cant get my to connnect to the server, I netstat the server and see that asterisk is running on 5060`
`21:28.53`	`aw22`	`I create 2 sip users and 2 extensions`
`21:30.14`	Trystan`	`greetings, have problems with Musiconhold, we get it to play but the output is just aweful (loud scratching as if bad mp3), tried with mpg123 0.59 and 0.60 same results, tried madplay same thing, the file plays fine on other systems, just not when coming through asterisk`
`21:30.40`	`ido`	`[TK]D-Fender: the spa-3102...i'd need to buy 8 of them, correct? and then in addition to that an x100p?`
`21:30.55`	`ido`	`oh oops`
`21:31.01`	`ido`	`nevermind it does both`
`21:31.03`	`ido`	`but...`
`21:31.06`	`intralanman`	`trystan: what's the sample rate of the mp3's?`
`21:31.08`	`ido`	`okay i get it`
`21:31.43`	`ido`	`[TK]D-Fender: i will not be able to make individual people's phones ring with an spa-3102, only all at once, since i won't be splitting the line into 8, right?`
`21:32.20`	Trystan`	`intralanman, one sec, should be in the format the site said but I will double check`
`21:33.04`	Trystan`	`intralanman, 128kbit, 10407 frames`
`21:33.46`	`tzafrir_laptop`	`ido, 8 sipura 3ks? why no 4 sipura 2ks?`
`21:34.26`	`tzafrir_laptop`	`How man phones you need to connect? how many telco lines?`
`21:34.27`	`n0n0x`	`hmmhesays, i'm back, i'm really sorry for the delay, i executed 1000@demo and went to an echo test`
`21:34.47`	`ido`	`tzafrir_laptop: lo nira li she'ani ikneh yoter me ehad...gam ani ba'aretz btw ;)`
`21:34.54`	`n0n0x`	`hmmhesays, i can't hear the dial tones as i should`
`21:35.59`	`ido`	`tzafrir_laptop: 1 telco line, 8 phones, preferably with the ability to individually ring each phone separately.`
`21:36.05`	`intralanman`	`trystan: i had the same problem before and resampling to 8khz mono helped`
`21:36.26`	Trystan`	`intralanman, hmmm okay let me try again`
`21:36.47`	`[TK]D-Fender`	`ido : quite backwards.....`
`21:36.59`	`[TK]D-Fender`	`ido : What are you using for phones?`
`21:38.00`	`ido`	`[TK]D-Fender: regular old single line analog phones, but if it's cheap, i could switch to ip phones`
`21:38.52`	`*** join/#asterisk P4C0 (n=ash@200.124.22.34)`
`21:39.26`	`[TK]D-Fender`	`ido : Then for all analog, get 4 x SPA-2002 (2 port FXS), and 1 x SPA-3102 (1 FXS, 1 FXO). You'll have room for 1 extra phone that way`
`21:39.41`	`P4C0`	`hello guys, is there a way to see what are the rtp ports a sip channel is using??`
`21:39.42`	`*** join/#asterisk meshuga (i=meshuga@c-71-231-141-145.hsd1.or.comcast.net)`
`21:39.47`	`ido`	`[TK]D-Fender: so, $1k? :P`
`21:40.25`	`meshuga`	`anyone used that telegate.net GSM PCI board?`
`21:40.38`	`[TK]D-Fender`	`ido : $370 USD`
`21:40.41`	`hmmhesays`	`waste my day`
`21:41.01`	`hmmhesays`	`n0n0x: you shouldn't hear tones, I sent you to the wrong extension`
`21:41.13`	`ido`	`[TK]D-Fender: that actually is reasonable.`
`21:41.16`	`ido`	`[TK]D-Fender: thanks!`
`21:41.26`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`21:41.28`	`sevard`	`hmmhesays: any experience using AudioCodes and * with a fax?`
`21:42.01`	`hmmhesays`	`some`
`21:42.08`	`hmmhesays`	`but you were a jackass earlier`
`21:42.11`	`hmmhesays`	`so bite me`
`21:42.26`	`hmmhesays`	`n0r0x local/s@demo`
`21:42.27`	`sevard`	`:)`
`21:42.28`	`justinu\|laptop`	`lol, you two rock`
`21:42.35`	`sevard`	`hmmhesays: come on baby i'll do it how you like it`
`21:43.00`	`hmmhesays`	`there are plenty of guys to hit on in here`
`21:43.47`	`hmmhesays`	`what do you want to know?`
`21:43.56`	`sevard`	`Have you ever tried the fax?`
`21:43.58`	`hmmhesays`	`yes`
`21:44.06`	`sevard`	`How did you do it?`
`21:44.13`	`[TK]D-Fender`	`ido : actually... make that 3 SPA-2002 and 2 SPA-3102 for $390 total and that takes in both lines with nothing wasted`
`21:44.17`	`hmmhesays`	`what a fantastically vague question`
`21:44.36`	`ido`	`sweet, thanks [TK]D-Fender`
