IRC log for #asterisk on 20090309

00:01.24*** join/#asterisk tlyng (n=torkel@84-52-246.62.3p.ntebredband.no)
00:02.03tlyngdoes anyone here have a skeleton plugin for asterisk using autotools?
00:03.13*** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)
00:09.58*** join/#asterisk dude7064 (n=dude7064@78-86-79-212.zone2.bethere.co.uk)
00:10.27*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk)
00:12.54*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
00:14.45dan__tpdfhacker, thanks again for the tips
00:19.20dude7064How can I have too many calls routed to one number without it being busy all the time ?
00:21.08k-manhow well does SIP work over a wireless network?
00:23.09jblackfine.
00:23.54k-manwhen i tried putting my asterisk server on the wireless network, it seemed to cause a bit of dropping out in the audio
00:24.06jblackmaybe you have a piss poor wireless network.
00:26.24k-manjblack: maybe
00:39.00*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
00:44.14mattbhi, I'm looking for some pointers on how I can send an outbound call I initiated to a context to await user DTMF input
00:44.23mattbeg. I want to dial someone and then present them with a menu
00:44.41mattbcan someone point me at where I should go RTFM ? :p
00:48.06*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-237-246.lns10.mel6.internode.on.net)
00:49.30*** join/#asterisk nargon (n=nargon@217.194.139.4)
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01:09.55jaytee~book
01:09.56jbotrumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
01:10.04jayteemattb, ^^^^^^^
01:10.32nargonsorry can anyone tell me how to contcatenat a string onto the end of an existing string in a dialplan
01:11.06nargonexten => _7XXXXXXX,10,Set(peerconcat = ${peerconcat}${peername})
01:11.12nargonsomething like this
01:11.58nargonI'm trying to build up a string of sip/peers to dial()
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01:13.03*** join/#asterisk friendly12345 (n=friendly@ppp118-208-145-44.lns10.mel4.internode.on.net)
01:14.43jayteenargon, try this instead Set(peerconcat = $["${peerconcat}${peername}"])
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01:19.27dwery'morning. just to let you all know.. it took me a almost week to workaround a series of bugs in the latest wanpipe drivers for the usbfxo. the usb driver in particular has some very nasty bugs and I wonder how such code could work at all. I'll see if I can make sangoma fix them in a saner way.
01:20.52nargonjaytee I notice when I set(peerconcat = somthing) and then NoOp ${peerconcat} on the nextline the NoOp variable echo's empty
01:20.56dwerydon't eve try it, you'e been warned :)
01:22.33k-manis there any linksys ATA's with more than 1 fxo? (to plug analogue phone into)
01:22.45k-manor is that fxs? i get confused
01:23.01k-manits FXS
01:23.09dweryk-man: fxs is to plug analog phones, fxo to plug analog land lines from the telco
01:23.24k-manyeah, so i meant FXS
01:23.30k-mani want to plug 2 analogue phones into it
01:23.54dweryk-man: there must be some.. haven't checked.. I'm pretty sure ZyXEL has one
01:24.04k-manare they a good brand?
01:24.37dweryk-man: not worst than any other brand :)
01:25.00*** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid)
01:28.22k-manthe SPA2102 seems to have dual FXS
01:30.47*** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net)
01:31.10Ritzeriskhow would i tell what codec im using if i have a inbound sip trunk
01:32.36dweryRitzerisk: try iax2 show channels
01:32.52dweryor iax2 show perr <peername> to see all the available codecs
01:32.56dwerypeer*
01:37.26Ritzeriskeven though its a sip trunk from the carrier
01:37.53dweryRitzerisk: sorry, it's late. I confused sip with iax :(
01:38.51dwerysip show channels will probably work
01:39.34Ritzeriski did a call while it was in session with that before u said hehee :) it was a ulaw
01:40.00Ritzerisknow my issue is im not getting any dtmf nor any audio of the ivr
01:40.33Ritzeriskbut if i direct a call to a ext it works (xlite)  and if i make a call it works carrier is les.net and i have g.711 checked
01:42.32Ritzeriskoxc
01:42.43dwerysip has three different ways to send/receive dtmf
01:43.01dwerybut the audio is sent via RTP
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01:43.45dwerygtg... see you
01:45.21Ritzeriskeeeeek
01:45.25Ritzeriskkk thanks ;)
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02:01.10ZX81hey all, anyone know if there is a bug open discussing file descriptor leaks?  We're getting too many open files on a system with about 5 concurrent calls - 3 crashes in last week
02:03.48ZX81hmmm
02:05.58ZX81looks like gsm files left open
02:06.25*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
02:09.12Ritzerisknow my issue is im not getting any dtmf nor any audio of the ivr im at ulaw too
02:10.59jayteesounds like a NAT issue
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02:23.31[T]ankwhat version what the first to introduce the dahdi package? I am looking for the latest version that did not include dahdi
02:24.29thehari'm using 1.4.22 with latest zaptel
02:24.39[TK]D-Fender[T]ank: 1.4.21 was the last pre-DAHDI IIRC
02:24.47[T]ankthank you
02:24.55thehar[TK]D-Fender: ^_^
02:25.08*** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net)
02:25.19theharit does use chan_dahdi, tho
02:25.59*** part/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com)
02:29.31Ritzeriskim open up all the way dmz
02:29.53drmessanoDMZ is your problem
02:29.59drmessanoOpen the proper ports
02:30.03drmessanoSet NAT correctly
02:30.13drmessanoThe DMZ in your router is screwing you
02:30.27Ritzeriskis there a quick port guide eeeeek
02:30.37drmessano~sipnat
02:30.38jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
02:32.47*** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar)
02:33.52Ritzeriskdoesnt a dmz forward all ports to an ip
02:35.02drmessanoIn theory, but theres more to it, and DMZ just doesnt work
02:35.09drmessanoOpen the correct ports
02:35.12drmessanoTake DMZ off
02:36.24Ritzeriskhmm k k
02:36.41Ritzerisklet me just try an open them all real quick haha ;)
02:39.03[TK]D-FenderDMZ is fine.  Overkill, but fine
02:39.19jblackanyone know of a voice chat network?
02:39.20[TK]D-FenderBut isn't enough by itself to configure * to operate behind NAT.
02:39.34[TK]D-FenderA point thats been beaten ove Ritzerisk's head COUNTLESS times.
02:40.12jblackritzerisk: The sip protocol embeds the ip, so port forwarding alone doesn't handle it.
02:41.10Ritzeriskwell bam bam :)
02:42.16Ritzeriski have ulaw outbound works fine just the inbound with the ivr i followed les.net to a T on the setup
02:42.40jblackthere must be some free conference calling Out There
02:45.48Ritzeriskyou miss me :)
02:46.08Ritzeriskthis ones gonna keep me up all night ....
02:46.40[TK]D-FenderRitzerisk: Yes.. but our aim is improving...
02:46.54[TK]D-Fendertrims the main battery another 2 degrees
02:47.31Ritzeriskhaha :)
02:47.35jblackjust go with iax for your provider  and be done with it. ;)
02:47.38Ritzeriskwooot wooot
02:47.50Ritzeriskreally i can change that quick with les
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03:00.53drmessanoLES's IAX is horrid
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03:01.09drmessanoWorks about 20% of the time
03:01.18drmessanoOption 2 is correct setup
03:01.27Ritzeriski cant even get it to pass to my system hehe
03:01.40Ritzeriskso i reverted back to sip trunk
03:01.41[TK]D-Fenderdrmessano: That's greater than Ritzerisk's average success rate, so I wouldn't balk at that!
03:01.54[TK]D-Fenderplays the odds
03:01.55Ritzeriskwooot wooot
03:01.57drmessanoI dont see why this is so difficult
03:02.06drmessanoOpen ports, follow the guide, move on
03:02.15[TK]D-Fenderdrmessano: It's not the odds.. its that playa y0!
03:02.25drmessanolol
03:02.30Ritzeriskthey are all open or i did the dmz route
03:02.53[TK]D-FenderRitzerisk: What part of "you need to CONFIGURE *" is still not getting in your head?
03:03.05[TK]D-FenderRitzerisk: Pile'o'settings without which you are screwed/.
03:03.15[TK]D-FenderRitzerisk: Follow the damn guide already
03:04.13Ritzeriskthe part of not the codec but maybe on why dtmf is not even passing do i need some sort of proxy from the pox to the carrier
03:04.34[TK]D-FenderRitzerisk: No, you just need to set your mode right
03:05.08Ritzeriski did a dtmfmode=rfc2833 even though i saw taht in the setup of the les trunk
03:05.49Ritzeriskno audio so its somethign when it hits inbound not coming back to the carrier but im googling more stuff as we speak :)
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03:14.00Ritzeriskohhh poop on a stick
03:15.34*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
03:15.53[TK]D-FenderRitzerisk: Get yours while quantities last...
03:17.00*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
03:17.15jchonigAnyone familiar with configuring an Aastra 480i with *?
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03:50.26Ritzeriskhah
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04:07.29drmessanoGoogling
04:08.04drmessanoStill a good excuse for not RTFMing
04:09.51[TK]D-Fendersharpens the edges of his "spoonfeeding" cutlery
04:10.44OctothorpeThat's such a hot expression. Consider it stolen. :)
04:11.15*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
04:11.46Octothorpe[TK]D-Fender, my previous statement was directed at you.
04:12.35*** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid)
04:12.50[TK]D-FenderOctothorpe: My works are all CC licensed :)
04:13.05OctothorpeFantastic, thanks muchly :)
04:13.20OctothorpeWait, which license?
04:13.22Octothorpe:P
04:13.49Octothorpespeeds up his plans for commercialization and derivation before [TK]D-Fender can answer
04:14.04[TK]D-FenderOctothorpe: Creative Commons
04:14.14OctothorpeThere are several of them.
04:14.31OctothorpeHence the question
04:15.23Octothorpehttp://creativecommons.org/licenses/
04:15.44[TK]D-FenderOctothorpe: Derivation of a joke tends to create substantially different results thus qualifying as a "different work".  Go right ahead :)
04:16.14OctothorpeYes, but commercialization doesn't.
04:16.22Octothorpespeeds up his T-shirt screenprinting equipment
04:16.36Octothorpe:)
04:17.17[TK]D-FenderOctothorpe: /me will make sure Octothorpe is wearing one of them when the body is found...
04:17.38[TK]D-Fenderheh
04:17.56OctothorpeHeh, niiiiiice. :)
04:18.25[TK]D-FenderOctothorpe: Where poetic licence meets poetic justice.
04:18.54OctothorpeSo true!
04:19.08[TK]D-Fender"Man killed by meteorite : Why so Sirius?!"
04:19.09Octothorpenonchalantly hides another pilfered line up his shirt and walks out the door
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04:34.27tlyngi've created an extension which i can call with my softphone. When I call this extensions the call is answered, then i do Dial(Mobile/my_phone/somenumber) and then WaitForSilence....
04:35.18tlyngmy intension is that WaitForSilence should stop when it's silent in the cellphone... but the dialplan seems to stop execution after the Dial command
04:35.27tlynghow should i do this?
04:40.26[TK]D-Fendertlyng: if you continue past dial that means you are not in that Dial'd call
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04:43.05richardlynchI'm looking for somebody with experience in writing EAGI scripts, preferably in PHP, for contract work.  Can I post here?  If not, where?
04:43.26tlyng[TK]D-Fender: ok, do you know how I'm supposed to do that kind of functionallity?
04:45.04[TK]D-Fendertlyng: What functionality?  You haven't described any real action to take.
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04:47.35tlyng[TK]D-Fender: I want to call an extension using my softphone. When that call is answered I want asterisk to dial another number using my cellphone. The number which is dialed is external and I want to have the possibility to control that channel with dialplan applications.
04:48.56[TK]D-Fendertlyng: You want * to call out and dump the caller into the dialplan basically?  So far I don't see any purpose being filled by your softphone in this
04:51.23drmessanoDo I need to do anything special to enable T.38 for a call, or on a specific peer?
04:52.03drmessanoFor a hot date
04:52.04drmessanoT38
04:52.07drmessanoBurma shave
04:52.15drmessano:(
04:52.50drmessanoI have an ATA with T38 enabled.. T38 passthru all nice and set up in my configs
04:54.11tlyng[TK]D-Fender: actually i only use the softphone to trigger the outgoing Dial. I was thinking about using the softphone while i create the dialplan, so that I can see what asterisk does at the same time hearing the conversation. Better ways to do this?
