irclog2html for #asterisk on 20061019

`00:01.41`	`ambriento`	`XSMerlin, why are you using in zaptel.conf: span=1,1,1,ccs,hdb3,crc4 instead of span=1,1,0,ccs,hdb3,crc4 ?`
`00:02.25`	`ambriento`	`Did you enable debug on pri? where's the pastebin of its output?`
`00:02.51`	`XSMerlin`	`those were the settings i use on systems that work fine witha single pri card. ill give it a try withthe other way`
`00:03.09`	`XSMerlin`	`i didnt get any new output after enabling the debugging`
`00:03.46`	`*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`00:03.47`	`kronic`	`if no context is specified in [general] of sip.conf is the last defined context used for inbound calls?`
`00:04.18`	`XSMerlin`	`no, changing htat 0 hasnt done the trick`
`00:05.01`	`*** join/#asterisk SplasPood (n=jwb@pool-71-125-210-154.nycmny.east.verizon.net)`
`00:06.14`	`ambriento`	`XSMerlin, let's go deeper. prin intense debug span 1`
`00:06.23`	`ambriento`	`you'll have a lot of stuff going on`
`00:06.24`	`XSMerlin`	`k`
`00:06.46`	`toerkeium`	`hello, does anyone know how JSPhone let you record a call ?`
`00:06.54`	`XSMerlin`	`ok, ill do a pastebin now`
`00:07.00`	`ambriento`	`then pri no debug to stop it`
`00:07.35`	`ambriento`	`XSMerlin, paste the output of pri show span 1 as well`
`00:07.58`	`XSMerlin`	`errr, pri no debug not workign`
`00:08.01`	`XSMerlin`	`:s`
`00:09.09`	`ambriento`	`o_O`
`00:09.13`	`XSMerlin`	`ahh, ok got it`
`00:09.22`	`XSMerlin`	`had to specify the span to stop on doh`
`00:09.36`	`ambriento`	`:) i missed that`
`00:09.48`	`XSMerlin`	`:)`
`00:10.46`	`XSMerlin`	`http://pastebin.ca/209063`
`00:10.51`	`*** part/#asterisk Primer (n=vi@sh.nu)`
`00:11.42`	`XSMerlin`	`http://pastebin.ca/index.php`
`00:12.25`	`*** join/#asterisk Buglouse (n=SourceRa@66.97.116.64)`
`00:12.36`	`XSMerlin`	`http://pastebin.ca/209065`
`00:13.50`	`ambriento`	`let's see it...`
`00:14.35`	`XSMerlin`	`?`
`00:14.55`	`XSMerlin`	`are those links not right?`
`00:15.28`	`ambriento`	`yes they are`
`00:15.35`	`ambriento`	`I'm looking at'em now`
`00:15.36`	`XSMerlin`	`k`
`00:15.40`	`XSMerlin`	`ta`
`00:15.56`	`ambriento`	`what is "ta" ?`
`00:16.16`	`XSMerlin`	`thanks (in british ;) )`
`00:16.55`	`ambriento`	`hmm.. I see`
`00:17.08`	`ambriento`	`tell me, do you have another boxes running * with PRI?`
`00:17.15`	`XSMerlin`	`i do`
`00:17.22`	`XSMerlin`	`but only witha pri card in`
`00:17.31`	`XSMerlin`	`not done one with a fxs b4`
`00:17.57`	`XSMerlin`	`why's that?`
`00:18.00`	`niZon`	`anyone ever used an audiocodes SIP gateway with asterisk`
`00:18.01`	`ambriento`	`and do they, by any chance, have less than 30 voice channel?`
`00:18.10`	`niZon`	`?`
`00:18.12`	`XSMerlin`	`yep`
`00:18.13`	`XSMerlin`	`8`
`00:18.36`	`ambriento`	`even with 8 channels only, the channel 16 is always the D-Channel?`
`00:18.43`	`XSMerlin`	`yer`
`00:18.47`	`kronic`	`http://pastebin.com/809416`
`00:18.59`	`XSMerlin`	`i can chack, hang on`
`00:19.19`	`kronic`	`in the above, I'd like inbound calls for mydid1 to go to inbound_mycontext1, it seems however they jump to the last specified context: outbound_mycontext3`
`00:19.32`	`ambriento`	`I never did less than 30 B-Chan b4 :)`
`00:19.45`	`XSMerlin`	`heh`
`00:19.53`	`*** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net)`
`00:20.02`	`XSMerlin`	`the company i work for insist on selling pri lines for some reaon`
`00:20.44`	`JT`	`uses heaps less cable`
`00:20.49`	`JT`	`that's probably a good reason`
`00:20.58`	`*** join/#asterisk saftsack (n=oliver@p54A7D1BC.dip.t-dialin.net)`
`00:21.06`	`XSMerlin`	`yeah, iv configed 16 as the b channel on the other boxes`
`00:21.27`	`XSMerlin`	`jsut looged in to check em`
`00:21.34`	`ambriento`	`D you meant, right`
`00:21.42`	`XSMerlin`	`yer, sorry`
`00:21.47`	`XSMerlin`	`d channel b layer`
`00:21.58`	`JT`	`ts16 is always the d channel on e1`
`00:22.09`	`ambriento`	`ty JT`
`00:22.28`	`ambriento`	`I wasn't that sure about it`
`00:22.46`	`XSMerlin`	`so this config should be workign....`
`00:22.57`	`JT`	`unless the line from the telco isn't working`
`00:23.05`	`ambriento`	`yep, the last resort is talk to the carrier`
`00:23.05`	`JT`	`is the line known good?`
`00:23.08`	`XSMerlin`	`its plugged into t atrend device`
`00:23.20`	`XSMerlin`	`an NT emulator`
`00:23.23`	`XSMerlin`	`so i know its workign`
`00:23.24`	`saftsack`	`whats the difference between sangoma and digium pri cards?`
`00:23.50`	`JT`	`you're lucky to have one of those :)`
`00:23.55`	`XSMerlin`	`if i were to plug this into the pstn BT would kick me off after 6 reconnects in an hour`
`00:23.57`	`saftsack`	`loel`
`00:24.12`	`XSMerlin`	`yeah, borrowed it from work`
`00:24.24`	`XSMerlin`	`top piece of kit, id be lost withut it`
`00:24.40`	`ambriento`	`I wish I could had one of those :)`
`00:24.42`	`saftsack`	`does someone of you has a patton gw?`
`00:25.05`	`XSMerlin`	`:)`
`00:26.54`	`ambriento`	`saftsack, sangoma has like 20 years of experiece on making wan cards`
`00:27.23`	`saftsack`	`ambriento, sounds good. i didnt want to ask here why sangoma cards are better ... but now i does. so can you tell me, why?`
`00:28.49`	`ambriento`	`Digium cards are OK too`
`00:29.02`	`ambriento`	`they work pretty nice, specially the new ones`
`00:29.24`	`ambriento`	`but if you take one sangoma in your hands you can feel the difference`
`00:29.36`	`saftsack`	`<PROTECTED>`
`00:29.40`	`saftsack`	`so they are not perfect ^^`
`00:30.00`	`saftsack`	`what do you mean feel? do you hear the difference? i mean they have the same ec`
`00:30.08`	`ambriento`	`I said the new ones because I had some trouble with old ones`
`00:30.28`	`saftsack`	`that was that what i tried to say`
`00:30.32`	`ambriento`	`by feel, I was talking about the looking of the card`
`00:30.38`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:30.38`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:31.24`	`saftsack`	`ambriento, why can you feel the difference?`
`00:32.11`	`ambriento`	`sangoma gives you a lot more of debugging options than digium`
`00:32.24`	`saftsack`	`thats a good thing. what about the sound quality?`
`00:32.38`	`ambriento`	`you can feel that you have more control over what's happenning in your box`
`00:33.09`	`saftsack`	`sounds good. and furthermore sangomas are compatible to more systems than digium cards, right?`
`00:33.17`	`ambriento`	`i never tested a digium with Hardware EC`
`00:33.23`	`XSMerlin`	`w00t`
`00:33.33`	`XSMerlin`	`restartign the trend seems to have cleared`
`00:33.33`	`ambriento`	`only sangoma`
`00:33.35`	`XSMerlin`	`<PROTECTED>`
`00:33.47`	`XSMerlin`	`thanks loads ambriento`
`00:33.50`	`saftsack`	`ambriento, you bought a digium card without embedded ec? Oo`
`00:33.53`	`ambriento`	`XSMerlin, that trend is broken`
`00:34.01`	`XSMerlin`	`i really couldnt have done it without you`
`00:34.01`	`kronic`	`anyone?`
`00:34.09`	`bsdfreak`	`:)`
`00:34.12`	`ambriento`	`send it to me. I'll throw it away for you. go get a new one`
`00:34.22`	`kronic`	`sorry to push the button, I'm under pressure here :(`
`00:34.37`	`ambriento`	`saftsack, yep.`
`00:34.50`	`XSMerlin`	`lol, sure :P`
`00:34.59`	`ambriento`	`kronic, let me try to help you, gimme a second. that wil be the last for the night`
`00:35.09`	`saftsack`	`ambriento, you had mass echo, didnt you?`
`00:35.39`	`kronic`	`thanks mate`
`00:35.58`	`JT`	`saftsack: why do you need echo cancellation on a digital card?`
`00:36.17`	`ambriento`	`I can say when I change from plain digium to a HEC sangoma iit was a huge difference`
`00:36.33`	`saftsack`	`because i need echocancellation on a bri card, too. this was my experience`
`00:36.47`	`Mooriz`	`I got a te110p card`
`00:36.51`	`JT`	`must be a far end echo problem`
`00:36.57`	`JT`	`digital doesn't have AF echo`
`00:37.02`	`*** join/#asterisk drfreeze (n=Jim@www.freeze.org)`
`00:37.06`	`drfreeze`	`hi`
`00:37.09`	`saftsack`	`JT, why do you need EC on ip telephony? its digital technology too`
`00:37.17`	`saftsack`	`Mooriz, do you experience echo?`
`00:37.17`	`*** join/#asterisk remmo (n=chatzill@smack.isp.net.au)`
`00:37.21`	`drfreeze`	`Anyone know if a TDM board will fit in a 1U or 2U server?`
`00:37.46`	`saftsack`	`ambriento, do you know sangomas prices?`
`00:37.53`	`JT`	`saftsack: echo is usually caused by shit handsets, or analogue lines`
`00:38.12`	`Mooriz`	`I did have some echo, turned on echo cancellation and the echo has gone... My external party now gets scratchy audio`
`00:38.18`	`saftsack`	`why do i have echo then with a snom 320 ip telephone connected over bri to the pstn?`
`00:38.24`	`JT`	`drfreeze: that'd probably depend upon your 1 or 2RU server, wouldn't it`
`00:38.40`	`JT`	`saftsack: what's on the other end?`
`00:38.55`	`Qwell`	`saftsack: echo comes from the far end`
`00:39.29`	`*** part/#asterisk Juxt (n=juxt@66.151.46.42)`
`00:39.47`	`saftsack`	`whats on the other hand? :> my crystal bowl just falled down.`
`00:40.00`	`JT`	`err okay`
`00:40.05`	`saftsack`	`there are a lot of different people on the other end.`
`00:40.08`	`ambriento`	`saftsack, idk about prices :(`
`00:40.16`	`saftsack`	`idk?`
`00:40.17`	`*** join/#asterisk JoseBravo (n=jbravo@ut-cable-126-44.uniweb.net.co)`
`00:40.18`	`saftsack`	`ok`
`00:40.45`	`*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)`
`00:40.50`	`saftsack`	`JT, did you get me?`
`00:41.04`	`ambriento`	`kronic, can you pastebin a entire call? and please use pastebin.ca if possible :)`
`00:41.15`	`ambriento`	`saftsack, I don't know`
`00:41.23`	`XSMerlin`	`i need to be heading bedwards, thanks again`
`00:42.59`	`saftsack`	`Qwell, you have many different far ends, so you must have an ec ...`
`00:43.00`	`kronic`	`ambriento: will do`
`00:43.16`	`Mooriz`	`I am using a te100p. I am running a dual core cpu, messed around with acpi settings etc. when I make a call over PRI the remote party hears scratchy audio. Calls internally are fine. Besides a complete replacing of hardware can anyone think of a reason of why this is so?`
`00:44.08`	`Mooriz`	`te110p even`
`00:44.41`	`JT`	`Mooriz: any irq sharing?`
`00:44.44`	`Mooriz`	`none`
`00:45.17`	`Mooriz`	`Im stumped... I have gone through most of the "audio problem" faqs.`
`00:45.39`	`Mooriz`	`SIS chipset on the MB though...`
`00:46.04`	`Mooriz`	`tried all the hdparm settings`
`00:46.17`	`JT`	`Mooriz: what does zttest say?`
`00:46.40`	`Mooriz`	`99.987793% 99.987793% 99.987793% 99.987793% 99.328613% 99.987793% 99.987793%`
`00:46.57`	`Mooriz`	`Results after 9 passes Best: 100.000000 -- Worst: 99.328613 -- Average: 99.915907`
`00:48.05`	`JT`	`that's decent`
`00:48.08`	`Mooriz`	`yeah...`
`00:48.17`	`Mooriz`	`hence my confusion`
`00:48.42`	`drfreeze`	`JT: why would it depend upon the server? I thought all 1U's were the same`
`00:48.48`	`kronic`	`ambriento: http://pastebin.ca/209108`
`00:48.49`	`Mooriz`	`Its a core 2 6300.....`
`00:49.07`	`JT`	`drfreeze: i dunno, some might have bigger 1U slots than others`
`00:49.12`	`JT`	`you also said 2U`
`00:49.18`	`Mooriz`	`(overkill for what it is doing :)`
`00:49.27`	`JT`	`i don't think you can get a tdm400p in half size`
`00:49.34`	`JT`	`considering it needs the space for the modules`
`00:49.38`	`JT`	`half height that is`
`00:49.39`	`drfreeze`	`JT: 1U or 2U. Haven't bought yet`
`00:49.47`	`JT`	`it's not a full length card though`
`00:50.04`	`drfreeze`	`Yeah, it's the height I'm worried about.`
`00:50.11`	`JT`	`drfreeze: it won't fit in a half height pci slot as a lot of 1U servers commonly have`
`00:50.29`	`Mooriz`	`Tried setting up SMP IRQ Affinity to stick the Te110p on its own cpu`
`00:50.52`	`drfreeze`	`JT: figured as much for the 1U. Would be nice if it would fit in a 2U`
`00:50.58`	`JT`	`differen't manufacturers do different things with 2U, that's for sure`
`00:51.09`	`JT`	`i've seen 2U servers that fit 6 full height pci cards`
`00:51.21`	`Mooriz`	`2RU also sometimes have riser cards...`
`00:51.31`	`JT`	`with risers obviously...`
`00:51.45`	`drfreeze`	`hmm, not familar with RU`
`00:52.07`	`JT`	`RU = U`
`00:52.09`	`Mooriz`	`Rack unit`
`00:52.10`	`JT`	`rack unit`
`00:52.32`	`ambriento`	`kronic, I think it's too little for a entire call... for now, get rif sip debug and set verbose to 10 at least and show me the console output`
`00:53.11`	`drfreeze`	`k`
`00:54.34`	`kronic`	`its not an entire call, basically the call comes in, and instead of matching the desired context inbound_mycontext1, it jumps to the last defined context`
`00:55.01`	`Mooriz`	`drfreeze: check the info for the case/mb you are looking at. If it has a riser card included generally most cards will fit. 2RU servers are more likely to come with a riser/pci slots, however I have seen them for 1Ru cases.`
`00:55.17`	`JoseBravo`	`I have a X100P, when I type: ztcfg -vvv I can see the channel 01 is configured: "Channel 01: FXS Kewlstart (Default) (Slaves: 01)"`
`00:56.02`	`JoseBravo`	`At moment I can call from exten to exten, but how can I call from my x100p?`
`00:56.07`	`JoseBravo`	`Its aleady connected`
`00:56.14`	`Mooriz`	`JT: so you think Im nearly at the point of throw out MB/CPU and try another one?`
`00:56.15`	`ambriento`	`kronic, rif = rid of*`
`00:56.38`	`JT`	`Mooriz: i have no idea, you should probably speak to some other people first :)`
`00:57.25`	`Mooriz`	`yeah, I have some more that I can try it on... (P4 where I can disable hyper threading). I might try that first and see how it goes.`
`00:58.20`	`kronic`	`ambriento: that's the thing, nothing is produced in the console, I had both debug and verbose at 99`
`00:58.28`	`kronic`	`ambriento: I had to use sip debug to find the issue`
`01:00.41`	`ambriento`	`hmm`
`01:00.43`	`ambriento`	`I see`
`01:01.01`	`drfreeze`	`Mooriz: thx`
`01:01.14`	*** join/#asterisk blue`sky (n=crazy_gu@ppp-70-225-183-47.dsl.chmpil.ameritech.net)
`01:04.29`	`*** join/#asterisk Yarrick40k (n=test@adsl-68-77-241-138.dsl.milwwi.ameritech.net)`
`01:06.38`	`ambriento`	`kronic, sip show settings will tell you the default context`
`01:07.56`	`kronic`	`its default`
`01:08.34`	`kronic`	`I think manipulated the context setting in [general], this had no effect`
`01:08.36`	`ambriento`	`what do you have in your default context?`
`01:08.43`	`kronic`	`nothing`
`01:08.54`	`kronic`	`like I said, it jumps to the last context specified`
`01:08.59`	`kronic`	`*defined, rather`
`01:09.31`	`*** join/#asterisk BigBadHoss_Work (n=hoss@68-191-130-114.static.dctr.al.charter.com)`
`01:09.51`	`BigBadHoss_Work`	`hey guys whats a good SIP softphone to use with *`
`01:11.26`	`*** join/#asterisk ucfMethod (i=ucfMetho@69.143.112.70)`
`01:11.34`	`saftsack`	`i would guess polycoms`
`01:11.48`	`ambriento`	`kronic, pm`
`01:11.56`	`BigBadHoss_Work`	`a softphone anyone?`
`01:12.15`	`justinu\|laptop`	`~softphone`
`01:12.16`	`jbot`	`something that should be drug out into the street and shot`
`01:12.31`	`BigBadHoss_Work`	`comeon guys`
`01:12.45`	`justinu\|laptop`	`there aren't any really good ones`
`01:12.48`	`BigBadHoss_Work`	`unless someone wants to send me a polycom 4000`
`01:12.55`	`Snake-Eyes`	`lol`
`01:13.10`	`BigBadHoss_Work`	`that ekiga is shit for asterisk`
`01:13.23`	`BigBadHoss_Work`	`x-lite is bollocks but does work alright`
`01:14.13`	`justinu\|laptop`	`x lite is probably the best free one`
`01:14.13`	`Snake-Eyes`	`sjphone and xlite alright`
`01:14.13`	`BigBadHoss_Work`	`havent tried sjphone`
`01:14.13`	`Mooriz`	`Yeah I have gone through http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html`
`01:14.13`	`Snake-Eyes`	`the linux version doesnt have a fancy gui`
`01:14.13`	`saftsack`	`whats with linphone? or is it for windows?`
`01:14.39`	`Snake-Eyes`	`i think linphone is for lindows :P`
`01:14.57`	`BigBadHoss_Work`	`lol`
`01:15.01`	`BigBadHoss_Work`	`thats great`
`01:15.06`	`BigBadHoss_Work`	`i remember trying that os`
`01:15.20`	`Snake-Eyes`	`hehe`
`01:15.56`	`*** part/#asterisk drfreeze (n=Jim@www.freeze.org)`
`01:17.11`	`Snake-Eyes`	`~x-lite`
`01:17.26`	`Snake-Eyes`	`~sjphone`
`01:17.27`	`jbot`	`sjphone is, like, a VOIP softphone that allows you to speak with any PC, PDA, stand-alone IP-phone, legacy wired or mobile phone (using VOIP gateway or purchasing service). It supports both SIP and H.323 standards. http://www.sjlabs.com/sjp.html`
`01:17.47`	`Snake-Eyes`	`~linphone`
`01:17.48`	`jbot`	`somebody said linphone was a SIP VOIP phone. To configure it to use sip.handhelds.org, ask ibot about linphone config . not working with fwd.pulver.com`
`01:18.56`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`01:20.17`	`*** join/#asterisk lule (i=lule@host129.201-253-77.telecom.net.ar)`
`01:20.19`	`shmaltz`	`I'm having trouble with a Polycom 501 that has been upgraded from 2.x.x bootrom to 3.2.1, at first they were having sip to sip echo, so I changed the sip.cfg to the one that comes with the 1.6.x firmware, but now they complain that the volume is too low, what am I missing?`
`01:20.54`	`BigBadHoss_Work`	`using different config files probably wasnt the problem`
`01:20.58`	`*** part/#asterisk lule (i=lule@host129.201-253-77.telecom.net.ar)`
`01:24.57`	`shmaltz`	`BigBadHoss_Work, I disagree, because the echo went away just by changing it.`
`01:25.34`	`BigBadHoss_Work`	`maybe AEC was disabled in the old config`
`01:25.49`	`BigBadHoss_Work`	`i have a fleet of 500s and they work fine`
`01:26.00`	`shmaltz`	`BigBadHoss_Work, then why with the old bootrom there was no echo?`
`01:26.09`	`JoseBravo`	`If my x100p have a red alaram with zttol It will not work?`
`01:26.21`	`shmaltz`	`BigBadHoss, same here, but have you updated one from 2.xx to 3.x.x?`
`01:26.40`	`BigBadHoss_Work`	`no, i dont use the features of it`
`01:27.25`	`shmaltz`	`BigBadHoss_Work, so here is you answer, I would have not upgraded either, but I got a new phone that didn't want to downgrade and it came preloaded to 3.x.x`
`01:27.37`	`BigBadHoss_Work`	`ooh too bad`
`01:27.48`	`BigBadHoss_Work`	`they wont downgrade`
`01:28.07`	`shmaltz`	`no, it can't you can't go from 3.x.x to 2.x.x`
`01:30.15`	`BigBadHoss_Work`	`ok another question`
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`01:30.31`	`shmaltz`	`what time is it?`
`01:30.33`	`BigBadHoss_Work`	`whats a really badass hardphone`
`01:30.34`	`shmaltz`	`~time`
`01:30.38`	`jbot`	`You are educated stupid and therefore too dumb to understand nature's perfect time cube! (2006.10.19 1:30:38 GMT)`
`01:30.50`	`shmaltz`	`BigBadHoss, grandstream,`
`01:30.53`	`BigBadHoss_Work`	`naw`
`01:30.57`	`BigBadHoss_Work`	`too cheap`
`01:31.11`	`Malawar`	`have it inlayed with diamonds then`
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`01:33.05`	`Malawar`	`wow`
`01:33.09`	`Malawar`	`grandstream phones ARE pretty cheap`
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`01:35.12`	`Snake-Eyes`	`yea`
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`01:36.18`	`BigBadHoss_Work`	`im used to polycoms`
`01:36.25`	`BigBadHoss_Work`	`i dont want some buggy thing`
`01:36.37`	`Snake-Eyes`	`grandstream has alot of firmware updates`
`01:36.48`	`Snake-Eyes`	`its not that buggie`
`01:37.04`	`Malawar`	`heh`
`01:37.12`	`Malawar`	`if they update firmware that much there must be some bugs somewhere`
`01:37.17`	`BigBadHoss_Work`	`yeah`
`01:37.17`	`Snake-Eyes`	`good for where some one just needs a phone with a few features`
`01:37.47`	`Snake-Eyes`	`updates introduce bug fixes but also features which create new bugs`
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`01:39.26`	`Snake-Eyes`	`http://www.voip-info.org/wiki/view/GXP-2000`
`01:39.26`	`shmaltz`	`gs phones are not worth $50`
`01:39.42`	`shmaltz`	`I got 2 on my desk you want them come and pick em up, its urs free`
`01:40.27`	`Snake-Eyes`	`i think there are too many ppl here who live on cloud 9`
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`01:40.42`	`BigBadHoss_Work`	`where u @?`
`01:40.46`	`ShadowHntr`	`shmaltz: if price weren't an issue, i'd go cisco all the way of course. :P`
`01:41.05`	`shmaltz`	`NJ`
`01:41.11`	`BigBadHoss_Work`	`cisco's sip is bullshit`
`01:41.16`	`shmaltz`	`nah, Polycom is better than Cisco`
`01:41.21`	`sumasuma`	`hi friends, anyone familier with the stun type terms ?`
`01:41.22`	`BigBadHoss_Work`	`yeah`
`01:41.38`	`SplasPood`	`Polycom > Cisco`
`01:41.41`	`SplasPood`	`but Cisco Config`
`01:41.45`	`SplasPood`	`> Polycom`
`01:41.48`	`ShadowHntr`	`hehehehe`
`01:41.53`	`BigBadHoss_Work`	`yeah`
`01:42.05`	`BigBadHoss_Work`	`i've provisioned all of my phones`
`01:42.13`	`Snake-Eyes`	`so shmaltz tell me what phone you could get for $50 ?`
`01:42.13`	`Snake-Eyes`	`thats better`
`01:42.22`	`sumasuma`	`IndependentFilter, DependentFilter, PortDependentFilter, DependentMapping !!! ??? anyone familier with these terms ?`
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`01:42.34`	`shmaltz`	`Snake-Eyes, a panasonic desk phone for pots`
`01:42.36`	`SplasPood`	`To be honest, I've found the grandstream budgetone POS to be actually pretty reasonable for the price`
`01:42.45`	`BigBadHoss_Work`	`i got my polycom 500s for $105`
`01:42.46`	`BigBadHoss_Work`	`shipped`
`01:42.54`	`SplasPood`	`good deal`
`01:42.55`	`SplasPood`	`from whom?`
`01:42.58`	`SplasPood`	`although 500 is old`
`01:42.59`	`BigBadHoss_Work`	`ebay`
`01:43.02`	`SplasPood`	`ahh.`
`01:43.02`	`shmaltz`	`the gs phones don't do their job, so it's not a good phone`
`01:43.04`	`SplasPood`	`not such a good deal`
`01:43.08`	`BigBadHoss_Work`	`the only difference is ram`
`01:43.13`	`SplasPood`	`shmaltz: eh?`
`01:43.17`	`BigBadHoss_Work`	`as long as it does sip and MWI im fine`
`01:44.52`	`Snake-Eyes`	`yea`
`01:44.52`	`SplasPood`	`ya freenode`
`01:44.52`	`BigBadHoss_Work`	`NETSPLIT!`
`01:44.52`	`shmaltz`	`I gtg, bbs`
`01:44.52`	`Snake-Eyes`	`with gs they do there job, at the end of the day you get what you pay for. I think to many ppl expect the gs = some top cisco or polycom lol`
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`01:45.42`	`Snake-Eyes`	`hmm seems only half the ppl came back`
`01:46.35`	`BigBadHoss_Work`	`anythign better than polycom`
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`01:47.18`	`Snake-Eyes`	`i hate the web interface on the polycoms, made me stop using it`
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`01:47.56`	`BigBadHoss_Work`	`who uses the web interface`
`01:47.56`	`Snake-Eyes`	`it feels the model i used just didnt have enough cpu grunt to do more than 1 op at a time, and the webserver on the phone was the slowest thing in history`
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`01:48.00`	`BigBadHoss_Work`	`i agree it blows`
`01:48.00`	`Snake-Eyes`	`for testing features`
`01:48.22`	`BigBadHoss_Work`	`yeah i guess`
`01:48.50`	`Snake-Eyes`	`i wouldnt use it for mass deployment (most likly turn it off), but the fact that polycom found that acceptable.....`
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`01:51.23`	`Snake-Eyes`	`gs problems but at least they try to fix them, and are getting less of my money for there efforts compared to polycom`
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`01:51.44`	`BigBadHoss_Work`	`guess what just happened`
`01:51.52`	`BigBadHoss_Work`	`i called charter tech support right`
`01:52.05`	`BigBadHoss_Work`	`and they transferred me to Nu Wave Ovens!!!`
`01:52.29`	`BigBadHoss_Work`	`insane`
`01:52.55`	`Snake-Eyes`	`...`
`01:52.56`	`Druken`	`did you order a Nu Wave Oven ?`
`01:53.24`	`Snake-Eyes`	`lol`
`01:53.24`	`BigBadHoss_Work`	`i said what the fuck is this shit\`
`01:53.26`	`BigBadHoss_Work`	`they hund up on me :)`
`01:53.29`	`Micko113`	`hey guys, I have spare P2 500mhz desktop, what else do I need to experiment with asterisk ?`
`01:53.40`	`Snake-Eyes`	`hmm Nu Wave Ovens = hippies maybe :P`
`01:53.44`	`BigBadHoss_Work`	`a fxs card`
`01:53.58`	`BigBadHoss_Work`	`and an fxo card if you want to use the public network`
`01:54.25`	`ShadowHntr`	`are the Tornado phones any good?`
`01:54.39`	`Druken`	`tornado phones?`
`01:54.40`	`Micko113`	`are those cards expensive ?`
`01:54.51`	`ShadowHntr`	`www.lenoxa.com`
`01:54.59`	`ShadowHntr`	`http://www.voip-info.org/wiki/view/VOIP+Phones+Reviews`
`01:55.10`	`Druken`	`Micko113: no.. but get yourself either an ATA or a voip telephone`
`01:55.17`	`Druken`	`not an FXS port...`
`01:56.08`	`BigBadHoss_Work`	`cisco ata`
`01:56.16`	`BigBadHoss_Work`	`or the iaxy`
`01:56.19`	`BigBadHoss_Work`	`from digium`
`01:56.29`	`Druken`	`linksys works fine...`
`01:57.03`	`Micko113`	`are any of those avalible in local stores?`
`01:57.07`	`Snake-Eyes`	`~tornado`
`01:57.08`	`jbot`	`rumour has it, tornado is A powerful twisting storm, often with rain and hail.`
`01:57.20`	`Snake-Eyes`	`hehe`
`01:57.28`	`BigBadHoss_Work`	`linksys~cisco`
`01:57.53`	`Druken`	`BigBadHoss: I know that, but the commoners don't...`
`01:58.00`	`JT`	`yay for excel spreadsheets`
`01:58.07`	`Snake-Eyes`	`Micko113, they should be`
`01:58.19`	`JT`	`with any luck, i will have designed this junghanns octoBRI crossover cable correctly`
`01:58.20`	`Druken`	`Micko113: no... you won't find voip stuff at staples or the source... hehehe`
`01:58.28`	`Micko113`	`cool thanks guys!`
`01:58.28`	`Snake-Eyes`	`fear the new cisco/linksys-One products coming`
`01:58.58`	`BigBadHoss_Work`	`especially not in washington :)`
`01:59.19`	`Snake-Eyes`	`dont get a dlink what ever you do (they're every where here)`
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`01:59.33`	`Micko113`	`hehe, how about in alabama BigBadHoss_Work`
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`02:02.41`	`Micko113`	`interesting guide for newbies like me http://www.automated.it/guidetoasterisk.htm`
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`02:02.50`	`BigBadHoss_Work`	`well, i live about 10 miles from digium`
`02:04.27`	`napkin`	`can anyone suggest a good canadian voip provider? i need 4 trunks in montreal. reliability and price are the important factors, not support/features.`
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`02:06.09`	`Adam12`	`napkin: I've had good service from les.net`
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`02:06.30`	`napkin`	`Adam12: i called them today actually and was really impressed just by their honesty`
`02:06.58`	`Adam12`	`napkin: Yeah. I'm suprised Les hasn't ignored me after all the emails with questions I sent him :)`
`02:07.08`	`napkin`	`his name is les?`
`02:07.14`	`Adam12`	`Yeah`
`02:07.22`	`napkin`	`where are your trunks and where are you located?`
`02:08.14`	`Adam12`	`I have a St. Catharines DID, and a Toll-free DID. I'm physically located in the Niagara region.`
`02:08.21`	`napkin`	`i thought babytel would be good, as they're in montreal - but they act like a real phone company - $45 per trunk!`
`02:08.54`	`Adam12`	`I use les.net for both origination/termination.`
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`02:09.20`	`napkin`	`cool, you find their servers are pretty quick?`
`02:09.21`	`Adam12`	`There's a company up in Cornwall (or Kingston, cant' remember) that sounded fairly decent as well, but no personal experience with them.`