`21:44.38`	`sevard`	`:D`
`21:44.38`	`n0n0x`	`i dialed 1000, started the demo, then i dialed 600 for an echo test and I started dialing some digits and i badly heared them.`
`21:44.57`	`[TK]D-Fender`	`ido : That gives you 8 FXS total, 2 FXO`
`21:45.03`	`sevard`	`Did you use T.38 Relay? Bypass?`
`21:45.08`	`hmmhesays`	`n0n0x you won't hear them good`
`21:45.27`	`n0n0x`	`ohp, i thought i should hear them good. :(`
`21:45.48`	`hmmhesays`	`if you aren't having any problems with local menus then the problem is with your carrier`
`21:45.54`	`P4C0`	`guys, I think asterisk is not using the rtp.conf... I have some range of ports there but rtp debug shows he is using other ports...`
`21:46.12`	`sevard`	`hmmhesays: you won't hear them WELL`
`21:46.15`	`sevard`	`:P`
`21:47.00`	`CunningPike`	`P4C0: Run ethereal/wireshark`
`21:47.03`	`sevard`	`hmmhesays: Did you use T.38 Relay? Bypass? Disable?`
`21:47.24`	`hmmhesays`	`whatever the ulaw setting is`
`21:47.28`	`hmmhesays`	`it has been a long time`
`21:47.45`	`hmmhesays`	`i need a marshall stack like in the darkness video`
`21:47.46`	`sevard`	`Well, there are t.38 settings and Fax/Modem Bypass Coder Type`
`21:47.50`	`sevard`	`if you disable them all they use ulaw`
`21:48.12`	`sevard`	`but with the sipura 2002s I use ulaw and i get UNKNOWN RTP 100 CODEC and fax works great`
`21:48.22`	`*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)`
`21:48.24`	`n0n0x`	`hmmhesays, it also could be my internet connection? or i'm wrong?`
`21:48.33`	`hmmhesays`	`n0n0x: highly unlikely`
`21:48.42`	`P4C0`	`CunningPike, what for? Got RTP packet from voipproviderip:11994 and Sent RTP packet to voipproviderip:11994 in the rtp debug... I have set a limit of 10001 to 11000 in rtp.conf`
`21:48.44`	`sevard`	`on the audiocodes if I 'disable' all the t.38 options it shoul dbe using ulaw, i don't get the unknown warning, but fax is flaky`
`21:49.01`	`hmmhesays`	`over a lan?`
`21:49.02`	`hmmhesays`	`or not`
`21:49.20`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`21:49.24`	`CunningPike`	`P4C0: You asked if there was a way to see what ports a SIP channel was using`
`21:49.48`	`P4C0`	`CunningPike, yes, but i figure that out, rtp debug :D`
`21:49.51`	`sevard`	`hmmhesays: over a lan`
`21:49.53`	`dlynes_laptop`	`Good afternoon, cp`
`21:49.55`	`*** join/#asterisk areski (n=areski@ip-213-49-76-28.dsl.scarlet.be)`
`21:49.59`	`CunningPike`	`P4C0: ;)`
`21:50.01`	`sevard`	`hmmhesays: same amount of hops with the sip2002`
`21:50.03`	`CunningPike`	`Hey, dlynes_laptop`
`21:50.11`	`sevard`	`hey dlynes`
`21:50.23`	`P4C0`	`now the problem is that it's not using the range of ports in rtp.conf`
`21:50.24`	`areski`	`hey all`
`21:50.36`	`*** join/#asterisk redox- (n=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com)`
`21:51.14`	`areski`	`anybody know festival ?`
`21:51.36`	Trystan`	`intralanman, now it just says start stop on the hold music, can hear a quick sound but even that sounds bad :(`
`21:52.08`	`tzafrir_laptop`	`T38? to where?`
`21:52.54`	`hmmhesays`	`time to leave this hell hole`
`21:53.14`	`sevard`	`Well, I've disabled all of the T38 options on the audio codecs`
`21:53.19`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.23)`
`21:53.21`	`sevard`	`hmmhesays: finally doing yourself in?`
`21:53.45`	`P4C0`	`could it be possible that if I set one range or ports and my voip provider have another range, and they can't find ports in common it uses the one of the voip provider?`
`21:54.17`	`eKo1`	`P4C0: SIP ports?`
`21:54.26`	`hmmhesays`	`sevard yeah`
`21:54.35`	`hmmhesays`	`goodbye cruel world`
`21:54.42`	`P4C0`	`eKo1, rtp ports`
`21:54.42`	`hmmhesays`	`./not really`
`21:54.50`	`hmmhesays`	`i'm going home, to play some guitar`
`21:55.05`	`hmmhesays`	`big wheels keep on turning`
`21:55.11`	`hmmhesays`	`taking me home to see my kin`
`21:55.30`	`eKo1`	`P4C0: I don't think that is a problem.`