04:54.51[TK]D-Fendertlyng: Yes, espcially since your callee isn't IN the dialplan when you call him with "Dial"
04:55.01tlyng:)
04:55.02[TK]D-Fendertlyng: Lookup "call files" and "AMI Originate" on the WIKI
04:55.12tlyngok, thanks
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04:59.37pcrackhi is alcatel omniPCX is based on asterisk?
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05:04.53[TK]D-Fenderpcrack: wHAT GIVES YOU THAT IMPRESSION?
05:05.35pcrackjust asking...
05:05.50pcracki just wanted to know...
05:05.58pcrackis it?
05:06.22[TK]D-Fenderpcrack: a proprietary PBX by a proprietary PBX maker.... I'd say **NO**
05:06.39pcrackic...
05:06.53pcrackhave you tried PBX?
05:07.12[TK]D-Fenderpcrack: I've tried all sorts of PBX's
05:07.37pcrackreally cool
05:08.34drmessanoI've tried PBX too
05:08.38drmessanoand network
05:08.58pcrackmaybe you could help me on this...i put the alcatel omniPBX behind the firewall and i already configured the port forwarding for the SIP and RtP ports..it can call but it cannot heard the recepient nor no ringing sound on the IP phone
05:09.45drmessanoThis is #asterisk
05:10.05drmessanoTry #alcatel, #omnipbx, or #pbx
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05:11.20voxterunreal
05:11.31drmessanoThats an IRCd
05:11.35drmessanoTry #unrealircd
05:11.47voxterhahaha
05:11.54drmessanoUmm
05:11.59drmessano#humor for that
05:12.01voxteryeah, nothin there.
05:12.12voxterhad i said lol, you could have pointed me to aol
05:12.14drmessano#0 if you need nothin
05:12.43[TK]D-Fenderpcrack: Your PBX is not supported here.  Go read its documentation or call up a consultant who specializes in them.
05:13.10drmessanoHmm
05:13.19drmessanoSeems like I have an old copy of internet around here too
05:13.22drmessanoMaybe in the same box with pbx
05:14.07drmessanoI stopped buying internet since I could get cracks for it on bittorrent
05:14.28drmessanoI need a good keygen for pbx though, if anyone has it
05:14.46[TK]D-Fenderok, checkout time, later all
05:14.50drmessanoLater
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05:15.28drmessanoPBX-Corporate-Edition-FINAL-1.0-Z0Mg.rar
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05:29.13[T]ankgetting the following when I create a call file. http://pastebin.ca/1356325 havent got a clue where to start looking to fix it. regular dialed calls work ok
05:33.02[T]ankis there anybody out there....
05:33.06[T]ankjust nod if you can hear me
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05:51.05richardlynchDo any of the Playback() Background() etc functions take an ongoing stream input, rather than a fully-formed audio file?  I want to pipe real-time audio output to a channel somehow...
05:56.19ikevin_richardlynch, i don't know if you can, so, you can use moh function for that
05:56.40ikevin_going at work
05:56.41ikevin_cya
05:57.06richardlynchThanks!
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05:58.04[T]ankanyone here know anything about call files?
05:58.24BeeBuu[T]ank: i got a little bit
05:58.51BeeBuu[T]ank: maybe i can help,if you like
05:59.29[T]anki can dial out from my system just fine... but when I try to do a callfile, this is the output I get on the cli: http://pastebin.ca/1356325
05:59.37[T]ankBeeBuu: can I pm you?
06:00.00BeeBuuprivite chat?
06:00.05[T]ankyep
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06:10.43MiccI'm unable to get color with asteirsk -vvvcr, but asterisk -vvvc works fine.
06:10.55MiccI set my TERM variable to xterm-color, but still no color.
06:11.16MiccI've read everything I can find on google, and nothing seems to help.
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06:15.44Miccdrmessano, I'm also working on getting customers setup with a T.38 ATA.
06:16.12Miccdrmessano, I haven't found a provider that does t.38 to pstn yet that will return my calls.
06:26.49drmessanoFlowroute does
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06:39.15Miccdrmessano, you like flowroute? pretty good service?
06:39.38drmessanoSo far
06:39.40drmessanoYep
06:40.06Micchow long have you been using them?
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06:49.52MiccI like the look of their site, very nice layout and I like the colors.
06:50.18bobsaccamanohi..how do i configure asterisk to forward anonymous calling from a sip channel??
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07:00.45mmlj4NOTICE[32564]: chan_sip.c:18390 handle_request_register: Registration from '<sip:0004f2144c31-a@192.168.2.1>' failed for '192.168.2.40' - No matching peer found    | my sip.conf has an entry for [0004f201079d-a]  # this is a polycom phone on * 1.6.0.3
07:14.15mmlj4ok, I'm an idiot
07:20.09mmlj4I'm up apparently way too late, trying to think, and it's not working
07:29.48*** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net)
07:31.07mmlj4all better now, and I'm off to bed
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07:55.13k-manmurio?
07:55.21k-manare you around?
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09:12.07tamielhello, bugs.digium.com is out ? I tried to browse it from two differents sites and browser timeout ...
09:13.18tamielnow bugs.digium.com is responding slowly ...
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10:46.34xmlsoaphi all. I have put wW in both DIAL_OPTIONS and TRUNK_OPTIONS. I have enabled the recordings with record_out=Adhoc,record_in=Adhoc
10:46.40xmlsoapbut i can not record the calls
10:46.44xmlsoapMar  9 11:41:41 VERBOSE[3022] logger.c:  [app_record.so]Mar  9 11:41:41 VERBOSE[3022] logger.c:  [app_record.so] => (Trivial Record Application)
10:46.44xmlsoapMar  9 11:43:02 VERBOSE[3091] logger.c:   recordingcheck|20090309-114302|1236595382.0: Outbound recording not enabled
10:46.49xmlsoapi can see this in the logs
10:46.54xmlsoapany help please?
10:47.16xmlsoap(yes, i;m calling *1 in middle of a call to record stuff)
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11:20.24double_cheesburgWhere are the asterisk playback sounds located? Does anyone know? i.e. transfer, tt-weasels
11:20.34double_cheesburgThe directory?
11:20.50Nobbieasterisk-addons
11:20.55double_cheesburgthx
11:20.58Nobbie<PROTECTED>
11:21.26Nobbiei still extract the old discontinued asterisk-sounds packages
11:22.02double_cheesburgsweet
11:23.25double_cheesburgLooks like the audio format is .gsm, right? Anyone know of a good converter for MP3 or WAV to .GSM ?
11:24.12mort_gibsox
11:26.11double_cheesburgmort_gib : Thank you
11:30.22Nobbieasterisk shows incoming calls from a SIP peer as SIP/x.x.x.x where x.x.x.x is it's IP Address.  How can i make it show as SIP/Name-Of-Peer-$id instead ?
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12:36.31richardlynchI would like to dynamically add a class to the musiconhold.conf file.  I'm sure it's in the docs somewhere, but I'm having a tough time figuring out which bits to re-read.  Any pointers?
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12:46.09[TK]D-Fenderrichardlynch: nothing "dynamic".  change the config file, issue a reload
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12:56.19rashed2020_Hello everyone
12:56.34rashed2020_Is there a way to route an FXO port over the internet
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12:57.10rashed2020_Actually nevermind. I realized that's not really relevant here
12:57.43[TK]D-Fenderrashed2020_: You can provide ACCESS to it using a VoIP protocol
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13:10.36matrix1233hello
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13:11.29rashed2020_[TK]D-Fender: I'm trying to get a DID in a country that none of the VOIP providers I looked at offer. Is that the way to go?
13:11.46rashed2020_The Asterisk box is going to be half way around the world
13:11.56matrix1233i have a problem with installing my B410P
13:12.04matrix1233any one can help
13:12.29beek~ask
13:12.30jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
13:13.20Gido-Ematrix1233 isn't that a isdn card, running on mISDN?
13:13.45GreyFoxxCan anyone recommend any open source SIP load testing software?  I'm looking for something that I could use to simulate 500-1000 or more sip clioent connections and then start placing calls between each other ?
13:14.20Gido-EGreyFoxx sipp
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13:21.10*** join/#asterisk pcrack (n=pcrack@122.53.131.237)
13:21.55pcrackcan anyone recommend what model of HP server asterisk installed can be use for 50 users?
13:23.00Gido-Epcrack i think, anny new hardware wil do the trick for 50 uers.
13:23.17MaliutaLapjust ensure sufficient RAM
13:23.45Gido-Ei think 1 gig is enough.
13:24.14pcrackim thinking using this http://h20341.www2.hp.com/integrity/cache/452196-0-0-101-121.html
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13:24.47pcrackand 4 gig of ram so for further expansion, and featured to be used like call recordings etc...
13:24.58brad_msswerr, i don't know if I'd go itanium
13:25.37MaliutaLapis that arch still alive?
13:25.50tzafrir_laptopYes
13:26.07brad_msswjust go x86 (or x86_64, amd64), you're less likely to have issues as that is what most people use
13:26.15tzafrir_laptopIn fact, HP still pushes it. And SGI's monster machines use it
13:26.29tzafrir_laptopThe rest of the world has mostly abandoned it
13:26.33MaliutaLapI thought it died when x86_64 (or AMD64 ... whatever you want to call it) became the defacto standard
13:26.46pcracknop ill go to dual core
13:27.00MaliutaLapdual core on which arch?
13:27.17MaliutaLapyou are likely to have issues with build and support on and ia64 arch
13:27.34pcracksorry i didnt see it..
13:27.45pcrackmaybe ill go on IBM..xSeries...
13:27.53MaliutaLapit's like going and buying a digium card and hoping to run it on openbsd ... the OS is great but has no support for those cards
13:28.03pcrackHP uses itanium
13:28.18MaliutaLapHP kit is still fine, you can get x86_64 machines from them
13:28.19brad_msswhp has their proliant line which uses x86_64
13:28.43brad_msswif you want to stick with HP
13:28.52MaliutaLapI would recommend HP kit ... ILO is much better than say the Dell offering
13:29.01pcrackany other server recommendations?
13:29.07MaliutaLap!Dell
13:29.12MaliutaLaphates on Dell
13:29.19brad_msswpcrack: have you determined which cards you might need, or are you just needing a pure SIP/IAX solution with no need for bringing in a PRI, or POTS lines?
13:29.45pcrackim just going to use TDM cards
13:29.50pcrack800P to be exact
13:29.54brad_msswpcrack: as I'd be more concerned if the cards will physically fit in the machine (enough slots), rather than the manufacturer of said machine
13:29.56pcrack8 fxo port
13:30.10pcrackic...
13:30.19MaliutaLappcrack: you don't want some PRI/BRI solution?
13:30.55brad_msswpcrack: you might be better off with the AEX800, rather than the TDM800 ... as most servers these days provide PCI-e risers, not PCI risers
13:31.17pcrackhow about both...coz maybe it the future we will use PRi/BRI currently we are just using standard POTS
13:32.33mort_gibMaliutaLap: Why do you hate Dell??
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13:33.45brad_msswpcrack: as long as you make sure the server can take 2 full-height PCI-e cards, you should be fine ...
13:34.23MaliutaLapmort_gib: I have had to work with their servers in the past and getting some of the DRAC stuff and the monitoring stuff working under linux wasn't easy ... and I just didn't like the way they did things. I much prefer working with HP kit, even if it's not perfect
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13:34.27brad_msswpcrack: other than that, you may want at least a raid 1 for reliability, maybe dual redundant power supplies ... can't really suggest much else ... minimum specs from the manufacturer would be sufficient
13:35.11pcrackcan u give me a good server link on HP that we can use based on my requirements..
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13:35.27mort_gibMaliutaLap: Hmm, I'm stuffed here, HP support is done by the biggest crooks in town
13:35.49tzafrir_laptoppcrack, besides the analog trunk, what else does the server need to do?
13:36.12brad_msswpcrack: you'd probably need to call them, HP gives basically 0 information on their expansion slots ... like the ProLiant DL160 G5 Server lists 2 expansion slots ... doesn't say PCI-e, doesn't say full or half height, etc ... useless info
13:36.32[TK]D-Fenderrashed2020_: Sure you can put your own server in there.  Plenty of other gateway devices you could use as well without a full server to manage
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13:37.31pcrackits just for small business use..the usual office type features...call recordings,queue,sip/iax trunking and analog trunks
13:37.43MaliutaLapmort_gib: I tend to work in areas where there is more than one choice
13:38.04Kattyhmm
13:38.22KattyObama is lifting the restrictions on federal funding of stem cell research today.