`02:09.24`	`Adam12`	`Yeah`
`02:09.45`	`napkin`	`this will be my first voip experience, i'm afraid of latency/echo :)`
`02:09.56`	`Adam12`	`And he's adding new features every few days. I noticed one 1-hour downtime but he emailed me shortly after I noticed it (automatically).`
`02:10.07`	`Adam12`	`Yeah, he's my first DID provider. So far so good.`
`02:10.30`	`napkin`	`do you notice a second or two of latency?`
`02:10.43`	`napkin`	`i suppose it's rare to get less.`
`02:11.02`	`Adam12`	`.. not too much, I suppose.`
`02:11.08`	`Adam12`	`Actually, nothing at all`
`02:11.20`	`Adam12`	`The routes I have to him arent' too bad.`
`02:11.46`	`napkin`	`which IP should i trace to get a feeling`
`02:11.53`	`Adam12`	`He uses did.voip.les.net`
`02:12.18`	`napkin`	`hm, i get 22ms latency :)`
`02:12.24`	`napkin`	`that's pretty good`
`02:12.43`	`Adam12`	`At the price, it wouldn't hurt to get one just to test. He has instant provisioning once you are signed up. You can pick the number you want from the stock he has on hand.`
`02:12.46`	`Druken`	`les.net is good`
`02:12.49`	`Druken`	`i use it`
`02:14.59`	`napkin`	`Adam12: that's true.`
`02:16.18`	`*** join/#asterisk digijohn (n=john@hiro.rh.rit.edu)`
`02:16.20`	`*** part/#asterisk digijohn (n=john@hiro.rh.rit.edu)`
`02:18.05`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`02:18.28`	`Adam12`	`Hey benjk.`
`02:18.51`	`napkin`	`what about unlimitel? anybody?`
`02:20.52`	`napkin`	`we only need 1 phone number, but need to be able to receive up to 4 calls at once - how do i describe that? we only need 1 DID right? (i'm new to the terms)`
`02:21.12`	`napkin`	`1 DID but 4 trunks?`
`02:21.34`	`Adam12`	`Hmm I think those are referred to as 'channels'`
`02:24.12`	`*** join/#asterisk Crashsys (n=kumba@146-210.124-70.tampabay.res.rr.com)`
`02:24.50`	`benjk`	`hi Adam12`
`02:24.50`	`Crashsys`	`Would a 1-gig processor be enough for 12-lines w/ MOH/VM/Queues?`
`02:25.05`	`Crashsys`	`No recording/meetme/conferencing...`
`02:27.18`	`Druken`	`Crashsys: what about translation?`
`02:27.53`	`Crashsys`	`the phones are u-law, and the lines are T1...`
`02:27.59`	`Crashsys`	`so u-law to u-law`
`02:28.08`	`Crashsys`	`1-gig be pushing it?`
`02:30.43`	`Druken`	`should do it fine....`
`02:31.14`	`Druken`	`napkin: where do you need the DID ?`
`02:31.21`	`napkin`	`montreal`
`02:31.29`	`Druken`	`ahh... 514`
`02:31.37`	`Crashsys`	`Druken: Thanks for the input...`
`02:31.48`	`Druken`	`np`
`02:32.24`	`Crashsys`	`Think there'd still be enough CPU left over for MySQL CDR logging?`
`02:32.38`	`shmaltz`	`I'm having trouble with a Polycom 501 that has been upgraded from 2.x.x bootrom to 3.2.1, at first they were having sip to sip echo, so I changed the sip.cfg to the one that comes with the 1.6.x firmware, but now they complain that the volume is too low, what am I missing?`
`02:32.42`	`Crashsys`	`system might, at max, handle 400 calls per day`
`02:33.07`	`Druken`	`Crashsys: logging doesn't take alot of cpu, it's the lookups that take the power`
`02:33.21`	`Crashsys`	`Hmmm...`
`02:33.46`	`Crashsys`	`well, I might just store the SQL Db on the other server... since it's not doing anything realtime...`
`02:35.10`	`Crashsys`	`too bad zaptel wont work on OBSD...`
`02:35.53`	`*** join/#asterisk jo3sm1th (i=jo3sm1th@204.sub-75-195-237.myvzw.com)`
`02:35.55`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`02:37.29`	`Crashsys`	`Anyone interested in a Digium TE205p 3rd Gen card that's 2-months old?`
`02:37.48`	`JT`	`fire sale?`
`02:37.57`	`Crashsys`	`Yeah, $600 shipped...`
`02:38.13`	`JT`	`ah, still not enough of a firesale for me ;)`
`02:38.24`	`Crashsys`	`No biggie...`
`02:38.26`	`JT`	`considering it would be for personal use...`
`02:38.30`	`Damin`	`shmaltz: Sounds like your users need to turn up the volume. ;)`
`02:38.36`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`02:38.48`	`shmaltz`	`Damin, thanks, I wish it would be taht easy`
`02:38.57`	`Crashsys`	`jt: Still about $300 under retail... if I dont find a buyer by the weekend i'll just e-bay it...`
`02:39.41`	`benjk`	`hi JT`
`02:39.50`	`benjk`	`did you manage to get your HFC card working?`
`02:40.22`	`JT`	`benjk: still working on it, i have actually been doing other things`
`02:40.23`	`Crashsys`	`I'd trade the TE205p for a Sangoma Remore with 12 ports on it... straight trade...`
`02:40.55`	`*** join/#asterisk lule (i=lule@host167.200-117-184.telecom.net.ar)`
`02:41.04`	`Crashsys`	`12fxo`
`02:41.08`	`JT`	`benjk: i just crimped an OctoBRI crossover cable`
`02:41.19`	`JT`	`of my design, since there are none of the net`
`02:41.22`	`JT`	`hopefully it works`
`02:41.26`	`*** part/#asterisk lule (i=lule@host167.200-117-184.telecom.net.ar)`
`02:43.14`	*** join/#asterisk blue`sky (n=crazy_gu@ppp-70-225-178-198.dsl.chmpil.ameritech.net)
`02:43.18`	`benjk`	`I keep my fingers crossed ;)`
`02:45.50`	`JT`	`i make the pairs cross :D`
`02:47.00`	`Crashsys`	`I tried making an AC cross-over cable once... it didn't work out too well...`
`02:47.19`	`JT`	`it's AC... why does it matter...`
`02:48.23`	`Crashsys`	`Cause hot to ground = bad`
`02:48.38`	`JT`	`well i suppose if you put it that way, yes, it is`
`02:48.50`	`JT`	`i thought you meant you crosswired an equipment lead`
`02:48.57`	`JT`	`which is a moderate safety concern`
`02:49.11`	`Crashsys`	`Naaa... that just makes things act funky...`
`02:49.13`	`JT`	`but not as major concern as a dead short across the circuit`
`02:49.17`	`Crashsys`	`and/or tempermental`
`02:49.31`	`Damin`	`I crossed over a cross over cable once..`
`02:49.38`	`Damin`	`That didn't work out too well either..`
`02:50.03`	`JT`	`benjk: if you're interested, i can send you the .xls or something similar of the pinout if it works`
`02:50.04`	`Adam12`	`darmin: Hmm, you made a straight through cable ? :P`
`02:51.30`	`*** part/#asterisk hyphen (n=hyphen@71.224.213.97)`
`02:51.42`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`02:51.42`	`*** mode/#asterisk [+o mog] by ChanServ`
`02:53.40`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`02:56.57`	`Micko113`	`guys, would good configured asterisk give me free long distance or international calls ?`
`02:57.21`	`*** join/#asterisk switch (n=switch@saya.attrition.jp)`
`03:02.10`	`james_`	`if you find a good, free long distance or international provider to configure it with, sure`
`03:03.44`	`Druken`	`napkin: if you have a problem finding a good carrier, give me a shout, i can help ya out`
`03:04.22`	`napkin`	`Druken: problems??? $!@!!@$@!$!$@$@! telecommunication companies are nothing but problems!!!!`
`03:04.35`	`Druken`	`generally....`
`03:05.14`	`napkin`	`i'm refering to the insanity that entails looking through site after site showing photos of manic corporate drones holding telephones up to their ears`
`03:05.42`	`napkin`	`equally manic women usually looking at them and laughing - why?`
`03:05.52`	`Adam12`	`Haha`
`03:05.54`	`james_`	`haha, it's terrible isn't it`
`03:06.00`	`napkin`	`don't forget the perspective shots that make everyone look like midgets`
`03:06.07`	`Druken`	`cause they just found out just how small his dick really is?`
`03:06.33`	`james_`	`"i know what will set us apart, being and looking exactly like all the others!"`
`03:06.34`	`napkin`	`maybe, yeah, and he thinks voip is SO COOL :P`
`03:09.31`	`shmaltz`	`I'm having trouble with a Polycom 501 that has been upgraded from 2.x.x bootrom to 3.2.1, at first they were having sip to sip echo, so I changed the sip.cfg to the one that comes with the 1.6.x firmware, but now they complain that the volume is too low, what am I missing?`
`03:11.52`	`benjk`	`JT, yeah, that might come in handy at some point, thanks`
`03:12.36`	`*** join/#asterisk damanivu (n=damanivu@ip68-4-207-173.oc.oc.cox.net)`
`03:16.59`	`JT`	`benjk: i'll let you know if it works later :)`
`03:17.34`	`BigBadHoss`	`shmaltz`
`03:17.38`	*** join/#asterisk blue`sky (n=crazy_gu@ppp-70-225-186-126.dsl.chmpil.ameritech.net)
`03:17.45`	`shmaltz`	`BigBadHoss, yes`
`03:17.46`	`BigBadHoss`	`pastebin your current config`
`03:17.49`	blue`sky	`anyone know the SIP programming tutorial website ?`
`03:17.58`	`*** join/#asterisk nowwhat (n=ARF@24-148-32-14.nwb-bsr1.chi-nwb.il.cable.rcn.com)`
`03:17.58`	`shmaltz`	`BigBadHoss, what config?`
`03:18.12`	`BigBadHoss`	`sip.cfg and phone1.cfg`
`03:18.21`	`BigBadHoss`	`or w/e the name is`
`03:18.42`	`BigBadHoss`	`i've got a 200 page printed manual, mayeb i can check it over`
`03:18.54`	`*** join/#asterisk tengulre (n=tengulre@221.11.5.182)`
`03:19.38`	`*** join/#asterisk mds2 (n=mds@thewife.inspirednetworks.co.nz)`
`03:19.45`	`shmaltz`	`BigBadHoss, I doubt it will help you much but anyhow, I'll do it`
`03:19.48`	`mds2`	`hola`
`03:20.18`	`nowwhat`	`anyone have time to answer a (more then likely) simple question?`
`03:20.27`	`BigBadHoss`	`always`
`03:20.28`	`mds2`	`Can anyone point me at some documentation for programming softkeys on Cisco 7940 handsets?`
`03:20.47`	`shmaltz`	`mds2, you can't program them`
`03:20.49`	blue`sky	`anyone know the SIP programming tutorial website ?`
`03:20.49`	`BigBadHoss`	`the echo did go away didnt it`
`03:20.57`	`shmaltz`	`mds2, you can disable some of them, but thats it`
`03:21.17`	`shmaltz`	blue`sky, its here: http://www.voip-info.org/
`03:22.09`	blue`sky	`thanks : shmaltz`
`03:22.34`	`nowwhat`	`I've been goofing around with the configuation files all day. Getting instructions via asteriskguru and similar websites, but when I try to connect an outgoing call the CLI on asterisk responds 8472516592@test not found. I'm pretty sure I have the _9xxxxxxxxxx setup correctly in extention.conf. although, mabey not.`
`03:23.05`	`mds2`	`shmaltz: you're kidding? what's the point of softkeys if you can't program them?`
`03:23.07`	`nowwhat`	`any number dialed from a soft client get "@test" included and "not found"`
`03:23.25`	`BigBadHoss`	`did you type test somewhere`
`03:23.47`	`shmaltz`	`mds2, ask Cisco`
`03:23.48`	`mds2`	`shmaltz: to set DND on these things I have to press 4 keys in sequence. I'm trying to simplify that, is there a better way?`
`03:23.54`	`nowwhat`	`I've grep all the files that I've touched and I don't remember inserting test anywhere`
`03:24.05`	`shmaltz`	`mds2, not to my knowledge`
`03:24.15`	`shmaltz`	`cisco is way overated when used with sip`
`03:24.18`	`FuriousGeorge`	`hey shmaltz`
`03:24.20`	`BigBadHoss`	`definitely`
`03:24.28`	`BigBadHoss`	`get a polycom :)`
`03:24.51`	`shmaltz`	`hey Furious`
`03:24.58`	`mds2`	`shmaltz: does CallManager do things to cisco handset softkey behaviour? or is it all in the fancy xml services menu?`
`03:25.06`	`mds2`	`which is also a few keypresses until you get to anything useful`
`03:25.19`	`BigBadHoss`	`its all in the skinny version`
`03:25.23`	`BigBadHoss`	`or sccp i think`
`03:25.31`	`BigBadHoss`	`dont use cisco`
`03:25.47`	`mds2`	`too late, already got a stack of them :(`
`03:25.49`	`bungalow`	`anyone know of a function that would return the Local/foo,1 channel if I have the Local/foo,2 channel?`
`03:25.54`	`shmaltz`	`mds2, never tried either`
`03:25.55`	`mds2`	`polycom handsets > cisco?`
`03:26.07`	`mds2`	`shmaltz, thanks`
`03:26.12`	`mds2`	`I'll harass some Cisco heads`
`03:26.23`	`BigBadHoss`	`dont bother`
`03:26.33`	`BigBadHoss`	`you might as well go to #freebsd`
`03:26.43`	`mds2`	`haven't Cisco converted CallManager to SIP?`
`03:26.55`	`BigBadHoss`	`yes but the verswion is differrnt`
`03:27.00`	`mds2`	`and therefore you'd expect the SIP code to get a whole lot more featured in the near future?`
`03:27.06`	`BigBadHoss`	`no`
`03:27.06`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)`
`03:27.13`	`mds2`	`oh, it's callmanager-sip, not everyone-else-sip?`
`03:27.18`	`BigBadHoss`	`they want you to buy thier stuff`
`03:27.29`	`benjk`	`vendor lock-in`
`03:27.29`	`BigBadHoss`	`call manager uses sccp i think`
`03:27.31`	`mds2`	`I want to use * :p`
`03:27.44`	`BigBadHoss`	`mgcp i mean`
`03:27.51`	`BigBadHoss`	`or whatever its called`
`03:27.53`	`benjk`	`no its SCCP`
`03:27.59`	`BigBadHoss`	`yeah mgcp is polycom`
`03:28.05`	`benjk`	`as in Skinny Client Control Protocol`
`03:28.53`	`nowwhat`	`anyone have a suggestion for a better soft client then MediaX?`
`03:28.57`	`benjk`	`not to be confused with SCCP as in SS7's Service Connection Control Part`
`03:28.57`	`mds2`	`yeah, I thought I read something recently that said they'd dropped skinny/sccp and were going with SIP`
`03:29.11`	`mds2`	`ahh, acronyms`
`03:29.28`	`benjk`	`right now, Cisco are doing both`
`03:29.31`	`*** join/#asterisk hb2hb (i=hb2hb@wishmaster.vttc.donpac.ru)`
`03:29.33`	`*** part/#asterisk hb2hb (i=hb2hb@wishmaster.vttc.donpac.ru)`
`03:29.52`	`BigBadHoss`	`but sccp is older, more mature`
`03:30.15`	`mds2`	`benjk, yes with separate loads. does callmanager use either protocol to talk to endpoints? or sccp only?`
`03:30.24`	`mds2`	`how mature is * sccp support?`
`03:30.31`	`BigBadHoss`	`its ok`
`03:30.32`	`shmaltz`	`I'm having trouble with a Polycom 501 that has been upgraded from 2.x.x bootrom to 3.2.1, at first they were having sip to sip echo, so I changed the sip.cfg to the one that comes with the 1.6.x firmware, but now they complain that the volume is too low, what am I missing?`
`03:30.32`	`benjk`	`not very mature`
`03:30.49`	`benjk`	`there are actually two versions`
`03:30.50`	`BigBadHoss`	`shmaltz, upload the file to pastebin.ca`
`03:30.55`	`benjk`	`chan_skinny and chan_sccp`
`03:31.10`	`benjk`	`the folks who made chan_sccp weren't happy with chan_skinny`
`03:31.22`	`mds2`	`yeah, I remember reading about it ages ago and it wasn't really viable`
`03:31.40`	`mds2`	`but if it gives the 79xxs a better feature set on the handsets maybe it's worth investigating`
`03:31.56`	`benjk`	`not through * though`
`03:32.03`	`mds2`	`pants`
`03:32.16`	`mds2`	`what's the * sip handset of choice then?`
`03:32.33`	`BigBadHoss`	`polycom or grandstream (yuck)`
`03:32.38`	`benjk`	`that's almost a religious question`
`03:32.48`	`mds2`	`:)`
`03:32.53`	`mds2`	`I have a few grandstreams`
`03:32.53`	`benjk`	`ask three people and you get four answers`
`03:32.57`	`mds2`	`and the users actually love them`
`03:32.58`	`BigBadHoss`	`how much dough do you have`
`03:33.19`	`shmaltz`	`don't even say polycom and grandstream in the same sentance`
`03:33.21`	`mds2`	`well, we've already bought Cisco + licences`
`03:33.28`	`BigBadHoss`	`i said yuck!`
`03:33.39`	`mds2`	`so something in that $ range`
`03:33.46`	`benjk`	`Cisco's are quite nice but rhey are a pain to set up`
`03:33.52`	`BigBadHoss`	`how many ;omes do you need`
`03:33.55`	`BigBadHoss`	`lines`
`03:34.05`	`benjk`	`especially if you have to convert the firmware to sip first`
`03:34.31`	`BigBadHoss`	`i would get polycom 501 or 650`
`03:34.49`	`BigBadHoss`	`make sure you order SIP firmware though`
`03:34.55`	`BigBadHoss`	`they dont support changign`
`03:35.02`	`mds2`	`2 would be nice`
`03:35.10`	`mds2`	`but 1 is really all we need`
`03:35.10`	`BigBadHoss`	`from mgcp to sip`
`03:35.37`	`BigBadHoss`	`well then a 301 will work, but you would like the 501 better`
`03:35.42`	`BigBadHoss`	`its a nicer phone`
`03:35.48`	`mds2`	`benjk, the ciscos are crappy from an end-user point of view.`
`03:35.56`	`mds2`	`what sort of price range?`
`03:36.21`	`BigBadHoss`	`well, i got some old 500s for 105, but the 501s are more`
`03:36.22`	`benjk`	`like I said, it's almost religious`
`03:36.28`	`BigBadHoss`	`i think like 200-250`
`03:36.29`	`BigBadHoss`	`lemm google`
`03:36.38`	`benjk`	`many of my clients will never ever consider anything other than Cisco phones`
`03:36.52`	`BigBadHoss`	`190-200`
`03:36.59`	`BigBadHoss`	`just because of the name, right?`
`03:37.05`	`benjk`	`if I was to tell them they are no good, that would be the end of doing business with them`
`03:37.14`	`BigBadHoss`	`who cares`
`03:37.21`	`BigBadHoss`	`if they want shit let'em have it!`
`03:37.41`	`mds2`	`not sure who`
`03:37.45`	`benjk`	`they want them because they like the big screens, the design and the robustness`
`03:37.50`	`mds2`	`+distributes polycom here`
`03:37.54`	`benjk`	`you can beat the shit out of them`
`03:38.02`	`benjk`	`they won't crack open or fail`
`03:38.07`	`benjk`	`they can take heavy beatings`
`03:38.19`	`benjk`	`very important on trading floors`
`03:38.22`	`BigBadHoss`	`what about with a sledgehammer or cinderblock?`
`03:38.28`	`mds2`	`how does linksys/sipura rate?`
`03:38.36`	`benjk`	`trading floors feature immense equipment abuse`
`03:38.41`	`BigBadHoss`	`grandstream quailty`
`03:38.49`	`BigBadHoss`	`a little better`
`03:38.53`	`mds2`	`seriously?`
`03:38.57`	`wmandra`	`benjk: i can vouch for that.. had a 7940 drop 4 feet, not a scratch and still works perfect`
`03:39.16`	`BigBadHoss`	`if you want quality go with a cisco or polycom`
`03:39.38`	`BigBadHoss`	`for quantity go for a gstream or linksys`
`03:39.43`	`shmaltz`	`nah, don't use Cisco`
`03:39.50`	`BigBadHoss`	`not mentioned is snom`
`03:39.59`	`benjk`	`never go with Grandstream for business use`
`03:40.00`	`BigBadHoss`	`which make pretty good phones i've heard`
`03:40.10`	`benjk`	`only for domestic use, when you only make calls on the weekend`
`03:40.24`	`BigBadHoss`	`polycom needs to make a color screen phone`
`03:40.24`	`Crashsys`	`Someone needs to create a polycom config-file generator... where you can set all the options in a nice neat lil gui and it spits out the phoneX.cfg for ya :)`
`03:40.27`	`benjk`	`like putting an IP phone in your parents' house or so`
`03:40.32`	`BigBadHoss`	`there is one`
`03:40.45`	`BigBadHoss`	`its a module for freepbx i think`
`03:40.48`	`BigBadHoss`	`its buggy though`
`03:40.50`	`Crashsys`	`Polycom's just need a backlight :)`
`03:40.56`	`BigBadHoss`	`definuitely`
`03:41.04`	`Crashsys`	`IMHO anyways`
`03:41.24`	`BigBadHoss`	`and they've listened and improved nat support`
`03:41.28`	`BigBadHoss`	`on recent releases`
`03:41.29`	`benjk`	`no backlight is a no go for me`
`03:42.04`	`BigBadHoss`	`yeah, its a bummer`
`03:42.05`	`Crashsys`	`Polycom told a friend of mine that it couldn't put in a backlight because they were worried about breaking the 15w limit on PoE...`
`03:42.19`	`BigBadHoss`	`they must be crazy`
`03:42.31`	`Crashsys`	`yet their IP601 can be expanded to like 96-ports or something... with LED's on the keys...`
`03:42.32`	`BigBadHoss`	`how much is peak draw now?`
`03:42.45`	`Crashsys`	`The adapters are 12v@1amp... so 12w...`
`03:42.57`	`BigBadHoss`	`peak`
`03:43.05`	`Crashsys`	`dunno`
`03:43.07`	`Crashsys`	`never measured`
`03:43.09`	`BigBadHoss`	`an led is like`
`03:43.23`	`BigBadHoss`	`a picowatt`
`03:43.39`	`Crashsys`	`cant imagine a backlight adding more then a watt or two...`
`03:44.44`	`JT`	`maybe they're already riding the limit`
`03:44.46`	`BigBadHoss`	`http://www.theledlight.com/120-VAC-LEDbulbs.html`
`03:44.53`	`BigBadHoss`	`not if thier supply is 15w`
`03:45.11`	`BigBadHoss`	`and thier adapter outputs 12`
`03:45.26`	`JT`	`Crashsys said the PoE limit of 15W`
`03:45.39`	`JT`	`voltage is irrelevant, PoE uses -48VDC anyway`
`03:45.54`	`BigBadHoss`	`well how much is it drawing if not under 12w`
`03:46.15`	`BigBadHoss`	`the ac-dc couldnt power it if it was more`
`03:46.33`	`BigBadHoss`	`i bet its because they don't want to change the design`
`03:46.37`	`BigBadHoss`	`for one little thing`
`03:46.45`	`BigBadHoss`	`pity they didnt put them in newer ones`
`03:47.38`	`Crashsys`	`http://www.networkworld.com/details/4681.html`
`03:48.08`	`Crashsys`	`12.95 watts is the 802.3af spec for max sustained load`
`03:49.02`	`BigBadHoss`	`they are pushing it`
`03:51.13`	`BigBadHoss`	`the new 650 DOES have a backlight`
`03:51.34`	`BigBadHoss`	`about the price of a cisco high-end`
`03:51.47`	`Crashsys`	`LOL...`
`03:51.52`	`Crashsys`	`$400 phone?`
`03:52.23`	`BigBadHoss`	`# Display 320 x 160 backlit grayscale graphical LCD`
`03:52.23`	`BigBadHoss`	`# White LED backlight with custom intensity control`
`03:52.27`	`Crashsys`	`I use grandstream phone for people who dont make their business with phone sales...`
`03:52.34`	`Crashsys`	`polycom's for others...`
`03:52.41`	`Crashsys`	`too bad the 430 doesn't have a backlight`
`03:52.44`	`BigBadHoss`	`yeah but if youre going to get a 7960 or a polycom 650`
`03:53.13`	`*** join/#asterisk bmg505 (n=leon@c1-237-4.rndf.isadsl.co.za)`
`03:53.17`	`BigBadHoss`	`that would be te sweet spot`
`03:53.42`	`Crashsys`	`I'd prolly go for the polycom...`
`03:53.46`	`BigBadHoss`	`speaker+blight`
`03:53.49`	`BigBadHoss`	`yeah`
`03:53.56`	`Crashsys`	`Everything i've ever seen about cisco's is they are flaky...`
`03:54.11`	`BigBadHoss`	`with SIP yes`
`03:54.19`	`BigBadHoss`	`sccp they are pretty good`
`03:54.22`	`Crashsys`	`polycom's just have a PITA config setup...`
`03:54.35`	`BigBadHoss`	`its easy if you're doing lots`
`03:54.41`	`BigBadHoss`	`just use provisioning`
`03:54.50`	`Crashsys`	`Yeah, I do...`
`03:54.58`	`Crashsys`	`but that first config file is still a PITA`
`03:55.02`	`BigBadHoss`	`yeah`
`03:55.06`	`bungalow`	`anyone here have any familiar with the asterisk api?`
`03:55.11`	`BigBadHoss`	`well`
`03:55.20`	`BigBadHoss`	`time to play the splinter cell demo on 360`
`03:55.23`	`BigBadHoss`	`bbl`
`03:57.49`	`*** join/#asterisk maverickbna (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`04:03.15`	`DoDaT69`	`anyone done any lcs and asterisk shiat?`
`04:05.28`	`*** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au)`
`04:06.46`	`mosty`	`i want something like a GotoIf from my dialplan that checks if a particular phone is set to DND, is there a way to do that?`
`04:07.56`	`*** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-b40a8978344a4702)`
`04:09.31`	`*** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net)`
`04:09.55`	`wmandra`	`DoDaT69: What are you looking for??`
`04:11.02`	`wmandra`	`mosty: yes, you can do it all in the extensions.conf file`
`04:11.22`	`mosty`	`wmandra, how can i check if a phone is set to DND?`
`04:12.26`	`wmandra`	`use the asterisk database`
`04:12.27`	`*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)`
`04:14.21`	`mosty`	`what family/key do i look for?`
`04:14.31`	`wmandra`	`most: create a special extension say *77 to enable / disable DND ......`
`04:14.45`	`wmandra`	`you will need to come up wtth your own`
`04:14.56`	`wmandra`	`say DND/exten = on or off`
`04:15.03`	`mosty`	`this phone (snom 320) has a DND button, can i use that?`
`04:15.45`	`mosty`	`hmm ok`
`04:16.03`	`mosty`	`that would work as a fallback, but i would prefer to use the DND button if possible`
`04:16.12`	`wmandra`	`actually just looking at the wiki it's built into * as 78 79`
`04:17.21`	`wmandra`	`scratch that... someone needs to update the wiki..... i guess i'll add it to my list`
`04:18.07`	`mosty`	`looks like that was only for zap lines, but you could implement it in the dialplan`
`04:18.37`	`wmandra`	`anyway, the DND buttons on the phones all behave differently... each manufacturer has their own implementation... since * has to support such a wide base of devices it would really be unwise to add code for each case`
`04:18.57`	`wmandra`	`i actually have it implemented in my dialplan using the * DB and it works fine`
`04:19.22`	`benjk`	`can you not program the button?`
`04:19.27`	`JT`	`err wouldn't people usually be able to program what the DND button does?`
`04:19.38`	`wmandra`	`although, I'm using Communicator and LCS to control it`
`04:19.57`	`mosty`	`benjk, i've been looking through the phone's web interface, it doesn't seem to be programmable`
`04:20.07`	`wmandra`	`JT, sometmes yes, sometimes no... it depends on the manufacturer`
`04:20.13`	`JT`	`ouch`
`04:20.17`	`benjk`	`what does the button do?`
`04:20.26`	`Crashsys`	`On my polycom when I hit DND it sends a sip response of 486... wouldn't that make * move to the next priority task on the dialplan?`
`04:20.36`	`Crashsys`	`Right now it causes it to fallthrough to congestion`
`04:21.04`	`mosty`	`benjk, it sends a SIP response 480 "Do Not Disturb"`
`04:21.10`	`JT`	`damn sip phones really need to standardise`
`04:21.20`	`mosty`	`which asterisk then sees as congestion`
`04:21.44`	`benjk`	`then you can modify chan_sip to pick up that response and set the astdb entry from there`
`04:22.12`	`wmandra`	`480 is Temporarily unavailable not DND`
`04:22.57`	`mosty`	`ok so it looks simplest to just put 78 / 79 in my dialplan then, if it is to work with other phones as well`
`04:23.08`	`Crashsys`	`err 486... when I hit DND on my polycom it sends back a sip 486...`
`04:23.22`	`wmandra`	`reading further i see the RFC mentions DND under 480, but based on the entire description i can see why * would disconnect the call`
`04:23.25`	`Crashsys`	`which causes asterisk to fallthrough to congestion instead of picking up on the next priority exten`
`04:24.48`	`wmandra`	`486 is Busy Here .... callee is currently not willing or able to take any additional calls..... it would make sense to enable DND on that response`
`04:25.29`	`Crashsys`	`wmandra: My polycom's send back a sip 486 response... but then * just gives congestion to the caller and hangs up...`
`04:25.47`	`Crashsys`	`and my next priority is to send to voicemail`
`04:25.52`	`mds2`	`how simple is it to replace the * DB with MySQL?`
`04:26.07`	`wmandra`	`looking at chan_sip.c .......`
`04:26.22`	`mosty`	`mds2: look at the asterisk realtime stuff on the wiki`
`04:26.22`	`Corydon76-home`	`mds2: you cannot replace it`
`04:26.33`	`benjk`	`of course you can`
`04:26.47`	`Corydon76-home`	`However, you can supplant much of its functionality with an external database`
`04:27.22`	`Crashsys`	`How do I make asterisk look like a cute blonde secretary with very loose morals?`
`04:27.28`	`benjk`	`some people have replaced astdb's berkeley db with sqlite`
`04:27.46`	`benjk`	`you need to get your hands dirty a bit, but its doable`
`04:27.46`	`wmandra`	`crashsys, when you find out let me know :)`
`04:28.18`	`benjk`	`point is that you can change the core so it uses whatever db engine you like`
`04:28.32`	`mds2`	`so you can replace the backend that astdb talks to?`
`04:28.47`	`benjk`	`yes, but like I said, you need to get your hands dirty`
`04:28.48`	`mds2`	`or you can use different function calls that talk to xSQL backend`
`04:28.53`	`Corydon76-home`	`That's like saying that you can fly with a handgun; you just need to get your own plane and a pilot license`
`04:28.58`	`benjk`	`its not a configuration thing`
`04:29.02`	`mds2`	`the latter being the cleaner way`
`04:29.06`	`JT`	`and a gun license?`
`04:29.26`	`benjk`	`people decide for yourselves`
`04:29.47`	`benjk`	`open source is all about being able to modify the code so it does what you want it to do for you`
`04:29.51`	`Corydon76-home`	`The question was if it was easy. The answer is no.`
`04:30.29`	`Corydon76-home`	`benjk: that is a next to useless answer when it comes to users`
`04:30.55`	`benjk`	`if you can't do it yourself, you can always pay somebody to do it for you`