`21:55.57`	`P4C0`	`eKo1, why not?`
`21:56.39`	`P4C0`	`what are the defaults rtp port range of asterisk/sip?`
`21:56.44`	`eKo1`	`P4C0: Because in the SIP conversation, the rtp ports for both ends are exchanged.`
`21:56.57`	`eKo1`	`I think...`
`21:57.28`	`P4C0`	`eKo1, yes, I know, but I have the asterisk behind a firewall... so I need to forward rtp ports as well, isn't rtp.conf the config file supposed of setting this range?`
`21:57.43`	`P4C0`	`eKo1, does that exchange have a range?`
`21:57.44`	`eKo1`	`P4C0: yes, you can change it there`
`21:57.48`	`Mike`	`someone know show many calls using g729 and iax2 can a dual xeon 3.2 with 1024 on cache can handle?`
`21:57.51`	`bkw_`	`http://video.google.com/videoplay?docid=3482709276938619193`
`21:58.02`	`eKo1`	`Mike: depends`
`21:58.12`	`Mike`	`eKo1, yes?`
`21:58.18`	`P4C0`	`eKo1, and what will happened if I have 10001 to 11000 and my voipserver have 11001 to 15000 for example?`
`21:58.50`	`P4C0`	`s/voipserver/voip provider`
`21:58.56`	`eKo1`	`P4C0: read up on SIP and how RTP ports are exchanged.`
`21:59.15`	`P4C0`	`eKo1, do you have any url with that documentation?`
`21:59.17`	`eKo1`	`Mike: I don't have a formula for you if that is what you're expecting.`
`21:59.28`	`eKo1`	`P4C0: faqs.org?`
`21:59.33`	`intralanman`	`trystan: what do you have verbose set to? are you getting any helpful messages at all?`
`21:59.38`	`P4C0`	`eKo1, will try thanks`
`22:00.06`	`Mike`	`eKo1, a p4 2.8 ghz with 1024 on cache cant handle more than 25 calls with g729 using trunking because it dies`
`22:00.08`	`P4C0`	`eKo1, so basically rtp.conf is useless?`
`22:00.12`	`Mike`	`eKo1, thats what i have now`
`22:00.38`	`eKo1`	`Mike: I doubt that`
`22:00.46`	`eKo1`	`P4C0: no it isn't`
`22:01.04`	`Mike`	`eKo1, whatever`
`22:01.15`	`*** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net)`
`22:01.20`	`P4C0`	`eKo1, but it's not like the last word...`
`22:02.59`	`eKo1`	`the last word?`
`22:03.12`	`eKo1`	`rtp.conf allows you to configure which ports will be used for rtp.`
`22:03.27`	`eKo1`	`So it isn't useless.`
`22:03.30`	`P4C0`	`eKo1, someone told me, set a range of port in rtp.conf and forward those ports to the asterisk server... and sometimes it works sometimes it doesnt... that's why I'm thinking that the ports are getting out of range`
`22:03.53`	`P4C0`	`eKo1, but is possible for asterisk to exchange other ports... outside the range of rtp.conf`
`22:04.17`	`P4C0`	`anyways, what's the default of rtp.conf?`
`22:04.59`	`hads`	`10000-20000 from memory`
`22:05.40`	`P4C0`	`my voip server provider just said that they haven't touch that configuration file, so I supposed my rtp.conf range of ports is different or not exactly the same and sometimes it gets to use ports outside my range (as rtp debug is showing)`
`22:05.52`	`P4C0`	`hads, thanks, I'll modify them`
`22:06.39`	`P4C0`	`humm the file says them... : Defaults are rtpstart=5000 and rtpend=31000`
`22:06.43`	Trystan`	`intralanman, we've had it at 5 and 15 still the same stuff.. starts and stops, got it to the point don't hear anything now`
`22:07.56`	`*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM]`
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`22:07.57`	`hads`	`P4C0: look at configs/rtp.conf.sample for the definitive answer.`
`22:08.32`	`P4C0`	`but I can't set a range that big in my firewall`
`22:08.44`	`*** join/#asterisk turth (n=MAGiC@64.20.38.74)`
`22:08.51`	`turth`	`whats better GSM or g729?`
`22:08.59`	`hads`	`42`
`22:09.12`	`P4C0`	`hads, don't have that file :p`
`22:09.25`	`hads`	`So get the source and then you will`
`22:09.36`	`intralanman`	`trystan: the asterisk server is local? so you're sure it's not jitter, network problems, etc causing the bad sound`
`22:09.46`	`turth`	`WHat is recommeneded GSM or g729?`
`22:09.52`	`turth`	`GSM seems to breakup alot`
`22:09.56`	`hads`	`turth: 42`