13:38.38MaliutaLapthat's a good thing
13:38.53Kattyvery.
13:38.56MaliutaLapstem cell research holds promise for a number of areas
13:39.17mort_gibMaliutaLap: Uhm, don't much like HD servers, seems slow and heavy and expensive
13:39.20pcrackhow about IBM servers? any model recommendations on my requirements
13:39.22MaliutaLapas the recipient of a Bone Marrow Transplant I appreciate that more than most
13:39.24mort_gibs/HD/HP
13:40.14KattyI wish people would do a little homework on stem cell research before bad-mouthing it
13:40.19jayteehugs Katty
13:40.23jayteegood mornin!
13:40.42beekmorning jaytee , [TK]D-Fender , et al
13:40.48jayteemornin beek
13:40.49MaliutaLapKatty: I wish some of the US states would lift bans on giving GCSF to healthy people, it makes it more likely to get donors for BMT (given that most transplants these days are adult stem cell transplants) and better for the recipients
13:40.56Kattyhugs jaytee
13:41.44MaliutaLapmy donor was a native american woman, she had to have a dual asperate (hence I got actual marrow not adult stem cells) because of where she lived
13:41.47KattyMaliutaLap: i'm just tired of people calling it an ethical debate.
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13:42.01KattyMaliutaLap: it is NOT an ethical debate, we've found ways to reprogramming skin cells.
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13:42.17KattyMaliutaLap: but southern missouri wants nothing of that. they want to turn this into a Killing Babies debate.
13:42.43brad_msswpcrack: doesn't seem like IBM discloses full vs half height in their specs either
13:42.44jayteewithout things like stem cell research we won't be able to overcome the genetic diseases that we've only encouraged through science letting people with chronic illness survive to adulthood and reproduce passing on those genetic traits, i.e. diabetes
13:42.49MaliutaLapthe christian right is the christian right, no matter where they are
13:42.50brad_msswpcrack: otherwise the x3350 looks fine
13:43.13Kattythe christian's can stuff it "where the sun don't shine" for all i care (=
13:43.35MaliutaLapKatty: you might need to explain that one to them
13:43.56MaliutaLapKatty: I suggest showing them a replica of goatse in explanation ;)
13:43.56jayteeJesus is just all right with me but most of his followers are a stuffy, self-righteous bunch I have no use for.
13:44.15mmlj4I get off on insulting folks I don't know and don't understand
13:44.23MaliutaLapI used to argue with the evangelicals for fun
13:44.44Kattyreligion is fine as long as you keep it a personal thing.
13:44.54Kattyi don't appreciate hearing about it. lol
13:44.56beekAmen to that!
13:45.21Kattyand i REALLY don't appreciate it when it gets in the way of curing diseases
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13:45.57MaliutaLapthere were these brothers that I did history with at uni ... they understood that in history you need more than one source so things can be verified, but still argued the bible was the literal word of god and all the proof that was needed
13:46.00Kattyhow do you feel now missouri?! you just banned the cure of HIV
13:46.31MaliutaLapI don't live in US, so I don't give a rats what they think they won or didn't
13:46.40Kattywell i can understand how a christian could sorta kinda believe the bible is still in tact.
13:46.51Kattybut any reasonable person would have to agree it's been altered.
13:47.01MaliutaLapCSIRO can still do it's research ... that's the same CSIRO that bought us 802.11G
13:47.06Kattywhether by accident in translation, or on purpose by the church
13:47.28Kattyyou know in Denmark, it's more embarassing to talk about God than it is to go running naked through the streets?
13:47.35coppicejaytee: diabetes has more to do with diet than genes
13:47.43mmlj4well, I'm not embarrassed
13:47.53Kattyjust some perspective ;)
13:48.08Kattycoppice: it really depends on the type.
13:48.19Kattycoppice: your'e right, a lot of diabetics are over weight and don't watch what they eat...
13:48.24jayteeI have alot of the "literal word of God" types around me. I point out that in Mark when Jesus came before Pilate he was wearing a scarlet robe and in Matthew it was a purple robe. How can an omniscient being make a mistake like that?
13:48.26Kattycoppice: but there are still children born diabetic.
13:48.38Kattycoppice: and that's a different level of severity
13:48.58mmlj4what embarasses me is when newbies who think they know something technical try to tell a guru how things ought to be
13:49.02coppicetrue, but until countries become affluent enough to eat badly, diabetes is a very rare disease
13:49.05mmlj4like you guys
13:49.27Kattyjaytee: i'm still trying to figure out that All Loving god bit.
13:49.28jayteecoppice, the genetic trait that allows for early onset diabetes is passed through generations via the genes. Although the exact cause is unknown and suspected to be viral, it's a genetic susceptibilty.
13:49.45coppicethe market for blood glucose meters in the west is huge. with india and china becoming more affluent, markets are expanding quickly :-)
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13:50.06fcois93how can I use fastAGI ?
13:50.30Kattythat's a very vague question
13:50.32jayteecoppice, I will agree that western diets exacerbate the situation but that is adult onset diabetes, which is a different form of the disease.
13:51.02Kattycoppice: i think what jaytee's trying to say is it's a REAL disease. not an eating disorder :P
13:51.28fcois93I saw how to send an agi to another server. how another server listen for that? it need asterisk ?
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13:52.59jayteeKatty, right and there are two distinct forms of the disease, one that strikes during childhood and is due to damage of the Islets of Langerhass section of the pancreas that produces insulin, the other is adult onset diabetes and is generally due to obesity, poor diet high in glucose and carbohydrates, i.e. American food in general.
13:53.09[TK]D-Fenderfcois93: http://www.voip-info.org/wiki-Asterisk+FastAGI
13:53.19tamielfcois93: fastagi : you need a socket server listening
13:53.46coppicekatty: that's like saying stupidity is a REAL disease, because it runs in families
13:53.47fcois93[TK]D-Fender: I saw it, but how to listen for ?
13:54.05[TK]D-Fenderfcois93: As you were told, YOU write a program that listens on a socket
13:54.10Kattycoppice: i think stupidity is a disease that runs in families :P
13:54.32Kattyhugs [TK]D-Fender
13:54.48[TK]D-Fenderhugz teh Katty
13:54.51Katty[TK]D-Fender: obama lifts the stem cell funding ban today! horay!!!
13:55.23[TK]D-FenderKatty: thats only so he can resume the creation of his Socialist clone Army!  the Empire begins!
13:55.39Katty[TK]D-Fender: that's an HIV FREE empire, thankyouverymuch!
13:55.42mmlj4if that weren't true, it would be funny
13:56.12coppicefew people susceptible to blood sugar problems don't need medical care unless they abuse themselves. their susceptibility is just like being genetically tall. that's something relatively harmless, as long as you watch out for low beams.
13:57.19pcrackon the forum someone used AEX800 on a HP proliant DL360 Server? is that fine
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14:02.57jayteecoppice, I know you're a very intelligent person, especially with regard to DSP, voip, et.al but go read up on diabetes a little more. Your claim that's ALL due to diet is sadly misinformed and misunderstood. There are people on this planet that regardless of their diet would die without daily injections of insulin. 99.999999% of doctors would agree diabetes is a disease. Show me your P.H.D. in medicine.
14:03.41Kattyshow me your phd
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14:03.44Kattyor the teddy gets it
14:03.48KattyWITH A BANANA
14:03.49jayteehehehe
14:04.28Kattyteddy goes on to have an abortion, and gets excommunicated from the catholic church.
14:04.42Kattybanana is supported by the vatican.
14:05.11coppicejaytee: I didn't say it was all diet, but the majority are. that's what diabetes rates closely match bad diet patterns. Its something I happen to follow for slightly obtuse reasons.
14:05.45Kattyhttp://www.gearfuse.com/wp-content/uploads/andrew/2_jan07/154735364_c175cda85b_1.jpg <- coppice
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14:07.02jayteecoppice, I would agree that in the U.S. most cases of adult onset diabetes (still one form of the disease) ARE caused by bad diet and especially because of obesity. (BIGGIE SIZE ME!!! I'll take the double half-pounder and 3 extra orders of cheese fries please! Oh, and a 96oz size soda full of high-fructose corn syrup!")
14:07.47jayteehahahaha, a headless teddy USB drive? priceless.
14:07.56Kattyisn't it cute?
14:07.56jayteeI still like my TARDIS usb hub
14:08.07[TK]D-Fenderjaytee: Actually its more like the bacon-triple-cheezeburger, uber cheeze fries, apple turnover, oh... and a Diet Coke... I'm trying to watch my weight!
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14:08.36coppicejaytee: do you realise the bonanza in blood glucose meters and insulin the pharma companies are expecting from China, as they eat worse and worse? glucose meters are all in English right now, but everyone is preparing Chinese display models :-\
14:08.50jaytee[TK]D-Fender, ummm, considering poutine I'd say you guys are just one step behind us if it weren't for all the red wine you consume. :-)
14:08.55[TK]D-Fenderjaytee: And you're right it is a disease, and there are people who are going to have it regardless.  Its the other 99.999% that give them a bad name :p
14:09.02jayteecoppice, so now you know which stocks to buy :-)
14:11.31Kattydrop kicks polycom501 through window
14:11.53jayteeI am not a WoW addicted couch potato, I'm an office chair, internet Zynga Mafia Wars addict. I think that's worse due to the lower back strain and the poor circulation in my legs.
14:12.00Kattyhey now
14:12.11Kattylet's not stereotype wow addicts.
14:12.28jayteeHas anyone else here ever had their ass fall asleep? It's an interesting sensation.... or lack thereof.
14:12.44Kattycant' say i've had that happen
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14:13.44Kattynot even during a 5 hour nax raid :P
14:13.53jayteeKatty, oh c'mon! "everyone" knows ALL WoW players are magenta mohawked, multiple face peircings tribal tattooed pot smokers.
14:14.02jaytee:-)
14:14.05Kattylooks at her hair
14:14.17Kattyhmm. i still seem to be brunette
14:14.25jayteeI was kidding!!!
14:14.28Katty;)
14:15.17jayteeThe only reason I'm not a WoW addict is because I deliberately avoid playing the game. It rocks too much that I know that after the "second toke" I'm hooked for life.
14:15.43jayteeand God forbid I ever buy a console with Guitar Hero
14:15.56Kattynever could get into guitar hero
14:16.34coppiceWoW is a true killer application. it has killed several in korea, china and here
14:19.41[TK]D-FenderKatty: I tried it twice, my hand cramped really bad.... never again :p
14:20.16[TK]D-Fenderis doing fine with 19 years playing the real thing :)
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14:20.57coppicehaven't you seen South Park? real guitars are gay
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14:22.08[TK]D-Fendercoppice: No.... I haven't :)
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14:56.03pcrackwhich is more good TDM800P or AEX800?
14:56.19[TK]D-Fenderpcrack: YES
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14:59.09pcracki mean which more you recommend TDM800P or AEX800?
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15:00.45[TK]D-Fenderpcrack: Depends what kind of slot you have.
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15:03.01pcrackim going to use this server?
15:03.54pcrackhttp://www-03.ibm.com/systems/x/hardware/rack/x3250m2/specs.html
15:05.19[TK]D-Fenderpcrack: Well what do YOU think?
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15:07.23pcracki think so..im asking just to make sure
15:07.48pcrackis it?
15:08.01[TK]D-Fenderpcrack: What slots does it have?
15:09.22djMaxis it possible to transfer an inbound call using Asterisk with a T1 and NOT take up any channels afterwards?
15:09.46pcrackits said this Riser assembly PCI-E (4911)
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15:11.29[TK]D-Fenderpcrack: So what do you think about the TDM800P then?
15:11.52[TK]D-FenderdjMax: With PRI and a carrier that supports 2BCT, yes.
15:12.15pcrackit will work also?
15:12.43[TK]D-Fenderpcrack: its &#^$ing PCI.  there are no PCI slots in that damn server.
15:13.18[TK]D-Fendergrumbles that for some even giant flashing neon signs aren't enough.
15:13.43pcrackwhat do you recommend?
15:14.03pcracksorry im very noob on hardware server
15:14.29djMaxso in theory if I ask XO (our T1 provider) if they support 2BCT they'll not think I'm from Mars?
15:14.53angryuserdjMax, they will if you ask the wrong person ;)
15:14.54pcrackso ill go on AEX800P
15:15.03pcrackAEX800 I mean
15:15.15[TK]D-Fenderpcrack: Server?  This is a stupid basic slot on any PC.  PCI is NOT PCI-E
15:15.27djMaxI think they will even if I do, but basically was saying is there a "higher level question" that will get me to the right person while still being in Telco speak.