`04:31.08`	`benjk`	`another benefit of open source`
`04:31.22`	`benjk`	`its not that much effort`
`04:31.44`	`benjk`	`somebody who knows what they are doing can probably do it over a rainy weekend`
`04:31.44`	`Corydon76-home`	`mds2: were you planning on either modifying the source or paying someone to do it, or simply to use existing methods of configuration to change the behavior?`
`04:32.40`	`Corydon76-home`	`Are you implying it can't be done on a sunny weekend?`
`04:32.47`	`Corydon76-home`	`What about a snowy weekend?`
`04:33.01`	`benjk`	`every heard about the expression "figure of speech" ?`
`04:33.23`	`Corydon76-home`	`Not since you starting splitting hairs about what a user asked`
`04:33.58`	`mds2`	`corydon: the simplest approach. supplanting astdb's functionality with an external database sounds like me, what should I read?`
`04:34.01`	`*** join/#asterisk J4k3 (n=jsuter@dhcp-12-197-128-58.intrastar.net)`
`04:34.22`	`Corydon76-home`	`mds2: depends on which functionality you are interested in using`
`04:34.48`	`Corydon76-home`	`mds2: generally the easiest method is to use func_odbc on svncommunity`
`04:34.51`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`04:35.12`	`Corydon76-home`	`mds2: that will give you a generic interface to querying and updating an external database from the dialplan`
`04:35.31`	`mds2`	`right`
`04:36.13`	`Crashsys`	`Here's my dial macro... when my polycome replies back with a sIP 486, it plays congestion and hangs up, instead of voicemail... Any idears? http://pastebin.ca/209417`
`04:36.15`	`Corydon76-home`	`mds2: however, there are a lot of applications that use the astdb implicitly. These cannot be altered in their choice of astdb, except by rewriting the source`
`04:37.09`	`Corydon76-home`	`chan_sip, for example, saves details of registered clients in the astdb for persistence purposes`
`04:40.00`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`04:41.49`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`04:42.45`	`wmandra`	`crashsys: what version of ast erisk are you using`
`04:43.38`	`Crashsys`	`1.2.12.1`
`04:43.45`	`wmandra`	`s,101 is not needed.....`
`04:44.01`	`Crashsys`	`Hmmm... 1.2.13 just posted... interesting...`
`04:44.08`	`wmandra`	`with a 486 chan_sip.c send back AST_CONTROL_BUSY`
`04:44.22`	`Crashsys`	`But why is it auto-fallthrough to congestion then hangingup?`
`04:44.23`	`JT`	`Crashsys: chan skinny had a local root exploit i believe`
`04:46.27`	`Crashsys`	`Is it because i'm running it as a macro?`
`04:47.17`	`wmandra`	`crashsys: try setting verosity to 3 and see if you get any notifications in the CLI`
`04:47.46`	`Crashsys`	`The notification I get is "Sip response 486 (Busy Here), autofallthrough`
`04:47.54`	`Crashsys`	`then it plays congestion, and disconnects`
`04:49.07`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`04:49.17`	`hmmhesays`	`sweet geebus i'm drunk`
`04:49.44`	`J4k3`	`awwww yeah`
`04:49.56`	`mosty`	`thanks guys, back to work`
`04:50.08`	`hmmhesays`	`just had band practice`
`04:50.09`	`hmmhesays`	`then the bar`
`04:50.49`	`hmmhesays`	`now i'm eating wendy's chili`
`04:51.11`	`Crashsys`	`Drunk and Chili...`
`04:51.18`	`hmmhesays`	`yeah...`
`04:51.19`	`hmmhesays`	`bad`
`04:51.21`	`Crashsys`	`good combination... if cars ran on methane you'd be set :)`
`04:51.32`	`hmmhesays`	`i'm waiting for my episode of battlestar galatica to download`
`04:51.37`	`hmmhesays`	`cause starbuck is hot`
`04:52.02`	`*** join/#asterisk phileeeep (n=philip@203.63.126.9)`
`04:52.44`	`hmmhesays`	`katee sackhoff and gretchen wilson, hot`
`04:52.52`	`Crashsys`	`I'm trying to figure out why asterisk keeps doing an autofallthrough when my polycom replies back with a sip 486...`
`04:53.07`	`hmmhesays`	`um.. because that is a rejection code`
`04:53.23`	`hmmhesays`	`so that means jump to the next priority`
`04:53.33`	`Crashsys`	`Wouldn't asterisk go to the next priority tho?`
`04:53.36`	`phileeeep`	`does anyone know much about using the manager interface to log agents in and out of a queue?`
`04:53.43`	`Crashsys`	`http://pastebin.ca/209417`
`04:53.49`	`Crashsys`	`that's what I got... and it fall's through`
`04:53.52`	`wmandra`	`crashsys: somethings definately f**ked.... I'm looking at chan_sip.c from 1.2.12.1 and in 486 it send back a BUSY and 480 it sends back CONGESTION......`
`04:54.49`	`Crashsys`	`Yeah... O`
`04:54.57`	`Crashsys`	`I've got the forrest gump edition of asterisk :)`
`04:55.24`	`wmandra`	`see what happens if you change s,2 to Goto(s-${DIALSTATUS}) then create a s-CONGESTION,1 extension`
`04:55.40`	`Crashsys`	`Hummmmmmmmmmm`
`04:56.02`	`Crashsys`	`Tomorrow I will find out... cause i'm not at the office to hit a DND button :)`
`04:56.12`	`wmandra`	`ooops... it's late s-CHANCONGESTION`
`04:57.03`	`phileeeep`	`anyone manager interface?`
`04:57.25`	`mds2`	`is FlashOperator panel the best app for watching calls/extensions in real time?`
`04:58.11`	`wmandra`	`my best guess is that chan_sip.c handle_response is sending back AST_CONTROL_BUSY but dial is setting the ${DIALSTATUS} to either CANCEL CHANUNAVAIL or CHANCONGESTION`
`04:58.39`	`wmandra`	`phileeeep: what's your question about the Manager??`
`04:59.04`	`*** join/#asterisk aadilismail (n=adilisma@202.38.55.114)`
`04:59.12`	`phileeeep`	`i have a problem with loging agents in and out of a queue using the manager interface`
`04:59.38`	`phileeeep`	`being with Hints, if an agent is on a call, logging them out, kills the current call`
`05:00.15`	`aadilismail`	`hi`
`05:00.47`	`wmandra`	`phileeep: i would guess that's probably by design`
`05:00.59`	`phileeeep`	`without hints, the agent is logged out and the current active call is uneffected.`
`05:01.11`	`phileeeep`	`we rely on hints for status information from the manager interface`
`05:01.29`	`wmandra`	`if their no longer logged in then they aren't there..... thus the call should be hunug up`
`05:02.20`	`phileeeep`	`correct, but without hints being used. the active call is uneffected by the logout, the agent is allowed to complete the call.`
`05:02.38`	`*** join/#asterisk linagee (n=linagee@unaffiliated/linagee)`
`05:04.29`	`phileeeep`	`so is there anyway to remove agents from the queue without killing active calls, still using hints.`
`05:08.18`	`aadilismail`	`new to asterisk .. i want to dial two different number and want both different response?`
`05:12.55`	`wmandra`	`phileeep: it's late here so i'm way to tired to really dig into this, but I'm thinking that when hints are enabled * is handling the active call correctly, and there's possibly a bug when hints aren't being used....`
`05:15.38`	`Crashsys`	`Anyone familiar with polycom's? (got a question for what it does during a transfer)`
`05:16.31`	`Crashsys`	`When you transfer on a polycom, is it creating a new call, or direct-dialing sip/exten?`
`05:17.02`	`Crashsys`	`like does it go back through extensions context assigned with that phone for the s group?`
`05:18.22`	`phileeeep`	`wmandra: im going to post more detail in the digium asterisk forums. maybe you would like to check the user section for my post when you get a chance? that would be most appreciated`
`05:18.44`	`wmandra`	`phileeeep: will do`
`05:19.09`	`[TK]D-Fender`	`Crashsys : All dialing is through the phone's context.`
`05:19.16`	`stephane_`	`jour all`
`05:19.17`	`Crashsys`	`ok`
`05:19.38`	`[TK]D-Fender`	`Crashsys : And yes it is pretty much like a new call except that it reattaches the audio stream of the previous call`
`05:20.07`	`Crashsys`	`Question 1, Part 2... :) Is there a way to change the callerID on a transfered call to the ID of the transferred party, not the phone it was transferred at?`
`05:20.15`	`JT`	`benjk: hmm`
`05:20.56`	`[TK]D-Fender`	`Crashsys : to do that you should be doing a "blind" transfer, not an "attended" transfer`
`05:21.12`	`Crashsys`	`Hmmm... i'll have to test that...`
`05:21.28`	`Crashsys`	`a blind transfer on a polycom will forward the caller's ID?`
`05:21.47`	`[TK]D-Fender`	`Crashsys : Press [transfer], then immediatly hit the [blind] soft-key. THEN enter your destination and the caller will be "thrown" over to the destination with the call carrying their CID`
`05:22.00`	`Crashsys`	`Cause nothing sucks like picking up the phone going "Yeah, what the hell do you want now?" and find out it's a customer :)`
`05:22.23`	`Crashsys`	`d-fender: Kewl, thank... right now they are doing attended transfers and hitting transfer before you even pick up...`
`05:23.25`	`Crashsys`	`Fender as in guitar?`
`05:23.35`	`[TK]D-Fender`	`Crashsys : Yeah, the uninitiated tend to do things like that`
`05:23.52`	`[TK]D-Fender`	`Crashsys : No, its from my FPS gaming days starting with CTF Tribes`
`05:24.02`	`[TK]D-Fender`	`Crashsys : I play Dean guitars ;)`
`05:24.06`	`Crashsys`	`Ahhh... Q2-CTF`
`05:24.11`	`Crashsys`	`Charvel here`
`05:24.20`	`Crashsys`	`I'm a crappy rhythm guitarist`
`05:25.12`	`Crashsys`	`Michaelangel Batio plays dean's... I run into him every now and then...`
`05:25.20`	`Crashsys`	`I live in the Tampa Bay area, Dean Guitar's is just up the road...`
`05:25.33`	`Crashsys`	`err angel = angelo`
`05:26.12`	`[TK]D-Fender`	`Lemme host something for ya... 1 sec`
`05:28.21`	`[TK]D-Fender`	`Crashsys : http://aocomputing.net/newguitarsolo.mp3`
`05:28.35`	`*** join/#asterisk CoffeeIV (i=rgr@rrcs-67-79-2-146.sw.biz.rr.com)`
`05:28.57`	`Crashsys`	`guitar Solo?`
`05:29.02`	`Crashsys`	`hmmm... Metal?`
`05:29.08`	`[TK]D-Fender`	`nope`
`05:29.16`	`[TK]D-Fender`	`just me goofing off :)`
`05:29.19`	`Crashsys`	`Ahhhh`
`05:29.24`	`file`	`moo`
`05:29.25`	`CoffeeIV`	`what "sip show" command will tell me what codec a channel is using ?`
`05:29.29`	`file`	`[TK]D-Fender: these are so bright >_<`
`05:29.41`	`[TK]D-Fender`	`file : They DO have controls you know :)`
`05:29.53`	`file`	`yes, they are set at 0 for brightness now`
`05:29.58`	`Crashsys`	`d-fender: Decent sustain...`
`05:30.04`	`file`	`so cool though!`
`05:30.05`	`Crashsys`	`Pretty clean runs`
`05:30.39`	`[TK]D-Fender`	`Crashsys : thx`
`05:31.09`	`Crashsys`	`And the appregios...`
`05:31.19`	`*** join/#asterisk RestLessGemini (n=rLg@202.61.49.248)`
`05:31.23`	`[TK]D-Fender`	`<- 2-bit hack :)`
`05:31.29`	`Crashsys`	`It's pretty good...`
`05:31.51`	`Crashsys`	`Better then 80% of the crap that floats around here claiming to be a guitarist...`
`05:32.58`	`Crashsys`	`http://www.youtube.com/watch?v=FXJuTsHGGmQ The lead guitarist and singer is a good friend of mine...`
`05:32.58`	`[TK]D-Fender`	`Crashsys : http://www.geocities.com/tk_dfender/Tone.mp3`
`05:33.45`	`Crashsys`	`pretty heavy pick in that one... dunno if that's what you were going for... personal pref :)`
`05:33.54`	`Crashsys`	`sounds good tho`
`05:35.07`	`[TK]D-Fender`	`Crashsys : started liking that bit from the start :)`
`05:35.26`	`Crashsys`	`which bit? the song video?`
`05:35.33`	`Crashsys`	`damn i'm tired`
`05:35.37`	`Crashsys`	`cant half type/think`
`05:35.39`	`[TK]D-Fender`	`Crashsys : the"tone" one was a deeping resonnace test to see how "creamy" I could get it.`
`05:35.45`	`[TK]D-Fender`	`Crashsys : the vid`
`05:36.08`	`Crashsys`	`Ahhh... sounds pretty good... you play for fun or what?`
`05:36.28`	`Crashsys`	`well lemme rephrase... you play for recreation or what?`
`05:37.03`	`[TK]D-Fender`	`Crashsys : Yeah I jsut pick up for R&R. Same with my work on the keyboard.`
`05:37.05`	`Crashsys`	`There's a second solo after the second chorus that's pretty good...`
`05:37.30`	`Crashsys`	`dont happen to know any decent drummers in tampa, florida area do ya?`
`05:37.42`	`[TK]D-Fender`	`Crashsys : Sorry, Montreal here :)`
`05:37.46`	`Crashsys`	`ahhhh`
`05:38.14`	`*** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net)`
`05:38.16`	`Crashsys`	`We are having a hard time finding a drummer who can play the drums on that song you just heard...`
`05:38.20`	`Crashsys`	`and it's pretty basic drums...`
`05:38.26`	`docelmo`	`Does the manager use TCP or UDP?`
`05:38.27`	`[TK]D-Fender`	`Well I've hit the limit. Gotta crash but back tomorrow. Later all.`
`05:38.34`	`Crashsys`	`later mang`
`05:38.48`	`tengulre`	`docelmo: TCP`
`05:38.50`	`[TK]D-Fender`	`Crashsys : Use "Hot For Teacher" as your test track ;)`
`05:38.54`	`docelmo`	`Thats what I thought.`
`05:39.03`	`Crashsys`	`d-fendeR: Yeah, that's the first thing I ask 'em to play...`
`05:39.34`	`Crashsys`	`Another band I deal with has some drums on it that are very difficult...`
`05:40.09`	`aadilismail`	`new to asterisk ... how to dial and configure digits ?`
`05:40.17`	`[TK]D-Fender`	`yawn later peeps...`
`05:40.22`	`Crashsys`	`later...`
`05:41.38`	`docelmo`	`thanks`
`05:45.58`	`JT`	`hrm, how do i shutup all warnings etc in the console?`
`05:46.44`	`Crashsys`	`two vv's instead of three?`
`05:46.55`	`JT`	`1v is still too much`
`05:47.13`	`JT`	`getting about 40 lines of whingeing about d channels every 2 seconds`
`05:47.53`	`carrar`	`JT, type exit`
`05:48.03`	`carrar`	`or fix them`
`05:48.05`	`JT`	`err i know how to get out`
`05:48.19`	`JT`	`i want to shutup the errors so i can do stuff like read help :)`
`05:49.13`	`JT`	`fix them, yes i'd like to do that`
`05:49.23`	`JT`	`but i'd probably need to do a bri debug`
`05:49.31`	`JT`	`without 100000 useless messages appearing`
`05:50.01`	`carrar`	`Whtas making all the errors?`
`05:50.14`	`carrar`	`exit out and then scroll back`
`05:50.29`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`05:50.32`	`carrar`	`or look at your logs`
`05:52.01`	`JT`	`probably the fact i have no bris hooked up to my octoBRI card`
`05:52.18`	`JT`	`except i do have 2 BRIs connected to 2 other BRIs by crossover cable`
`05:52.24`	`JT`	`they should in theory not have alarms`
`05:52.27`	`JT`	`but they do`
`05:52.29`	`JT`	`so hmm`
`05:57.47`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`06:03.11`	`JT`	`bri intense debug is pretty useless`
`06:03.27`	`JT`	`will have to look elsewhere to see why my crossover cable isn't working`
`06:04.44`	`*** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no)`
`06:06.38`	`aadilismail`	`new to asterisk how to dial digits ?`
`06:08.51`	`*** join/#asterisk lorinc (n=ang@caracas-4901.adsl.interware.hu)`
`06:17.17`	`*** part/#asterisk Crashsys (n=kumba@146-210.124-70.tampabay.res.rr.com)`
`06:17.32`	`*** join/#asterisk Assid (i=assid@203.115.83.215)`
`06:20.44`	`J4k3`	`I didn't know you could go back-to-back with BRIs`
`06:20.55`	`J4k3`	`I know you can't with PRIs`
`06:21.18`	`J4k3`	`you can with CT1s, but they're comparatively stupid`
`06:23.27`	`JT`	`i don't see why it can't`
`06:23.35`	`JT`	`unless its signalling is somehow deficient`
`06:24.35`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`06:29.24`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)`
`06:32.23`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`06:36.20`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`06:41.15`	`*** join/#asterisk Gunde (n=spamyous@82.153.170.213)`
`06:41.59`	`*** join/#asterisk oej (n=oej@217.148.148.248)`
`06:44.53`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`06:46.28`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`06:51.55`	`*** join/#asterisk DEader (n=dead@ool-44c6a5c2.dyn.optonline.net)`
`06:52.28`	`aadilismail`	`new to asterisk ... want to call through xpro to two different country code .. and want both different response from asterisk`
`06:52.34`	`DEader`	`hello does any one have this problem where when call from asterisk box using pstn trunk the number comes up private`
`06:52.45`	`DEader`	`can someone help me show my caller id`
`06:54.00`	`aadilismail`	`<DEader> asterisk -r`
`06:54.31`	`DEader`	`asterisk -r connects to asterisk`
`06:54.32`	`DEader`	`?`
`06:55.52`	`*** join/#asterisk af_ (n=af@ip-200-10.sn2.eutelia.it)`
`06:56.00`	`aadilismail`	`connecting "asterisk CLI"`
`06:56.38`	`DEader`	`then what`
`07:01.02`	`*** join/#asterisk The_Gus (n=Augusto@81.200.133.119)`
`07:01.40`	`The_Gus`	`hi all :-)`
`07:06.25`	`*** join/#asterisk The_Gus (n=Augusto@81.200.133.119)`
`07:07.19`	`The_Gus`	`hi :-) any1 here?`
`07:10.08`	`The_Gus`	`good morning everybody. I've got a problem with asterisk. Is there any1 available for kindly help me?`
`07:11.31`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`07:15.52`	`*** join/#asterisk xnon (n=xnon@200.8.30.163)`
`07:16.03`	`*** join/#asterisk Corydon76-home (i=silver@pdpc/supporter/sustaining/Corydon76-home)`
`07:16.03`	`*** mode/#asterisk [+o Corydon76-home] by ChanServ`
`07:17.15`	`The_Gus`	`any1 please?`
`07:17.48`	`Rhizome`	`I'm afraid of commitment.`
`07:17.48`	`Rhizome`	`;)`
`07:19.15`	`The_Gus`	`:-) dont' fully get u because I0m not english native speaker...but at least I saw an answer ;-)`
`07:25.49`	`kaldemar`	`The_Gus: you're more likely to get an answer by just asking the question instead of asking for a helper.`
`07:26.16`	`The_Gus`	`k`
`07:27.55`	`The_Gus`	`I'm dealing with a registration of a sip trunk. I've changed dozen of settings but still I receive connection timeout when trying to register the trunk`
`07:28.01`	`The_Gus`	`I paste som elines:`
`07:28.28`	`The_Gus`	`Oct 19 09:21:43 DEBUG[2649] chan_sip.c: Stopping retransmission on '275d510a1bfd6b220faf9cc61d493afa@81.200.133.119' of Request 102: Match Found`
`07:28.29`	`The_Gus`	`Oct 19 09:21:48 NOTICE[2649] chan_sip.c: -- Registration for '04321845472@voip.eutelia.it' timed out, trying again (Attempt #21)`
`07:28.29`	`The_Gus`	`Oct 19 09:21:48 DEBUG[2649] acl.c: ##### Testing 195.62.225.244 with 192.168.1.0`
`07:28.58`	`*** join/#asterisk skywriter (n=test@mail.splendor.net)`
`07:29.09`	`The_Gus`	`any suggestion? NAT is set to yes and my router forwards some ports to the asterisk server`
`07:30.24`	`The_Gus`	`system is TrixBox with freePBX`
`07:31.18`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-160-82.columbus.res.rr.com)`
`07:32.41`	`The_Gus`	`I can ping voip.eutelia.it so I suppose dns is set correctly`
`07:33.22`	`*** join/#asterisk [1]tobbe (n=tobbe@212.247.4.149)`
`07:34.23`	`intralanman`	`anyone familiar enough with the SIP_HEADER function to give me some help that would be willing to do so?`
`07:34.55`	`kaldemar`	`The_Gus: i'd first check if sip port (5060 by default) is forwarded to your asterisk box in the router.`
`07:35.15`	`The_Gus`	`yes sir, it is`
`07:35.18`	`The_Gus`	`5060`
`07:35.23`	`sumasuma`	`can anyone please help me with the stun messaage types ?`
`07:35.24`	`The_Gus`	`10000-12000`
`07:35.37`	`The_Gus`	`I meant Kaldemar : yes it is`
`07:35.47`	`The_Gus`	`sorry all, being a long time I'not chatting`
`07:35.51`	`sumasuma`	`can anyone please explain the stun terms ? TypeIndependent ? Typedependent? PortDependentFilter? DependentMapping?`
`07:37.12`	`*** join/#asterisk jaike (n=jaike@203.115.188.120)`
`07:38.47`	`jaike`	`anyone with experience load balancing asterisk using openser?`
`07:40.44`	`*** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)`
`07:44.10`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`07:45.08`	`jaike`	`anyone run asterisk queues with 300 simultaneous calls?`
`07:45.57`	`The_Gus`	`thank you very much anyway :-)`
`07:48.12`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`07:50.14`	`FlatFoot`	`morning all`
`07:50.29`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`07:50.29`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`07:50.33`	`FlatFoot`	`anyone come across a weird thing with the snom 320 ?`
`07:50.46`	`FlatFoot`	`using the menu system ie press 1 for x`
`07:51.05`	`FlatFoot`	`this only works when on speaker phone not when using the handset`
`07:51.35`	`FlatFoot`	`i have followed the destructions on the snom site for tuning the phones to work with * but it no work`
`07:51.52`	`FlatFoot`	`anyone got a suggestion , other than throwing the phones away`
`07:52.26`	`*** join/#asterisk pif (n=ldm@zenon.apartia.fr)`
`07:54.15`	`sbingner`	`mail them to me? ... it's not throwing them away!`
`07:54.55`	`FlatFoot`	`sbingner: don't think the client would be very happy with that`
`07:57.31`	`DarKnesS_WolF`	`wow the lookupCIDname is very very easy !`
`07:57.47`	`DarKnesS_WolF`	`one line in the extenstion.conf and few enteries in the astdb and viola !`
`08:04.25`	`*** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net)`
`08:06.29`	`*** join/#asterisk Givur (n=mail@p54BCF214.dip.t-dialin.net)`
`08:06.31`	`Givur`	`Good morning`
`08:06.35`	`*** join/#asterisk oej (n=oej@62.92.148.159)`
`08:11.36`	`DEader`	`my caller id for my asteris box to pstn don't work`
`08:11.44`	`DEader`	`i check the log and it said`
`08:11.49`	`DEader`	`No application 'SetCallerID' for extension ....`
`08:11.53`	`DEader`	`what does that mean`
`08:12.34`	`jaike`	`DEader: its deprecated on 1.4 i think`
`08:14.02`	`DEader`	`i have trunk latest version installed`
`08:16.10`	`jaike`	`try Set(CALLERID(all)=name <number>)`
`08:18.18`	`DEader`	`cool`
`08:18.21`	`DEader`	`that work`
`08:18.38`	`DEader`	`but the caller id on the phone i'm calling comes up private`
`08:18.40`	`DEader`	`?`
`08:24.53`	`*** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty)`
`08:25.33`	`skywriter`	`does any one have a step by step how to configure asterisk behind a NAT? AsteriskTFOT and Wiki didnt work`
`08:27.46`	`*** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com)`
`08:32.05`	`THX2000`	`Ring/Off-hook in strange state 6 ... ne1 besides me ever come across this?`
`08:35.16`	`*** join/#asterisk oQPa (i=Ftv@86.Red-83-33-148.dynamicIP.rima-tde.net)`
`08:36.14`	`*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`08:38.45`	`*** join/#asterisk pipwerk (n=remco@145.100.6.122)`
`08:39.15`	`pipwerk`	`morning`
`08:44.00`	`jaike`	`skywriter: if your provider uses IAX, nat is a non-issue`
`08:44.08`	`*** join/#asterisk RoyK (n=roy@213.160.242.90)`
`08:45.00`	`*** join/#asterisk xin01oz (i=askme@c-68-63-69-49.hsd1.al.comcast.net)`
`08:45.24`	`*** join/#asterisk maverickbna (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`08:45.54`	`xin01oz`	`Can someone based on experience say whether i should buy a Rhino or Digium t1 interface card`
`08:46.35`	`RoyK`	`xin01oz: methinks sangomas are really good`
`08:47.11`	`RoyK`	`xin01oz: but digium cards may work well as well, except if you have 'incompatible hardware', that is, hardware that isn't in sync with digiums faulty hardware`
`08:47.34`	`RoyK`	`both IBM and dell servers are not recommended. you know, IBM only makes shite`
`08:48.20`	`*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)`
`08:50.02`	`sumasuma`	`RoyK: familier with stun types?`
`08:50.50`	`RoyK`	`me? I only use stun as in stun as in stun. i didn't know there were several types`
`08:51.31`	`sumasuma`	`IndependentFilter, DependentFilter, PortDependentFilter, DependentMapping !!! ??? These are stun types. I'm not much familier with the terms`
`08:52.37`	`sumasuma`	`IndependentFilter, DependentFilter, PortDependentFilter, DependentMapping !!! ??? These are stun types. I'm not much familier with the terms. Can you help to understand them?`
`08:52.56`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`08:54.35`	`*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)`
`08:55.29`	`chapeaurouge`	`hello.. got a sangoma a101 card, and compiling the wanpipe TDM driver... got this question:`
`08:55.34`	`chapeaurouge`	`Compile TDMV DCHAN Native HDLC Support (y/n) ?`
`08:55.40`	`chapeaurouge`	`what does it relate to?`
`08:55.49`	`chapeaurouge`	`echo cancellation?`
`08:58.31`	`skywriter`	`Jaike: my provider only support NAt`
`08:58.36`	`skywriter`	`Jaike: my provider only support Sip`
`08:58.53`	`*** join/#asterisk [shodan] (n=shodan@ip164.96-113-216.pppoe1.joliette.intermonde.net)`
`09:01.08`	`[shodan]`	`beside the nokia , which cellphone supports wifi ?`
`09:03.41`	`aadilismail`	`hi`
`09:04.03`	`*** join/#asterisk yxa (n=diablo@58.185.90.101)`
`09:05.58`	`yxa`	`can several lines of Saydigits be nested inside cmd Read() ?`
`09:06.05`	`[shodan]`	`damn, the nokia e70 is one sweet looking phone`
`09:06.47`	`pipwerk`	`hmmm, in AEL, do all applications set DIALSTATUS to reflect success or failure? or how do I eg. detect failure of a DBGet?`
`09:07.11`	`*** join/#asterisk drega (n=drega@mail.amcat.co.uk)`
`09:08.33`	`*** join/#asterisk misc-- (n=misc@60-242-43-58.static.tpgi.com.au)`
`09:09.22`	`misc--`	`hi, I have installed asterisk@home but for some reason, it logs all php debugging to /var/log/messages. Anyone know what is causing this? Is it something to do with /etc/asterisk/logger.conf (full => notice,warning,error,debug,verbose) ?`
`09:09.40`	`misc--`	`I don't want to log any php stuff. Well I do, but not in the messages file`
`09:10.29`	`jaike`	`anyone with experience using asterisk with openser?`
`09:13.54`	`*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)`
`09:20.36`	`*** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se)`
`09:20.44`	`jeremy_g`	`hi need a hand with asterisk rpm installs`
`09:20.58`	`jeremy_g`	`asterisk requires this libtime.a`
`09:21.08`	`jeremy_g`	`and stddef.h which rpm has it`
`09:21.14`	`jeremy_g`	`running suse 10.1`
`09:27.00`	`*** join/#asterisk FlatFoot (n=simon@80.88.192.113)`
`09:27.58`	`FlatFoot`	`ok here's a weird result`
`09:28.05`	`DarKnesS_WolF`	`jeremy_g: what distro ?`
`09:28.10`	`DarKnesS_WolF`	`ahh sorry suse 10.1`
`09:28.15`	`FlatFoot`	`just thought you all might like to know in case you ever use snom 320`
`09:28.29`	`DarKnesS_WolF`	`hmm dosn't suse have a packegmanager to manage the dependencies?`
`09:28.30`	`FlatFoot`	`don't specify dtmfmode per sip entry`
`09:28.45`	`DarKnesS_WolF`	`FlatFoot: why ?`
`09:28.55`	`FlatFoot`	`make it global then the key press' are recognised for voicemail entry`
`09:28.57`	`xnon`	`hello`
`09:29.17`	`drega`	`I thought suse had a tool similar to yum if not yum its self`
`09:29.19`	`xnon`	`can i change the port 5060 to 5061 near shell CLI console¿?`
`09:29.22`	`xnon`	`can i change the port 5060 to 5061 near shell CLI console¿?`
`09:29.24`	`FlatFoot`	`i had specified for each sip entry and the voicemail key press would only work on speaker`
`09:29.48`	`FlatFoot`	`but now as a global it works on the handset aswell , WEIRD huh`
`09:29.52`	`DarKnesS_WolF`	`FlatFoot: i think this has something to do with the DTMFmode itself`
`09:30.06`	`DarKnesS_WolF`	`xnon: edit /etc/asterisk/sip.conf change it and then in the CLI`
`09:30.11`	`DarKnesS_WolF`	`reload chan_sip.so`
`09:30.15`	`xnon`	`i was do it`
`09:30.21`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
`09:30.22`	`FlatFoot`	`perhaps but it's the only way i found to fix a 3 week old fault , been scratchin my head for a while`
`09:30.31`	`xnon`	`but in my console say that the port is 5060`
`09:30.56`	`DarKnesS_WolF`	`FlatFoot: :-) i was lucky ! i have phone able to acess the voicemail and some not ... so i checked the configs and find it was a dtmf mode in the phone.`
`09:31.36`	`FlatFoot`	`DW: i had followed the destructions on tweaking the phone ( Snom www )`