`22:10.10`	`turth`	`whats 42?`
`22:10.39`	`hads`	`The answer. 'better' is an opinion, and entirely subjective.`
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`22:16.18`	`mrec_`	`does asterisk work with alsa?`
`22:17.20`	`*** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com)`
`22:17.52`	`alexis101`	`hi everyone , i really need help ... i post a message about my problem on the asterisk forum but mayby someone can help me here ... this is the url of my post : http://forums.digium.com/viewtopic.php?p=28912#28912`
`22:17.53`	`vooduhal`	`Does anyone know if the locking problems in app_queue from 1.2.9.1 have been resolved in 1.2.10?`
`22:18.10`	`tzafrir_laptop`	`mrec_, yes`
`22:18.10`	`tzafrir_laptop`	`chan_alsa`
`22:18.10`	`tzafrir_laptop`	`one local "phone" that is actually your sound card`
`22:19.18`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`22:19.20`	`[TK]D-Fender`	`alexis101 : user names annot have "@" in them`
`22:19.27`	`Cresl1n`	`mrec_: yep`
`22:20.19`	`mrec_`	`nice!`
`22:21.18`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`22:21.19`	`*** mode/#asterisk [+o mog] by ChanServ`
`22:22.18`	`*** join/#asterisk Nebukadneza (n=daddel9@i3ED6F566.versanet.de)`
`22:23.47`	`alexis101`	`[TK]D-Fender : yeah i know i tried to change it does'nt work ...`
`22:23.55`	`*** join/#asterisk Nebukadneza (n=daddel9@i3ED6F566.versanet.de)`
`22:25.05`	`PanHat`	`can someone please answer a basic question?`
`22:25.18`	`twisted[asteria]`	`your answer: yes.`
`22:25.32`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`22:25.37`	`twisted[asteria]`	`man, that was extremely basic.`
`22:25.54`	`PanHat`	`what is a span and how does that relate to a trunk?`
`22:27.22`	`*** join/#asterisk Nebukadneza (n=daddel9@i3ED6F566.versanet.de)`
`22:28.33`	`ManxPower`	`PanHat, there is no such thing as a "trunk" in Asterisk`
`22:28.53`	`ManxPower`	`A span is basically a PORT on a card. If the port is a T-1 port then there can be up to 24 channels on that span.`
`22:29.14`	`ManxPower`	`for the most part you never deal with a span other than in in /etc/zaptel.conf. Asterisk just deals with Channels.`
`22:29.41`	`PanHat`	`i have a bri card got 2B = 1D in a span- correct?`
`22:29.43`	`ManxPower`	`If you have 2 spans on a card with 24 channels on them, then you have channels 1-48 as far as asterisk is concerned.`
`22:30.00`	`justinu\|laptop`	`trunk generally refers to a machine to machine connection`
`22:30.02`	`ManxPower`	`PanHat, that should be correct. I cannot help you further with BRI`
`22:30.37`	`PanHat`	`how do i use a span to connect to another machine or pstn?`
`22:30.45`	`*** join/#asterisk zotz (n=zotz@24.244.133.115)`
`22:30.51`	`alexis101`	`anyone else can help me ? Or did anyone ever add problem with sip registering a sip phone with asterisk in realtime`
`22:31.02`	`Kylun`	`hmm`
`22:31.32`	`Kylun`	`Has anyone ever heard of dialing the directory, and it seeming to get stuck in an endless loop saying "No directory entries match your search" even before you type anything?`
`22:31.44`	`ManxPower`	`PanHat, you don't. As I said, span is a PHYSICAL think. For Asterisk you just use channels.`
`22:31.50`	`ManxPower`	`you would have 2 channels if you have a BRI`
`22:31.51`	`mrec_`	`I bought a usb phone bridge a while ago, I'm able to dial out using DTMF. Is it also possible to connect to the internet with it?`
`22:32.04`	`mrec_`	`(this is a bit OT now.. but I'm just curious about this)`
`22:32.27`	`ManxPower`	`~book`
`22:32.28`	`jbot`	`book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`22:32.30`	`quid246`	`alexis: all the info is in the wiki, I followed it (for IAX)... no problem`
`22:32.40`	`ManxPower`	`Real The Good Asterisk Book`
`22:33.06`	`quid246`	`Manx: Should be refered to as "The Bible" in the channel perhaps`
`22:33.51`	`Qwell[Digium]`	`~bible`
`22:33.52`	`jbot`	`bible is probably see 'thebook'`