15:15.52[TK]D-FenderdjMax: Prepare to say "get me a level 2+ tech"
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15:20.46Kattyhugs anthm
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15:21.33anthmhi
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15:23.25qpafternoon
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15:25.34qphas anyone heard of conference room issues with garbled sound but not during normal calls?
15:25.55mort_gibqp:Timing issue??
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15:26.32qpworried about this. normal calls are fine. but I am running it in vmware, and have been told its fine these days for *
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15:26.41obdzhi
15:27.34mort_gibqp: You need to get some kind of timing device up and running, like ztdummy or similar
15:27.56qphow can I test if there is one or not?
15:28.24obdzasterisk 1.4.18. i have a h323 trunk to a definity but when i call TO the definity i got a 80secs DNS SRV lookup timeout. do you know how to resolv this ?
15:28.51obdzDNS server ofcourse is set and from the os i have no problem with it
15:28.51mort_gibqp: You just need to set it up....
15:29.19qpmodprobe ztdummy gives me nothing
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15:30.16mort_gibqp: There you go! Go set it up :-)
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15:30.42qp:)
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15:35.48qpmort, dmesg seems to show its running:
15:35.55qpZaptel Version: 1.4.10.1
15:35.55qpztdummy: RTC rate is 1024
15:36.18angryuserqp, it's rather goot if modprobe ztdummy gives you no output, to be sure if ztdummy is loaded type "zap show status" in cli or "dahdi show status"
15:36.23angryusergood*
15:37.51mort_gibqp: Then I don't know, it would have been a classic scenario for "missing timing" that normal calls were fine...
15:38.04qpyeah :/
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15:39.30angryuserqp, some kernels work bad with zaptel :( i had some situation where the only solution was to add a souce of timing
15:39.45angryuserzaptel dummy i mean*
15:39.52qpodd that this is only in conf rooms?
15:40.11angryuserqp, not at all, what do you call a normal call ? voip ?
15:41.15qpyeah, we use conference rooms to connect 2 voip calls, like a managed transfer from our softphone.
15:43.28qpasterisk01*CLI> zap show status
15:43.28qpDescription                              Alarms     IRQ        bpviol     CRC4
15:43.28qpZTDUMMY/1 (source: RTC) 1                UNCONFIGUR 0          0          0
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15:44.32qpnot sure if thats a good thing angryuser :)
15:44.38angryuserqp, yes it's classic, you have a timing issue, try to execute zttest
15:44.46qpok
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15:45.51qploads of 99.x %
15:46.08angryuserqp, paste the average min and max
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15:47.47angryuserqp, ctrl +c to complete
15:48.16qpheh one result went mad
15:48.17qp--- Results after 47 passes ---
15:48.18qpBest: 99.998 -- Worst: -6359.631 -- Average: -37.562894, Difference: 237.414602
15:48.30qpmost are 99.9x
15:49.09qpi'll do another 20
15:49.20angryuserqp, something is definatly wrong , normally values dont ever go under 99.
15:49.31angryuserdefinetely
15:50.03angryuserqp, pastebin all output
15:50.54qphttp://pastebin.com/d152fc152
15:50.57qp2 runs
15:54.11angryuserqp, second one is not so bad , but not great either , do you use g729 ?
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15:57.30murraytmis there any trick to getting the i option for the Dial app to work on a PRI?  i'm using asterisk 1.4.22 with sangoma hardware.
15:58.32qphmm, I think so angry
15:59.13angryuserqp, try to use meetme without transcoding in local environement
15:59.30brunneris there a valid way to comment something out in extensions.conf?
15:59.38Qwellbrunner: ;
15:59.41brunnerthanks
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16:01.00qpthanks angry, will look into that. would timer issues cause calls to not come in properly too? ie sometimes get number unavaialble? or is that down to the voip provider
16:02.32angryuserqp, not a time issue, voip provider or your config
16:02.37angryusertimer*
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16:04.30qpg711 we use
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16:50.38dweryhi. I need to make a Dahdi channel to go off-hook, dial a Flash and then a number. Is that possible?
16:51.19[TK]D-Fenderdwery: Yes
16:51.51dwery[TK]D-Fender: nice to know. now let's hope the usbfxo supports that!
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17:01.41dwery[TK]D-Fender: should I use Flash() after Dial() ?
17:01.48theharis SetCallerPres() from 1.0 now Set(CALLERPRES()=) in 1.4.22?
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17:05.11[TK]D-Fenderdwery: No, because after Dial is too late.
17:05.38dwery[TK]D-Fender: mm then I'm missing something.. :(
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17:05.52[TK]D-Fenderdwery: call-file -> Dial(zap/1), dump into dialplan, then call flash, SendDTMF, then bridge to wherever
17:06.31dwery[TK]D-Fender: it seems that Dial() waits for the other party to answer
17:06.53[TK]D-Fenderdwery: Depends
17:07.34dwery[TK]D-Fender: should I try the timeout option?
17:08.06[TK]D-Fenderdwery: I have jsut told you the process for this.
17:08.07mitchGaffiganWere there any changes made to the MailboxExists function in 1.6?
17:08.13[TK]D-Fenderdwery: Please actually read it
17:08.26dwery[TK]D-Fender: I've read it but I'm probably missing the critical part.
17:08.28[TK]D-FendermitchGaffigan: What does the ChangeLog say?
17:08.33[TK]D-Fenderdwery: CALL-FILE
17:09.20eppigyhello
17:09.22eppigyi am dave
17:10.30dwery[TK]D-Fender: ty, I'll read docs on call-file
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17:12.48mitchGaffigan[TK]D-Fender: only that it was converted to a dialplan function
17:13.12[TK]D-FendermitchGaffigan: Then there you ahve it
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17:14.55mitchGaffiganwould that cause it to not respond to the j argument?
17:15.37mitchGaffiganor require any chagnes to the dialplan for it to work correctly?
17:15.52mitchGaffigan(relative to it's usage in 1.4)
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17:20.38[TK]D-FendermitchGaffigan: Show us what you're doing.
17:23.19mitchGaffiganhttp://pastebin.com/m565687d
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17:25.05[TK]D-FendermitchGaffigan: And this is 1.6?
17:25.14mitchGaffiganyes
17:25.23[TK]D-FendermitchGaffigan: go read that apps instructions again.
17:26.24mitchGaffiganwhere would I find that (other than on a site like voip-info)?
17:26.36[TK]D-FendermitchGaffigan: "core show application mailboxexists"
17:28.19mitchGaffigan"Options: (none)" meaning do your own jump?
17:29.59[TK]D-FendermitchGaffigan: meaning there are no "options" any more and you should be suing the channel variable for this.  Also that "|" is not a valid delimiter in 1.6 which you seemed to have missed
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17:30.26mitchGaffiganthanks
17:30.33ZerylHey all, not sure how to phrase my question for google, or even the correct terminology, so I'm curious if someone could assist me in lettingme know if what I want to do is possible
17:30.38*** part/#asterisk Magicblaze007 (n=sony@garpc.cs.fsu.edu)
17:30.59Zerylthe company i work for (6 of us) all work from home, and currently outsource our PBX to a company, so that when someone calls for support, it round robins the support staff
17:31.23Zerylis that something asteisk could do i.e. each person logs in w/ a sip phone, and when someone calls the main support number, it'd go out to one of us?
17:31.33kerxany suggestions on a good method to integrate an Order Entry system and call recording w/ Monitor() so that call recordings from agents who place entries into the Order Entry web-interface have an attached call recording in the database for that person who they were speaking to?
17:31.49mitchGaffiganZeryl: look at http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
17:32.08Kattyi hurt my hand swapping motherboards )=
17:32.10ZerylmitchGaffigan: ty! I'll take a look now
17:32.14Kattyshakes fist
17:32.19Kattydamn you manual labor!
17:32.48eppigyD:
17:32.56Talkradioheh
17:33.02Talkradioyou go girl
17:33.16Talkradionice to see someone not afraid of alittle hard work
17:33.50[TK]D-FenderZeryl: Quick answer : Yes
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17:34.35mitchGaffiganKerx: Record the call, then move the file to a defined location and have your entry system attach that file once the call is completed.
17:34.38Zerylty [TK]D-Fender, going to look into it then, as it's gotta be cheaper than what we/they are paying currently (already have servers sitting around doing nothing)
17:35.09KattyTalkradio: heh ;)
17:35.15mitchGaffiganZeryl: if you are trying to do something quickly, take a look at Trixbox
17:35.22[TK]D-Fender...
17:35.24[TK]D-FenderEW
17:35.29mitchGaffigansorry...
17:35.40mitchGaffiganIt works to a point.
17:36.01[TK]D-FendermitchGaffigan: Sure thing... Vlad
17:36.04Zerylwe have pretty basic requirements, so i'm sure almost anything would work
17:36.08mitchGaffiganlol
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17:37.12Talkradioi have 4 trixbox setups out there :) all working fine but watchout for phones not working when internet goes down
17:39.03kerxmitchGaffigan, how would I know if the order through the web-based order entry has been completed to know to attach it to example order id #1243
17:40.33Zerylbased on my limited knowledge, i'd say have asterisk save the file out to a drive, with the telephone number + timestamp, and when you click "complete order" or something, it checks for the presnece of that file, and attaches it
17:40.37Zerylimo
17:40.59kerxZeryl, i like this suggestion, i'm only a bit worried about people who call in w/ Blocked Numbers
17:40.59Zerylfrom a programming perspective at least (don't know the full capabilities of asterisk tho, so it may not be possible)
17:41.20kerxyeah, i'm trying to figure out that
17:41.22Zeryldepending on your situation, you may be able to get ANI delivery (especially if it's an 800 number)
17:42.27kerxhrm.  i'm not sure i know what that is, also quickly reading the Wikipedia page.  Is it some sort of caller id un-block? ;-)
17:43.03Zerylit's automatic number identification (essentially), it's the REAL number that calls.  Since the 800 numbers are paying for you to call them, they apparently have the right to know who's calling them, so they use ANI
17:43.34kerxawesome
17:43.42kerxunfortunately 800 numbers are expensive
17:43.43Zeryli would imagine that some sip providers also allow you to see ANI, but I can't guarentee that, or have any resoning to back that up
17:43.55kerxyeah, i hear ya.  i'll check that out
17:44.08Zeryli'd be willing to bet your telephone provider offers it (probably at a price though)
17:44.11kerxi read a bit on AGI's, and it seemed promising....
17:44.37mitchGaffiganI would have the exension of the user attached to their user account and use manager to get the call id when the user clicks a button on the UI to retrive the filename.
17:44.44kerxi was thinking of the possibility of having agent's sit in a Meet-me room, and when a inbound call comes in then through some AJAX pop there web-interface w/ the call
17:44.49[TK]D-Fenderkerx: Expensive?  Never seen them cost anything more than normal LD...
17:45.07kerx[TK]D-Fender, AFAIK 800,866,877, etc.  are per-minute basis
17:45.24[TK]D-Fenderkerx: Yes, just like normal LD
17:45.47ZerylmitchGaffigan's idea is better.  if asterisk supports querying current calls based on extension, seems like a much safer method
17:46.00kerxAlso, I can't be bound to just a 800 number for the inbound calls.
17:46.08kerxI'm going to be routing them lots of SIP and IAX2 calls
17:46.28kerxYeah, I like mitch's Idea also
17:46.33kerxI think that's going to be the route
17:46.35mitchGaffiganso, the complete thing is: call comes in, user clicks button to say they have a call, order entry server logs in to AMI to get the call id, the call finishes recording and is saved to a web accessible directory with the call id as the filename, the filename is stored in the database for later retreval
17:47.03[TK]D-Fenderkerx: Just 800 number?  HUH?  And then SIP & IAX2 are PROTOCOLS.  this says nothing of the kind of calls being passed over it.
17:47.08kerxSeems promising
17:47.23*** join/#asterisk neurosys (n=vinix@sheltercorp.net)
17:47.45kerx[TK]D-Fender, What I mean is that the call center will be receiving inbound calls through various methods.  They do have 800 number but also they have other forms of receiving calls.
17:47.45*** join/#asterisk socram (i=c86c8936@gateway/web/ajax/mibbit.com/x-c1829a15676f4885)
17:48.02kerxI'm trying to find an all-around solution if one exists
17:48.05[TK]D-Fenderkerx: "800 #" is not a way of receiving calls.
17:48.13kerxAnd so far I like this idea
17:48.16[TK]D-Fenderkerx: Stop mixing this up.