`09:31.40`	`yxa`	`how do I workaround this: I have several lines saydigits like to play but anytime a user enters a string of numbers during any of the lines, I want it to do something else with that string`
`09:31.48`	`FlatFoot`	`but this did not make it work`
`09:32.04`	`xnon`	`is a trixbox system`
`09:32.19`	`xnon`	`and say that the port open is the 5060`
`09:33.11`	`xnon`	`but in the sip.conf is 5061`
`09:34.16`	`skywriter`	`anyone know why i m getting this warning:`
`09:34.27`	`skywriter`	`WARNING[1304]: chan_sip.c:3592 process_sdp: Unknown SDP media type in offer: video 41090 RTP/AVP`
`09:34.59`	`yxa`	`is it possible to output saydigits() to a sound file?`
`09:38.18`	`kaldemar`	`yxa: you could try to combine it with application monitor`
`09:38.49`	`jeremy_g`	`DarKnesS_WolF:suse 10.1`
`09:40.10`	`FlatFoot`	`sip no debug`
`09:40.17`	`FlatFoot`	`oops wrong screen`
`09:40.48`	`yxa`	`kaldemar more hints appreciated :)`
`09:41.20`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`09:41.33`	`*** join/#asterisk backblue (n=igor@82.102.1.42)`
`09:42.50`	`jeremy_g`	`is this valid Set(B1=EXTEN:5) or should i use ,Set(B1={EXTEN:5})`
`09:45.29`	`kaldemar`	`yxa: monitor is an application that records channels. start the application before saydigits in your dialplan`
`09:46.41`	`yxa`	`kaldemar what's the different between Monitor and Record`
`09:47.29`	`aadilismail`	`hi`
`09:48.26`	`aadilismail`	`new to asterisk "i did recover during changes in asterisk" received this msg ... E45: 'readonly' option is set (add ! to override) ...what does it mean??`
`09:49.10`	`kaldemar`	`yxa: if you start record, it won't move on in the dialplan unless you press # and record can't record what other applications do, or a dialog between two channels.`
`09:50.17`	`kaldemar`	`yxa: wiki would have told you that too.`
`09:53.28`	`jeremy_g`	`this muffing suse sucks at compiling asterisk, has anyone managed to compile an asterisk 1.2 on it`
`09:54.30`	`kaldemar`	`i've compiled 1.0.X on 9.1 and 10.0 and CVS and SVN versions on 9.1 with no problems.`
`09:56.08`	`skywriter`	`where do i configure my ip gateway in asteriak?`
`09:57.40`	`qdk`	`SkymeyeR: you dont.`
`09:57.57`	`*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`09:59.27`	`*** join/#asterisk LittleBastard (n=BOfH@toepp.odn.de)`
`09:59.30`	`LittleBastard`	`hi`
`10:00.27`	`skywriter`	`he ll get it automatically?`
`10:02.36`	`xnon`	`anybody was work with the trixbox?`
`10:06.06`	`backblue`	`how can i do echo cancelation on a sip to sip call?`
`10:08.27`	`*** part/#asterisk skywriter (n=test@mail.splendor.net)`
`10:11.40`	`Dr-Linux`	`question, i taking calls from pstn on 2 zap ports and i directly fwd them to particular extensions. problem is that, in few calls both channels get busy.`
`10:12.06`	`Dr-Linux`	`i'm sure 4 lines are allowed`
`10:12.08`	`CtRiX`	`backblue, there sholdnt be echo on sip -> sip`
`10:12.13`	`Dr-Linux`	`LHR-PBX*CLI> show channels`
`10:12.14`	`Dr-Linux`	`Channel Location State Application(Data)`
`10:12.14`	`Dr-Linux`	`Zap/3-1 s@pasport:6 Busy Congestion()`
`10:12.14`	`Dr-Linux`	`Zap/4-1 s@pasport:6 Busy Congestion()`
`10:12.27`	`CtRiX`	`unless the sip phone sucks or you are using speackers`
`10:12.34`	`Dr-Linux`	`so each time i "soft hangup the calls"`
`10:14.07`	`Dr-Linux`	`anybody any clue`
`10:14.17`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`10:15.46`	`*** join/#asterisk toasteriskornott (n=test@mail.splendor.net)`
`10:17.51`	`backblue`	`CtRiX: i'm having a lot of troubles with echo`
`10:18.15`	`backblue`	`and i can find any how to do echo cancel`
`10:18.30`	`CtRiX`	`backblue, sip -> * -> sip ?`
`10:18.40`	`CtRiX`	`or there is some other things in the middle ?`
`10:18.57`	`toasteriskornott`	`are there trial g729 g723 license for asterisk to test?`
`10:24.46`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
`10:37.01`	`xin01oz`	`anyone got any comments for running asterisk on a supermicro server`
`10:42.23`	`backblue`	`CtRiX: there are cards in the midle.`
`10:43.09`	`backblue`	`toasteriskornott: you have the source the free version, just for testing it works ok.`
`10:45.19`	`toasteriskornott`	`once i register asterisk with my VOIP provider,i should be able to make international call from any extension?????????`
`10:47.21`	`LittleBastard`	`a small question: is it posible to play music on hold while asterisk is playing LIMIT_CONNECT_FILE? at the moment, my asterisk stops moh before it plays the limit_connect_file.`
`10:49.59`	`*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`10:50.53`	`*** join/#asterisk Inez (i=faceoff@devel4.net)`
`10:50.55`	`Inez`	`Hi`
`10:51.27`	`Inez`	`How Can I simulation loop back call in dialplan.`
`10:51.42`	`Inez`	`I want go from one extension to another using Dial & Answer (not goto)`
`10:52.34`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`10:52.56`	`*** join/#asterisk cortez_ (n=cortez@85.221.54.254)`
`10:54.52`	`*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)`
`11:01.13`	`Inez`	`I found it: http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels`
`11:07.44`	`jeremy_g`	`asterisk compiles real well on suse, it was my fault!!! do not blame suse`
`11:09.50`	`Inez`	`ok`
`11:11.06`	`kaldemar`	`Inez: Dial(Local/exten@context)`
`11:12.14`	`jeremy_g`	`Inez:what difference does it make if u use dial and answer vs goto`
`11:12.16`	`jeremy_g`	`??`
`11:14.06`	`Inez`	`take a look at url which I paste`
`11:15.07`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-160-82.columbus.res.rr.com)`
`11:16.24`	`jimmy1234`	`when i try to resgister xlite with asterisk using a password i got authentication failed while i m sure i putting the write password`
`11:16.31`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`11:17.09`	`Inez`	`configure: error: installation or configuration problem: C compiler cannot create executables.`
`11:17.12`	`Inez`	`make: *** [editline/libedit.a] Error 1`
`11:17.16`	`Inez`	`do anyone know that error?`
`11:18.42`	`RoyK`	`have you tried installing editline by any chance?`
`11:18.47`	`RoyK`	`oh`
`11:18.51`	`RoyK`	`or installing gcc?`
`11:18.52`	`RoyK`	`:D`
`11:19.23`	`RoyK`	`~docsa~docs`
`11:19.34`	`RoyK`	`~book`
`11:19.35`	`jbot`	`rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`11:19.35`	`RoyK`	`~rtfm`
`11:19.36`	`jbot`	`[rtfm] 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://en.wikipedia.org/wiki/RTFM`
`11:19.36`	`jimmy1234`	`chan_sip.c:11084 handle_request_register: Registration from '"whateveryouwant"<sip:mike@192.168.0.166>' failed for '192.168.0.101' - Username/auth name mismatch`
`11:19.37`	`RoyK`	`~docs`
`11:19.38`	`jbot`	`from memory, docs is 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`
`11:19.38`	`Inez`	`right :)`
`11:19.38`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:19.40`	`Inez`	`hehe, I forget gcc :)`
`11:20.08`	`Inez`	`RoyK but still that same problem`
`11:20.11`	`RoyK`	`jimmy1234: it MIGHT mean that you have an invalid username and/or password......`
`11:20.33`	`jimmy1234`	`i put the password in secret field`
`11:20.40`	`*** part/#asterisk RoyK (n=roy@80.239.107.70)`
`11:20.44`	`jimmy1234`	`if i ommit it work`
`11:20.49`	`Inez`	`I have gcc 3.35, isit ok?`
`11:21.13`	`jimmy1234`	`and in xlite in the password field`
`11:21.27`	`jimmy1234`	`how could possible password be wrong`
`11:28.10`	`*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)`
`11:29.08`	`*** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.6.57.revip2.asianet.co.th)`
`11:30.30`	`Druken`	`jimmy1234: what do you have in auth ?`
`11:32.17`	`HaMYaI`	`anyone knows how to obtain the remote IP address in the dialplan`
`11:32.29`	`HaMYaI`	`for ooh323`
`11:33.29`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`11:34.06`	`Inez`	`Who install successful asterisk on debian?`
`11:34.45`	`HaMYaI`	`I saw that the IP can be shown in "ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr)" of rtp.c`
`11:35.57`	`HaMYaI`	`but don't know how to get it in the diaplan`
`11:36.06`	`HaMYaI`	`anyone?`
`11:37.23`	`drega`	`Inez I have several times on ubuntu`
`11:37.50`	`*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)`
`11:38.27`	`Inez`	`drega which gcc you used?`
`11:39.24`	`drega`	`ah... good question.`
`11:39.54`	`drega`	`I can look. but I'm sure I just did the standard apt-get gcc so whatever version it is.`
`11:40.00`	`drega`	`hold on and I can find out.`
`11:41.43`	`Inez`	`great`
`11:41.53`	`*** join/#asterisk Nix (n=Nix@195.175.21.46)`
`11:42.21`	`*** join/#asterisk m3_luk3 (n=jsicxruf@dslb-088-073-205-178.pools.arcor-ip.net)`
`11:42.32`	`m3_luk3`	`hello`
`11:42.54`	`drega`	`Inez 4.1.2`
`11:43.16`	`m3_luk3`	`If anyone is able to answer some questions about sip please let me know.`
`11:43.26`	`m3_luk3`	`I would greatly appriciate any help.`
`11:43.34`	`drega`	`m3_luk3: ask your question and if anyone can help they will.`
`11:43.41`	`m3_luk3`	`Ok`
`11:43.52`	`Inez`	`drega wow, I have 3.3.5`
`11:44.15`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`11:44.44`	`m3_luk3`	`I have an internal sip trunk to connect a german pbx system but there are some reliablity issues and I would like to add a fall back sip trunk just as you would in the outbound routes.`
`11:45.09`	`m3_luk3`	`The issue is that it seems impossible to register more than one sip trunk to the same device`
`11:45.25`	`m3_luk3`	`Tried creating a secondary ip and still same result.`
`11:53.06`	`*** join/#asterisk alexcf (i=alexcf@lexi.me.uk)`
`11:53.07`	`alexcf`	`hi`
`11:53.15`	`alexcf`	`anyone played with asterisk+7970G?`
`11:54.15`	`*** join/#asterisk tengulre (n=tengulre@221.11.5.182)`
`11:55.10`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`11:55.16`	`RoyK`	`~smop`
`11:55.18`	`jbot`	`from memory, smop is just a simple matter of programming or encrypted floor cleaning tool`
`11:59.26`	`*** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net)`
`12:01.30`	`*** join/#asterisk loopy44 (n=sub@e178114183.adsl.alicedsl.de)`
`12:01.44`	`*** part/#asterisk loopy44 (n=sub@e178114183.adsl.alicedsl.de)`
`12:01.57`	`alexcf`	`~7970`
`12:02.29`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`12:04.56`	`*** join/#asterisk sysreq (n=sysreq@142-217-213-125.telebecinternet.net)`
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`12:05.05`	`jeremey_g`	`hi`
`12:05.15`	`alexcf`	`hi`
`12:05.16`	`jeremey_g`	`:) working from my suse text mode bitchx terminal`
`12:05.20`	`alexcf`	`fun`
`12:05.21`	`jeremey_g`	`:) after 6 years`
`12:05.22`	`alexcf`	`Irssi 0.8.10 (20051211) - http://irssi.org/`
`12:05.26`	`alexcf`	`;)`
`12:06.40`	`jeremey_g`	`if i dial(sip/101@192.168.1.1,,o) and 192.168.1.1 is one of my other * box. is there a way to disable authentication at all`
`12:07.44`	`jeremey_g`	`was there some guestinvte=yes sort of thing in sip.conf`
`12:09.31`	`jeremey_g`	`if i do dial(SIP/101@account-in-ast-box,,o) my callerd id to remote * box becomes the username with which i logged in`
`12:10.43`	`*** join/#asterisk McLazarus (n=mcallist@64.94.197.130)`
`12:11.27`	`lilalinux`	`How do I use Asterisk behind NAT? Is it possible without port forwardin?`
`12:12.38`	`cortez_`	`anyone know of asterisk based hosted pbx solution providers?`
`12:16.32`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`12:16.42`	`hi365`	`hello. how do i convert a .bmp to base64 for the grandstream 2000?`
`12:18.33`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`12:23.36`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`12:23.48`	`hi365`	`<hi365> how do i convert a bmp to base64 (for the grandstream xml)?`
`12:26.36`	`*** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)`
`12:26.57`	`kaldemar`	`hi365: perl -MMIME::Base64 -0777 -ne 'print encode_base64($_)' < pic.bmp > pic64`
`12:27.01`	`toerkeium`	`hello guys, how can I do to make every agent to record a call ?`
`12:27.22`	`kaldemar`	`hi365: took me 30 seconds to find that.`
`12:27.24`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:29.39`	`*** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br)`
`12:41.07`	`hi365`	`kaldemar: do u have experiance with the grandstream?`
`12:42.20`	`*** join/#asterisk DanTMG (n=danielga@203-217-84-193.dyn.iinet.net.au)`
`12:42.27`	`kaldemar`	`hi365: none`
`12:42.57`	`*** part/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`12:43.22`	`*** join/#asterisk akoch (n=chatzill@mail.gk-soft.de)`
`12:43.36`	`*** join/#asterisk msa20 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`12:43.41`	`*** join/#asterisk Curi (n=la@pc-75-51-104-200.cm.vtr.net)`
`12:44.32`	`Curi`	`hello, anyone knows something about this error:`
`12:44.33`	`Curi`	`ERROR[30345]: chan_sip.c:11011 handle_request_subscribe: Got SUBSCRIBE for extension 97614438@from-inside from 192.168.1.206, but there is no hint for that extension`
`12:44.43`	`Curi`	`and i'm not using hints`
`12:45.43`	`akoch`	`hello, I got after upgrading to 1.2.13 a crash, after hanging up the call`
`12:45.45`	`toerkeium`	`my "One Touch Record" isn't working.. any idea how can I troubleshoot this?`
`12:47.16`	`*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)`
`12:48.17`	`akoch`	`this crash I can reproduse -> start asterisk -> place a call -> asterisk crash`
`12:48.40`	`akoch`	`after going back to 1.2.12.1 the things works fine`
`12:49.09`	`alexcf`	`what's usually the cause for . Got SIP response 400 "Bad Request" back from 84.*..**`
`12:50.01`	`*** join/#asterisk leandroaugusto (n=leandroa@201.21.219.33)`
`12:51.10`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`12:51.23`	`*** join/#asterisk heison (n=heison@ns.somanetworks.com)`
`12:51.39`	`jimmy1234`	`i tried to register xlite with asterisk using a password where secret(in sip.cont)=password(in xlite) but it didn t work?????`
`12:52.03`	`[TK]D-Fender`	`Curi: Well one of your phones is looking for a hint.`
`12:54.55`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`12:56.17`	`Curi`	`[TK]D-Fender: how could that be?, i'm not using hints at all. I read that it had something to do with real time which i'm using`
`12:56.31`	`*** join/#asterisk vgster (n=vgster@170.252.64.1)`
`12:56.56`	`[TK]D-Fender`	`Curi: : pay attenition. THAT PHONE is trying to do it because it was TOLD to look for a hint. Go fix your PHONE.`
`12:58.38`	`Curi`	`[TK]D-Fender: i have 100 phones configured using the exact same configuration, and just a couple have the problem`
`12:58.42`	`LittleBastard`	`a small question: is it posible to play music on hold while asterisk is playing LIMIT_CONNECT_FILE? at the moment, my asterisk stops moh before it plays the limit_connect_file.`
`12:59.31`	`[TK]D-Fender`	`Curi: its something done locally on the phone. You even have the IP. Go to the damned phone and remove the subscribe on it.`
`13:00.11`	`*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)`
`13:03.05`	`*** part/#asterisk zaide (n=zaide@script-kiddy.fr)`
`13:05.22`	`*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)`
`13:09.17`	`*** join/#asterisk mpls-eric (n=ejo1@user-12l39ej.cable.mindspring.com)`
`13:10.19`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net)`
`13:10.45`	`[shodan]`	`my asterisk box runs with 2x x100p clones , and I have just about every documented problem all at once with those cards ! so if I switch to a pair of linksys spa-3102 , all my problems will disappear ?`
`13:11.44`	`[shodan]`	`or the linksys fxos have problems too ?`
`13:12.31`	`cortez_`	`what other suppliers of pbx'es are big other than cisco and nortel?`
`13:12.39`	`cortez_`	`that might have somewhat better pricing`
`13:14.13`	`alexcf`	`anyone got cisco 7970 working on asterisk?`
`13:14.13`	`*** part/#asterisk mpls-eric (n=ejo1@user-12l39ej.cable.mindspring.com)`
`13:14.45`	`p1p`	`cortez_ there is a laundry list of pbx suppliers but none of them are very cost friendly`
`13:15.22`	`DanTMG`	`cortez_: what exactly were you looking for? featureset, size of deployment etc...`
`13:17.05`	`[TK]D-Fender`	`cortez_: Panasonic, Toshiba, Avaya, InterTel......`
`13:17.08`	`[shodan]`	`cortez_ I just stopped by some cie I do service at , they had a new panasonic hybrid ip/pbx`
`13:17.35`	`[shodan]`	`owner said it costed in the thousands for a 5 user system (!)`
`13:17.42`	`[TK]D-Fender`	`cortez_: Of course why are you asking about other stuff that * replaces the need for?`
`13:18.45`	`[TK]D-Fender`	`[shodan]: Stop looking for the cheap way out and buy a real card.`
`13:19.28`	`*** join/#asterisk roving_prole (n=Harper@72-254-127-253.hq.ibahn.com)`
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`13:21.54`	`*** join/#asterisk pingwin[work] (i=pingwin@gateway/tor/x-e97412d0609cf8df)`
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`13:22.55`	`akoch`	`has anyone same problems like me with the new version 1.2.13?`
`13:24.46`	`[TK]D-Fender`	`akoch: Maybe you should actually give some details about your problem.`
`13:26.33`	`JT`	`0pbxs usually cost in thousands even for half a dozen users`
`13:26.37`	`JT`	`to be expected really`
`13:26.55`	`akoch`	`hello, I got after upgrading to 1.2.13 a crash, after hanging up the call`
`13:26.56`	`akoch`	`this crash I can reproduse -> start asterisk -> place a call -> asterisk crash`
`13:26.58`	`akoch`	`after going back to 1.2.12.1 the things works fine`
`13:28.29`	`cortez_`	`DanTMG: moving three locations with about 25 users each, from pstn to ip pbx`
`13:29.34`	`[TK]D-Fender`	`akoch: Are you deaf? DETAILS.`
`13:29.36`	`cortez_`	`got an offer for nortel based solution for about $85000`
`13:29.42`	`drega`	`there are tons of supliers other than those two.`
`13:29.59`	`*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`13:30.00`	`[TK]D-Fender`	`cortez_: Describe your setup requirements`
`13:30.07`	`drega`	`Typically cheap or cost friendly isn't in the same vocabulary though :)`
`13:30.09`	`JT`	`they're about the most expensive suppliers :P`
`13:30.26`	`drega`	`asterisk is cost friendly compared to the others.`
`13:31.12`	`DanTMG`	`cortez_: so the offices are being consolidated? so 75 users, are there any specific requirements in terms of security, features or intergration with existing system?`
`13:31.28`	`*** part/#asterisk pipwerk (n=remco@145.100.6.122)`
`13:31.56`	`cortez_`	`DanTMG: no there will still be three locations connected via vpn`
`13:32.12`	`cortez_`	`all audio needs to be recorded`
`13:32.23`	`cortez_`	`security is "critical"`
`13:34.53`	`cortez_`	`how many users are generaly needed before a pri line is required?`
`13:34.58`	`DanTMG`	`did you have any requirements for unified messaging or other features like presence status'?`
`13:35.30`	`tzanger`	`apparently formatting the filesystem of an ip501 does NOT unfuck the boot settings. you need to manually unfuck them`
`13:35.31`	`cortez_`	`to be honest i'm not really sure what features might be available`
`13:35.50`	`cortez_`	`interoperation with windows systems are much as possible`
`13:35.51`	`*** join/#asterisk ambriento (n=ambrient@201-27-80-82.dsl.telesp.net.br)`
`13:35.54`	`DanTMG`	`a PRI line is not determined by user count but by maximum concurrency (so maximum in/out linesrequired at any given time)`
`13:36.04`	`cortez_`	`yeah`
`13:36.44`	`DanTMG`	`so I would say with your users PRI would be needed`
`13:38.50`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`13:40.11`	`[TK]D-Fender`	`tzanger: Do an up/down grade and that will fluch the boot-rom.`
`13:40.19`	`[TK]D-Fender`	`flush*`
`13:40.35`	`cortez_`	`well instead of having a pri at each location wouldn't it be a better solution to just have pbx at hosting center?`
`13:41.03`	`cortez_`	`that might also provide better termination costs to pstn`
`13:41.13`	`[TK]D-Fender`	`cortez_: a very nice * setup would cost less than a 3rd of the Nortel solution.`
`13:41.24`	`*** join/#asterisk CrashSys1 (n=kumba@office.crashsys.com)`
`13:41.29`	`cortez_`	`what hardware would you have in mind for that`
`13:41.49`	`drega`	`or have the pri's at one location and use sip across the vpn; OR have all the PRI's at one location put * at each location and use IAX trunking between sites to save bandwith and provide quality.`
`13:42.00`	`cortez_`	`will dependent on a third party integrator`
`13:42.02`	`cortez_`	`anyway`
`13:42.08`	`[TK]D-Fender`	`cortez_: * server with 2-port HWEC PRI card, and Polycom IP phones across the board.`
`13:42.31`	`akoch`	`D-Fender: nothing in the log exept -->`
`13:42.33`	`akoch`	`Oct 19 11:54:13 DEBUG[14771] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE`
`13:42.34`	`akoch`	`Oct 19 11:54:13 DEBUG[14771] chan_sip.c: Setting NAT on RTP to 0`
`13:42.36`	`akoch`	`Oct 19 11:54:13 DEBUG[14771] chan_sip.c: Setting NAT on VRTP to 0`
`13:42.37`	`akoch`	`Oct 19 11:54:13 DEBUG[14771] chan_sip.c: Checking SIP call limits for device 746`
`13:42.39`	`akoch`	`Oct 19 11:54:13 DEBUG[14771] chan_sip.c: Updating call counter for incoming call`
`13:42.40`	`CrashSys1`	`So when I turn DND on, it sends back a Sip 486 response... then my server plays congestion and drops the call... I cant get it to send the call to VM... here's the dial macro I use... http://pastebin.ca/210012`
`13:42.40`	`akoch`	`Oct 19 11:54:13 DEBUG[14771] chan_sip.c: build_route: Contact hop: <sip:746@192.168.1.220:5060;transport=udp>`
`13:42.42`	`akoch`	`Oct 19 11:54:13 DEBUG[14748] chan_sip.c: Checking device state for peer 746`
`13:42.43`	`akoch`	`Oct 19 11:54:13 DEBUG[14748] channel.c: Avoiding initial deadlock for 'SIP/746-00720030'`
`13:42.45`	`akoch`	`Oct 19 11:54:13 DEBUG[14790] pbx.c: Launching 'Dial'`
`13:42.47`	`akoch`	`Oct 19 11:54:13 DEBUG[14790] channel.c: Not copying variable STACK-fromsip-46-1.`
`13:42.48`	`akoch`	`Oct 19 11:54:13 DEBUG[14790] channel.c: Not copying variable SIPCALLID.`
`13:42.50`	`akoch`	`Oct 19 11:54:13 DEBUG[14790] channel.c: Not copying variable SIPUSERAGENT.`
`13:42.51`	`akoch`	`Oct 19 11:54:13 DEBUG[14790] channel.c: Not copying variable SIPDOMAIN.`
`13:42.52`	`CrashSys1`	`Uhhh`
`13:42.53`	`akoch`	`Oct 19 11:54:13 DEBUG[14790] channel.c: Not copying variable SIPURI.`
`13:42.54`	`akoch`	`Oct 19 11:54:13 DEBUG[14748] devicestate.c: Changing state for SIP/746 - state 2 (In use)`
`13:42.56`	`akoch`	`Oct 19 11:54:13 DEBUG[14748] devicestate.c: Changing state for CAPI/ISDN2/46 - state 2 (In use)`
`13:42.57`	`akoch`	`Oct 19 11:54:13 DEBUG[14791] app_queue.c: Device 'SIP/746' changed to state '2' (In use) but we don't care because they're not a member of any que`
`13:43.00`	`akoch`	`ue.`
`13:43.02`	`akoch`	`Oct 19 11:54:13 DEBUG[14748] devicestate.c: Changing state for CAPI/ISDN2/46 - state 2 (In use)`
`13:43.04`	`akoch`	`Oct 19 11:54:13 DEBUG[14792] app_queue.c: Device 'CAPI/ISDN2/46' changed to state '2' (In use) but we don't care because they're not a member of a`
`13:43.05`	`CrashSys1`	`stop`
`13:43.06`	`CrashSys1`	`please`
`13:43.07`	`akoch`	`ny queue.`
`13:43.09`	`akoch`	`Oct 19 11:54:13 DEBUG[14793] app_queue.c: Device 'CAPI/ISDN2/46' changed to state '2' (In use) but we don't care because they're not a member of a`
`13:43.12`	`akoch`	`ny queue.`
`13:43.16`	`akoch`	`Oct 19 11:54:20 DEBUG[14748] devicestate.c: Changing state for CAPI/ISDN2/46 - state 2 (In use)`
`13:43.16`	`akoch`	`Oct 19 11:54:20 DEBUG[14790] channel.c: Set channel SIP/746-00720030 to read format slin`
`13:43.18`	`akoch`	`Oct 19 11:54:20 DEBUG[14790] channel.c: Set channel CAPI/ISDN2/46-0 to write format slin`
`13:43.20`	`akoch`	`Oct 19 11:54:20 DEBUG[14794] app_queue.c: Device 'CAPI/ISDN2/46' changed to state '2' (In use) but we don't care because they're not a member of a`
`13:43.23`	`akoch`	`ny queue.`
`13:43.25`	`akoch`	`Oct 19 11:54:32 NOTICE[14806] cdr.c: CDR simple logging enabled.`
`13:43.27`	`akoch`	`Oct 19 11:54:32 WARNING[14806] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy.`
`13:43.30`	`akoch`	`Oct 19 11:54:32 WARNING[14806] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy.`
`13:43.33`	`akoch`	`Oct 19 11:54:32 WARNING[14806] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy.`
`13:43.36`	`akoch`	`Oct 19 11:54:33 NOTICE[14806] config.c: Registered Config Engine mysql`
`13:43.44`	`ambriento`	`~pb`
`13:43.52`	`jbot`	`it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`13:43.53`	`drega`	`ack!! no more!`
`13:43.57`	`DanTMG`	`cortez_: the next question is how much do you want to spend, if you need high 9's reliability it can be done with asterisk but you will more than likely need consultants time to pull it off so that will need to be taken into account in your costs as well as the maintanence that needs to be done, with commercial solutions they will have higher reliability and security out of the box but will cost more, it is a matter of looking at the`
`13:44.08`	`tzanger`	`[TK]D-Fender: it's working now... just took a bit to figure out`
`13:44.23`	`ambriento`	`copy that akoch ?`
`13:44.56`	`CrashSys1`	`ok... so back to square 1... when I turn DND on, it sends back a Sip 486 response... then my server plays congestion and drops the call... I cant get it to send the call to VM... here's the dial macro I use... http://pastebin.ca/210012`
`13:45.34`	`DanTMG`	`cortez_: what I am saying i determining your solution is not in the scope of an IRC chatroom but you need to sit back and look at it as a business investment, look at both your options and don't buy on purchase cost`
`13:46.06`	`CrashSys1`	`Yeah... cost is a poor metric to use...`
`13:46.32`	`[TK]D-Fender`	`CrashSys1: the "n"'s will be numbered as 102 onwards because of their ordering. Also using 101 at all for priority jumping is DEAD. you should not be doing that.`
`13:46.32`	`CrashSys1`	`If that's all that mattered then grandstream would rule the world...`
`13:46.45`	`[TK]D-Fender`	`CrashSys1: And its hanging up because their is no priority 2.`
`13:46.50`	`DanTMG`	`cortez_: look at open source on-site, commercial on-site and hosted solutions`
`13:46.50`	`CrashSys1`	`hmm...`
`13:47.51`	`CrashSys1`	`d-fender: ok, I took the 101 out, left the rest of the code the same... but I get the same result...`
`13:48.43`	`[TK]D-Fender`	`CrashSys1: Show me the new code, that you did a reload and the CLI output of the call.`
`13:49.24`	`CrashSys1`	`http://pastebin.ca/210026`
`13:50.12`	`ambriento`	`am I the only one not seeing what cortez_ is writing?`
`13:50.14`	`[TK]D-Fender`	`CrashSys1: Small tip for you.... you aren't even USING that macro.....`
`13:50.31`	`CrashSys1`	`...`
`13:50.35`	`*** join/#asterisk zeppelin_ (n=gdiilog@201.21.219.33)`
`13:51.02`	`[TK]D-Fender`	`CrashSys1: #`
`13:51.03`	`[TK]D-Fender`	`-- Starting simple switch on 'Zap/26-1'`
`13:51.05`	`[TK]D-Fender`	`#`
`13:51.06`	`[TK]D-Fender`	`<PROTECTED>`
`13:51.10`	`CrashSys1`	`lol`
`13:51.13`	`CrashSys1`	`durh...`
`13:51.25`	`[TK]D-Fender`	`Your first statment should eb CALLING the macro. You have not been paying attention to your inbound context :)`
`13:51.41`	`CrashSys1`	`It's ok... i'm a genius...`
`13:52.17`	`CrashSys1`	`but that macro is what's called on the incoming side...`