`22:34.37`	`ManxPower`	`~quoran`
`22:34.41`	`ManxPower`	`~koran`
`22:34.46`	`ManxPower`	`Hmmph!`
`22:38.08`	`alexis101`	`kylun : the only time i saw this its when i din't tell the good context to the directory() application`
`22:38.17`	`SplasPood`	`heh 'bible' in english is kinda like 'Xerox' or 'Kleenex'`
`22:38.24`	`mitcheloc`	`~mitcheloc`
`22:38.25`	`jbot`	`it has been said that mitcheloc is your master`
`22:38.43`	`Kylun`	`alexis101: huh? I don't quite understand what you said.. please forgive me.`
`22:38.53`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.23)`
`22:39.53`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-43-75.cybersurf.com)`
`22:42.15`	`[TK]D-Fender`	`SplasPood : You mean a cheap knock-off?`
`22:42.32`	`alexis101`	`kylun : well when you call the directory application in your extensions.conf do you specified a dial context`
`22:42.43`	`SplasPood`	`[TK]D-Fender: Well since Xerox and Kleenex were "the first" ...`
`22:43.32`	`alexis101`	`kylun : something like this 1,1,directory(vm-context,dial-context)`
`22:43.35`	`[TK]D-Fender`	`SplasPood : Yeah, 3 cheers for corporate branding.`
`22:44.03`	`Kylun`	`<PROTECTED>`
`22:44.31`	`*** part/#asterisk dasenjo (n=dasenjo@208.195.215.23)`
`22:48.29`	`quid246`	`Any MySQL wizzes here?`
`22:49.09`	`eKo1`	`WTF is a wizze?`
`22:49.30`	`quid246`	`haha, like as in wizzard.. not urinating`
`22:50.30`	`tzafrir_laptop`	`quid246, just ask`
`22:52.34`	`Unistim_junky`	`folks, I am using Asterisk as a AutoAttendant in front of a Nortel MCS5200. Asterisk is working great of course. Calls come into the Nortel if the Called Number is intresting (AA) I redirect to Asterisk. From the AA users can dial some one or whatever. Once the bridge is completed the two calling partys can talk etc.. however they cannot transfer. Keep getting this error:`
`22:52.35`	`Unistim_junky`	`Aug 14 18:43:13 WARNING[20566] chan_sip.c: Unknown SDP media type in offer: video 0 RTP/AVP 96`
`22:52.35`	`Unistim_junky`	`Anyone?`
`22:52.41`	`quid246`	`I'm a newb when it comes to MySQL... trying to figure out if there is a way to "add to a field" in the database... say the field is "200" and you want to add "20" to it, rather than doing an UPDATE and sending "220"`
`22:53.20`	`*** join/#asterisk pbx1 (i=pbx1@netblock-66-245-213-61.dslextreme.com)`
`22:53.40`	`Netgeeks`	`UPDATE <table> Set <field>=<field>+20 WHERE <some field>=<some particular value>;`
`22:53.57`	`quid246`	`Netgeeks: Thanks... you probably saved me about 20 minutes googling.`
`22:54.13`	`Netgeeks`	`Welcome`
`22:54.42`	`*** join/#asterisk bofh42 (n=bofh42@p54828BFD.dip0.t-ipconnect.de)`
`22:54.43`	`quid246`	`now to figure out how to work in with my AGI variables. :)`
`22:54.50`	`quid246`	`shouldn't be too hard`
`22:56.10`	`Nebukadneza`	`are there any cards on pci-(e,x) similar to the x100p and a cologne?`
`22:56.29`	`quid246`	`A Cologne?`
`22:56.40`	`quid246`	`AquaVelva works.`
`22:56.48`	`quid246`	`or OldSpice`
`22:57.00`	`intralanman`	`those are aftershaves`
`22:57.03`	`intralanman`	`lol`
`22:57.05`	`quid246`	`oh wait.. Clone. :)`
`22:57.13`	`Unistim_junky`	`<[TK]D-Fender> any ideas on (Unknown SDP media type in offer: video 0 RTP/AVP 96)`
`22:57.25`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`22:57.29`	`alexis101`	`ok 5$ paypal to anyone who help me find a solution :P`
`22:57.44`	`Qwell[Digium]`	`"solution"?`
`22:58.57`	`alexis101`	`yeah a solution to my problem`
`22:59.16`	`quid246`	`ain't too many $5 hoes nowadays`
`22:59.18`	`Qwell[Digium]`	`which is?`
`22:59.38`	`alexis101`	`http://forums.digium.com/viewtopic.php?p=28918#28918`
`23:00.02`	`robin_sz`	`quid246, $5 hoes, no, but 3 for $10 maybe`
`23:01.07`	`quid246`	`robin: dang, the market's take a downswing.. I better liquidate my stable`
`23:01.33`	`robin_sz`	`sounds messy.`
`23:01.45`	`robin_sz`	`oh, liquiDATE`