17:48.21*** join/#asterisk ]technophreak[ (n=]technop@modemcable048.23-81-70.mc.videotron.ca)
17:48.37[TK]D-Fenderkerx: 800 # is a BILLING concern
17:48.42mitchGaffiganI don't think the call protocol or sourse is an issue
17:48.48mitchGaffigan*source
17:48.53kerxOk, I may have mixed it up sorry
17:49.16kerxI like the above idea though :)
17:49.24]technophreak[Is there a way to have registration keep trying when handle_response_register gets a 404 ?
17:49.29mitchGaffiganthanks... it is similar to what we use
17:49.47dwery[TK]D-Fender: it seems to work, I had to play a bit with the timings. I now need to understand how to open a SIP channel toward a registered user and bridge to it
17:50.04[TK]D-Fenderdwery: ... DIAL
17:50.26dwery[TK]D-Fender: d'oh...
17:50.35kerxAgent 123 receives call, customer seems interested, he begins a order-entry.  The web script logs in through AMI to receive the call ID being recorded.  Then keeps that recorded in the database.  When the call is completed and the order is completed, some daemon that monitors the directory comes in and grabs the recording and attaches it into the database
17:50.53kerxmitchGaffigan, What do you use in AMI to see which call ID the Rep is on?
17:50.58kerxIf you don't mind me asking
17:51.45socrami need to make an ivr for a bank-like org. People will call and make transactions over the phone, check account status, billing status, etc. how much is asterisk suitable for this?
17:52.37mitchGaffigansocram: AGI
17:52.41dwery[TK]D-Fender: but if I make an agi or something like that in order to generate the call-file on-the-fly, the SIP channel to the phone should already be established. So I need to find it and bridge to it, right?
17:52.45kerxsocram, seems very suitable once you become pretty efficient and knowledgeable of the AGI
17:53.12kerxmitchGaffigan, maybe all of the IVR i'm doing should be an AGI script instead of the dialplan?
17:53.42mitchGaffiganyou could... but why?
17:54.23mitchGaffigankerx: Command with the command of core show channels
17:54.51*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
17:54.52kerxgot it
17:54.53mitchGaffigansorry, sip show channels
17:54.58mitchGaffiganthat has call id
17:55.02kerxyea
17:55.03kerxi see now :0
17:55.05kerxslaps himself
17:55.10kerxok i'm going to go through w/ this
17:55.19kerxi would prefer not to make an AGI
17:55.28mitchGaffiganwhat is your order entry system written in?
17:55.41kerxit's  PHP and Dojo javascript framework
17:56.03guaxkerx, zend?
17:56.10kerxguax, no PHP framework
17:56.21guaxhumm
17:56.23kerxi'm not good at any of those unfortunately
17:56.27mitchGaffiganyou're in luck, PHP has a full API already written
17:56.27kerxjust basic PHP
17:56.42Zeryli don't blame you there kerx, i've tried to get into the whole MVC w/ zend/cake/etc, and I can't
17:56.51mitchGaffigansee http://code.google.com/p/asterisk-php-api/
17:56.58guaxuses Zend Framework, PostgreSQL and PhpAgi
17:57.20kerxmitchGaffigan, the only reason i didn't want to make an AGI is because I like to continue to use asterisk's queue mechanism w/ rrmemory
17:57.24Zeryldon't get me wrong, the zend framework looks to be really strong, I just can't do it :(
17:57.30guaxZeryl, do you program in java or another OO language?
17:57.31kerxi'd hate to have to create my own ACD queue inside the agi
17:58.04Zeryli'm a php dev, know a little bit for a couple other languages, but none as comfortably as php
17:58.29mitchGaffiganwho said anything about AGI?
17:58.41kerxwhat are u talking about then?
17:58.48kerxOH!
17:58.50kerxslaps himself
17:58.51kerxyou said API
17:58.56mitchGaffiganoh, that was in response to dwery about accessing a bank system
17:58.58kerxok, it looks like my 4 hours of night sleep wasn't enough
17:59.01Talkradiowow for a sec i felt my wallet getting tugged on when i thought you typed aig heh
17:59.03mitchGaffigansame here
17:59.05kerxnevermind me
17:59.06kerxsorry
17:59.56siera08Using cmd dial with 'm' or 'r' option, i have problem in asterisk 1.4.18.
18:00.23siera08if i call internal phone and he cancel the call, i can hear some voice message.
18:00.49siera08but when i call external phone through sip/zap/iax trunk, i can't hear voice message which other pbx(asterisk, too) plays.
18:01.07siera08In this case, after i heared "connecting..." from other pbx, called party should be called.
18:01.35siera08but i can't hear anything(if 'm' option is enabled, moh plays) before called party accepts or cancels the call.
18:01.58mitchGaffigansiera08: check that your codecs are compatible
18:02.02siera08i  want to hear sound message that other pbx plays before calling the phone.
18:02.12Gido-Ein sip.conf     is it    limitonpeer or limitonpeers ?
18:02.22siera08sorry, me is very poor english.
18:02.45mitchGaffiganGido-E: plurarl
18:02.56mitchGaffigan*limitonpeers
18:03.06RobHAnyone offhand know what the entry is to increase the default wait time for entering a name in the directory?
18:03.10siera08mitchGaffigan: thank u for your kind. codec has relation with it?
18:03.20]technophreak[Is there a way to have registration keep trying when handle_response_register gets a 404 ?
18:03.25Gido-Eok, i see a lot of examples with limitonpeer option.    But that doesn't do annything for me, i thought.
18:03.28siera08sorry [TK]D-Fender, yesterday.
18:03.32mitchGaffiganif asterisk cannot convert between two codecs, it will not give any sound
18:03.50*** join/#asterisk |AbsyntH| (n=never@host216-187-dynamic.16-79-r.retail.telecomitalia.it)
18:04.25*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
18:05.06siera08mitchGaffigan: en, but if the called party accepts the call, i can have conversation with him..
18:05.17mitchGaffiganah
18:05.36|AbsyntH|a little question from an asterisk n00b ...which linux distribution do you suggest ?
18:05.46mitchGaffiganmost
18:06.43*** join/#asterisk wbw (n=caboose@c-66-229-196-96.hsd1.fl.comcast.net)
18:07.00kerx|AbsyntH|, speaking on behalf of myself CentOS, Debian, Ubuntu
18:07.04hapsout of curiosity, does anyone here use freebsd/asterisk?
18:07.05|AbsyntH|yep but for example i can manage gentoo,debian,ubuntu,opensuse or centoos but for this work a don't wont to spend a lot of time
18:07.24|AbsyntH|so i think about debian/ubuntu
18:08.15kerxyes, that's a good one
18:08.24kerxbig community backing
18:08.26kerxgo for it !
18:08.36mitchGaffigansiera08: can you show the relevant dialplan
18:08.44*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
18:08.57|AbsyntH|right...and a good gui?
18:09.44]technophreak[Is there a way to have registration keep trying when handle_response_register gets a 404 ?
18:10.05|AbsyntH|for a quick manage...i know that the big work i've to do by shell,but for some quick change
18:13.32[TK]D-Fender|AbsyntH|: Generally no
18:13.57[TK]D-Fender|AbsyntH|: the GUI's out there are largely complete solutions and working outside the box is challenging.
18:14.14[TK]D-Fender|AbsyntH|: the more direct phrase would be "owns your sorry ass"
18:14.21|AbsyntH|ahah
18:14.25|AbsyntH|ok ok
18:14.42*** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net)
18:15.23|AbsyntH|the shell is the way i understand ;)
18:16.50|AbsyntH|i've to go...tnx and goodbye
18:18.30]technophreak[Does anyone even know what I am talking about ?
18:19.01*** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw)
18:19.04]technophreak[Is there a way to have registration keep trying when handle_response_register gets a bad password response or a 404 ?
18:20.42siera08mitchGaffigan: http://pastebin.com/d45ec4941.
18:21.18*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
18:21.27RobHHrmm, suddenly since I switched to .6 my voicemail dialing from phones is odd.  It now always prompts the user to enter both the extension and password, when before it only asked for the password.  I have the extension listed here:  http://pastebin.com/m776111f
18:21.29edoceois it possible to build a geo-redundant * setup?  I want my 800 number to go to San Jose server if on/available or my Baltimore server if not
18:21.32RobHAnyone see what I am doing incorrectly?
18:22.35[TK]D-FenderRobH: Yes... you are not showing us the failed call or your VM config
18:23.29]technophreak[Is there anyway for asterisk registration to keep trying when it gets a 404 or bad password instead of giving up ?
18:24.12RobH[TK]D-Fender: http://pastebin.com/m1c4e518d  now shows the CLI
18:24.30RobHbut my question is can you still pass the callerid(num) into the voicemailmain application?
18:25.20[TK]D-FenderRobH: Try following my actual request in its ENTIRETY
18:25.41RobHYou want the contents of my voicemail.conf?
18:25.59[TK]D-FenderRobH: CLEARLY
18:26.02*** join/#asterisk Jabka (n=jabka@CBL217-132-73-4.bb.netvision.net.il)
18:26.16*** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com)
18:28.00*** join/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net)
18:28.39*** part/#asterisk Zeryl (n=Zeryl@97-87-122-210.dhcp.stls.mo.charter.com)
18:28.41*** join/#asterisk docid (n=eris@69.196.68.142)
18:29.22Jabkawhat should i read to set asterisk between two as a relay (any incoming call transfer to some sip server (ekiga for exanple))
18:29.23docidok, so i need to strip all the leading digits from incoming calls whose last 4 digits begin with 68 or 69, haveing trouble figgureing out how to write that
18:29.26RobH[TK]D-Fender: http://pastebin.com/m59471db5
18:29.32*** join/#asterisk Khratos (n=khratos@190.166.103.111)
18:29.51RobHI am not sure why the voicemail.conf is needed, when I am not sure why it is not passing the callerid number to the voicemailmain application is all =[
18:29.54RobHbut its there now
18:30.30[TK]D-FenderRobH: Its asking for the box #?
18:30.34RobHYep
18:30.52RobHIt prompts for user to enter voicemail box, then pass.  Before it was just taking the callerid number, and prompting for the pass
18:30.53[TK]D-FenderRobH: Do another call and fully enter the box
18:30.56mostyJabka, the book would be a good start
18:30.58RobHok
18:30.58mosty~thebook
18:30.59jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
18:31.11Jabkawent ot amazon
18:31.34[TK]D-FenderJabka: Dial(SIP/12345@ekiga.net)
18:31.40[TK]D-FenderJabka: There.  thats it.
18:31.59RobHbah, found the issue
18:32.27Jabka[TK]D-Fender that easy ?
18:32.54[TK]D-FenderJabka: Yes
18:33.07[TK]D-FenderRobH: namely?  Didn't reload your VM config?
18:33.23RobHdid not reload the sip after fixing a users callerid
18:33.26RobH>_<
18:33.33*** join/#asterisk op3r (n=op3r@114.108.201.205)
18:33.56[TK]D-FenderRobH: Yeah, I missed the 620/602 myself
18:34.25*** part/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net)
18:35.05RobHif only all my problems were a result of my own inattention, I would be set.
18:36.17Jabkais reading thebook
18:36.24*** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl)
18:37.07op3rcan anyone point me to the right direction why asterisk kept on playing default for music on hold while I only have a girl.raw on /var/lib/asterisk/mohmp3 which i also set on musiconhold.conf?
18:37.34[TK]D-Fenderop3r: I don't see any backup for your problem....
18:37.45op3r:(
18:37.47op3rlol
18:38.24]technophreak[Is there anyway for asterisk registration to keep trying when it gets a 404 or bad password instead of giving up ?
18:38.43RobHAnyone know off the top of their head what I can put into either my dialplan or elsewhere to increase the timeout a user has to enter someones name in the directory?  I do not see any option for this kind of thing in the Directory application call itself.
18:39.37*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)
18:45.45mosty]technophreak[, registertimeout
18:47.35]technophreak[you sure ?
18:47.55]technophreak[it doesnt seem to be timing out, just abandoning trying to register
18:51.11]technophreak[I found registerattempts  ... will try messing with that
19:02.03*** join/#asterisk Jabka (n=jabka@CBL217-132-73-4.bb.netvision.net.il)
19:02.08]technophreak[It doesnt seem to change anything
19:02.39]technophreak[asterisk literally stops trying to register as soom as it gets a 403
19:04.29mostywhat combination of registertimeout and registerattempts did you try?