`13:52.27`	`CrashSys1`	`it differs from what's done internally`
`13:52.41`	`CrashSys1`	`I dunno why... but it sounded good at the time...`
`13:53.12`	`CrashSys1`	`ok... that macro is a #include file-a.conf, and the dialing part for internal is a #include file-b.conf...`
`13:53.13`	`*** join/#asterisk Ebola (n=Ebola@host81-152-204-158.range81-152.btcentralplus.com)`
`13:53.37`	`CrashSys1`	`Can I use a macro from file-a's include in file-b's dialplan?`
`13:53.45`	`ambriento`	`an uncomprehended genius ;)`
`13:53.55`	`CrashSys1`	`course, I could find out the hard way...`
`13:54.27`	`CrashSys1`	`course, I could just put the macro in extensions before the includes... and use it wherever...`
`13:55.20`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net)`
`13:59.11`	`CrashSys1`	`yay... it works...`
`14:00.53`	`[TK]D-Fender`	`CrashSys1: Doesn't matter where teh MACRO is, what matters is the location of what CALLS it.`
`14:01.30`	`CrashSys1`	`well, the macro needs to be defined before it can be referenced elsewhere right?`
`14:01.36`	`*** join/#asterisk saftsack (n=oliver@p54A7C621.dip.t-dialin.net)`
`14:01.50`	`CrashSys1`	`or does asterisk not do that kind of parsing?`
`14:02.53`	`*** join/#asterisk Avochelm (n=damien@CPE-60-226-238-40.sa.bigpond.net.au)`
`14:03.34`	`[TK]D-Fender`	`CrashSys1: : Nope. Largely irrelevent.`
`14:03.53`	`CrashSys1`	`ok...`
`14:05.07`	`CrashSys1`	`Hmm...`
`14:06.49`	`sevard`	`Does anyone know of a free conf room to try out?`
`14:06.55`	`sevard`	`i need to make a quicky`
`14:07.09`	`sevard`	`didn't gizmo have some available...`
`14:07.26`	`*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)`
`14:08.26`	`sevard`	`[TK]D-Fender: didn't you knwo of one?`
`14:09.18`	`[TK]D-Fender`	`sevard: Nope. How big?`
`14:09.24`	`sevard`	`four or five connections`
`14:10.16`	`sevard`	`i suppppppppppppppose i could go ahead and figure out meetme`
`14:11.30`	`[TK]D-Fender`	`sevard: "load app_motivation.so"`
`14:11.44`	`[TK]D-Fender`	`sevard: "unload app_slacker.so"`
`14:12.12`	`sevard`	`:(`
`14:12.50`	`sevard`	`sometimes i feel like i need motivation pills`
`14:13.02`	`sevard`	`the guy on the street tried to sell me "horse", he said it'd make me feel good.`
`14:13.07`	`sevard`	`I said I don't need a horse.`
`14:15.03`	`*** join/#asterisk n3tim (i=eth0@BHE200150062060.res-com.wayinternet.com.br)`
`14:19.12`	`JoseBravo`	`How can I do, that the called choose what exten want to call. When it call from the land line?`
`14:19.20`	`JoseBravo`	`caller`
`14:19.34`	`sevard`	`holy crap, what the hell does that question even say?`
`14:24.03`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.163)`
`14:26.48`	`saftsack`	`do i have to be aware of some things, if i create more than one line for a telephone in sip.conf?`
`14:27.44`	`*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)`
`14:29.07`	`saftsack`	`do i have to create two full accounts with host options, and so on?`
`14:31.26`	`ManxPower`	`saftsack: no, but it is a good idea to do so.`
`14:31.53`	`saftsack`	`because i created to full accounts but see here:`
`14:32.13`	`saftsack`	`pattonvaussen/pattonvauss 10.10.10.23 D 5070 UNREACHABLE`
`14:32.13`	`saftsack`	`patton/patton 10.10.10.23 D 5060 OK (15 ms)`
`14:32.14`	`saftsack`	`<PROTECTED>`
`14:32.14`	`saftsack`	`how is this possible? Oo`
`14:34.52`	`tzanger`	`wtf`
`14:35.04`	`tzanger`	`does exten => _98XXX,1,noop() not match 98843?`
`14:35.52`	`Corydon76-home`	`It does`
`14:36.09`	`Corydon76-home`	`Of course, if you have _*98., that matches first`
`14:36.28`	`tzanger`	`I don't`
`14:36.32`	`*** join/#asterisk p1p (i=pip@mail.comp911.com)`
`14:36.32`	`tzanger`	`I have 98 and _98XXX`
`14:36.47`	`Corydon76-home`	`In the same context?`
`14:37.55`	`tzanger`	`the matching code does not like _*`
`14:38.38`	`tzanger`	`exten => _998. will not match 998, _998XXX will not match 998843, but 998 matches and 998843 matches.`
`14:38.47`	`*** part/#asterisk RoyK (n=roy@80.239.107.70)`
`14:38.58`	`tzanger`	`unless someone added '*' to the matching code now`
`14:40.20`	`ManxPower`	`Actually _9*98 will match exactly.`
`14:40.26`	`ManxPower`	`* is not a wildcard, it's a DTMF tone.`
`14:40.43`	`ManxPower`	`so it SHOULD work as you expect, tza`
`14:41.29`	`tzanger`	`I know`
`14:41.35`	`tzanger`	`I know * isn't supposed to be a wildcard`
`14:41.43`	`tzanger`	`I'm writing a message to -dev as we speak`
`14:48.59`	`JoseBravo`	`How can the caller choose the exten that want to talk?`
`14:49.40`	`ManxPower`	`JoseBravo: you start by reading The Book`
`14:49.42`	`ManxPower`	`~book`
`14:49.44`	`jbot`	`somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`14:50.51`	`tzanger`	`hmm interesting`
`14:51.06`	`tzanger`	`_998. does not* match 998, but it does match 998123`
`14:51.37`	`*** join/#asterisk designdream (n=designdr@rrcs-71-40-49-30.sw.biz.rr.com)`
`14:52.05`	`tamp4x`	`eith the . it expects 1 or more chars`
`14:52.08`	`tamp4x`	`with`
`14:52.55`	`tzanger`	`oh it is a '1 or more' ? I thought it was 0 or more`
`14:52.56`	`designdream`	`anyone here using the asterisk plugin with wildfire?`
`14:53.19`	`*** join/#asterisk DarKnesS_WolF (n=wolf@82.129.240.253)`
`14:54.57`	`tzanger`	`oh for fuck sakes`
`14:55.02`	`tzanger`	`I had a stupid dialplan error`
`14:55.09`	`tzanger`	`exten => *98,1,foo`
`14:55.13`	`tzanger`	`exten => *98,n,bar`
`14:55.15`	`Qwell[]`	`. is zero or more`
`14:55.16`	`tzanger`	`exten => *98,n,baz`
`14:55.26`	`tzanger`	`exten => _*98XXX,n,foo`
`14:55.28`	`ManxPower`	`No. . is ONE or more. try it and see.`
`14:55.32`	`Qwell[]`	`tzanger: oops, heh`
`14:55.37`	`tzanger`	`(i.e. no _*98XXX,1)`
`14:55.44`	`Qwell[]`	`ManxPower: hmm`
`14:55.47`	`tzanger`	`good thing I didn't send that off yet`
`14:55.57`	`Qwell[]`	`ManxPower: really? I'll have to believe you`
`14:56.48`	`tzanger`	`Qwell[]: I just tested it as well... . is one or more.. _. matches one or more, but also matches oshiat`
`14:56.57`	`tzanger`	`_X. matches 2 digits or more, not 1 or more`
`14:57.14`	`tzanger`	`but yes, I thought _X. matched any single digit`
`14:57.15`	`tzanger`	`it does not`
`14:57.48`	`[TK]D-Fender`	`tzanger: you = silly`
`14:58.01`	`tzanger`	`[TK]D-Fender: indeed. :-(`
`14:58.19`	`tzanger`	`[TK]D-Fender: I need to start working with the ip601 minihttp browser`
`14:59.02`	`[TK]D-Fender`	`tzanger: Lots of fun to be had in there...`
`14:59.27`	`*** join/#asterisk mboehn (i=mathias@65.23.156.248)`
`14:59.48`	`[TK]D-Fender`	`Qwell : So how's the chan_skinny botnet coming? ;)`
`14:59.58`	`Qwell[]`	`heh`
`15:01.34`	`*** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe)`
`15:01.55`	`*** join/#asterisk seele_ (n=seele@208.35.117.195)`
`15:02.27`	`seele_`	`<PROTECTED>`
`15:02.44`	`*** join/#asterisk Trakkasure (n=Trakk@adsl-146-14-250.mia.bellsouth.net)`
`15:03.17`	`*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)`
`15:04.09`	`xheliox`	`am I the only one having an issue with the latest zaptel 1.4 tarball on asterisk.org? gzip: stdin: decompression OK, trailing garbage ignored`
`15:05.09`	`ambriento`	`seele_, recording another sound prompt?`
`15:05.43`	`backblue`	`does anyone know how do i configure the voicemail number in a polycom phone?`
`15:05.43`	`seele_`	`ambriento, no I need to remove them`
`15:05.52`	`seele_`	`ambriento, hablas espanol`
`15:06.07`	`ambriento`	`un ratito. puedo intentar si quieres`
`15:06.49`	`seele_`	`ambriento,ok, mira cuando me logeo en el sistema el me pregunta numero de agente`
`15:06.57`	`ambriento`	`vale`
`15:07.00`	`seele_`	`ambriento, password`
`15:07.15`	`seele_`	`ambriento, y luego la extencion ....`
`15:07.31`	`ambriento`	`seele_, en el primer logon?`
`15:07.51`	`seele_`	`ambriento, necesito eliminar la pregunta de extension`
`15:08.10`	`seele_`	`ambriento, si es el primer login como un agente comun`
`15:08.20`	`seele_`	`ambriento, mira mi configuracion http://pastebin.ca/210132`
`15:08.23`	`ambriento`	`vale`
`15:09.33`	`toerkeium`	`guys, how could I troubleshoot when agents hear with poor performance and voice cuts ?`
`15:09.53`	`ambriento`	`seele_, el contexto local existe, vale?`
`15:10.02`	`seele_`	`ambriento, me entiendes el problema ??`
`15:10.27`	`seele_`	`ambriento, ya miro`
`15:11.34`	`seele_`	`ambriento, si tengo un error ... ya lo verifico`
`15:12.09`	`ambriento`	`seele_, sí entiendo. Mira, la mensage que pide extension solo va tocar cuando el agente ya está logado`
`15:12.33`	`seele_`	`ambriento, ese era el error .... muchas gracias`
`15:12.46`	`seele_`	`ambriento, Thanks`
`15:12.51`	`ambriento`	`por nada :)`
`15:14.34`	`seele_`	`ambriento, bueno ahora me funciona para el login ... pero para el logout sigue preguntando la extencion`
`15:15.16`	`backblue`	`who the hell, work with computers and dont know english?`
`15:16.18`	`[TK]D-Fender`	`backblue: Be nice.... some people are just much more comfortable in another language.`
`15:18.05`	`ambriento`	`seele_, sí es verdad.. dejame ver unas cosas por aca`
`15:18.15`	`ambriento`	`backblue, I know :)`
`15:18.41`	`ambriento`	`btw, spanish isn't my first language :)`
`15:18.45`	`seele_`	`backblue, when I try to logoff one agent the system ask for a extension I need to remove that prompt`
`15:19.08`	`Juggie`	`bonjour comment ca va? :)`
`15:19.11`	`*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-169-13.buff.east.verizon.net)`
`15:19.26`	`SuPrSluG`	`hi all`
`15:19.26`	`tzanger`	`yo quiero taco bell?`
`15:20.25`	`ambriento`	`hahahaha`
`15:20.49`	`ambriento`	`ca va bien?`
`15:20.50`	`SuPrSluG`	`having an issue w/ the caller not hearing ringing,just silence. The callee hears it just fine. Is this a service provider issue or config ?`
`15:20.50`	`[TK]D-Fender`	`Juggie: j'ai ASSEZ tabarnac ;)`
`15:20.53`	`seele_`	`ambriento, jajaj meti las patas`
`15:21.24`	`*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)`
`15:21.25`	`Juggie`	`:)`
`15:21.28`	`tzanger`	`[TK]D-Fender: i know j'ai and tabarnac... assez?`
`15:21.30`	`*** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com)`
`15:21.31`	`*** join/#asterisk dpro (n=dpro@80.109.121.47)`
`15:21.38`	`dpro`	`hi`
`15:21.43`	`Juggie`	`assez = try`
`15:21.54`	`heison`	`~seen JerJer`
`15:22.08`	`jbot`	`jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 2d 22h 49m 4s ago, saying: 'apt-file search lspci'.`
`15:22.08`	`[TK]D-Fender`	`tzanger: Closest translation "I've fucking had ENOUGH" :)`
`15:22.17`	`dpro`	`anyone can point me in the right direction to get IM a la ekiga to work via our asterisk server ?`
`15:22.27`	`benjk`	`essayer == try, assez == enough`
`15:22.28`	`ambriento`	`seele_, jejejje`
`15:22.33`	`tzanger`	`[TK]D-Fender: ahhh okay I'll have to remember that`
`15:23.02`	`ambriento`	`french is the next in line to me to learn`
`15:23.04`	`hi365`	`my grandstream gxp-2000 doesnt want to take an update. the tftp server seems to be working fine. what to do?`
`15:23.31`	`Juggie`	`kick it`
`15:23.38`	`seele_`	`ambriento, ok in english now ... when I try to logoff the system still asking for extension number`
`15:23.52`	`Juggie`	`ich bin muda`
`15:24.06`	`ambriento`	`and german comes right after french`
`15:24.08`	`hi365`	`uh, didnt help`
`15:24.29`	`Juggie`	`thats the only german i nkow`
`15:24.36`	`Juggie`	`and as i recall it means 'i am tired'`
`15:24.42`	`Juggie`	`which i was, tired of that german class`
`15:25.00`	`hi365`	`sounds about right`
`15:25.03`	`seele_`	`Juggie, in spanish = estoy cansado`
`15:25.56`	`hi365`	`my grandstream gxp-2000 doesnt want to take an update. the tftp server seems to be working fine. what to do?`
`15:25.58`	`ambriento`	`in portuguese = estou cansado`
`15:27.51`	`hi365`	`hebrew = ani ayeif`
`15:28.20`	`hi365`	`any love for a grandstream?`
`15:28.53`	`[TK]D-Fender`	`h8 @ GrandSuck`
`15:29.41`	`ManxPower`	`seele_: just press # as the extension number`
`15:29.44`	`hi365`	`my grandstream gxp-2000 doesnt want to take an update. the tftp server seems to be working fine. what to do?`
`15:29.59`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`15:32.07`	`[TK]D-Fender`	`hi365: Make sure your GS is even looking at the right server.`
`15:32.13`	`*** join/#asterisk Inkubot (n=fgutierr@200.74.182.15)`
`15:32.28`	`*** part/#asterisk Inkubot (n=fgutierr@200.74.182.15)`
`15:32.56`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com)`
`15:33.17`	`hi365`	`[TK]D-Fender, of course it is xx.xx.xx.99`
`15:34.05`	`ambriento`	`ManxPower, the thing is: he doesn't want to hear that prompt`
`15:34.06`	`[TK]D-Fender`	`hi365: You confirmed that the phone is looking at the right TFTP server?`
`15:34.25`	`seele_`	`ManxPower, I can automate this?`
`15:34.25`	`hi365`	`the log actualy show the phone hitting`
`15:34.48`	`[TK]D-Fender`	`seele_: There is a script on the WIKI that shows passing the DTMF to it automatically.....`
`15:35.00`	`ambriento`	`automagically`
`15:35.02`	`[TK]D-Fender`	`seele_: Go searth the WIKI`
`15:35.20`	`ManxPower`	`ambriento: too bad.`
`15:35.33`	`hi365`	`[TK]D-Fender, the log actualy show the phone hitting`
`15:35.52`	`seele_`	`[TK]D-Fender, ok but for all commands ... I need only for extension question`
`15:36.24`	`hi365`	`[TK]D-Fender, Oct 19 17:34:30 asterisk1.local atftpd[3086.-1208407120]: Serving gxp2000a.bin to xx.xx.xx.109:61756`
`15:37.23`	`[TK]D-Fender`	`seele_: Promt it yourself and pass the parameter for where to dial. it would then only ask for the password which you can automate as per the sample on the WIKI`
`15:37.42`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`15:37.42`	`*** mode/#asterisk [+o mog] by ChanServ`
`15:37.45`	`toerkeium`	`guys, I have installed trixbox and I see asterisk processes bein shutdown, do you know what could be wrong?`
`15:37.46`	`*** join/#asterisk Arnar (n=arnarb@landi.oddi.is)`
`15:37.56`	`Arnar`	`hi folks..`
`15:38.23`	`Arnar`	`I'm having some trouble connecting Asterisk to a Nortel Meridian1 Option 11 via a E1 PRI (EuroISDN)`
`15:38.32`	`seele_`	`I can verify the hour of access of the agent?`
`15:38.38`	`Arnar`	`does anyone here have any experience or insight?`
`15:39.37`	`drega`	`what are your issues Arnar, it might prompt additional response if you provide more information.`
`15:39.50`	`seele_`	`[TK]D-Fender, ok how can I make this in this lines http://pastebin.ca/210132`
`15:39.59`	`ambriento`	`I second drega`
`15:40.19`	`drega`	`one thing I can think of off hand would be to make sure that you have the timing set up correctly.`
`15:40.37`	`ambriento`	`and the endpoint`
`15:40.38`	`Arnar`	`drega: well, the link comes up (indicated zap show status) - but the pri trunk doesn't (indicated by "Provisioned, Down, Active" in pri show spans and stat dch on the nortel side)`
`15:40.45`	`ambriento`	`CPEvsNET`
`15:41.05`	`Arnar`	`the nortel is usr (cpe) and asterisk is net`
`15:41.39`	`Arnar`	`nortel has two tdm cards, one connected to the telco, the second card takes timing from there and I set up asterisk to take timing from nortel`
`15:42.03`	`ambriento`	`so far so doog`
`15:42.17`	`ambriento`	`oops :) good I meant`
`15:42.20`	`Arnar`	`:)`
`15:42.32`	`ambriento`	`is doog actually a word?`
`15:42.41`	`Arnar`	`we removed the clock daughter board from the second tdm card in the meridian`
`15:43.01`	`hi365`	`[TK]D-Fender, any advice?`
`15:43.27`	`ambriento`	`Arnar, do you have any output on * console?`
`15:44.04`	`Arnar`	`if I do "pri enable debug" (or intense debug) it's indicated that asterisk is sending a "set asynchronous balance mode extended" every second or soe..`
`15:44.05`	`Arnar`	`*so`
`15:44.12`	`Arnar`	`otherwise, no output no..`
`15:44.23`	`[TK]D-Fender`	`hi365: You need your Grandstream to accelerate at 10m/s^2 for approx 25ft and land at an incline of about 45 deg from the front edge`
`15:44.23`	`Arnar`	`well..`
`15:44.32`	`Arnar`	`I should see the output with "asterisk -r" ?`
`15:44.40`	`Arnar`	`or do I need to start asterisk with "-c" ?`
`15:44.59`	`hi365`	`[TK]D-Fender, well obviously u have issues with them. what do u recomened instead?`
`15:45.12`	`[TK]D-Fender`	`hi365: Polycom or Aastra`
`15:45.21`	`*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)`
`15:46.10`	`[TK]D-Fender`	`hi365: I personally suggest the IP 501 or 480i for general full-use`
`15:46.36`	`Arnar`	`if I start asterisk with "-C" I get two warnings, related to SMDI (or SDMI, spelled in two different ways in the same message) and "iax2 show provisioning" already loaded -- and one error, parse error for extensions.ael`
`15:46.47`	`Arnar`	`none of these should affect the pri trunk should they?`
`15:46.57`	`hi365`	`[TK]D-Fender, am o wrong for thinking that the GS has the most features? BLF, sip-notify for auto on speaker`
`15:46.59`	`hi365`	`?`
`15:47.55`	`ambriento`	`not really Arnar`
`15:48.06`	`[TK]D-Fender`	`hi365: Correct. Polycom has all that, supports more calls, mor provisioning methods, BLF & SLA, much better LCD usability (minus backlight), much better audio quality, etc.`
`15:48.21`	`ambriento`	`omg! whats happenning to me today! grrrr`
`15:48.23`	`[TK]D-Fender`	`hi365: The only thing GS wins out on is cost, but clearly "you get what you pay for"`
`15:48.53`	`Arnar`	`ambriento: I have a digium te405 .. I saw the sangome cards have an option to "trace" to see the traffic on the card.. is there something similar for the digium cards?`
`15:48.54`	`ambriento`	`Arnar, no. it won't affect`
`15:49.13`	`ambriento`	`I don't know such thing for digium cards`
`15:49.31`	`*** join/#asterisk QbY (i=user@cm-64-221-173-108.dhcp.southerncoastalcable.net)`
`15:49.41`	`Arnar`	`any pointers on how I can debug this further?`
`15:49.45`	`QbY`	`anyone have a good annotated SEP.xml file for the Cisco 7971`
`15:49.49`	`ambriento`	`with sangoma, you have a network interface visible by ifconfig`
`15:49.55`	`Arnar`	`oh, IC`
`15:50.19`	`Arnar`	`so the "trace" is just ethereal in that case?`
`15:50.42`	`Arnar`	`(or similar)`
`15:50.50`	`*** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net)`
`15:50.54`	`ambriento`	`kind of... sangoma provides its own "tcpdump"`
`15:51.39`	`ambriento`	`Arnar, I remember once connecting 1 * box to a Meridian`
`15:52.07`	`Arnar`	`ambriento: ok.. T1 or E1?`
`15:52.09`	`ambriento`	`I have no idea of how Nortel works or even any other legacy PBX`
`15:52.11`	`ambriento`	`E1`
`15:52.28`	`Arnar`	`I have a Nortel technician helping us..`
`15:52.37`	`ambriento`	`my part of the job was only at the * side`
`15:52.43`	`Arnar`	`he has set up connections to various other IP-PBX, for example Cisco.. but not *`
`15:53.19`	`Arnar`	`We followed setup-guides for meridian-* and meridian-cisco found online.. and everything seems normal on the Nortel..`
`15:53.31`	`Arnar`	`except of course the d-channel is "awaiting establishment"`
`15:53.45`	`ambriento`	`that's weird`
`15:54.03`	`[TK]D-Fender`	`Arnar: Make sure your Nortel is using PRI signalling`
`15:54.06`	`ambriento`	`are you sure the Nortel is using EuroISDN?`
`15:54.15`	`Arnar`	`Yes.. pretty sure`
`15:54.40`	`ambriento`	`I wish I could have access to that box`
`15:54.55`	`*** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com)`
`15:55.01`	`*** part/#asterisk dpro (n=dpro@80.109.121.47)`
`15:55.13`	`*** join/#asterisk Blafasel (n=bpodszun@pd95b71ae.dip0.t-ipconnect.de)`
`15:55.15`	`Arnar`	`well - I can dump any config if you want to see it..`
`15:55.18`	`ambriento`	`I would check it for you if I have something different over there that I'm not aware by heart`
`15:55.21`	`BlepsoaF`	`can anyone recommend a hardware device that is described in http://www.voip-info.org/wiki/index.php?page=Asterisk+failover+case1`
`15:55.34`	`ambriento`	`I meant my box connected to Nortel`
`15:55.40`	`BlepsoaF`	`Looking to have 2 machines with a hot failover incase 1 server goes down`
`15:55.55`	`Arnar`	`oh right :)`
`15:56.12`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net)`
`15:56.41`	`*** join/#asterisk p1p (i=pip@mail.comp911.com)`
`15:56.44`	`Blafasel`	`Hi there. I'm trying to build asterisk-ss7 from svn to see if libss7 solves my issues with chan_ss7. Unfortunately it just doesn't want to accept my zaptel installation and fails with "The Zaptel installation on this system appears to be broken.".. I end up with an asterisk from the ss7 branch without ss7.. My autoconf foo is useless, can anyone help me out?`
`15:57.35`	`Arnar`	`ambriento: I'm using * 1.4 beta.. should I try 1.3 instead?`
`15:57.47`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`15:58.17`	`*** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net)`
`15:58.18`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`15:58.35`	`*** part/#asterisk drega (n=drega@mail.amcat.co.uk)`
`15:59.22`	`ambriento`	`wow`
`15:59.30`	`ambriento`	`that's new thing :)`
`15:59.51`	`ambriento`	`my * is 1.2`
`16:00.08`	`HarryR`	`BlepsoaF: HP Proliant DL380 G4`
`16:00.43`	`Arnar`	`I mean 1.2 instead of 1.3 :)`
`16:01.18`	`*** join/#asterisk knarfly_wk (n=bwatson@12.42.132.26)`
`16:01.24`	`BlepsoaF`	`HarryR: I already have 2 dell servers`
`16:01.26`	`ambriento`	`can you try 1.2?`
`16:01.27`	`BlepsoaF`	`1850's`
`16:02.18`	`*** join/#asterisk blitz[dallas] (n=blitzrag@216-110-89-130.static.twtelecom.net)`
`16:02.20`	`blitz[dallas]`	`morning all`
`16:02.28`	`blitz[dallas]`	`actually.. guess its afternoon now (here)...`
`16:02.38`	`HarryR`	`BlepsoaF, but the DL380 g4 is soo much nicer than any dell server ;)`
`16:03.07`	`BlepsoaF`	`HarryR: well thats now what I'm interested in, I'm looking for hot failover to a new server incase one fails`
`16:03.19`	`blitz[dallas]`	`HarryR: those are nice servers`
`16:03.19`	`ambriento`	`ist noon in here too`
`16:03.20`	`HarryR`	`oh, you'll want to use heartbeat`
`16:04.07`	`Arnar`	`ambriento: downloading 1.2 now.. :)`
`16:04.16`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`16:04.19`	`ambriento`	`Arnar, good :)`
`16:04.21`	`BlepsoaF`	`HarryR: heartbeat?`
`16:04.37`	`HarryR`	`http://www.linux-ha.org/`
`16:04.49`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`16:05.26`	`blitz[dallas]`	`so I want reset the billsec value in my CDR, but keep the duration value -- any ideas?`
`16:05.44`	`blitz[dallas]`	`ResetCDR() seems to reset both`
`16:08.08`	`*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)`
`16:08.25`	`Arnar`	`ambriento: should I be using Linux 2.4 or 2.6?`
`16:08.25`	`BlepsoaF`	`HarryR: hmm I guess that would work from the server side, but I dont know how the PRI's would then know how to goto failover server`
`16:08.29`	`*** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)`
`16:08.38`	`ambriento`	`I say 2.6`
`16:09.08`	`benjk`	`BlepSoaF, they won't failover`
`16:09.09`	`tmccrary`	`Does anyone know what would cause Polycom 501 handsets to have interference and static during discussions? The speaker works decent, there is a little background noise but thats it`
`16:09.34`	`Arnar`	`ambriento: good.. otherwise I'd have to downgrade :)`
`16:09.36`	`Blafasel`	`Is there any * & ss7 related channel? Or can anyone tell me where I'd look for help with this * issue?`
`16:09.44`	`mosty`	`how can i answer a call and then forward it to another extension? ie how do i do it on the phone, and what do i need in my dialplan?`
`16:09.49`	`ambriento`	`=D`
`16:09.56`	`BlepsoaF`	`tmccrary: cell phones?`
`16:10.34`	`tmccrary`	`No, it has all kinds of static and noise all the time`
`16:10.45`	`benjk`	`Blafasel, which ss7 stack?`
`16:10.45`	`tmccrary`	`Polycom's voice quality seems to be horrible so far`
`16:10.47`	`blitz[dallas]`	`Blafasel: best off just asking the -ss7 mailing list`
`16:10.58`	`*** part/#asterisk knarfly_wk (n=bwatson@12.42.132.26)`
`16:11.01`	`*** part/#asterisk blitz[dallas] (n=blitzrag@216-110-89-130.static.twtelecom.net)`
`16:11.16`	`*** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net)`
`16:12.02`	`Blafasel`	`benjk: libss7 - but the issue is the asterisk-ss7 branch of the "mattf" guy at digium for me`
`16:12.24`	`Blafasel`	`I'm currently running chan_ss7 and have lots of problems, so I thought I could give libss7 a try.`
`16:12.38`	`benjk`	`mattf's IRC nick is Creslin`
`16:13.13`	`Blafasel`	`Thanks a lot.`
`16:13.25`	`benjk`	`he doesn't seem to be online right now`
`16:13.44`	`Blafasel`	`Yep, /whois'd as well ;)`
`16:13.52`	`benjk`	`there is also ss7box and a channel #ss7box`
`16:14.13`	`ambriento`	`Arnar, is that a brand new * box you just made it to start with nortel integration?`
`16:14.15`	`tamp4x`	`who would put * on an ss7 link that would be lolers`
`16:14.36`	`Blafasel`	`benjk: Free? This is only a proof of concept, so I have no budget yet.`
`16:15.11`	`benjk`	`not its about 4800 USD but they may well be willing to let you have a test ride`
`16:15.18`	`Arnar`	`ambriento: yup.. brand new`
`16:15.39`	`[TK]D-Fender`	`tmccrary: In the middle of a staticy call, take the handset from another phone and swap them. If it stops then you've ID'd the culprit. I've had 1-2 bad handsets before.`
`16:15.41`	`Arnar`	`ambriento: the nortel is our current pbx, and were just trying to hook them up to be able to test asterisk`
`16:15.42`	`Blafasel`	`benjk: Thanks a lot for your help. Just found the site and I'm already reading the docs`
`16:16.14`	`Arnar`	`ambriento: we are at the limits with the nortel, and we'd like to be able to migrate some extensions over to the * box`
`16:16.43`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`16:17.00`	`ambriento`	`Arnar, what I did was pretty much what you're doing now. Call to PSTN goes thru Nortel`
`16:17.03`	`[TK]D-Fender`	`tmccrary: Also, which model are you using, and which firmware?`
`16:17.06`	`ambriento`	`some extensions on * box`
`16:17.40`	`ambriento`	`and I installed the unistim channel to use some nortel phones on *`
`16:18.32`	`[TK]D-Fender`	`BCM... lol... cough`
`16:18.34`	`qdk`	`tamp4x who would put * on an ss7 link that would be lolers <- could you please elaborate?`
`16:19.46`	`qdk`	`Blafasel: chan_ss7 works fine as far as i know.`
`16:20.07`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`16:20.08`	`mosty`	`does anyone know how to do attended call transfers?`
`16:20.28`	`Blafasel`	`qdk: I have a running sample with chan_ss7 here, but it's very, very instable. It works. Sometimes. ;)`
`16:21.18`	`ambriento`	`mosty, check features.conf and use the option 't' or 'T' for Dial() cmd`
`16:21.28`	`[TK]D-Fender`	`BBIAB`
`16:21.53`	`*** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se)`
`16:21.54`	`qdk`	`Blafasel: ast2*CLI> show uptime`
`16:21.54`	`qdk`	`System uptime: 16 weeks, 6 days, 2 hours, 26 minutes, 9 seconds`
`16:21.55`	`qdk`	`Last reload: 16 weeks, 6 days, 2 hours, 24 minutes, 11 seconds`
`16:22.12`	`skopii`	`I am setting up a PBX for my company. I am wondering if I could get a toll free number with my SIP provider or if there is some third party (stable) service that just forwards those 1800 calls to the SIP provider?`
`16:22.12`	`Arnar`	`ambriento: nice..`
`16:22.17`	`Blafasel`	`Not the system is instable. The ss7 link is. At least for me`