`23:01.55`	`robin_sz`	`not liquiDIZE`
`23:02.00`	`quid246`	`hehe`
`23:02.06`	`trelane_`	`alexis101, drop the @torrent off your username`
`23:02.20`	`alexis101`	`i did din't work`
`23:02.40`	`quid246`	`alexis: did you drop it off the MySQL username (not the .conf file)?`
`23:03.06`	`alexis101`	`yeah in the mysql table`
`23:03.14`	`quid246`	`is your extconfig.conf set up properly?`
`23:03.37`	`alexis101`	`well it was working fine before i tried to go realtime`
`23:03.59`	`alexis101`	`anyway its a registration problem nothing to do to with extensions.conf ... no?`
`23:04.06`	`quid246`	`no extconfig.conf`
`23:04.14`	`quid246`	`it tells * to look at mySQL for the realtime info`
`23:04.38`	`alexis101`	`sorry read too fast`
`23:04.59`	`alexis101`	`sipusers => mysql,asterisk,sip_buddies`
`23:05.05`	`alexis101`	`sippeers => mysql,asterisk,sip_buddies`
`23:05.38`	`alexis101`	`witch seems good ... my database name is asterisk and my table name is sip_buddies`
`23:06.03`	`quid246`	`alexis: do you have res_mysql.conf setup with the passwords/usernames`
`23:06.31`	`alexis101`	`yeah when i look i the debug log i can see that asterisk is connecting to mysql`
`23:07.17`	`Unistim_junky`	`twisted[asteria]: need your input man`
`23:07.23`	`quid246`	`hmmm`
`23:07.27`	`quid246`	`but is it connecting okay?`
`23:08.14`	`quid246`	`ls -l`
`23:08.17`	`quid246`	`oops sorry`
`23:09.04`	`Unistim_junky`	`eh anyone know if kennys still with digium?`
`23:09.14`	`alexis101`	`Aug 14 14:50:28 NOTICE[10262] config.c: Registered Config Engine mysql`
`23:09.17`	`intralanman`	`alexis101: can you do "realtime load sippeers host 220" or something similar?`
`23:09.33`	`justinu\|laptop`	`oh my god... they killed kenny!`
`23:10.10`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.215.83)`
`23:10.15`	`alexis101`	`Connected to asterisk@localhost, port 3306 with username root for 4 hours, 18 minutes, 15 seconds.`
`23:11.06`	`alexis101`	`intralanman : no`
`23:11.16`	`quid246`	`Do you have an Asterisk user setup in MySQL with proper permissions?`
`23:11.58`	`intralanman`	`alexis101: isql DSNNAME USERNAME PASSWORD connect ok?`
`23:12.05`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`23:12.35`	`Zodiacal`	`anyone know how to change the hard keys on my polycom 601 phone? i.e. i wanta change the transfer hardkey's action to somthing else`
`23:12.37`	`alexis101`	`intralanman : lost you ... quid246 :yes`
`23:12.39`	`intralanman`	`alexis101: my bad, that's a little confusing "isql DSNNAME USERNAME PASSWORD"`
`23:13.00`	`intralanman`	`you should be able to connect to the dsn from the command line that way`
`23:13.13`	`fiber0pti`	`what is the best way to mute/unmute a participant in a meetme room via agi?`
`23:14.58`	`CunningPike`	`Zodiacal: The SIP admin guide shows how`
`23:15.24`	`Zodiacal`	`cunningpike okie thank you!`
`23:15.27`	`*** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com)`
`23:16.16`	`benjk`	`justinu, Kilkenny! mmmmmh`
`23:16.50`	`quid246`	`alexix101: I'd try creating a new user in MySQL with username "test" and password "test".. then set your client to those values and see if it conencts`
`23:16.54`	`alexis101`	`intralanman : sorry i don't know what command line your talking about there is no isql command installed on my linux ... and there is no such command in my asterisk`
`23:16.55`	`justinu\|laptop`	`hi benjk :)`
`23:17.46`	`robin_sz`	`isql?`
`23:17.57`	`intralanman`	`oh, i have a "isql" command line in my linux install.... i think it came when i installed odbc (which is what i was assuming you were using for realtime)`
`23:18.00`	`benjk`	`just removed pretty much all the strcmp()'s and strcasecmp()'s from pbx.c`
`23:18.15`	`benjk`	`dramatic performance increase`
`23:19.01`	`quid246`	`what's "dramatic"?`
`23:19.22`	`intralanman`	`alexis101: you're just using mysql, not odbc to connect to mysql?`