19:05.14]technophreak[registerattempts=0
19:07.35]technophreak[I just tried with registerattempts=100 and registertimeout=60
19:07.38]technophreak[same problem
19:09.01*** join/#asterisk Shaun2222 (n=shaun@ip68-5-154-128.oc.oc.cox.net)
19:09.11]technophreak[I put up a trace, and iets just not retrying
19:09.12Shaun2222how would i match what phone number sombody called in on?
19:09.23Shaun2222i have some did's figured i should use them :)
19:09.35mitchGaffigan${EXTEN}
19:09.47Shaun2222lol, i guess that makes sense.
19:09.48]technophreak[shaun2222: is your carrier providing you multiple trunk or just one ?
19:10.13Shaun2222]technophreak[: it's a PRI T1, just one circuit.
19:10.30]technophreak[then you`ll need to match it at the dialplan
19:10.36[TK]D-FenderShaun2222: It lands on a exten of the DID they dialed
19:10.53[TK]D-FenderShaun2222: You already have this, you just weren't looking
19:11.50*** join/#asterisk dwery (n=dwery@nslu2-linux/dwery)
19:12.29*** join/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net)
19:12.49minteeis there another way besides IAX2 and SIP to trunk 2 asterisk servers together?
19:13.16minteeor to fwd a number to another voip number without using the PRI?
19:13.21jplankpri
19:13.31jplankBRI?
19:13.47minteeO_o
19:14.09pdmmmmintee: what are you trying to do?
19:14.13minteeyeah, i'm not trying to use any of my zap channels
19:14.28minteejust route about 100 numbers over to another provider
19:14.55jplankSo you dont want to use a ZAP channel, and you don't want to use SIP or IAX, do you have strings and a tin can?
19:14.56Shaun2222[TK]D-Fender: yep, i just had a global match though _X. for that context.
19:15.04pdmmmuse a laplink cable!
19:15.05pdmmmhaha
19:15.14minteeI have a 10mB ethernet connection, and would prefer to use that.
19:15.16minteepdmmm: hehe
19:15.38jplankbut you don't want to use VOIP, so how are you going to use a ethernet connection?
19:15.47minteejplank: no, I will use IAX or SIP, just making sure there are no other options
19:15.52jplankH323
19:15.54jplank:)
19:15.54mitchGaffiganchan_console
19:16.13mostymintee, do IAX over the ethernet connection
19:16.17Shaun2222gtalk ;)
19:16.32jplankI'm sure you if you were a good enough coder, there's probably a million ways to do it
19:16.37jplankmaybe write your own protocol?
19:16.43mitchGaffiganSpeech Recognition -> serial
19:16.43minteemosty: yeah, that's my main goal, but i don't know if they support IAX on  the other side...
19:16.52jplankthen why not SIP?
19:17.03docidany way to use gotoif to check incoming trunk and direct from there? im feelin a bit lost on this one
19:17.08mostymintee, ask what they do support, then you know your options
19:17.13jplanklol
19:17.17minteelol, yeah, i tried
19:17.29Shaun2222docid: what are you trying to match?
19:17.32minteethey are indian, and i couldn't understand a goddamn word they were saying
19:17.37minteeso i'm composing an email to them
19:17.37jplankthen use SIP
19:17.52mostymintee, email and ask for authentication details
19:18.00jplankI'm sure they support SIP, I've never worked with a customer who had an office in India that didn't support SIP
19:18.06mitchGaffiganlol
19:18.17docidwell, if a call comes in on g0 and doesnt match a sip extention of voicemailbox, then it needs to be dialed out on g1, and the reverse is also true
19:18.18minteemosty: that's what i'm doing ;)
19:18.31minteethanks kids...
19:18.36jplankkids?
19:18.39mitchGaffigandocid: what are you trying to do?
19:18.46jplank!kickban mintee
19:18.49jplank:)
19:19.11minteezoooooo Nooooooooes
19:19.19docidmitchGaffigan, Shaun2222    well, if a call comes in on g0 and doesnt match a sip extention of voicemailbox, then it needs to be dialed out on g1, and the reverse is also true
19:19.20*** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net)
19:19.33docidor not of
19:19.55Shaun2222docid: so you want a call that comes in that doesnt have a voicemailbox setup to dial out on DAHDI/g1/${EXTEN})?
19:20.20docidyes,
19:20.28docidif it comes from g0
19:20.37docidif it comes from g1 iut needs to dial out on g0
19:21.13Shaun2222sounds simple enough to do.
19:21.17mitchGaffiganso use _#. for the extension, use MailboxExists(${EXTEN}) and Dial(DAHDI/g1/${EXTEN}) otherwise
19:21.40Shaun2222tis tis... MailboxExists is deprecated :)
19:21.50docidwhat is it replaced by?
19:21.50mitchGaffiganto be replaced with what?
19:21.53jplankI was just looking for the updated command
19:22.05Shaun2222probably somthing like.... exten => _X.,1,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?:1)
19:22.10mitchGaffiganI was under the impression that the j option was just depreciated
19:22.19Shaun2222you would need to modify where it goes obviously
19:22.28[TK]D-FenderShaun2222: Umm.... NO.
19:22.29docidok, also in this i need to strip any leading number besides the last 4 if the last four match 68XX or 69XX
19:22.44edoceoDo I need to get a special carrier to get geo-redundant * servers?
19:22.51jplankI thought it was also, but it doesn't say so in the eiki
19:22.57jplankhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MailboxExists
19:23.00Shaun2222[TK]D-Fender: no?
19:23.05[TK]D-Fenderdocid: Fo read the CHANNELVARIABLES docs in your source tarball.  You seem to lack the basics
19:23.18mitchGaffiganedoceo: just one that supports SRV records
19:23.20[TK]D-Fender15:22]<Shaun2222>probably somthing like.... exten => _X.,1,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?:1) <- tragic logic flaw
19:23.21docidokiez, will do.....
19:23.30edoceomitchGaffigan: thx
19:24.04Shaun2222[TK]D-Fender: well, ya i know that wont work as it.. he would have to change peices...
19:24.05jplankoh, that would just keep looping wouldn't it?
19:24.22jplankif not boxes exist
19:24.29[TK]D-Fenderjplank: SHHHHHH!!!!
19:24.32jplankgrrr
19:24.34Shaun2222i use it like 'exten => _2XX,7,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?:9)' to jump over 8 which is voicemail()
19:24.35[TK]D-Fender:p
19:24.54jplankshaun look what you have and what you told him to do
19:25.22mitchGaffiganShaun2222: if that is correct... i'm going to cry
19:25.39Shaun2222jplank: yes i know, i see the loop, i just put _X. and 1 at the end, it's his task to make it work..
19:26.08[TK]D-FenderShaun2222: its not the "_X." and "1" that is the problem.  It is the "1" and the "1"
19:26.12docidwell, thanks for the bait, mabey  can catch something with it :)
19:26.19jplankI've noticed, no matter how much you tell people to not take your suggestions verbatim, they will
19:26.19Shaun2222mitchGaffigan: if whats correct?
19:26.31mitchGaffiganusing Mailbox_exists like that
19:27.01mitchGaffiganit looks absolutely horrendus... yet much shorter than the three line construct I am using
19:27.04Shaun2222mitchGaffigan: well the console yelled at me when i upgraded to 1.6 and my dialplan was using mailboxexists :)
19:27.08mitchGaffigansame here
19:27.15mitchGaffiganI was trying to fix that right now.
19:27.47Shaun2222mitchGaffigan: for me it actually cut out some lines
19:27.58Shaun2222mitchGaffigan: since i no longer had to check the mailbox and then check the var
19:28.19mitchGaffiganit would be the same number of lines as my 1.4 as I was using j to jump to n+101 on success
19:28.21Shaun2222it does it in one task.,
19:30.28Shaun2222jplank: ya i know people copy/paste exactly what you give them... hell i'm guilty at times... either way, if i would have pasted him mine he would have just been trying to figure out why it keeps jumping to 9... probably would have had a bunch of exten => 2__,3,noop(filler);exten => 2__,4,noop(filler);exten => 2__,5,noop(filler);etc... lol
19:31.02Shaun2222bah... s/2__/_2XX/g
19:31.41Shaun2222mitchGaffigan: ah, well then it's a simple change for you...
19:31.56Shaun2222i never liked the jump to n+101 though it was ghetto.
19:31.58Shaun2222:)
19:32.41mitchGaffiganI didn't either and it makes sense to make it that way... more flexible too
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19:39.19edoceoDo I need to use a carrier or is their a way (via DIDx for example) to have my * servers directly handle inbound w/o an upstream SIP provider
19:41.04*** join/#asterisk seanmh (n=johndoe@198.59.129.24)
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19:44.59mitchGaffiganedoceo: set sip.conf's context to be the one where you want to process an anonymous sip call
19:46.34mitchGaffiganedoceo: or if you are referring to a group of * boxes that are geographically distributed, you can use DUNDI or just have a dialplan that routes over trunks forming a mesh to the correct box
19:48.15dweryI'm getting a bunch of "dahdi: Cannot start tone until a tone zone is loaded."  I have both loadzone and defaultzone in system.conf . What am I missing?
19:48.26[TK]D-Fenderedoceo: What does it mean to get SIP calls without an "upstream SIP provider"?  Do packets appear out of thin air?  SOMEONE is sending you them...
19:48.53*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
19:53.07edoceo[TK]D-Fender: Right - but instead of Caller->AT&T (their carrier) -> XO (my carrier) -> [sip] -> My Asterisk PBX in office -> my phone
19:53.21*** part/#asterisk ]technophreak[ (n=]technop@modemcable048.23-81-70.mc.videotron.ca)
19:53.23edoceoI want to take XO out of the loop and have ATT know to SIP right to my machine
19:53.52[TK]D-Fenderedoceo: You would need to arrange that with AT&T
19:54.06[TK]D-Fenderedoceo: At which point... THEY become the unstream provider
19:54.12edoceoOh - so I have to elevate myself to a Tier1 provider like status?
19:54.37[TK]D-Fenderedoceo: Or have an arrangement directly with that telco, etc
19:54.48[TK]D-Fenderedoceo: basically a waste of time mostly.
19:54.50edoceoHmm - maybe more work than I expected
19:56.07jplankedoceo, whats wrong with XO?
19:56.51edoceoNothing - just exploring how to move up the chain of telephony providing
19:57.20jplanktheres nothing wrong with using XO over AT&T, hell, I'd personally rather XO over AT&T
19:57.38Jabkawonders if he can use his motorola e1000 (standart AT modem) as an FXO (maybe chan_mobile)
19:58.25jplankwell, unless you have to deal with their support, then I'd rather neither :P
19:58.42mitchGaffiganor if you are the caller and the receiver, just make a direct sip connection
20:08.02[TK]D-FendermitchGaffigan: If he's the caller & receiver he should speak to a psychologist.
20:08.22*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
20:08.51mitchGaffiganI suppose that would be an interesting occasion...
20:09.33mitchGaffiganperhaps I should rephrase that to be "If you are in charge of the dialplan of the caller and receiver..."
20:10.16*** join/#asterisk shinao1 (n=shinao1@62.173.46.239)
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20:18.56Gido-Eis there a log facility to she the manager commands?
20:19.09Gido-Es/she/see
20:23.17*** join/#asterisk [T]ank (n=ckwall@206.71.78.158)
20:26.24*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
20:29.47*** join/#asterisk Jabka (n=jabka@89-139-77-49.bb.netvision.net.il)
20:31.10ecretI have asterisk setup on my system and it can take sip calls.  I had hoped to get a single PTSN connection.  Can someone suggest a provider that preferably is instantly usable?
20:32.27pdmmmecret: www.vitelity.com
20:32.37ecretpdmmm: thanks
20:32.46pdmmmwelcome!
20:33.34*** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com)
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20:38.50[T]ankI am trying to learn how to use call files. I have entered in a callfile and had asterisk successful attempt to make a call, but I am running into some errors and could use some assistance. Please see pastebin for details. I can place calls from the T1s just fine. I am currently adding more details to another paste to show that.