`16:22.25`	`Arnar`	`unistim? that allows me to use nortel digital phones?`
`16:22.42`	`ambriento`	`yep`
`16:22.48`	`mosty`	`ambriento, i don't want to park calls though, i want to send calls to a specific context/extension/priority`
`16:22.50`	`Arnar`	`how's that working out?`
`16:22.55`	`qdk`	`Blafasel: define instable ss7 link?`
`16:23.00`	`Qwell[]`	`unstable..`
`16:23.05`	`ambriento`	`mosty, what version of * are you using?`
`16:23.12`	`tmccrary`	`D-Fender: They are Polycom 501's with 1.6 firmware. I can't use the newer firmware because it's incompatible with asterisk`
`16:23.16`	`mosty`	`ambriento, 1.2.something`
`16:23.27`	`FlatFoot`	`a`
`16:23.27`	`FlatFoot`	`r`
`16:23.29`	`ambriento`	`Arnar, you can even set up some keys`
`16:23.37`	`Arnar`	`ambriento: excellent..`
`16:23.58`	`Qwell[]`	`tmccrary: define "incompatible"`
`16:24.05`	`Blafasel`	`qdk: I get lots of protocol violations that confuse the other side, rejected calls, calls that aren't freed after hangup, lots of different warnings on the asterisk console and poor performance`
`16:25.04`	`ambriento`	`mosty, in features.conf you can setup like *2 for attended transfer`
`16:25.24`	`ambriento`	`mosty, you really need to read it :)`
`16:25.31`	`tmccrary`	`Qwell: Major features do not work with any stable version of Asterisk.`
`16:25.53`	`benjk`	`Blafasel, protocol errors on which layer?`
`16:26.12`	`qdk`	`Blafasel: Oct 19 18:26:39 NOTICE[3132]: chan_ss7.c:1889 ss7_write: Write buffer full on CIC=45 (wrote only 0 of 160), audio lost. <- thats about the only issue i have.`
`16:26.16`	`mosty`	`ambriento, i looked through that file before, the comment at the top says it's about parking, as did the wiki page. but parking is not what i want. do i just need to uncomment the "atxfer" bit in there?`
`16:26.33`	`qdk`	`Blafasel: what are you trying to do?`
`16:26.37`	`ambriento`	`mosty, we're not talking about the same thing`
`16:26.41`	`seele_`	`ambriento, como puedo especificar a que cola se registran los agentes ???`
`16:27.27`	`ambriento`	`mosty, does your file look like this? http://pastebin.ca/210212`
`16:27.40`	`Blafasel`	`qdk: I have a customer that just tries to start as a telco/carrier here and wants to see a sip <-> ss7 bridge`
`16:27.45`	`mosty`	`ambriento, i just want my phones to be able to do something like #123 to send the caller to a different extension (possibly in a different context)`
`16:27.53`	`ambriento`	`seele_, con AgentCallback() ?`
`16:28.01`	`Blafasel`	`So I'm doing nothing fancy, just ss7 -> * -> sip and vice versa.`
`16:28.07`	`*** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net)`
`16:28.10`	`benjk`	`Blafasel, which SS7 standard? are you sure you are using the right one?`
`16:28.22`	`mosty`	`ambriento, it looks like that, except mine has lots of it commented out`
`16:28.22`	`ambriento`	`mosty, and try CLI> show features`
`16:28.23`	`benjk`	`there are about 100+ different ones`
`16:28.55`	`ambriento`	`mosty, you're a lazy boy ;)`
`16:29.15`	`Blafasel`	`benjk: No, actually I'm not. Because I'm from the sip side of the fence and the guys at the customer aren't that bright/experienced either. The errors are ISUP related, as far as I read the traces`
`16:29.35`	`qdk`	`Blafasel: ok, works fine here.. i mostly use SIP <-> IAX <-> SS7... only did some init. SIP2SS7.`
`16:29.39`	`benjk`	`ok, which country are you in?`
`16:29.46`	`mosty`	`ambriento, i have looked through this stuff, but i haven't yet found definitions for many of these terms, it's hard to see what corresponds to the behaviour i want`
`16:29.47`	`Blafasel`	`germany`
`16:30.13`	`benjk`	`more precisely, which country is the SSP or STP you are connecting to?`
`16:30.39`	`qdk`	`Blafasel: Denmark here.`
`16:30.51`	`ambriento`	`mosty, I see... well atxfer stands for attended transfer. The first thing you asked in was about atxfer, which now seems that is not what you need`
`16:30.59`	`Blafasel`	`benjk: same`
`16:31.03`	`*** join/#asterisk x86 (n=x86@p3m/member/x86)`
`16:31.05`	`BlepsoaF`	`what are peoples thoughts on RED-PHONE`
`16:31.16`	`benjk`	`and this is a node on the national level or the international level?`
`16:31.31`	`Blafasel`	`benjk: But I'm not even sure if they managed to configure it right.. I know that their endpoint supports multiple protocol standards.`
`16:31.33`	`Blafasel`	`national`
`16:31.43`	`mosty`	`ambriento: from what i gather, attended transfer is where the called party dials something and then the extension to transfer to, right?`
`16:31.47`	`benjk`	`ok, that would be ETSI then`
`16:31.52`	`Arnar`	`ambriento: unistim only works for norte IP phones? - not digital sets?`
`16:31.57`	`ambriento`	`mosty, yep`
`16:32.10`	`*** join/#asterisk macli (n=vincent@mcli.brc.ubc.ca)`
`16:32.13`	`mosty`	`ambriento: but i don't understand what blind transfer is`
`16:32.19`	`benjk`	`you want to check if the stacks on both sides are configured to speak the same SS7`
`16:32.22`	`*** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)`
`16:32.31`	`Blafasel`	`benjk: I don't even see a way to tell chan_ss7 which standard to use?`
`16:33.09`	`benjk`	`maybe qdk can tell you about that`
`16:33.14`	`qdk`	`Blafasel: read the doc and see if the standard is even supported.`
`16:33.30`	`*** join/#asterisk Peperzaken (n=Zout@cust-02-52869dc8.adsl.scarlet.nl)`
`16:33.46`	`ambriento`	`Arnar, yes IP phones`
`16:33.51`	`Arnar`	`ambriento: no luck with * 1.2 :(`
`16:33.58`	`Peperzaken`	`Someone form Digium here?`
`16:34.06`	`Qwell[]`	`Peperzaken: several`
`16:34.09`	`ambriento`	`Arnar, same thing?`
`16:34.10`	`qdk`	`Blafasel: chan_ss7 is ITU only as far as i remember, and pretty much only "one way" through the stack.`
`16:34.22`	`Arnar`	`ambriento: yup.. the pri span is "Down"`
`16:34.30`	`Peperzaken`	`Qwell[]: Okay, including someone form Sales`
`16:34.46`	`Qwell[]`	`Peperzaken: If you need Sales, I highly recommend calling`
`16:34.56`	`benjk`	`if it is ITU-T only, then it is possible that it has troubles with your national ISUP`
`16:35.05`	`ambriento`	`~blindtransfer`
`16:35.12`	`benjk`	`ITU-T is only for the international network`
`16:35.20`	`mosty`	`ambriento, so with attended transfer i have two questions: what does the callee press to transfer to say extension 123, and in what context must that extension be?`
`16:35.45`	`Blafasel`	`Hmm..`
`16:35.52`	`*** join/#asterisk BrokenNoze (n=root@host81-159-209-149.range81-159.btcentralplus.com)`
`16:36.02`	`Peperzaken`	`Qwell[]: Only thing i'd like to know is why i didn't receive my licence code........`
`16:36.05`	`Blafasel`	`Maybe I can ask the customer to reconfigure their side and check if the situation improves`
`16:36.15`	`Qwell[]`	`Peperzaken: Call them, and they will help you.`
`16:36.34`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.163)`
`16:36.36`	`ambriento`	`mosty, blind transfer you will hit the sequence defined in features.conf, then you'll hear a prompt "Transfer" followed by a dialtone`
`16:36.43`	`benjk`	`blafasel, you may want to get yourself this book: Lee Dryburgh, Signaling System No.7, ISBN 1-58705-040-4, CiscoPress`
`16:36.43`	`qdk`	`Blafasel: have you tried the chan_ss7 2 chan_ss7 test setup?`
`16:36.47`	`Peperzaken`	`okay, i will then`
`16:36.49`	`*** part/#asterisk Nix (n=Nix@195.175.21.46)`
`16:36.54`	`ambriento`	`then you dial the extension you want and the call is dropped`
`16:37.04`	`BrokenNoze`	`Anyone help a newbie with a Monitor() issue?`
`16:37.09`	`Blafasel`	`qdk: No`
`16:37.15`	`ambriento`	`at that time, the caller was transfered to the extension you dialed`
`16:37.31`	`Arnar`	`ambriento: if I look at zttool.. there doesn't seem to be any activity on the link`
`16:37.37`	`qdk`	`Blafasel: that will show you that its quite stable, and that the issue is probably just protocol mismatch/negotiation problem with the "real" ss7 site.`
`16:37.40`	`Blafasel`	`benjk: I've borrowed it recently but have yet to read it.. Thank you`
`16:38.00`	`ambriento`	`Arnar, I think zttool does not show anything`
`16:38.01`	`Blafasel`	`qdk: I didn't state anything different.`
`16:38.02`	`Arnar`	`could there be some version/compile mismatch between zaptel, libpri and asterisk?`
`16:38.08`	`Arnar`	`ambriento: oh.. ever?`
`16:38.12`	`BrokenNoze`	`My monitor function is creating the file but only 44k, and neither SIP handsets are audible to one another`
`16:38.17`	`qdk`	`Blafasel: didnt say that you did. ;-)`
`16:38.29`	`hi365`	`my grandstream isnt taking any upgread's!!!!`
`16:38.30`	`ambriento`	`Arnar, as far as I remember, it only tells you if is up or down`
`16:38.35`	`Peperzaken`	`Qwell[]: Got it, thanks!`
`16:38.39`	`qdk`	`Blafasel: just confirming that it is possible to make ss7 work, and quite nice even.`
`16:38.43`	`Arnar`	`ok`
`16:38.44`	`mosty`	`ambriento: hmm, i am so confused now. what is the difference between blind and attended? does attended turn it into a 3-way call or something?`
`16:39.02`	`Blafasel`	`qdk: Thank you. I'll have to recheck that tomorrow`
`16:39.16`	`qdk`	`Blafasel: but the SS7 protocol stack is huge so i can be a bit finding the problem(s).`
`16:39.20`	`ambriento`	`mosty, attended transfer, when you dial the other extension, you can talk to the other party before sending'em the original caller and when you hang up the caller is transfered`
`16:39.35`	`Blafasel`	`Wish me luck ;)`
`16:40.11`	`qdk`	`so it*`
`16:40.14`	`qdk`	`Blafasel: gl ;-)`
`16:40.18`	`ambriento`	`Arnar, is it your first experience with * and PRI cards?`
`16:40.30`	`mosty`	`ambriento, ok then blind is what i want. i will play around with it myself before hassling the channel any more. thanks for your help`
`16:40.41`	`BrokenNoze`	`Anyone point me in direction of some help stuff I haven't already seen on voip-info at least?`
`16:40.44`	`Blafasel`	`Just called the customer. They are sure to use ITU-T though..`
`16:40.46`	`Blafasel`	`sigh`
`16:40.55`	`ambriento`	`mosty, in both cases, when you are dialing the other extension, the caller is put on hold`
`16:41.08`	`ambriento`	`good luck mosty :)`
`16:41.17`	`Arnar`	`ambriento: yes..`
`16:41.21`	`mosty`	`but #123 sends the caller to the extension 123 in the callee's default context?`
`16:41.40`	`Arnar`	`ambriento: does it show? :)`
`16:42.04`	`ambriento`	`not at all Arnar. :) I was just guessing :)`
`16:42.27`	`Arnar`	`ambriento: :)`
`16:42.43`	`ambriento`	`mosty, what CLI> show features tells you?`
`16:42.52`	`Arnar`	`ambriento: so I've exhausted all ways to debug this?`
`16:42.53`	`benjk`	`Blafasel, they may not know what they are doing`
`16:43.13`	`mosty`	`ambriento, Blind Transfer # #1, at the moment`
`16:43.18`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`16:43.21`	`Blafasel`	`benjk: Possible, yes. ;) But I cannot complain, since I'm only confident with sip as well..`
`16:43.23`	`Arnar`	`ambriento: one idea we had was to try qsig instead of euroisdn - but maybe that doesn't matter since it relies on the d-channel being up?`
`16:43.28`	`benjk`	`SS7 is split into two different worlds`
`16:43.45`	`benjk`	`national networks and the international network`
`16:43.58`	`ambriento`	`mosty, if your Dial() has the option 't', you'll be able to hit #1 and hear "transfer" + dialtone and dial other extension`
`16:44.06`	`benjk`	`ITU-T only specifies the international network`
`16:44.11`	`ambriento`	`Arnar, I agree`
`16:44.30`	`benjk`	`many of the things in the specs say something like "this part is up to the national administration to decide for themselves"`
`16:44.51`	`[TK]D-Fender`	`DTMF transfers are BS. horrible idea to be avoided.`
`16:45.00`	`Arnar`	`ambriento: sorry.. what do you mean? Agree we should try Q.Sig or that Q.Sig probably won't work since the d-channel is out?`
`16:45.27`	`ambriento`	`Arnar, I agree with no need to try qsig`
`16:45.33`	`Arnar`	`ok`
`16:45.34`	`benjk`	`so if your customer is connected to the national network in Germany, all those parts which are defined by the German standardisation body (whoever that is) may not be in line with your ITU-T implementation`
`16:45.35`	`mosty`	`ambriento, ok, i changed that back to #. i am sorted now. thankyou again`
`16:45.44`	`Arnar`	`ambriento: any other ideas that I could try?`
`16:46.39`	`ambriento`	`Arnar, does this box internet access? what do you think about I'm connecting to it?`
`16:46.39`	`benjk`	`anyway, I think you should read that book you borrowed over this coming weekend first ;)`
`16:46.53`	`BlepsoaF`	`whats everyones thoughs on RED-PHONE`
`16:47.00`	`ambriento`	`I missed the "has" :(`
`16:47.36`	`Arnar`	`ambriento: well.. I'd have to run it by my supervisor :)`
`16:47.58`	`Blafasel`	`benjk: Will do, thanks anyway.`
`16:48.19`	`BrokenNoze`	`please, could really do with 5 secs if someone has the time. totally stumped`
`16:48.28`	`benjk`	`Blafasel, the author of this book maintains an online SS7 forum at www.c7.com`
`16:48.38`	`ambriento`	`lets check some other stuff meanwhile`
`16:48.46`	`BlepsoaF`	`[TK]D-Fender: have you used any devices like RED-PHONE?`
`16:48.55`	`ambriento`	`Arnar, can you pastebin your zaptel.conf and zapata.conf?`
`16:49.07`	`ambriento`	`sure Arnar`
`16:49.09`	`Blafasel`	`Thank you again. Another thing to read ;)`
`16:50.41`	`Arnar`	`ambriento: sure.. I'll paste`
`16:50.41`	`ambriento`	`nice`
`16:50.41`	`benjk`	`once you've read the book, you should be able to ask intelligent questions and post them on that forum ;)`
`16:50.54`	`*** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com)`
`16:50.55`	`Arnar`	`that's zaptel.conf`
`16:50.55`	`[TK]D-Fender`	`BlepsoaF: TDMoE = who cares? virtually no-one uses it. I understand you desire for redundancy but that things purpose can be done by a NORMAL machine which is no less prone to failure and opens up better card options.`
`16:51.16`	`[TK]D-Fender`	`BlepsoaF: From there (a colo w/ your PRI's) you can then push out calls as you see fit.`
`16:51.19`	`Arnar`	`and that's zapata`
`16:51.28`	`benjk`	`BlepsoaF, TDMoE is very immature, lots of kernel panics and crashes`
`16:52.20`	`jgoo`	`is there a good set of agi tutorial anywherE?`
`16:52.26`	`*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)`
`16:52.28`	`drcode`	`hi all`
`16:52.31`	`jgoo`	`things like: how to forward a channel to voicemail`
`16:52.31`	`drcode`	`whats up`
`16:53.05`	`*** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net)`
`16:53.10`	`SuPrSluG`	`having an issue w/ the caller not hearing ringing,just silence. The callee hears it just fine. Is this a service provider issue or config ?`
`16:53.11`	`MrTelephone`	`does anyone here use PRI in canada?`
`16:53.41`	`MrTelephone`	`supprslug, ur phone isn't receiving an ack that ringing has began`
`16:53.48`	`Peperzaken`	`anyone know about a volume licence for the g729 codec?`
`16:53.49`	`BrokenNoze`	`Hi drcode any chance of a little help with my Monitor() problem?`
`16:53.50`	`MrTelephone`	`return path is failing on the sip signalling`
`16:54.14`	`MrTelephone`	`peperzaken, what about it? 10$/channel`
`16:54.16`	`SuPrSluG`	`MrTelephone:is that my end or the providers?`
`16:54.37`	`Peperzaken`	`MrTelephone: i need like 5000 licences.....`
`16:54.39`	`MrTelephone`	`supprslug, how are you setup? wheres your phone? do you have a local asterisk box?`
`16:54.46`	`MrTelephone`	`5000? u serious`
`16:54.56`	`[TK]D-Fender`	`jgoo: What does AGI have to do with voicemail? You don't forward anything to voicemail, you just have the call call the app at thats it.`
`16:54.58`	`MrTelephone`	`use 711 for local networks`
`16:55.18`	`Peperzaken`	`we are`
`16:55.19`	`[TK]D-Fender`	`Peperzaken: I highly doubt you need 5000 licenses...`
`16:55.25`	`MrTelephone`	`use 729 over wan links? you have 5000 concurrent calls?`
`16:55.30`	`MrTelephone`	`me neither`
`16:55.48`	`Peperzaken`	`I did not say it would be on one server`
`16:55.51`	`ambriento`	`Arnar, do I have to be logged in? cause I cannnot see anything`
`16:55.56`	`SuPrSluG`	`MrTelephone:it's a nufone 800 number. the ivr works. but when the call is passed the caller hears no ringing.`
`16:56.02`	`MrTelephone`	`you'd have to have a super datacenter server to manage 5000 calls at 729 with 100 cpus`
`16:56.06`	`Arnar`	`ambriento: hmm.. one moment`
`16:56.20`	`Peperzaken`	`1000 servers (aprox) and 5 per each`
`16:56.23`	`Arnar`	`ambriento: oh.. sorry..`
`16:56.30`	`msa20`	`SuPrSluG: did you troubleshoot with nufone?`
`16:56.31`	`[TK]D-Fender`	`Peperzaken: Still... explain how it is you think you'll need that many...`
`16:56.33`	`Qwell[]`	`Peperzaken: again, you're going to want to ask Digium sales.`
`16:56.41`	`Arnar`	`forgot to select what I wanted to paste :P`
`16:56.45`	`*** join/#asterisk macli (n=vincent@mcli.brc.ubc.ca)`
`16:56.54`	`ambriento`	`:)`
`16:57.14`	`MrTelephone`	`supprslug, nufone ---> asterisk or nufone ---> router ---> internet --> asterisk?`
`16:57.39`	`SuPrSluG`	`MrTelephone:I just sent in a support ticket. They sometimes take a while.`
`16:58.25`	`Qwell[]`	`Peperzaken: 1000 servers? Why so many?`
`16:58.28`	`MrTelephone`	`Do you have access on the asterisk box?`
`16:58.52`	`MrTelephone`	`1000 servers? I have 2 linux boxes runnin an entire isp`
`16:59.10`	`MrTelephone`	`thats a super dooper corporation`
`16:59.19`	`Arnar`	`ambriento: these work.. just select "mac classic" as the theme for less ugly colours`
`16:59.34`	`Peperzaken`	`we build small PBXes for small company's`
`16:59.38`	`MrTelephone`	`ahhh`
`16:59.44`	`Qwell[]`	`Peperzaken: ahh, okay`
`16:59.49`	`Qwell[]`	`That makes a whole lot more sense :)`
`16:59.53`	`ambriento`	`:)`
`16:59.57`	`Peperzaken`	`like the linksys boxex but with mutch more featres`
`17:00.01`	`mut`	`next time you're hiring give me a call Peperzaken`
`17:00.03`	`ambriento`	`looking at'em`
`17:00.14`	`Qwell[]`	`Peperzaken: yeah, definitely call Digium sales`
`17:00.15`	`Peperzaken`	`Live in the netherlands?`
`17:00.16`	`MrTelephone`	`should be 5 dollars a license`
`17:00.28`	`*** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net)`
`17:00.47`	`saftsack`	`hi`
`17:00.49`	`jgoo`	`[TK]D-Fender: to the initiated that sounded fairly simple :-)`
`17:00.55`	`mut`	`no, but i could move`
`17:00.56`	`mut`	`:P`
`17:01.09`	`jgoo`	`[TK]D-Fender: I worked it out actually, I was just looking for examples of some more 'complex' agi work`
`17:01.26`	`Peperzaken`	`Qwell[]: I will call them next week, first we are going to test the codec`
`17:01.27`	`saftsack`	`if i raise my handset on my analog telephone followed by a very fast dial i cannot hear any voice in the call. does someone has an`
`17:01.33`	`[TK]D-Fender`	`jgoo: ok....`
`17:01.51`	`[TK]D-Fender`	`Peperzaken: Again, what do you think you'll need that many licenses?`
`17:02.01`	`Qwell[]`	`[TK]D-Fender: <Peperzaken> we build small PBXes for small company's`
`17:02.17`	`jgoo`	`I don't know about you, but when I don't know everything about something, I like to see what other people are doing with it, and read code... the best form of documentation is code`
`17:02.40`	`[TK]D-Fender`	`Peperzaken: Ah, so you basically want to provide them the licenses local to their servers?`
`17:03.06`	`Peperzaken`	`[TK]D-Fender: Well... if we say to our clients that thay can put people on hold and stuf they need at least 5 licences`
`17:03.15`	`Peperzaken`	`[TK]D-Fender: yeah thats the idea`
`17:03.19`	`[TK]D-Fender`	`jgoo: Might have helped if I understood what you were trying to accomplish, but yeah :)`
`17:03.19`	`Qwell[]`	`Peperzaken: not entirely true.`
`17:03.35`	`Qwell[]`	`Peperzaken: If you had g729 music on hold files...you wouldn't(shouldn't...) need to transcode`
`17:03.42`	`*** join/#asterisk pingwin[work] (i=pingwin@gateway/tor/x-0a8347f0a5b2d834)`
`17:03.53`	`[TK]D-Fender`	`Peperzaken: So your clients are expecting to use VoIP to terminate calls outside their local LAN and would be using G.729 then?`
`17:04.06`	`*** join/#asterisk xnon_ (n=xnon@200.8.30.161)`
`17:04.28`	`Peperzaken`	`Qwell[]: that's true, we are going to do just that if the codec suits our needs`
`17:05.04`	`[TK]D-Fender`	`Peperzaken: Have you considered the larger Licencing issues such as transfer of ownership, long terms support, etc?`
`17:05.08`	`BlepsoaF`	`Can anyone recommend an a/b fallback swich for t1's... I really only need to support 4 pri lines that would failover to a hot swap server`
`17:05.32`	`ambriento`	`Arnar, nothing wrong with both files`
`17:05.46`	`Arnar`	`ambriento: well that's good :)`
`17:06.08`	`Peperzaken`	`[TK]D-Fender: As Qwell[] said, if the codec suits our needs, then we contact digium to have a chat about it`
`17:06.21`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`17:06.32`	`ambriento`	`Arnar, have you tried unplug the cable?`
`17:06.37`	`Peperzaken`	`[TK]D-Fender: so probablt we will end up with some kind of special licence`
`17:06.42`	`[TK]D-Fender`	`Peperzaken: Ok, well you are in a somewhat special category, so good luck with your plans for this.`
`17:06.52`	`Arnar`	`ambriento: not since we got the link to work..`
`17:07.03`	`Peperzaken`	`[TK]D-Fender: Thanks!`
`17:07.06`	`*** join/#asterisk pifiu (n=someone@216.5.79.1)`
`17:07.10`	`SuPrSluG`	`MrTelephone:Here's the message from nufone. We send session progress messages out of band. Also, make sure you are running Asterisk 1.2.13`
`17:07.11`	`pifiu`	`hey everyone`
`17:07.13`	`*** join/#asterisk xnon (i=xnon@200.8.30.161)`
`17:07.20`	`Arnar`	`ambriento: just unplug it and plug it back in?`
`17:07.25`	`ambriento`	`yep`
`17:07.53`	`[TK]D-Fender`	`BlepsoaF: You putting your termination server into a colo you won't have access to?`
`17:08.06`	`Arnar`	`I have powercycled the * box a few times.. would that do the same? (the box is in another building, but I could get my wife to do it - show works in the next room from it :o)`
`17:08.07`	`msa20`	`Nufone answers tickets as if the person receiving the response is an ITSP employee`
`17:08.26`	`ambriento`	`it's the same`
`17:08.27`	`BlepsoaF`	`[TK]D-Fender: no its in the office, but we want a complete failover solution because the asterisk server will be handling more than just office calls`
`17:08.28`	`*** part/#asterisk pingwin[work] (i=pingwin@gateway/tor/x-0a8347f0a5b2d834)`
`17:08.43`	`[TK]D-Fender`	`BlepsoaF: What else?`
`17:09.04`	`BlepsoaF`	`Outside conference calls, automated systems that use AGI`
`17:09.23`	`Arnar`	`ambriento: my wife is unplugging/plugging right now..`
`17:09.34`	`pifiu`	`hey fnender`
`17:09.40`	`ambriento`	`hehe`
`17:09.55`	`ambriento`	`omg, you have a 11c in your house!`
`17:09.59`	`ambriento`	`:)`
`17:10.00`	`[TK]D-Fender`	`BlepsoaF: Ok, weil I'd suggest you go for a "pure routing" box with the PRI card in it, and then just run 2 "pure-voip" * boxes for the rest.`
`17:10.23`	`[TK]D-Fender`	`pifiu: y0`
`17:10.31`	`[TK]D-Fender`	`pifiu: Mission Accomplished?`
`17:10.39`	`pifiu`	`i think so fender, want you to look it over`
`17:10.52`	`*** join/#asterisk leikh (n=chatzill@x1-6-00-13-46-cc-b1-ad.k217.webspeed.dk)`
`17:11.02`	`pifiu`	`i think its perfect! lol but who knows i know there could always be improvements or small un-noticeable mistakes`
`17:11.42`	`Arnar`	`ambriento: nope.. not really, my wife works at the same place I do :)`
`17:11.46`	`[TK]D-Fender`	`pifiu: Ok, PM me`
`17:11.58`	`BlepsoaF`	`[TK]D-Fender: do you know of any hardware switches that would just do a a/b switch over?`
`17:12.10`	`*** join/#asterisk dwrecktion (n=dwreckti@209.60.234.170)`
`17:12.26`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`17:12.44`	`[TK]D-Fender`	`BlepsoaF: Not offhand, nor have I heard mention of that kind of stuff around here. You'd be best served calling an interconnector. They're the ones who use that kinda gear`
`17:13.18`	`leikh`	`i'm trying to use the destar GUI, but I have no idea how to set up an account so that I can can login into this GUI. Any ideas?`
`17:14.19`	`jgoo`	`anyway of getting better debug? I see == Spawn extension (custom-isec, s, 1) exited non-zero on 'Zap/1-1'`
`17:14.19`	`jgoo`	`but, even though I know what line killed it... I don't see any reference to that... I am on -v9`
`17:14.22`	`[TK]D-Fender`	`leikh: GUI's are not supported in this channel. Try asking on their own if they have one, or on a message-board/mailing list`
`17:14.26`	`Arnar`	`ambriento: ok.. plugged out and in.. no change :(`
`17:14.46`	`Arnar`	`the link just went down and up again.. but no change to the pri span`
`17:15.04`	`[TK]D-Fender`	`jgoo: Perhaps you could pastebin your diaplan for us to debug along with the FULL CLI output of that call.`
`17:15.18`	`ambriento`	`Arnar, PM`
`17:15.21`	`leikh`	`[TK]D-Fender: so there's no setting for admin and pin in asterisk configs?`
`17:15.39`	`*** join/#asterisk oQPa (n=Spain@86.Red-83-33-148.dynamicIP.rima-tde.net)`
`17:15.42`	`jgoo`	`[TK]D-Fender: there is only one line that is non-normal. I just call Voicemail(s501) << like that`
`17:15.47`	`*** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se)`
`17:15.49`	`[TK]D-Fender`	`leikh: destar's problems have nothing to do with *.`
`17:16.16`	`jgoo`	`so, there, you have both lines :p if nothing looks weird there, I will try something else, I checked the voicemail is there (I can check it)`
`17:16.23`	`leikh`	`[TK]D-Fender: i haven't localized the problem, yet...`
`17:16.38`	`[TK]D-Fender`	`jgoo: Please just show what I asked for. If you really did know what you were doing then odds are you wouldn't be asking us about it and everything would be working. That does not appear to be the case...`
`17:17.26`	`backblue`	`there is anything more evoluted than dia?`
`17:17.39`	`teknoprep`	`anyone have any idea why a DID would go through right away on my asterisk box.. when i call my DID from whereever... it shows up instantly on my asterisk box.. but the person who calls won't hear a ring fro about 6 seconds`
`17:18.24`	`teknoprep`	`<PROTECTED>`
`17:18.41`	`teknoprep`	`keeps saying that ... on an incomming call.. is that normal ?`
`17:18.54`	`[TK]D-Fender`	`teknoprep: We'd need to see the full CLI output for the call to get a better idea`
`17:19.22`	`teknoprep`	`debug?`
`17:19.24`	`teknoprep`	`or just normal ?`
`17:20.03`	`teknoprep`	`seems like the ring group is taking a long time to start the ringing on all phones`
`17:20.05`	`*** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com)`
`17:20.14`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`17:20.20`	`jgoo`	`did anyone catch my quit message then? someone paste it to me`
`17:21.07`	`eKo1`	`For some reason, the D-channel on some of my spans keep going up and down. What could be the cause of this? The only way to fix it is to reboot the whole machine.`
`17:22.04`	`teknoprep`	`pastebin is really slow lately`
`17:22.29`	`eKo1`	`Rats. I guess I'm the only one with this problem.`
`17:22.49`	`Juggie`	`eKo1, check make sure the span has a timeing source`
`17:22.57`	`*** join/#asterisk seele_ (n=seele@208.35.117.195)`
`17:23.25`	`seele_`	`how can I configure AgentCallbackLogin to epecific queue`