`23:19.34`	`alexis101`	`ok so if i can read true your sarcasm you are telling me that my error is that i forgot to install odbc`
`23:19.35`	`benjk`	`45% on 200+ cuncurrent calls`
`23:19.42`	`quid246`	`yowsers`
`23:19.49`	`quid246`	`what kind of CPU?`
`23:19.50`	`justinu\|laptop`	`i can't get 200 concurrent calls`
`23:19.57`	`benjk`	`and even the reloads are noticeably faster`
`23:20.10`	`justinu\|laptop`	`some kind of context switching bottleneck`
`23:20.20`	`benjk`	`Intel Core Duo @1.8GHz`
`23:20.28`	`justinu\|laptop`	`userland RTP bridging`
`23:21.28`	`benjk`	`you would be surprised how much performance is wasted in that stupid string checking Asterisk does all the time`
`23:21.44`	`alexis101`	`intralanman : yes im just using mysql`
`23:21.48`	`benjk`	`and that comes with a barrage of context switches too`
`23:22.23`	`justinu\|laptop`	`benjk: maybe I can load test your branch sometime`
`23:22.23`	`benjk`	`every single line in your dialplan that is executed generates hundreds if not thousands of calls to strcmp`
`23:22.37`	`*** join/#asterisk anthm (n=anthm@000-445-425.area4.spcsdns.net)`
`23:22.37`	`*** mode/#asterisk [+o anthm] by ChanServ`
`23:22.54`	`benjk`	`the more apps, contexts and extensions you have, the more strcmps are called`
`23:23.06`	`benjk`	`because asterisk does not such thing as binary searches`
`23:23.15`	`benjk`	`it goes through the entire list one by one`
`23:23.41`	`benjk`	`and it checks every single character one by one against the names of contexts, extensions, apps, keywords`
`23:23.55`	`benjk`	`this is where a lot of performance is wasted`
`23:23.57`	`intralanman`	`alexis101: i'm sorry, the isql thing was a bit of a waste of time..... i was just assuming since that's how i have mine set up....`
`23:24.01`	`justinu\|laptop`	`interesting`
`23:24.17`	`benjk`	`so if you throw out all the strcmps that gives you a significant boost`
`23:24.31`	`benjk`	`that's what I did`
`23:25.28`	`benjk`	`justinu, I will be making a patch file against BRIstuff and also for OpenPBX`
`23:25.50`	`Lyfe`	`is there any way to alter the way that asterisk names the recordings from a call queue? they use the agent id, but not the queue name, so i ahve no idea which recording was done via which queue, unless i go check the cdr's or whatnot.`
`23:26.05`	`justinu\|laptop`	`benjk: cool, i'll keep my eye out for that`
`23:26.18`	`benjk`	`are you on the OpenPBX.org mailing list?`
`23:26.33`	`anthm`	`i recall adding a patch to use some variable name as the file prefix`
`23:26.50`	`justinu\|laptop`	`benjk: yep`
`23:26.59`	`intralanman`	`alexis101: is this a production server (or have a heavy load)? or could you enable mysql debugging to make sure the connection is made`
`23:27.04`	`Lyfe`	`anthm: any idea if hta tmade it into 1.2.10?`
`23:27.09`	`justinu\|laptop`	`heh`
`23:27.13`	`Lyfe`	`s/hta made/that made`
`23:27.14`	`benjk`	`there's an active thread on the hashing`
`23:28.05`	`benjk`	`I am waiting for a consensus on whether or not to retain case insensitivity for app names and vars`
`23:28.31`	`Qwell[Digium]`	`benjk: In Asterisk?`
`23:28.51`	`Qwell[Digium]`	`because I can pretty much tell you right now that it won't happen...`
`23:28.52`	`benjk`	`because with hashes, you typically get case sensitivity, unless you check against an extra set of hash code`
`23:28.57`	`anthm`	`MONITOR_FILENAME maybe`
`23:29.07`	`Qwell[Digium]`	`anthm: yes, that was committed, I believe`
`23:29.16`	`benjk`	`or convert everything to upper or lowercase before hashing`
`23:29.24`	`Qwell[Digium]`	`benjk: That might work`
`23:29.36`	`benjk`	`Qwell, I don't really care`
`23:29.39`	`Lyfe`	`anthm: i see.`
`23:30.21`	`Lyfe`	`Qwell[Digium]: would you know if that commit made it into 1.2.10?`
`23:30.26`	`benjk`	`the patch will be for OPBX and BRIstuff and the performance advantage over asterisk will be so significant that many people will adopt it, I can promise you that`