20:39.37Kattyohai
20:40.58*** part/#asterisk horvath (n=horvath@74-51-45-109.telnetcommunications.com)
20:42.18[T]ankI take that back... I can dial from the CLI, but when dialing from a softphone, I get the same error as the call file: http://pastebin.com/d37467c3b
20:42.40[T]ankjust noticed my past in the first post was omitted... hang on
20:42.53[T]ankhttp://pastebin.com/d4519feb7
20:43.07[T]ankthose are the details of all my configs and the outcome of attempting a callfile
20:44.07*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
20:48.45*** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
20:49.27Kattyjaytee: poke
20:49.42*** join/#asterisk torr (n=sylar@bzq-79-183-140-138.red.bezeqint.net)
20:49.45torrhello
20:49.49Kattyherro
20:49.55[T]ankso I guess the core issue is that the extension dialing out is failing: http://pastebin.com/d37467c3b what would cause this when I dial from the softphone? When I dial from the CLI it works: http://pastebin.com/d75e002b3
20:50.14jayteeKatty, ????
20:50.38Kattyjaytee: incoming query
20:50.42torrQ: how do I connect asterisk to a regular phone?
20:50.53Kattytorr: you stick a card in there.
20:51.03beektorr: or use an ATA
20:51.03torris there something like for skype phone?
20:51.18torrwhat card? modem?
20:51.59Kattya card that support digital or analog phones.
20:52.21*** join/#asterisk doolph (n=doolph@190.141.68.31)
20:52.30[T]anktorr: do you have a working server that can do sip phone? And now you are ready to tackle analog? It sounds like you might be in over your head
20:53.18Gido-E[T]ank if you see how the questions are formed.
20:53.34torr[T]ank, I have a linux server, without asterisk, and i think to install asterisk, and wonder if I can just exit a line from there to my operator
20:54.21ariel_hello everyone
20:54.27[T]anktorr: over simplified, yes you can. There are many places with the documentation on how to do that.
20:54.32doolphhi ariel_
20:54.50[T]ankGido-E: Are you referring to my questions or torr's?
20:54.54ariel_hello doolph
20:54.59torr[T]ank, I want to know if I can get the hardware for it for $50
20:55.20doolphanyone got any problem with asterisk 1.6? it got disconnected from all ip phones without any reason, but from the server I can ping the phones, once I restart the asterisk all gets back
20:55.31[T]anklook for an ATA device. That will fall in your price range. They are however not as quality as using an FXO/FXS card
20:56.14torrhow will this quality be manifested?
20:56.25[T]ankdifferent for everyone
20:56.39dude7064I want to have my own  calling card-business , and I'm wondering if there is a service/application commercially available to facilitate this ?
20:56.55[T]ankSo, regarding my questions above, is there anyone who may be able to identify what I have screwed up?
20:58.17torr80$ Cisco ATA 188 - VoIP phone adapter
20:58.20[T]ankmy softphone can dial other extensions just fine on the server, but if it tries to place a call to the t1, then it fails with a cause 99. However I can succesfully place calls from the console using the same context and t1 that the softphone is pointing to
20:58.22torrgood
20:58.32[T]ankany ideas why it would do that to me?
20:59.11jblackLooks like verizon is having a problem via alternet. Just in case anyone happens to be on the same provider and is suddenly finding sip broken.
21:00.36jblackdepending on hops, packet loss of 65-90%
21:01.26torris this " Cisco ATA 186" good
21:01.31torr?
21:01.44torrDo I need asterisk for it, or is it standalone?
21:02.10jblackThis is #asterisk. You always need asterisk. ;)
21:03.39torr:/
21:07.11*** join/#asterisk voxter (n=voxter@76.77.95.2)
21:11.32*** join/#asterisk af_ (n=getsmart@88-149-230-241.dynamic.ngi.it)
21:13.20*** join/#asterisk hi365_m (n=hi365@85.130.230.240)
21:13.34Gido-ETank it looks like you are dialing the wrong number
21:15.06*** part/#asterisk Jabka (n=jabka@89-139-77-49.bb.netvision.net.il)
21:15.34af_there any ip phone with an snmp agent?
21:15.45[T]ankGido-E: you mean that the telephone number I am dialing is wrong? Its my cell phone.
21:15.50Gido-E[T]ank how long are the numbers you dial? 11 or 10 digits?
21:16.01[T]ankworks when I dial the same number from the cli.
21:16.08[T]ank1+10 digit number
21:16.22Gido-E11 digits number.
21:18.06Gido-Eyou dont need to chop of the leading digit?
21:18.17[T]ankno. the carrier requires it
21:19.46*** join/#asterisk andalou (n=chatzill@190.233.2.179)
21:20.31*** part/#asterisk andalou (n=chatzill@190.233.2.179)
21:21.58*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:22.31*** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com)
21:23.17brunnerif my voip provider is sending calls to various extensions, how can I force the calls to go to extension s?
21:24.22[TK]D-Fenderbrunner: Got()
21:24.25[TK]D-FenderGoto()
21:24.43brunnerokay, I wasn't sure if there was something I could do in sip.conf
21:24.44brunnerthanks
21:25.04[TK]D-Fenderbrunner: Depends if you're the one telling them what extens to send
21:25.26brunnernope
21:28.07*** join/#asterisk double_cheesburg (n=chatzill@ip68-98-36-177.ph.ph.cox.net)
21:29.18[TK]D-Fenderbrunner: Absolutely sure?  Most people don't even know when they are.
21:29.32brunnerI'm probably wrong, then
21:29.40[TK]D-Fenderbrunner: PB it up
21:32.41brunner[TK]D-Fender: http://pastebin.com/d3aad5d87
21:34.36[TK]D-Fenderbrunner: Yup, its them.  So make a catch-all and Goto "wherever"
21:34.48brunnerokay, thanks
21:35.02brunneris it common for voip providers to do that?
21:35.15brunnerseems like they would just pass the dnis
21:38.46[TK]D-Fenderbrunner: they are dialing in the DID that was dialed on their end which is entirely normal
21:39.29[TK]D-Fenderbrunner: This is supposed to be a GOOD thing.  Why would you order 10 #'s from them only to have them all processed identically?
21:42.59brunnerI wouldn't process them identically.  I'd process them based on the dnis they pass
21:43.13[TK]D-Fenderbruthat IS the extension.
21:43.32brunnerI understand that, but how do things end up in CALLERID(dnis)?  is that PRI-only or something?
21:44.43dude7064when installing asteriskNow,, it's asking for username and password,,, which username should I choose ?
21:44.44[TK]D-Fenderbrunner: I'm not 100% on this, but I believe that because of ANI thats why that split exists.  Multiple ways to pass varying levels of info.  But Exten is the norm IIRC.
21:44.57brunnerI see
21:45.04dude7064I tried admin, but it's saying username/password is incorrect !!!
21:45.13dude7064what's the username I'm supposed to be using ?
21:45.31[TK]D-Fenderdude7064: Try asking in their channel.  Check the topic
21:46.25dude7064there are like 10 people over there !! Nobody answers anybody
21:46.50jplankdid you ask google?
21:47.18jplankI find it hard to believe that the default login/password isn't a google search away
21:48.00dude7064yes I did. It says "admin" is the username
21:48.03[TK]D-Fenderjplank: I got the answer in the FIRST LINK of my own search just now
21:48.05dude7064but it's not working for me.
21:48.10dude7064root is working fine,,
21:48.20dude7064but it's not taking me to the GUI
21:48.34dude7064It's taking me to the command prompt,, how can I view the usernames from there ?
21:49.03[TK]D-Fender"The boot sequence stops at a welcome screen that tells you the default login is "admin," "password." Hit the return key to let it finish booting. When you see the Console Menu it's done. "
21:49.46[TK]D-FenderAnd for FreePBX ... Default username/password is freepbx/fpbx
21:50.30dude7064just tried admin/password ,, still doesn't work
21:50.33awk_rdude7064, i'm offended by your 'Nobody answers anybody' comment
21:50.34brunnerwhy bother with FreePBX?  Just install ubuntu and apt-get install asterisk
21:50.56brunneror debian
21:50.57*** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com)
21:51.14jblackbrunner: That's what I do, but that's not necessarily a flawless solution.
21:51.25bmoracasome people like pretty things like freepbx
21:51.26jayteethis afternoon one of my coworkers was configuring a network printer and accidentally chose the same static IP address that our asterisk server uses. For 5 minutes until he disconnected the printer and reconfigured it's IP address our asterisk server could not respond to network requests. My messages log is full of  messages like chan_sip.c: Maximum retries exceeded on transmission a13245f0-4e1a542a@128.68.116.145 for seqno 101 (Critical Response)
21:51.26jayteeand 4 calls that I know of were dropped at that time. Unfortunately one of those calls was our CEO talking to someone outside.
21:51.27brunnerwhat's wrong with it?
21:51.50jblackcurrent is 1.4.21.2
21:52.07brunnerthen just compile your own
21:52.20[TK]D-Fenderjaytee: maul his punk ass
21:52.23bmoracajaytee: that's when you kill the idiot who statically set the IP at the printer and then show his decapitated corpse how to set up a DHCP reservation
21:52.27awk_rwhats with the *-GUI bashing :-/
21:52.46[8none1]Can anyone give me some insight into debugging this error : "Ring requested on channel 0/3 already in use or previously requested on span 1.  Attempting to renegotiating channel."
21:52.48dude7064any hints please ? what should I do to find out what the username is ? the password I remember it asked to input one,, but it didn't display any usernames !!
21:52.59[8none1]I'm having calls rejected on my PRI because of this.
21:52.59Gido-Eawk_r the asterisk_gui sucks :)
21:53.03[TK]D-Fenderjaytee: Better yet... new chew-toy for Kisa ;)
21:53.16awk_rGido-E, you'll eat those words
21:53.50jaytee[TK]D-Fender, that won't solve the problem that user expectations coming from a Nortel environment is that the phone system almost never, ever goes down and yet in less than 1 week we've had two incidents where calls were dropped. Prior to that we had over 9 months of uninterrupted service with Asterisk in the equation.
21:54.40jayteeand there's only so much you can do with building in redundancy. No way can I make this 100% fail proof
21:54.59[TK]D-Fenderjaytee: Its a NETWORK environment and network screwups = service failures.  Couple this with an incopetant admin and lose the fake surprise
21:57.01jaytee[TK]D-Fender, agreed but then you and I understand the complexity of networks and the potential for failure. Others are not so well informed and have shall we say, "unreasonable expectations". It has my boss almost apoplectic this afternoon. After he went off on me he later apologized and when I told him he didn't need to apologize he shoved 20 bucks in my pocket and said, "Dinner's on me tonight"
21:58.05jblack"Here's free mcdonald's for abuse." How well he expresses regret.
21:58.19jayteeHe's worried that the CEO will hold him accountable and that it will be held against him for going with VOIP/Asterisk despite all the cost savings, advances, convergence, etc. that we gain by it.
21:58.58jblackjaytee: You could put your phone system on a discrete network to avoid that sort of risk.
21:59.04jayteejblack, he can be a pain but he's basically a decent guy for the most part. He just gets too stressed and is worried because the CEO can be a major prick.
21:59.57jblacka switch and a pile of monkey hours.
22:00.25jayteejblack, all the phones are in one vlan and the server is in the primary vlan because it needs to talk to Exchange via sipX. I need to reconfigure to use both NICs in the server, 1 to talk to all the phones and the other as a route to Exchange is what I'm thinking.
22:01.23jayteeputting the entire VOIP system in it's own physical network is impractical and cost prohibitive in our environment.
22:01.24jblackI'm thnking if you go with physical seperation, mr. printer configuration guy can't cause an ip conflict.
22:01.27Kobaz[Mar  9 18:01:08] WARNING[24052]: chan_iax2.c:2169 __attempt_transmit: Max retries exceeded to host 192.168.24.31 on IAX2/2248-300 (type = 6, subclass = 11, ts=6409767, seqno=179)
22:01.44Kobazi keep getting a bazillion of those when i place an iax2 channel on park
22:02.14jblackPerfectly reasonable. So, tell the boss that you could prevent that sort of problem, but it would cost (hold arms out) that much.
22:05.00jblackThen, when _his_ boss gets snooty, he can say "We can solve that type of problem, but it would cost (holds arms out) that much"
22:05.25*** join/#asterisk RoPBX (n=nickserv@200.93.34.175)
22:06.20*** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com)
22:07.28jblackI need to find a good, c++ friendly tcp library for talking to the AMI
22:07.30jayteejblack, hahahaa, that sounds like a good idea!
22:07.43jayteejblack, what about C#?
22:07.47af_c++ and frindly?
22:07.47RoPBXHi All
22:08.02jayteethere's a nice little C# .NET library for that on Sourceforge
22:08.10beekjaytee: A simpler solution is crucifiction of the offending network tech.   That'll be an example to the others...
22:08.14jblackjaytee: Oh, you're serious?