`17:23.45`	`teknoprep`	`[TK]D-Fender, http://rafb.net/paste/results/5JA2ZE44.html`
`17:24.15`	`teknoprep`	`thats the IAX debug`
`17:24.29`	`[TK]D-Fender`	`teknoprep: Do you see * saying its answering pretty much immediately?`
`17:24.43`	`teknoprep`	`yes`
`17:24.44`	`ambriento`	`seele_, with callback, you wont specify queue`
`17:24.57`	`[TK]D-Fender`	`teknoprep: If you see the answer being immediate and then just taking long to actually start ringing phones, DITCH AMP.`
`17:25.00`	`jgoo`	`<PROTECTED>`
`17:25.06`	`jgoo`	`[TK]D-Fender: I posted that wrong chan`
`17:25.12`	`teknoprep`	`[TK]D-Fender, thats what it is...`
`17:25.21`	`ambriento`	`seele_, inside your queue you'll have which agents are part of it`
`17:25.50`	`seele_`	`ambriento, ok let me check`
`17:26.19`	`seele_`	`ambriento, like static agents`
`17:26.46`	`ambriento`	`yep`
`17:26.51`	`ambriento`	`the members =>`
`17:26.53`	`*** part/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`17:28.18`	`carrar`	`[TK]D-Fender, how fast do you usually set your polycoms to resubscribe to subscriptions?`
`17:29.04`	`carrar`	`(after a reload)`
`17:30.24`	`[TK]D-Fender`	`carrar: I leave mine to default so it could last a REALLY long time. Not sure really. They re-reg if failed, thats about all I've done to date`
`17:32.11`	`japerry`	`[TK]D-Fender: Do you ever see issues with your Polycom's hanging up a call after 60 seconds?`
`17:32.28`	`*** join/#asterisk queuetue (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca)`
`17:32.33`	`carrar`	`When I change the expire settings on the polycom configs to 1, it seems to be 1 min for the extension registration but not for blf subscriptions`
`17:32.38`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`17:32.52`	`carrar`	`still messing with it trying to figure out how to speed that up`
`17:32.54`	`eKo1`	`Quick ?: If I upgrade Zaptel, do I need to recompile Asterisk?`
`17:34.05`	`seele_`	`ambriento, ok works .... other problem now`
`17:34.08`	`[TK]D-Fender`	`japerry: Nope, sounds like an * problem to me.`
`17:34.09`	`seele_`	`ambriento, look`
`17:34.16`	`seele_`	`ambriento, *[JB LOG]* fixed jitterbuffer created on channel Local ...`
`17:34.34`	`japerry`	`[TK]D-Fender: hmm okay. yah the logs report that the call never got completed`
`17:34.38`	`[TK]D-Fender`	`carrar: Oh, presence subscriptions. Even LESS of a clue about that :)`
`17:34.43`	`queuetue`	`Hi. I have two standalone asterisk servers, and would like to be able to call one from the other over IP. I assume the protocol I'd use would be IAX2 - can anyone point me towards a tutorial on how to set both ends' IAX and extensions up?`
`17:34.47`	`japerry`	`which is odd because it did... only happens on the polycom phone, and on outgoing calls`
`17:35.42`	`*** part/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`17:35.45`	`[TK]D-Fender`	`queuetue: You can use any protocol really. IAX2 has a few specific advantages/requirements over SIP. look up "asterisk dual servers" on the WIKI. they provide various implementation samples for you to try`
`17:36.11`	`*** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it)`
`17:36.11`	`[TK]D-Fender`	`japerry: Then its DEFINATELY your * config`
`17:36.44`	`queuetue`	`[TK]D-Fender: Thank you. For the prompt response, and also for giving the clearest, most helpful response I've ever got in here. :)`
`17:36.46`	`japerry`	`heh I'm wondering if its our shity flexgrow`
`17:37.09`	`japerry`	`no one seems to have this type of T1 setup, and apparently its not that great--many little bugs in it`
`17:37.12`	`*** join/#asterisk soda_ (n=soda@www.dat.demokritos.gr)`
`17:37.20`	`soda_`	`any unmatched call should go to the context defined in [general] right ?`
`17:37.31`	`*** join/#asterisk msa20 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`17:37.41`	`jgoo`	`so [TK]D-Fender: I sorted it out, see, no need for config files!`
`17:37.48`	`[TK]D-Fender`	`japerry: Kick your telco firmly in the ass and tell them to give you PRI or they'll find themselves "in the doghouse" in no time flat..`
`17:37.49`	`ambriento`	`seele_, that's not really a problem. it's saying that a JB was fixed. thats good :)`
`17:37.50`	`jgoo`	`channel.exec("VoiceMail", "s501");`
`17:38.04`	`jgoo`	`I was using sendCommand, and that was stupid of me`
`17:38.22`	`[TK]D-Fender`	`jgoo: I see NOTHING, because thats what you've shown us so far...`
`17:38.23`	`jgoo`	`how sweet awesome is that. fastAgi development is the best way to develop imnsho`
`17:38.53`	`[TK]D-Fender`	`jgoo: AGI itself adds a big load...`
`17:38.55`	`toerkeium`	`guys, is it normal that asterisk restart itself so often? safe_asterisk, opserver, etc ?`
`17:39.24`	`jgoo`	`[TK]D-Fender: I pasted the exact two lines. shall I repaste them for the benefit of hindsight? I am not ragging you but, you seemed to be quite peeved and gave the impression I was some troll demanding attention, instead of someone asking if there is a more debug option :p`
`17:39.35`	`[TK]D-Fender`	`toerkeium: Its not normal for * to restart itself period. how often is this happening?`
`17:39.54`	`jgoo`	`[TK]D-Fender: a big CPU load?`
`17:39.56`	`[TK]D-Fender`	`jgoo: Sorry if I sound a little harsh....`
`17:40.08`	`toerkeium`	`[TK]D-Fender: about 4 hours more or less.. I have 2 boxes, Celeron 2,53 Ghz, with 1 GB of ram`
`17:40.14`	`jgoo`	`[TK]D-Fender: I always try and assume the best, irc is a cold interface :p`
`17:40.17`	`seele_`	`ambriento, but I can't make a call to agent`
`17:40.39`	`[TK]D-Fender`	`jgoo: Yes, the AGI envirnment adss quite a noticable load to a server and its recommended you only use it for the little bits you really HAVE to use it for.`
`17:40.59`	`[TK]D-Fender`	`toerkeium: OUCH. Something is tragically wrong then. What * versions?`
`17:41.25`	`toerkeium`	`[TK]D-Fender: it's 1.2.12.1`
`17:42.27`	`toerkeium`	`[TK]D-Fender: and I have changed hardware and reinstalled everything 5 times, I see this happening when I connect 10 SIP operators, if I use it with my regular line (only one ATA > SIP account) it doesn't happen`
`17:42.44`	`*** join/#asterisk Noky (n=dnardell@200.68.89.17)`
`17:42.45`	`Noky`	`hi`
`17:42.54`	`*** join/#asterisk chexum (i=chexum@gateway/tor/x-569217a033a0503c)`
`17:43.36`	`Noky`	`i have a problem with my asterisk 1.0.9, i read that is a bug to FreeBSD, but i run a Linux Slackware 10.2...`
`17:43.39`	`Noky`	`Oct 19 14:49:10 WARNING[8249]: file.c:1084 ast_waitstream_full: Wait failed (Resource temporarily unavailable)`
`17:43.42`	`Noky`	`i have this error:`
`17:43.56`	`Noky`	`when i try to use "say digits"`
`17:44.26`	`Noky`	`from my AGI.. i patch the say.c for use ast_waitstream()`
`17:44.38`	`Noky`	`and now it's works ok..`
`17:44.50`	`Noky`	`but i don't know if i will have some problem with this change... any help or idea?`
`17:44.59`	`ambriento`	`seele_, pastebin the output of a entire call`
`17:45.36`	`[TK]D-Fender`	`toerkeium: What are "10 sip operators"?`
`17:46.19`	`intralanman`	`can anyone tell me how to do this "Set(SIP_CALLID=${SIP_HEADER(Call-ID)})" in an AGI?`
`17:46.23`	`[TK]D-Fender`	`Noky: Never had any problem with * on Slackware 10.2 and thats what my server is running now for the past year...`
`17:46.37`	`toerkeium`	`[TK]D-Fender: 10 sip accounts in use, 10 real persons speaking`
`17:46.39`	`*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)`
`17:47.09`	`[TK]D-Fender`	`toerkeium: any transcoding? Describe whats on both ends of the call.`
`17:47.35`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`17:47.47`	`teknoprep`	`[TK]D-Fender, well that problem fixed... had users in a "Ring Group" that were not actual extensions`
`17:47.57`	`teknoprep`	`[TK]D-Fender, i will be doing my next install asterisk fully.. no AMP`
`17:48.20`	`[TK]D-Fender`	`teknoprep: Wise decision.`
`17:48.22`	`toerkeium`	`[TK]D-Fender: it's all pure VOIP, and I have buy 12 g729 licenses since my ITSP uses g720, and the sip accounts (softphones) uses g711 for connecting to asterisk`
`17:48.29`	`teknoprep`	`[TK]D-Fender, i don't want to admin this thing tho... and they have pretty much idiots for anything dealing with CLI`
`17:48.48`	`teknoprep`	`[TK]D-Fender, so general adding of extensions and the such will be done through AMP... FreePBX since its easy for them to do`
`17:49.13`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`17:49.15`	`[TK]D-Fender`	`teknoprep: Write your own "mini-gui" then`
`17:49.19`	`teknoprep`	`[TK]D-Fender, if i am going to be paid to admin it.. i am going to install it using just asterisk.. which is how i am going to setup my home today... so i can learn... need to read the book and do as i read`
`17:49.37`	`teknoprep`	`[TK]D-Fender, i only know dynamic page design with vb.net or c#`
`17:49.38`	`Damin`	`Dammnit..`
`17:49.44`	`teknoprep`	`[TK]D-Fender, i have not yet learned cgi or PHP`
`17:50.00`	`Damin`	`I hate it when I have something that is so simple.. and I just brain-fart...`
`17:50.03`	`[TK]D-Fender`	`teknoprep: I'm sure you'll figure it out in no time. I did, as a complete 2-bit hack....`
`17:50.12`	`BlepsoaF`	`use ruby`
`17:50.29`	`seele_`	`ambriento,`
`17:50.33`	`L\|NUX`	`hello every one`
`17:50.37`	`seele_`	`ambriento, http://pastebin.ca/210333`
`17:50.40`	`BlepsoaF`	`MVC might be overkill, but ruby is really nice`
`17:50.45`	`teknoprep`	`[TK]D-Fender, i am sure i will also... php is where i am pushing myself next after i learn asterisk`
`17:50.57`	`L\|NUX`	`can some one tell me that i do have a box in korea which have 600 DID on it. I am forwarding all my dids to another * box but i could not able to see DNIS. why this happening`
`17:51.01`	`BlepsoaF`	`teknoprep: learn ruby`
`17:51.06`	`toerkeium`	`[TK]D-Fender: any tip on what should I try to find in asterisk/full ?`
`17:51.07`	`teknoprep`	`why ruby?`
`17:51.08`	`generalhan`	`Im having an issue with the stupid echo on my PRI lines ... i keep getting this in /var/log/messages "Asterisk kernel: zaptel Disabled echo canceller because of tone (rx) on channel 6" can some one point me in the right direction on how i may start to trouble shoot this ???`
`17:51.18`	`teknoprep`	`i have only ever heard of it.. never seen it implemented`
`17:51.28`	`[TK]D-Fender`	`toerkeium: Not sure what you mean by that....`
`17:51.30`	`Noky`	`[TK]D-Fender: i have this error`
`17:51.36`	`Noky`	`and it is a bug for the FreeBSD`
`17:51.47`	`BlepsoaF`	`teknoprep: ruby is OO, so its not procedural code - alot cleaner to write in and easier`
`17:52.00`	`teknoprep`	`BlepsoaF, is it for web pages?`
`17:52.20`	`BlepsoaF`	`teknoprep: yes, ruby has a web framework called rails`
`17:52.22`	`[TK]D-Fender`	`generalhan: Thats normally because a fax/modem tone is detected on the line and is being disabled because EC would kill the connection for sure`
`17:52.22`	`teknoprep`	`BlepsoaF, if so i'll check itout`
`17:52.25`	`*** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se)`
`17:52.34`	`generalhan`	`hmm`
`17:52.40`	`jgoo`	`[TK]D-Fender: good advice on AGI, but I realise, I am using fastAGI...`
`17:52.49`	`toerkeium`	`[TK]D-Fender: I am reading the asterisk log files, but I am not famiiar with this log entries.. maybe there is something more specific to find errors.. like in postfix a grep '(erro\|warning\|panic)' etc..`
`17:52.50`	`Noky`	`i can use the ast_waitstream() instead of ast_waitstream_full()`
`17:52.55`	`jgoo`	`I am hosting my app over TCP/IP on a beefy machine... I guess that is ok then right`
`17:52.58`	`Noky`	`and with this change in say.c ... works KO`
`17:53.01`	`generalhan`	`well almost everytime some on complains about their call being VERY echo-y i can check that log and see another on of those messages`
`17:53.08`	`*** join/#asterisk astguest (n=j@pool-71-98-77-207.ipslin.dsl-w.verizon.net)`
`17:53.17`	`[TK]D-Fender`	`jgoo: Indeed I don't know all the details, just pointing out that you should implement with care. the large-scal impact can be considerable in many cases.`
`17:53.47`	`astguest`	`Anyone in the room have any DTMF experience with a Cisco 7970?`
`17:54.15`	`L\|NUX`	`any idea ?`
`17:54.26`	`generalhan`	`the echo is starting to KILL me. i have played with the rx/tx numbers little by little for about a month now, and i still get the worst echo some times.`
`17:54.44`	`jgoo`	`ah well, I hope to become an asterisk-java guru around here, this will be production in a few weeks, just running on an Oracle 10g machine, looking up zips and addresses via phone, just 10 channels, mean lead 4-5 live channels, should be fine, I'll keep peeps posted, cos fastagi is great`
`17:55.59`	`seele_`	`ambriento, can't answer a call to the agent ???`
`17:58.40`	`dan__t`	`Anyone ever had any problems with Teliax?`
`17:58.49`	`ambriento`	`seele_, sorry. I was away a little bit`
`17:58.54`	`ambriento`	`I'm reading your stuff now`
`17:58.57`	`msa20`	`dan__t: ever? or always? :b`
`17:59.30`	`dan__t`	`Just looking for background info :)`
`18:04.03`	`*** join/#asterisk saftsack (n=oliver@p54A7F11C.dip.t-dialin.net)`
`18:04.33`	`saftsack`	`hi are tdm experts here?`
`18:05.09`	`teknoprep`	`ask the question`
`18:05.14`	`teknoprep`	`there are alot of ppl here that can hel you`
`18:05.42`	`dan__t`	`I'm hunting for a provider that can give me a rediculous amount of service.`
`18:05.42`	`dan__t`	`I've had good experiences with Teliax in testing and such, and I'm interested in learning more.`
`18:06.31`	`dan__t`	`Well, I see most providers have a structured plan, i.e. N minutes of long distance, X minutes of toll free service, etc etc. I would prefer to pool all these items together to accomidate for the service which Provide.`
`18:06.40`	`msa20`	`what does "rediculous amount of service" mean?`
`18:06.46`	`saftsack`	`ok. if i raise the handset and dial shortly after that i cant hear anything. i have to wait 0.2ms everytime. can i shorten this time?`
`18:06.58`	`dan__t`	`For what I'm planning, there's a good chance that quite a lot of the numbers I'll be providing will not even be used 80% of the time.`
`18:07.21`	`dan__t`	`msa20, I mean I need several hundred, if not thousands, of toll free numbers which will be sold off to clients of mine as part of their service plans.`
`18:07.44`	`*** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941)`
`18:08.05`	`dan__t`	`I've got a LOT of technical stuff to learn, but before I progress to that ,I want to make sure that my master plan will actually work ;)`
`18:09.52`	`astguest`	`I have a small DTMF question. If anyone has any experience with the problem I have, i'd appreciate your input! I have a Cisco 7970 using RFC2833 DTMF to my asterisk server. I'm using inband DTMF with my VoIP provider. Does asterisk automatically send forward the digits to my provider as inband when it detects me sending rcf2833 to asterisk?`
`18:10.30`	`ambriento`	`seele_, that's a lot of stuff, isn't?`
`18:10.59`	`astguest`	`I am unable to use IVR's outside of my asterisk because of this, and my provider doesn't support rfc2833, I would use inband on my 7970, but I can't get a value to work the in the XML file for the phone`
`18:11.03`	`ambriento`	`what happens when you got those JB messages?`
`18:12.16`	`msa20`	`dan__t: voicepulse will let you use their api to add numbers to your account on the fly...`
`18:12.43`	`dan__t`	`That would be rather interesting.`
`18:13.35`	`seele_`	`ambriento, is resolved now ... a typo error jejeje`
`18:13.38`	`dan__t`	`I want to be able to purchse minutes in bulk, as part of the entire account - not part of individual numbers.`
`18:14.05`	`ambriento`	`:)`
`18:14.07`	`seele_`	`ambriento, how can I block outbound calls to some extensions ??`
`18:15.07`	`ambriento`	`seele_, you can put them on a context that does not have patterns to an outboundcall, for instance`
`18:15.31`	`seele_`	`ambriento, hmmmm`
`18:16.22`	`ambriento`	`seele_, that's what a context was made for ;)`
`18:16.45`	`ambriento`	`seele_, donde eres?`
`18:17.17`	`seele_`	`ambriento, de Colombia`
`18:17.31`	`*** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr)`
`18:17.51`	`ambriento`	`seele_, que bueno. entonces no estamos tan lejos :)`
`18:18.03`	`seele_`	`ambriento, y vos ??`
`18:18.34`	`dan__t`	`Thanks for the tip, msa20`
`18:18.35`	`ambriento`	`seele_, yo soy de Brasil`
`18:20.08`	`seele_`	`ambriento, que bien !`
`18:21.18`	`L\|NUX`	`can some one tell me that i do have a box in korea which have 600 DID on it. I am forwarding all my dids to another * box but i could not able to see DNIS. why this happening`
`18:22.42`	`*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es)`
`18:22.44`	`kink0`	`hello`
`18:22.52`	`ambriento`	`hola`
`18:23.03`	`kink0`	`would work fine grandstream h.264 in Asterisk ?`
`18:23.25`	`ambriento`	`1.4beta only. I guess`
`18:24.11`	`ambriento`	`seele_, mira, lo que es "meter las patas" ?`
`18:24.18`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:28.58`	`tzanger`	`[TK]D-Fender: what do you do for speed dials on the polycoms?`
`18:29.15`	`tzanger`	`or rather how have you found to implement it nicely?`
`18:30.32`	`*** join/#asterisk CrashSys1 (n=kumba@office.crashsys.com)`
`18:30.39`	`*** join/#asterisk DjPepse (n=pepse@71-223-116-141.phnx.qwest.net)`
`18:31.02`	`*** join/#asterisk boch (n=root@201.216.241.97)`
`18:31.08`	`dan__t`	`What's the status on using IAX for softphones? I've not seen it having been outright suggested, but I'm curious.`
`18:31.45`	`DjPepse`	`i use iax for my softphone`
`18:31.46`	`*** join/#asterisk nowwhat (n=ARF@24-148-32-14.nwb-bsr1.chi-nwb.il.cable.rcn.com)`
`18:32.03`	`DjPepse`	`seems to work well`
`18:33.17`	`boch`	`anyone familiar with phpagi ?`
`18:33.29`	`dan__t`	`hm`
`18:33.40`	`nowwhat`	`anyone familiar with digium hardware can answer a question about wcfxo?`
`18:34.03`	`*** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz)`
`18:34.07`	`DjPepse`	`can someone tell me if there's a built-in * command to pick up a call on another extension?`
`18:34.11`	`DjPepse`	`I seem to remember doing it before`
`18:34.46`	`qdk`	`Oct 19 20:28:36 WARNING[4360]: channel.c:2328 set_format: Unable to find a codec translation path from alaw to unknown <- does this mean that the channel (SIP) does not support or disallow alaw?`
`18:35.28`	`BlepsoaF`	`boch: just ask what you're trying to do`
`18:35.31`	`[TK]D-Fender`	`tzanger: not sure what you mean....`
`18:35.55`	`*** join/#asterisk toxap (n=toxap@213.227.193.75)`
`18:36.10`	`DjPepse`	`dan__t: I use idefisk`
`18:36.32`	`tzanger`	`[TK]D-Fender: speed dials... implementing a speed dial button`
`18:37.04`	`boch`	`BlepsoaF: i do an exec_dial(), then fetch the DIALSTATUS variable with get_variable(). but when i do again another exec_dial() in the same AGI i cannot fetch DIALSTATUS var with get_variable() neither exec("GET VARIABLE..`
`18:37.05`	`[TK]D-Fender`	`tzanger: Pick a free line key, add a contact, set its Index. The End`
`18:37.28`	`tzanger`	`hmm`
`18:37.28`	`[TK]D-Fender`	`tzanger: Set "buddy watch" if I feel like keeping track of them as well.`
`18:37.52`	`boch`	`BlepsoaF: get_variable() is returning ['result'] == -1, and the phpagi-homepage says nothing about this -1`
`18:37.54`	`kink0`	`well, appear h.264 does not works, at least you used the develoment version... but what is your opinions about h.263 support ?`
`18:38.06`	`tzanger`	`I'll have to play with it... no I don't mean a one-touch speed dial... but 2-3 keypress procedure... <speeddial>44 to dial index #44 type of thing.`
`18:38.08`	`*** join/#asterisk quellhorst (n=pro@unaffiliated/rend)`
`18:38.45`	`[TK]D-Fender`	`tzanger: At the office I use 3 line-keys @ 1 call/per for normal calls. of the 3 remaining, 2 are external dials, and 1 is a remote SPA attached to a branch's PBX as a VoIP tie-line. I have presence on that so I know when the "free-line" is busy.`
`18:38.50`	`kink0`	`anyone is ussing h263 with Asterisk 1.2.12.1 ?`
`18:38.56`	`quellhorst`	`anyone find a high quality sip provider?`
`18:39.07`	`[TK]D-Fender`	`tzanger: You mean like multi-touch, not one-touch via soft-key?`
`18:39.34`	`tzanger`	`yeah`
`18:39.38`	`kink0`	`quellhorst, all is depending how much you pretend pay, and also where you pretend to get termination`
`18:40.00`	`tzanger`	`I think I can do it by assigning a button to dial *0 or something and then using the dialplan to complete the match`
`18:40.28`	`*** join/#asterisk josherz01 (n=r00t@h-68-165-181-130.lsanca54.covad.net)`
`18:41.27`	`DjPepse`	`Do I have to make my own Pickup() rule in extensions.conf or is there something built in?`
`18:42.18`	`BlepsoaF`	`boch: hmm thought I would be able to help, I use perl AGI`
`18:42.43`	`BlepsoaF`	`but -1 is obviously a failure, or something is not getting set`
`18:43.07`	`boch`	`thats what i thought`
`18:44.13`	`boch`	`is it possible Dial() is not setting DIALSTATUS ?`
`18:45.18`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`18:47.21`	`kink0`	`nobody has a packet8 video phone connected thought ASterisk ?`
`18:47.53`	`[TK]D-Fender`	`tzanger: Add the index #, and then just press the "up" arrow. so [up] 44`
`18:47.55`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`18:48.23`	`*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
`18:48.35`	`tzanger`	`...`
`18:49.01`	`tzanger`	`interesting idea`
`18:49.13`	`tzanger`	`44[up] puts the cursor beofre the 44, then hit the speed dial button... interesting`
`18:49.55`	`boch`	`BlepsoaF: http://pastebin.ca/210410 - maybe you could tellme what im doing wrong`
`18:50.57`	`msa20`	`quellhorst: what're your requirements?`
`18:51.01`	`*** join/#asterisk Cephallus (n=Cephallu@206.40.216.150)`
`18:51.32`	`DjPepse`	`Hm, I guess Pickup() wasn't what i was looking for`
`18:51.51`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.80.124)`
`18:52.03`	`DjPepse`	`that's just for picking up a ringing extension. How do I snag a call from another extension?`
`18:52.19`	`DjPepse`	`aka, the phone doesn't have a transfer option`
`18:53.39`	`Cephallus`	`Hi all, I have a stumper relating to transferring an incoming call from a queue to voicemail. If I transfer to the extension, the call simply drops after timeout. If I transfer to voicemail, it drops immediately. I've seen a few mentions online, but no definitive answer...is this a common problem that everyone knows about?`
`18:54.08`	`*** join/#asterisk Cresl1n (i=matt@nat/digium/x-346f88a829f953bc)`
`18:54.08`	`*** mode/#asterisk [+o Cresl1n] by ChanServ`
`18:54.46`	`Cephallus`	`(Asterisk 1.2.9.1 on FC5, FreePBX front-end)`
`18:55.17`	`[TK]D-Fender`	`Cephallus: Please read the channel topic.`
`18:58.31`	`hmmhesays`	`dee dee dee`
`18:59.26`	`DarKnesS_WolF`	`<PROTECTED>`
`19:00.56`	`quellhorst`	`msa20: i have a server with a 100 mbit connect that i want to run asterisk on. i dont have any phone lines going to it... so i would want a high quality sip service. a plus would be to have a Houston TX phone # for inbound, and then free/low long distance outbound..`
`19:02.28`	`ambriento`	`Cephallus, /j #freepbx`
`19:05.12`	`Cephallus`	`tx`
`19:05.24`	`msa20`	`quellhorst: my personal recommendation is connect.voicepulse.com even if you use it just for your outbound... the quality and price can't be beat, they charge $11/month for a phone number, so you may want to get that elsewhere if that's a problem for you`
`19:06.55`	`quellhorst`	`msa20: any recommendations for the phone number in houston?`
`19:07.24`	`msa20`	`i use voicepulse for my incoming too, so i wouldn't know... someone else probably knows super-cheap phone numbers`
`19:07.57`	`*** join/#asterisk Fnc-1 (n=Fnc-1@63.99.187.98)`
`19:08.11`	`Fnc-1`	`afternoon all`
`19:08.44`	`Fnc-1`	`does anyone know how to unlock or defualt pw on verizon voicewing pap2`
`19:11.31`	`*** join/#asterisk zigman (i=zigman@irc.zigman.de)`
`19:13.07`	`*** join/#asterisk mountainm2k (n=mountain@165.236.183.1)`
`19:13.36`	`*** join/#asterisk Assid (i=assid@203.115.83.215)`
`19:13.40`	`mountainm2k`	`Oct 19 13:01:39 NOTICE[31086]: chan_zap.c:8207 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1`
`19:13.41`	`mountainm2k`	`Oct 19 13:01:39 WARNING[31087]: chan_zap.c:6347 handle_init_event: Detected alarm on channel 2: Red Alarm`
`19:13.45`	`mountainm2k`	`Meaning my span is down?`
`19:13.52`	`mountainm2k`	`or could it be a problem on my end?`
`19:16.01`	`kink0`	`any h263 phone that works with Asterisk ?`
`19:16.32`	`kink0`	`I had checked some from packet8.com but they told me are locked and that will not work without her call plans.`
`19:20.54`	`hmmhesays`	`my girls in the next room, sometimes I wish she was you`
`19:21.13`	`quellhorst`	`whats a good ethernet phone for astersk?`
`19:21.54`	`[TK]D-Fender`	`quellhorst: Any Polycom or Aastra`
`19:22.19`	`quellhorst`	`how much $ do they run?`
`19:22.26`	`ccesario`	`hi, somebody can indicate any goood book!?!`
`19:22.43`	`hi365`	`Grandstream`
`19:23.14`	`[TK]D-Fender`	`ccesario: ...`
`19:23.15`	`[TK]D-Fender`	`~book`
`19:23.17`	`jbot`	`[book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`19:23.31`	`Fnc-1`	`anyone know how to unlock a verizon pap2`
`19:23.42`	`[TK]D-Fender`	`quellhorst: Anywhere from around $95 USD through $350 USD depending on the model.`
`19:24.05`	`[TK]D-Fender`	`quellhorst: Describe your needs`
`19:24.49`	`ccesario`	`[TK]D-Fender, this book have concepts and praticals material!?!`
`19:25.08`	`quellhorst`	`looking for a phone where i can have speakerphone, callerid, ethernet, and handle maybe 2 calls at once.. so if i need to put one on hold to answer the other call.`
`19:25.18`	`[TK]D-Fender`	`ccesario: Yes`
`19:25.53`	`ccesario`	`[TK]D-Fender, thanks :)`
`19:26.04`	`*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)`
`19:26.10`	`PakiPenguin_`	`hello everyone`
`19:26.36`	`PakiPenguin_`	`i get the following [Oct 19 15:13:38] == Connect attempt from '127.0.0.1' unable to authenticate`
`19:26.36`	`PakiPenguin_`	`<PROTECTED>`
`19:26.49`	`[TK]D-Fender`	`quellhorst: I would then suggest the Polycom IP 501 ($170 USD) You could consider the Aastra 9133 at around $135, but its nowhere near as nice to use as the IP 501.`
`19:27.39`	`Fnc-1`	`unlock pap2?`
`19:29.24`	`quellhorst`	`[TK]D-Fender: ok, i may get the 501 then`
`19:29.56`	`*** join/#asterisk beehive (n=michael@pool-72-66-14-128.washdc.fios.verizon.net)`
`19:30.12`	`quellhorst`	`[TK]D-Fender: does it have a mini headset jack?`
`19:32.08`	`[TK]D-Fender`	`quellhorst: No, neither of these use 2.5mm. They only have the RJ9 style like used in call-centers. For a 2.5mm model you could consider the Linksys SPA-941 is this is a "must" for you, but the phone is inferior in virtually every other way to the IP 501`
`19:32.34`	`quellhorst`	`[TK]D-Fender: so maybe it would be better if i just got a new headset?`
`19:32.49`	`quellhorst`	`i want the best thats reasonable.`
`19:33.47`	`quellhorst`	`then again i dont use a headset much anymore.`
`19:35.00`	`*** join/#asterisk backblue (n=moo@87-196-96-248.net.novis.pt)`