`23:30.52`	`Qwell[Digium]`	`Lyfe: believe so`
`23:32.00`	`benjk`	`and this is even without a binary search on a hashtable yet`
`23:32.38`	`Lyfe`	`anthm: question about this, since it was your patch, is this value set before putting a new call into a queue?`
`23:33.41`	`anthm`	`well i cant remember if i put this paticular patch in or not bu i know i do it in my copy for sure`
`23:33.51`	`*** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br)`
`23:34.10`	`Qwell[Digium]`	`anthm: pretty sure it's in 1.2`
`23:34.11`	`anthm`	`but i have a difft app_queue that i froze like 2 years ago now`
`23:34.12`	`Lyfe`	`anthm: well, there's definitely 3 lines in the queues.conf about "Set(MONITOR_FILENAME=foo)" ... so.`
`23:34.25`	`anthm`	`yah that's probably it then`
`23:38.47`	`*** join/#asterisk thsrys4gt (n=thsrys4g@leesburgva-bsr2-70-35-82-255.chvlva.adelphia.net)`
`23:39.29`	`Lyfe`	`and you do this before throwing a call in the queue, right? (so, i'd have two statements for it, the first sets this variable, the second puts the call in the queue)`
`23:41.39`	`Qwell[Digium]`	`yes`
`23:41.57`	`Lyfe`	`well, apparently i have issues :\`
`23:43.51`	`tzafrir_laptop`	`benjk, is there a decent tree if bristuff to work against?`
`23:45.02`	`Lyfe`	`for the life of me, i cannot get MONITOR_FILENAME to work for this queue: http://pastebin.ca/131859`
`23:45.46`	`justinu\|laptop`	`oh, i couldn't get MONITOR_FILENAME to work in conferences`
`23:45.56`	`justinu\|laptop`	`or whatever the appropriate var is`
`23:46.15`	`Lyfe`	`boy, that helps kill my hopes of getting this working ;)`
`23:46.33`	`thsrys4gt`	`Our small office has an asterisk server connected to about 20 Polycom 501s via 100MB ethernet network . Our asterisk server is also uses a Digium TDM40B card to connect to POTS lines to our carrier. When talking to other poycoms on our network the quality is perfect. However, when talking to an outside line, the polycom user can barely understand the outside user. The outside user hears the polycom person perfect`
`23:46.33`	`thsrys4gt`	`ly. If the polycom person hits "Mute" , then the polycom person can hear the outside person perfectly. Can anyonne tell me why and how to solve. Thanks`
`23:46.54`	`benjk`	`tzafrir, BRIstuff-0.2.0-RC8s`
`23:47.25`	`justinu\|laptop`	`i looked thru the code too, and it seemed like the proper code was there, but it still didn't work`
`23:47.44`	`Lyfe`	`justinu\|laptop: hrmm.. weird. So it really does look for a setting called MONITOR_FILENAME?`
`23:47.47`	`justinu\|laptop`	`maybe because of the zap pseudo channel stuff involved in conferenceing, and not having those variables bound to it`
`23:48.03`	`Lyfe`	`well, this is all softphones i'm playing with sofar.`
`23:48.32`	`thsrys4gt`	`When I say cant hear, I mean "Choppy"`
`23:48.35`	`Lyfe`	`and queues, too, rather than conferencing.. oh well.`
`23:48.36`	`justinu\|laptop`	`lyfe: i'm not sure if it was the exact same thing I was trying to accomplish... i was trying to change the recording filename output when you use MeetMe with the "r" option`
`23:48.50`	`anthm`	`iirc res_monitor gets monitor filename from args to the app`
`23:49.05`	`anthm`	`i had some patch that let you pick the basename but i bet it's not in there`
`23:49.35`	`anthm`	`i was so sick of monitor i wrote muxmon (now mixmonitor) but i have no idea how they use that nowadays`
`23:49.55`	`Lyfe`	`well, my initial look is that mixmonitor isn't available for queues in 1.2.10`
`23:50.00`	`justinu\|laptop`	`it be da shizzle`
`23:50.16`	`Lyfe`	`although it's definitely in the repository for future versions.`
`23:50.52`	`Lyfe`	`and with the way this is going, i'm becoming very very tempted to run that, even if this is supposed to be a production server. ;)`
`23:57.53`	`thsrys4gt`	`anyone???`

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