`19:35.48`	`syzygyBSD`	`to be honest I prefer the linksys over the 501`
`19:36.17`	`quellhorst`	`syzygyBSD: why?`
`19:36.31`	`syzygyBSD`	`configuration is easier`
`19:36.51`	`syzygyBSD`	`the 501 tries to do too much for me`
`19:37.03`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`19:37.04`	`*** mode/#asterisk [+o denon] by ChanServ`
`19:37.43`	`syzygyBSD`	`I do have a 501 on my desk, and a sipura 841 next to it, both ring at the same time... I answer the 841`
`19:37.50`	`[TK]D-Fender`	`syzygyBSD: You'll get no argument from me on that, but ones its up an running I you're left with the fact that Polycom's call handling capabilities are considerably superior as is physical phone interface`
`19:37.54`	`syzygyBSD`	`my boss has the linksys though`
`19:38.01`	`quellhorst`	`im a gadget freak.. i print less than 100 pages per month, yet have a 31ppm color laser.. also went overkill on my 3 monitor display, including 1x30"`
`19:38.05`	`[TK]D-Fender`	`syzygyBSD: You use an 841? ICK`
`19:38.18`	`*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`19:38.34`	`syzygyBSD`	`[TK]D-Fender: I only ever get 1 call, and dont' ever need to transfer, put on hold...`
`19:38.43`	`dan__t`	`Anyone know of a softphone extension for Firefox?`
`19:39.03`	`syzygyBSD`	`dan__t: I have heard of them.. but I have to ask... why?`
`19:39.30`	`dan__t`	`It's all part of the master plan. I think I'd like to use something like that. My devs have started to dig around on the subject, but I'd rather not re-invent the wheel.`
`19:40.21`	`syzygyBSD`	`http://blog.tmcnet.com/blog/tom-keating/voip/firefox-voip-extension.asp`
`19:40.44`	`syzygyBSD`	`a google search for "sip firefox" will turn up a lot`
`19:41.21`	`syzygyBSD`	`well, I best go wire the rest of this 50 pair`
`19:41.23`	`syzygyBSD`	`bbl`
`19:41.26`	`dan__t`	`Excellent, thanks.`
`19:41.41`	`dan__t`	`I'd like to use AIX eventually. I'll throw this to my guys and see what they can do with it heheh`
`19:42.34`	`hmmhesays`	`aix?`
`19:42.46`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
`19:42.48`	`dan__t`	`erm, http://moziax.mozdev.org/`
`19:42.53`	`dan__t`	`iax, rather, sorry.`
`19:43.19`	`[TK]D-Fender`	`Any ops around?`
`19:44.17`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`19:45.42`	`hmmhesays`	`me`
`19:45.45`	`hmmhesays`	`arrrrrrrgh`
`19:45.47`	`hmmhesays`	`j/k`
`19:46.09`	`xheliox`	`http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.0-beta2.tar.gz - doesn't make the post sha1sum. Is someone aware of this?`
`19:46.14`	`xheliox`	`Someone who can do something about it? :(`
`19:46.18`	`xheliox`	`erm :)`
`19:46.50`	`xheliox`	`*doesn't match the posted sha1sum. I'm just gonna stop typing.`
`19:47.10`	`teknoprep`	`yeah i am not getting a call back when i park a call`
`19:47.12`	`teknoprep`	`exten => _*3, 1, ParkAndAnnounce(pbx-transfer:PARKED\|120\|SIP/${DIALEDPEERNUMBER}\|sip_incoming,${DIALEDPEERNUMBER},1)`
`19:47.16`	`teknoprep`	`using that to park it`
`19:47.58`	`hmmhesays`	`its really good to hear your voice, saying my name it sounds so sweet`
`19:50.18`	`CrashSys1`	`Can you hear me now?`
`19:50.26`	`tzanger`	`wtf... home* registers to office*. OK.`
`19:50.31`	`tzanger`	`home* registers to office*. 401`
`19:50.34`	`tzanger`	`home* registers to office*. OK`
`19:50.37`	`tzanger`	`??`
`19:51.58`	`tzanger`	`home*'s sending the exact same info three times, and the middle one fails`
`19:53.32`	`tzanger`	`heh... it looks like older svn trunk doesn't like sip MWI events... I get back 489 from home* whenever office* sends it :-)`
`19:55.04`	`*** part/#asterisk mountainm2k (n=mountain@165.236.183.1)`
`19:55.27`	`BlepsoaF`	`well I found a fail over solution for asterisk involving PRI T1...http://dataprobe.com/products/switch/ab_net/index.html using RJ-45 to DB25 converters`
`19:56.02`	`BlepsoaF`	`only about 600 bucks`
`20:02.57`	`BlepsoaF`	`CrashSys1:?`
`20:04.11`	`De_Mon`	`how do I get the user that has an active channel?`
`20:04.30`	`De_Mon`	`the SIP user`
`20:04.31`	`hmmhesays`	`show channnels`
`20:04.38`	`hmmhesays`	`with one less n`
`20:04.47`	`tzanger`	`haha`
`20:05.50`	`De_Mon`	`in the dialplan`
`20:06.00`	`De_Mon`	`it'd be ${SIP(user)} or something...`
`20:06.25`	`De_Mon`	`show channels shows the channel which is user-number.. I just want the user part`
`20:06.34`	`BlepsoaF`	`show hints?`
`20:06.51`	`BlepsoaF`	`although, you would need to setup hints in your dial plan`
`20:07.06`	`De_Mon`	`the hints are there, but again I need to access this from the dialplan`
`20:07.46`	`De_Mon`	`I'm trying to record who made the call in the accountcode, so I was going to dump the user who made the call`
`20:08.02`	`De_Mon`	`s!dump!log!`
`20:08.22`	`*** join/#asterisk lorinc (n=ang@caracas-0422.adsl.interware.hu)`
`20:09.15`	`beehive`	`Hey folks. When setting up a box to host multiple companies how does one keep the dialplans separate so that each company has its own extensions and the like?`
`20:10.07`	`De_Mon`	`beehive separate contexts for each company`
`20:11.04`	`beehive`	`thanks de... how does asterisk choose which dialplan to use?`
`20:11.08`	`*** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net)`
`20:11.15`	`beehive`	`ah... contexts... i see`
`20:11.21`	`De_Mon`	`http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels`
`20:11.35`	`De_Mon`	`beehive each account or number as an incoming/default context`
`20:12.30`	`De_Mon`	`each company will typically have 2 contexts an internal(calls from employee phones) and an inbound (calls from outside)`
`20:12.55`	`*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)`
`20:13.04`	`De_Mon`	`and they branch out from there...`
`20:13.28`	`beehive`	`so the phone 'knows' what context to use.`
`20:13.52`	`beehive`	`this is one box that hosts multiple companies.`
`20:14.33`	`De_Mon`	`the phone's account or phone number does, yeah.`
`20:14.53`	`*** join/#asterisk linagee (n=linagee@unaffiliated/linagee)`
`20:15.04`	`DasTech`	`but that is asterisk realtime`
`20:15.12`	`beehive`	`I see.`
`20:15.19`	`DasTech`	`and freepbx does not support realtime at the min`
`20:15.24`	`DasTech`	`it is in the works`
`20:15.43`	`DasTech`	`and realtime is all database driven`
`20:15.50`	`beehive`	`So if there were 7 companies each with 10 phones it would have a very long dialplan.`
`20:16.05`	`beehive`	`I see.`
`20:16.12`	`DasTech`	`so its a matter of setting up the db and the needed tables in the db`
`20:16.21`	`beehive`	`right`
`20:16.38`	`DasTech`	`but I have to read and learn realtime`
`20:16.40`	`hmmhesays`	`dialplan wouldn't be that long`
`20:16.56`	`hmmhesays`	`also, realtime wouldn't change the length of your dialplan`
`20:17.07`	`*** part/#asterisk chexum (i=chexum@gateway/tor/x-569217a033a0503c)`
`20:17.09`	`beehive`	`But as DasTech says it would be in a database.`
`20:17.13`	`hmmhesays`	`each line in the text file ia record in teh database`
`20:17.29`	`hmmhesays`	`holy shiat my typing is bad today`
`20:17.35`	`hmmhesays`	`dee dee dee`
`20:17.44`	`beehive`	`so just update the DB and the dialplan is updated. Does that work with multiple IVRs as well?`
`20:17.54`	`DasTech`	`the main base dial plan is in the conf files`
`20:18.09`	`hmmhesays`	`um.. .invalid question, next`
`20:18.31`	`De_Mon`	`the config that tells asterisk EVERYTHING is in the database is a config... yeah`
`20:18.54`	`hmmhesays`	`multiple ivr's would be written into your dialplan whether it is in a database or in the config file`
`20:18.56`	`De_Mon`	`I like the static dialplan myself, eash to backup, don't have to worry about a database server`
`20:19.17`	`hmmhesays`	`less overhead`
`20:19.23`	`De_Mon`	`I don't have customers wanting to update their dialplan every 5seconds either though :)`
`20:19.26`	`hmmhesays`	`although I use mysql in my static dialplan`
`20:19.35`	`hmmhesays`	`cmd mysql, and it works dandy`
`20:19.40`	`beehive`	`ok... I'm obviously in over my head... I'll just lurk for a while and educate myself. Thanks De_Mon and DasTech.`
`20:19.57`	`hmmhesays`	`what exactly are you trying to accomplish?`
`20:20.24`	`De_Mon`	`realtime dialplans is just one extra level of complexity. start simple...`
`20:20.53`	`hmmhesays`	`once you have your dp in a static config file, it is cake to add it to a database`
`20:21.42`	`*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)`
`20:22.58`	`teknoprep`	`what gives me the ability to trasfer calls by pressing # ?`
`20:23.09`	`teknoprep`	`this ability has gone away lol.. i upgraded and it no longer works`
`20:23.19`	`CrashSys1`	`analog call transfer?`
`20:23.35`	`teknoprep`	`?`
`20:23.42`	`hmmhesays`	`proper flag in the dial command, and features.conf set right`
`20:23.54`	`*** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com)`
`20:24.23`	`teknoprep`	`should it be`
`20:24.33`	`teknoprep`	`blindxfer => ## or blindxfer => #`
`20:24.40`	`teknoprep`	`if i only want to press it once`
`20:24.44`	`hmmhesays`	`#`
`20:24.49`	`teknoprep`	`tought so just making sure`
`20:26.35`	`[TK]D-Fender`	`teknoprep: DTMF transfers etc = ASS`
`20:26.44`	`[TK]D-Fender`	`teknoprep: What kind of phones are you working with?`
`20:26.47`	`hmmhesays`	`hot ass or skanky ass?`
`20:26.59`	`[TK]D-Fender`	`hmmhesays: "Total ho"`
`20:27.17`	`hmmhesays`	`so we got the subs hooked up last night, 2 18 in ev's`
`20:27.23`	`hmmhesays`	`powered by a behringer ep2500`
`20:27.32`	`hmmhesays`	`my guitar never had so much low end`
`20:30.12`	`[TK]D-Fender`	`Ok, checkout time in Insanityville... back in a bit...`
`20:30.17`	`CrashSys1`	`Behringer V-Amp's sound pretty damn good for what they are... surprisingly so...`
`20:30.23`	`teknoprep`	`[TK]D-Fender, gxp-2000 & spa-942`
`20:30.37`	`teknoprep`	`damn`
`20:30.40`	`teknoprep`	`he left`
`20:31.05`	`CrashSys1`	`Grandstream = Immediately upgrade to 1.1.1.14 bios or suffer severe death!`
`20:31.38`	`teknoprep`	`already did`
`20:31.40`	`CrashSys1`	`well, suffer a horrible death anyways...`
`20:31.45`	`*** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net)`
`20:31.53`	`*** join/#asterisk xnon (n=xnon@200.8.30.161)`
`20:31.59`	`teknoprep`	`CrashSys1, this used to work before i upgraded asterisk to 1.2.12.1`
`20:32.18`	`CrashSys1`	`Why do you want to use analog call transfer with a voip phone?`
`20:32.24`	`teknoprep`	`CrashSys1, my real problem is that when i hit trasfer 700 <--- parking ext... it doesn't call me back`
`20:32.57`	`teknoprep`	`it parks the call`
`20:33.11`	`teknoprep`	`but doesn't call me back and tell me the ext its on`
`20:33.42`	`CrashSys1`	`Have you scoured voip-info.org for code snippets of different ways to accomplish the callback?`
`20:34.03`	`CrashSys1`	`Plus, it's opensource... It could have just been broken... :)`
`20:35.08`	`hmmhesays`	`there is many ways to do callback`
`20:37.16`	`*** join/#asterisk Amilcar_ (n=xxxxx@201.34.202.17)`
`20:37.24`	`teknoprep`	`exten => _*3, 1, ParkAndAnnounce(pbx-transfer:PARKED\|120\|SIP/${DIALEDPEERNUMBER}\|sip_incoming,${DIALEDPEERNUMBER},1)`
`20:37.29`	`teknoprep`	`CrashSys1, thats the way i doit`
`20:37.48`	`teknoprep`	`it used to work till i upgraded to 1.2.12.1`
`20:38.21`	`CrashSys1`	`But if it's not working now, then you need to try something else... or downgrade...`
`20:38.51`	`hmmhesays`	`why did you upgrade?`
`20:39.38`	`teknoprep`	`i was getting a problem with VERY long delays... everyone told me to so i did`
`20:39.50`	`teknoprep`	`it actually fixed my ring groups delay's instantly`
`20:39.54`	`teknoprep`	`but... now i have this as a problem`
`20:40.27`	`*** join/#asterisk fiber0pti (n=John@207.114.199.107)`
`20:47.34`	`CrashSys1`	`tek: Are you implicity loading or not loading modules?`
`20:49.26`	`*** join/#asterisk riddlebox (n=blah@24-207-167-28.dhcp.stls.mo.charter.com)`
`20:50.47`	`*** join/#asterisk scurb (n=scurb@c-b7ade355.14-16-64736c13.cust.bredbandsbolaget.se)`
`20:50.48`	`*** join/#asterisk montoya (n=so@201.39.235.130)`
`20:51.10`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`20:51.13`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`20:51.15`	`scurb`	`Anyone has some idea of why my polycome 501-phone does not get any incoming sound?`
`20:51.45`	`ManxPower`	`scurb: reseat the handset cord until you hear the SECOND click`
`20:52.00`	`scurb`	`ManxPower.. I get sound when I call internal extensions..`
`20:52.08`	`scurb`	`but when I receive or place outgoing calls i don't :(`
`20:52.31`	`FuriousGeorge`	`im looking into ser integration. does anyone know if asterisk sends out a NOTIFY with event: /parked/ when i park someone? i assume 1.2 doesnt (thats what i use) but i hear parking is changes in 1.4 beta`
`20:52.40`	`FuriousGeorge`	`*is changed`
`20:55.05`	`FuriousGeorge`	`perhaps "has changed" would have been best there... in any case, i gotta run... anyone know?`
`20:55.30`	`*** join/#asterisk saftsack (n=oliver@p54A7F11C.dip.t-dialin.net)`
`20:56.24`	`scurb`	`humm.. some further investigations.. It's only incoming/placed calls on my SIP-trunk that transfers no sound.. calls over my tdm400-card are correct`
`20:57.28`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`20:58.57`	`saftsack`	`hi whats the price of an polycom ip 300 telephone?`
`21:00.48`	`quellhorst`	`saftsack: check froogle.com`
`21:01.25`	`[TK]D-Fender`	`saftsack :$115 USD for the IP301. Forget the IP 300 its the discontinued version.`
`21:02.18`	`ManxPower`	`scurb: sounds like a reinvite problem then`
`21:04.40`	`saftsack`	`ok do you know any dealers in europe?`
`21:04.55`	`scurb`	`ManxPower: yes, thought so too.. doesn't seem to help with canreinvite=no though.. well.. i'll try around with sip.conf ;)`
`21:05.02`	`saftsack`	`does the 301 has the same good qualitiy as the more expensive polycoms?`
`21:05.19`	`[TK]D-Fender`	`saftsack : yup`
`21:05.33`	`[TK]D-Fender`	`saftsack : What kind of use do you have in mond for it?`
`21:06.09`	`saftsack`	`i want to find a telephone that sounds better than an isdn telephone`
`21:06.42`	`[TK]D-Fender`	`saftsack : What kind of call volume? Don't need speakerphone or native PoE I take it?`
`21:07.03`	`saftsack`	`what is speakerphone?`
`21:07.17`	`saftsack`	`and why depends it on the call volume?`
`21:08.21`	`[TK]D-Fender`	`saftsack : If you don't know what a speakerphone is you've got some serious issues.... And if you take a lot of calls its worth investing in a phone with a better interfaces. PoE factors in as well for total cost-effectiveness`
`21:08.57`	`*** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net)`
`21:10.41`	`riddlebox`	`has anyone run asterisk on openwrt?`
`21:10.42`	`*** join/#asterisk MrTelephone (n=DeaLER25@Toronto-HSE-ppp3697167.sympatico.ca)`
`21:10.42`	`saftsack`	`maybe my english is to bad to know whats a speakerphone ^^ (it is something where i can speak to without having the handset on my ear? :> ) and what do you mean with interfaces? you mean a better configuration?`
`21:10.42`	`MrTelephone`	`is anyone here trying to get MGCP/NCS working with asterisk?`
`21:10.42`	`MrTelephone`	`Is there a software media converter from mgcp/ncs to sip?`
`21:14.47`	`[TK]D-Fender`	`saftsack : yes. Hearing the call on a speaker and having a MIC on the phone so you can talk handsfree (not suing a headset or handset)`
`21:15.01`	`BlepsoaF`	`does anyone see any issues with using a RJ45 -> db25 connecting that to a switch and going back to RJ5 then to a digium card`
`21:15.34`	`saftsack`	`this has my 80eu grandstream phone ^^ why doesnt a polycom doesnt have it? ^^`
`21:15.59`	`[TK]D-Fender`	`BlepsoaF : Pin-out compatability comes to mind....`
`21:16.00`	`[TK]D-Fender`	`BlepsoaF : plus signal integrity`
`21:16.31`	`BlepsoaF`	`[TK]D-Fender: I think the signal integrity will be OK because its just wiring`
`21:16.47`	`BlepsoaF`	`just trying to think of anythign before I buy this switch`
`21:16.51`	`[TK]D-Fender`	`BlepsoaF : next question... WHY?`
`21:16.53`	`BlepsoaF`	`the sales guy said it should work`
`21:16.59`	`BlepsoaF`	`let me get link`
`21:17.10`	`BlepsoaF`	`http://www.dataprobe.com/products/switch/ab_net/index.html`
`21:17.56`	`[TK]D-Fender`	`iP-AB-RJ8-L`
`21:18.10`	`[TK]D-Fender`	`that one seems to do RJ45 by itself. wh screw around with up/down conversion?`
`21:18.45`	`BlepsoaF`	`[TK]D-Fender: it only supports 1`
`21:18.50`	`BlepsoaF`	`DB25 model supports 4`
`21:19.18`	`BlepsoaF`	`tech guy at dataprobe recommended it`
`21:19.42`	`MrTelephone`	`who is developing asterisk and how do you pay them to create a module for you?`
`21:19.53`	`ManxPower`	`MrTelephone: Digium`
`21:20.14`	`[TK]D-Fender`	`BlepsoaF : glwaot`
`21:20.25`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`21:24.32`	`*** part/#asterisk xnon (n=xnon@200.8.30.161)`
`21:25.10`	`BlepsoaF`	`[TK]D-Fender: thanks :P`
`21:25.54`	`*** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il)`
`21:29.27`	`*** join/#asterisk snowy_owl (i=0@200.218.196.2)`
`21:31.14`	`snowy_owl`	`Hi friends. I would like to know if the codec 729 (256 (1 << 8) (0x100) audio g729 (G.729A)) supported by my asterisk is compliant with 729r8`
`21:31.43`	`*** join/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co)`
`21:31.55`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`21:33.05`	`snowy_owl`	`I'm having problems with a carrier and they told me about a incompability between them`
`21:33.25`	`Dovid`	`what did they say exactly`
`21:33.27`	`Dovid`	`?`
`21:34.16`	`snowy_owl`	`that the call cant be established between this codecs. They use a cisco router there, working with this 729r8`
`21:34.47`	`JoseBravo`	`Anoye can tell me where can I but for example $50 at minutes using SIP?`
`21:35.17`	`snowy_owl`	`the call is established, but I cant listen anything with my voip neither the PSTN side`
`21:36.35`	`snowy_owl`	`My asterisk dont receive any RTP packet from this Cisco router`
`21:37.04`	`snowy_owl`	`so, I just want to know if I need to buy a 729r8 license`
`21:37.20`	`MrTelephone`	`How long would it take someone to make chan_ncs using chan_mgcp and urthos ncs patch?`
`21:37.30`	`MrTelephone`	`Are there any coders in here`
`21:37.35`	`Dovid`	`snowy_owl: in order to use g729 u need to by a liscence`
`21:37.54`	`Dovid`	`JoseBravo: u wana get prepaid outbound sip ?`
`21:38.03`	`ManxPower`	`MrTelephone: This is not a developers channel`
`21:38.26`	`snowy_owl`	`yes, I know.`
`21:38.30`	`JoseBravo`	`Dovid yes`
`21:38.36`	`MrTelephone`	`where is the developers channel`
`21:38.56`	`Dovid`	`JoseBravo: voipjet.com is cheapest however thier customer service sux big time`
`21:39.05`	`Dovid`	`telix is a lil more but thier cust. service is great`
`21:39.21`	`Dovid`	`MrTelephone - doa channel search`
`21:39.29`	`Dovid`	`I believe it is #asterisk-devel`
`21:39.47`	`msa20`	`#asterisk-dev`
`21:40.37`	`*** join/#asterisk japerry (n=falc0n@216.231.51.209)`
`21:41.12`	`ManxPower`	`MrTelephone: There are not a lot of people interested in doing Asterisk developement for pay.`
`21:41.21`	`ManxPower`	`But if you pay enough, who knows.`
`21:41.35`	`*** join/#asterisk dlynes_laptop (n=dlynes@s64-180-109-134.bc.hsia.telus.net)`
`21:41.49`	`DasTech`	`what get payed for deving asterisk I did not know you could`
`21:41.51`	`DasTech`	`lol`
`21:41.56`	`Dovid`	`lol`
`21:42.08`	`Dovid`	`i went on to the dev list and got some one in 2 minutes`
`21:42.11`	`DasTech`	`I thought you did it for free to make a name for yourself`
`21:42.37`	`Dovid`	`lol`
`21:42.48`	`Dovid`	`i paid him $35.00 and had him add it to the tree`
`21:43.08`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`21:43.12`	`ManxPower`	`Dovid: writing a new channel driver is not a 2 min job.`
`21:43.22`	`Qwell[]`	`ManxPower: chan_null is!`
`21:43.24`	`ManxPower`	`SMALL things are pretty easy to get done.`
`21:43.40`	`Dovid`	`no i found some one to do the work in 2 min. meaning i got a fast reply ?`
`21:44.21`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com)`
`21:45.20`	`MrTelephone`	`anyways I'm willing to pay a couple thousand anyways`
`21:45.38`	`MrTelephone`	`and all they have to do is add a few lines of c code to the already working patch`
`21:45.40`	`Dovid`	`MrTelephone: email the devel list`
`21:46.35`	`Dovid`	`MrTelephone: email here Http://lists.digium.com`
`21:46.40`	`Dovid`	`and click on asterisk developers`
`21:46.58`	`X-Rob`	`Qwell, sorry I missed the discussion about VERSION() 8-)`
`21:48.36`	`mboehn`	`This is a bit offtopic, but it's worth a shot: Anyone worked with Tellabs DXX-equipment?`
`21:52.17`	`*** join/#asterisk justinu\|laptop (n=Justin@12.44.122.130)`
`21:56.26`	`*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`21:59.40`	`*** part/#asterisk MrTelephone (n=DeaLER25@Toronto-HSE-ppp3697167.sympatico.ca)`
`22:04.37`	`*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)`
`22:07.21`	`*** join/#asterisk cavediver (i=jonas@trimix.eklof.eu)`
`22:07.22`	`cavediver`	`Hello.`
`22:07.33`	`cavediver`	`Can I run asterisk on a sun netra1 ?`
`22:07.52`	`cavediver`	`is there a sparc-distribution ready with asterisk and perhaps some webgui ?`
`22:18.33`	`*** join/#asterisk Dude34 (n=Aces1UP@209.101.89.82)`
`22:18.40`	`Dude34`	`is anyone here in the calling card business?`
`22:20.02`	`*** join/#asterisk ucfMethod (i=ucfMetho@c-69-143-112-70.hsd1.va.comcast.net)`
`22:20.06`	`ucfMethod`	`hello`
`22:20.40`	`robin_sz`	`goodevening`
`22:22.09`	`robin_sz`	`these 190's seem OK, cheap enough`
`22:22.39`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`22:26.17`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`22:27.18`	`ucfMethod`	`is anyone here an extensions.conf master?`
`22:27.47`	`robin_sz`	`why not just ask your question`
`22:27.53`	`robin_sz`	`if we can answer it, we will`
`22:28.18`	`Amilcar_`	`snom 190 is a VERYGOOD phone.`
`22:28.27`	`Amilcar_`	`I really like it, very stable.`
`22:28.32`	`napkin`	`anyone here use les.net? i need a bit of help configuring a soft phone :)`
`22:28.50`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.168.78)`
`22:29.55`	`ucfMethod`	`I am having a bit a trouble setting up the "invalid extension" path in extensions.conf... right now if i dial an extension that doesn't exist, it just hangs up.... I thought it was the "i" as in "exten => i,1,Playback(pbx-invalid)"`
`22:29.57`	`X-Rob`	`murf, thanks`
`22:31.27`	`*** join/#asterisk mmurdock1 (n=vircuser@mail.kimballequipment.com)`
`22:31.37`	`mmurdock1`	`Howdy all.`
`22:31.54`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`22:33.47`	`mmurdock1`	`I'm looking for some information on setting up a fxs port on a tdm2400p. I have a analog phone connected to port 5. I can call that extension from a sip phone, but when I try calling an extension from the analog phone asterisk just hangs up on it.`
`22:35.14`	`*** join/#asterisk hyphen (n=hyphen@71.224.213.97)`
`22:35.38`	`robin_sz`	`ucfMethod, should be right, is it in the correct context?`
`22:38.00`	`JoseBravo`	`Why im getting this error? Oct 19 17:37:47 WARNING[2656]: chan_iax2.c:7300 socket_read: I don't know how to authenticate`
`22:43.53`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`22:43.53`	`*** mode/#asterisk [+o mog] by ChanServ`
`22:44.12`	`*** join/#asterisk Nixis (n=Nixis@196.204.152.72)`
`22:48.57`	`jarrod`	`hey, is there a way to have an extension playback a media stream to a remote server?`
`23:00.02`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`23:00.36`	`JonR800`	`jarrod: wouldn't you just have the extension dial the other server at a specified extension? have the other server's extension answer and playback the media stream`
`23:03.02`	`*** join/#asterisk DanTMG (n=danielga@203-217-84-193.dyn.iinet.net.au)`
`23:05.37`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`23:14.01`	`*** part/#asterisk cavediver (i=jonas@trimix.eklof.eu)`
`23:16.17`	`bungalow`	`anyone know of a way to break out of a long running wait via an AGI?`
`23:16.18`	`*** join/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`23:16.57`	`bungalow`	`outside of a loop of waits in the agi itself...`
`23:17.27`	`*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)`
`23:20.21`	`*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)`
`23:22.46`	`JoseBravo`	`~book`
`23:22.55`	`jbot`	`methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`23:24.32`	`*** part/#asterisk mmurdock1 (n=vircuser@mail.kimballequipment.com)`
`23:26.34`	`*** join/#asterisk adorah (n=admin@84.94.135.203.cable.012.net.il)`
`23:34.22`	`linagee`	`book is a book called?`
`23:34.27`	`linagee`	`grammar-bot!`
`23:34.42`	`tzanger`	`ok what the fuck`
`23:34.51`	`tzanger`	`I can register to my office* from home*`
`23:34.54`	`tzanger`	`but I can't make a call?`
`23:35.34`	`linagee`	`tzafrir: SIP/IAX, firewalls, what's touching the internet, what's firewalls, need more info`
`23:35.39`	`tzanger`	`but if I take out host=dynamic and the secret and use host=ip.of.home*, it works fine`
`23:35.47`	`tzanger`	`linagee: trust me this isn't a normal problem`
`23:36.20`	`linagee`	`tzanger: when is a problem normal?`
`23:37.36`	`*** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca)`
`23:40.51`	`*** join/#asterisk postel (n=jp@wikimedia/Postel)`
`23:41.56`	`*** join/#asterisk Netgeeks (n=chris@pbx5.netgeeks.net)`
`23:43.23`	`*** join/#asterisk zpertee (n=chatzill@cpe-65-25-121-5.neo.res.rr.com)`
`23:44.16`	`zpertee`	`does anyone have a minute to help me get asterisk .call files working?`
`23:49.39`	`*** join/#asterisk Florineluuuu (n=Florin@smtp-02-ro.nobelcom.ro)`
`23:49.47`	`Florineluuuu`	`hi`
`23:49.58`	`Florineluuuu`	`anyone online ?`
`23:54.31`	`murf`	`X-Rob: u welcome (what did I do?)`
`23:54.48`	`X-Rob`	`VERSION()`
`23:54.54`	`X-Rob`	`Is that going to get into 1.4?`
`23:55.26`	`murf`	`Don't think so, but it wouldn't do you any good even if it did.`
`23:55.37`	`X-Rob`	`whyzat?`
`23:56.11`	`murf`	`Because hardly any of the folks running 1.4 would have it...! Just a few brave updating souls.`
`23:56.31`	`X-Rob`	`HUrm?`
`23:56.58`	`X-Rob`	`Doesn't VERSION() report the same thing as asterisk -V?`
`23:57.50`	`murf`	`I'll put it differently: what percentage of those installations running 1.4 right now (beta 1 or 2), will upgrade when the next release comes out?`
`23:58.03`	`X-Rob`	`Probably none`
`23:58.07`	`X-Rob`	`but when 1.4 comes out, everyone will`
`23:58.14`	`X-Rob`	`well, everyone that I care about, anyway`
`23:58.26`	`X-Rob`	`at which point I want to be able to use one extensions.conf, rather than two 8)`
`23:58.52`	`murf`	`Well, even if it's 90%, that's 10% that won't have it, even if they are running 1.4...`
`23:59.15`	`X-Rob`	`well they won't be running a supported asterisk with freepbx and it's their own fault`
`23:59.22`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`23:59.43`	`murf`	`oops, sorry, gtg, dinner is calling... rather loudly, too! I'll be back in .5 hours.`
`23:59.58`	`X-Rob`	`<PROTECTED>`
`23:59.58`	`X-Rob`	`<PROTECTED>`
`23:59.58`	`X-Rob`	`<PROTECTED>`

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