22:08.16jayteeI've played with it in Visual C#
22:08.24RoPBXplease, someone to ask about TDM problem in Asterisk PBX
22:08.35beek~ask
22:08.36jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
22:09.07jblackTo put it kindly, I don't work on that platform.
22:12.32jayteebeek, he's an MCSE, CCNA and makes more money than me. He actually threw himself under the bus when I told everyone we had an outage that lasted 5 minutes and the time frame. "Oh, what's the IP of the server? Yeah, that was me!"
22:12.33beekWow -- that's ballsy.
22:12.33beekOkay, perhaps crucifiction is a bit harsh.
22:12.33beekDon't they teach DHCP in MCSE classes? ;-)
22:12.33jayteeand circumcision of an already circumcised person is cruel and unusual punishment although it's probably not against the law in Indiana.
22:12.33[TK]D-FenderYou're right... mercy killing... back of the head, capo-style
22:12.33dude7064anybody can tell me what's the username is ?
22:12.33RoPBXi have a Rhino tdm card, and the incoming calls get mixed with outgoing calls
22:12.33jayteehahahaaha
22:12.33dude7064I just installed AsteriskNow,, and it's asking for the username
22:12.34[TK]D-Fenderdude7064: WTF is "IT"
22:12.34jblackropbx: That's almost certainly caused by a configuration devised by you.
22:12.34dude7064Centos
22:12.34*** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net)
22:12.34[TK]D-Fenderdude7064: One would think that you'd sign in as ROOT
22:12.34dude7064I did
22:12.34RoPBXthanks jblack, how can I configure it to fix that problem?
22:12.34dude7064but I am not getting any GUI
22:12.39dude7064It is taking me to the command promp
22:12.44dude7064*prompt
22:12.45[TK]D-Fenderdude7064: And how did you you think you would get to a GUI?
22:13.03jblackRoPBX: I'd suggest you look at what contexts you're dropping calls into (both directions), and working up your extensions.conf to separate them according to your deepest, darkest desires.
22:13.27dude7064in the guide for AsteriskNow , it says simply boot the media with the ISO image and follow the instructions till you get the URL
22:13.33drmessano[TK]D-Fender --> c:\win.com
22:13.54[TK]D-Fenderdude7064: its a WEB GUI.  You access it via ANOTHER COMPUTER with a WEB BROWSER
22:13.54dude7064afterwards type this URL and it'll take you to the configuration page with a GUI interactive interface
22:14.30jblacklooks at libclaw-net1
22:14.30dude7064what is the url then ?
22:14.33RoPBXjblack, incoming and outgoing calls are in separates contexts now
22:14.50drmessanoipaddress or hostname
22:14.51jblackropbx: take it from there. don't forget to reload files after you change them.
22:14.51dude7064I'm using VirtualBox to run AsteriskNow
22:14.55[TK]D-Fenderdude7064: the IP of the server :80 or :8080 or something to that effect
22:14.59dude7064and was not presented with any URLs
22:15.07jblackI bet anything with "claw" in the name does a powerfully good job.
22:15.23drmessanojblack: Indeed
22:15.28[TK]D-Fenderdude7064: "presented"?  There is no "miracle announcement protocol" for this.  Its a WEB SERVER, not *MAGIC*
22:15.30drmessanojblack: Just ask Billy Mays
22:15.50jblacklol. "Claw is a C++ Library Absolutely Wonderful"
22:16.03jblackBilly must have a sourceforge account. :P
22:16.29drmessanoAny software that is both a description and a review in the same line is DEFINITELY teh awesum
22:16.30dude7064how can I tell that Asterisk is running after logging in ?
22:16.31*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk)
22:17.03drmessanoThe GUI will tell you
22:17.27jblackdude7064: w/ normal asterisk, ps aux | grep asterisk or  asterisk -r should suffice.
22:17.41jblacknote: running doesn't necessarily mean "working" =)
22:17.50RoPBXjblack, everything works perfect, but there are extrange cases that calls get mixed
22:17.55[TK]D-Fenderjblack: thats a grey area :)
22:18.32jblackwtf? why would a stream library have png/jpg/targa support?
22:18.40jblackphear teh libclaw
22:20.25drmessanoapp_CLAW
22:21.29jblackmaybe I should just make something up in perl or python.
22:21.53drmessanopython + claw would rock.. Thats as cool as a Bear with a rifle on its back
22:22.09[TK]D-Fenderspecifically request sharks with friggen lasers on their heads
22:23.18RoPBXjblack, it seems that asterisk is making a bridge with incoming and outgoing calls , that happens when there are a lot of incoming and outgoing calls at the same time, like asterisk didn't realize that the TDM line is busy
22:23.34drmessano....
22:23.58*** part/#asterisk Khratos (n=khratos@190.166.103.111)
22:24.48jblackropbx: Hmmm. So you think signalling with upstream is getting dropped somehow?
22:24.59*** join/#asterisk mib_5ozpug (i=d936d556@gateway/web/ajax/mibbit.com/x-eea7e9caaea58c3a)
22:26.10RoPBXjblack, i don't know what happens there, but you can even hear the other users dialing, and then you can talk
22:26.27jblackYou could check your switchtype, signalling and channel with your provider.
22:26.31mib_5ozpughow to istal hud pakage on the server
22:26.31mib_5ozpughow to install hud server
22:27.46jblackPerhaps your zaptel/dahdi configuration is wrong.
22:28.00mib_5ozpugi cant open the package page
22:28.04RoPBXi'm using my local phone line
22:28.07jblackpots?
22:28.11mib_5ozpugto install hud
22:28.33jblackw/ an ata, of course.
22:28.55RoPBXwhat parameter can i configure in my zaptel configuration to avoid this?
22:29.03dude7064I have AsteriskNow installed installed in a VirtualBox machine,,
22:29.26RoPBXi set the busydetect=yes, callinprogress=yes, with no change
22:30.28dude7064how can i access the AsteriskNow from my host machine ?
22:30.35dude7064it seems very complicated,
22:30.53mib_5ozpugany help me to install hud
22:31.04RoPBXits a very extrange case, this happens when the calls are incoming/outgoing at the very same time
22:31.05dude7064isn't possible to simply run the GUI from the VirtualBox machine ?
22:31.16Kobazdude7064: what about #asterisknow?
22:31.34brunnerdo I have to register if I'm only pushing outbound calls?
22:31.45Kobazdude7064: we don't know anything about asterisknow in here, so you're not going to get any help
22:31.49[TK]D-Fenderdude7064: what part of WEB SERVER do you not understand?
22:32.04RoPBXjblack: why with an ata???
22:32.13*** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net)
22:32.21mib_5ozpugany body help me to install hud-lite-2 on server
22:32.28jblackropbx: Let's start over. What equipment are you using?
22:32.36RoPBXjblack: i'm making test with 2 GS-2020
22:32.41jblacklol
22:32.44mib_5ozpugok
22:32.58jblack![TK]D-Fender grandstream-rant
22:33.08Kobazdude7064: and, honestly... if you think that accessing the gui interface is complicated... maybe setting up your own phone system is not for you
22:33.31seanmhmib_5ozpug, You could try iSymphony. It's pretty easy to install. www.i9technologies.com/isymphony
22:34.11RoPBXjblack: and one Rhino tdm card, 8 ports, but i have only one local line to test
22:35.03jblackRoPBX: Sorry, no suggestions. I'm not familiar with tdm cards.
22:35.10RoPBXok, thanks
22:35.38[TK]D-Fenderbrunner: Generally no.
22:35.40RoPBXany one familiar with TDM cards? or another channel?
22:36.37drmessanomib_5ozpug: Astassistant is free, and pretty decent
22:37.08jblackI feel like #asterisk is currently in the twilight zone
22:37.29[TK]D-FenderRod.... Rod Serling..... is that you?
22:37.33[TK]D-Fenderstares at the sky
22:38.04jblackIt's either that, or someone's cloned a whole pile of Bizarro's.
22:38.07*** join/#asterisk lanning (n=lanning@173.8.187.197)
22:38.24Dovidevening TK
22:43.01brunner[TK]D-Fender: thanks
22:43.34*** join/#asterisk isamar (n=isamar@server1.dw7.telegate-americas.com)
22:43.41eppigyhello
22:43.52isamarhi folks
22:44.05*** join/#asterisk killown (n=ukendt@unaffiliated/killown)
22:44.07isamaranybody using tor3e+astunicall ?
22:44.18*** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru)
22:44.28*** join/#asterisk `paul (n=kutimoy@121.97.99.151)
22:44.41RoPBXZap/1-1              (None)               Up      Bridged Call(SIP/6001-081feea8
22:44.41RoPBXSIP/6001-081feea8    1-dial@macro-trunkdi Up      Dial(Zap/g1/04141420663)
22:44.53RoPBXsorry
22:44.55isamarneed a hand for that to make zaptel running with astunicall patches + tor3e stuff
22:45.25brunnerWhen I call Asterisk from one SIP provider and Dial() out to another, do my RTP packets ever flow directly from one provider to the other, or do they always go through Asterisk?
22:45.28`paulis it possible for 3 people to do video conferencing.... (even if the other one has no video at all)
22:45.29RoPBXdude7064 did you check file /etc/asterisk/manager.conf ?
22:46.03isamaranybody rathern than coppice knows how to do that ? :-)
22:46.39[TK]D-Fenderbrunner: Depends if you allw reinvites or not
22:47.48brunnerhow does the authentication work if I allow reinvites?
22:48.32[TK]D-Fenderbrunner: After the call is accepted then * tries to reconnect both ends together.  "shouldn't" affect auth.
22:48.41brunnerokay, thanks
22:48.49[TK]D-Fenderbrunner: Would if you tired a raw Transfer() however
22:48.51[TK]D-Fendertried*
22:56.13*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
23:07.15*** join/#asterisk moy (n=chatzill@74.12.124.89)
23:07.35isamar.
23:11.55*** join/#asterisk joako (n=joako@opensuse/member/joak0)
23:12.10RobHI know this isnt the best place to ask, that being said.  Has anyone used a Polycom Soundstation IP 4000 with asterisk?  If so, did you experience any horrible echo?  (I have enabled echo cancellation and it does nothing that I can tell.)
23:13.10jayteedrmessano, you around?
23:14.00isamaranybody using mfc/r2 ??
23:14.24moyisamar: the best way to start is just asking, what's your question?
23:14.34jayteeRobH, I haven't used their IP model but I've used a Polycom Soundstation analog plugged into line off of an ATA without any echo issues unless I'm talking on my cell phone over the channel while in the same room.
23:14.55RobHdamn
23:15.02RobHI have horrible echo
23:15.02RobH=[
23:15.35RobHI am messing around with the thresholds now.  I have little to no idea what they do, but they are part of the echo cancellation.  Worst case I set them wrong, cannot hear anything, and have to reboot the phone ;]
23:15.49RobHGotta love polycom and the two minute boot process.
23:16.14Kobaztotally
23:18.04isamarmoy: I am using tor3e board and I am trying to use astunicall with that...
23:18.26isamarmoy: my doubt is if zaptel-tor3e must be patched also for unicall or not...
23:19.04moyisamar: not that I know of, zaptel does not need any patch at all for R2 signaling
23:20.27k-manRobH: no idea if this helps, but my billion ADSL modem's viop ATA thingo also suffered from bad echo, and the solution there was to reduce the mic and speaker gain settings in the modem/ATA
23:20.44k-manRobH: and that solved the problem on the billion
23:20.52isamarmoy: ok.. cool.. dude.. that's what I needed to know
23:20.58isamarmoy: thanks a lot
23:21.00RobHk-man: I will check that out, thanks!
23:21.23k-manRobH: hope it helps - i wish I uderstood why these devices get echo? its bizzare
23:23.08RobHthe worst part is my soundpoint 501 sounds better on handsfree than my expensive soundstation
23:23.19RobHbut it doesnt have enough pickup to use throughout the conference room.
23:24.12joakoI know this doesn't help much, but I hear echo some times from IP501 <-> IP501 if one is on speakerphone
23:27.59voxteris there a way to disable sip uri dialing on polycom phones?
23:28.12joakovoxter: Yes... look in the config files
23:28.29voxterjoako: im looking. they're pretty friggin big, and searching for URI has turned up nothing so far. Hmm. maybe url.
23:28.38voxterfound it!
23:28.38voxter:)
23:29.24*** join/#asterisk jchonig (n=jch@firewall.honig.net)
23:30.41`paulis there a way to automatically just mute the call (disable sound) i just want to use asterisk as video conference server
23:31.28[TK]D-Fender`paul: Disallow audio codecs
23:31.44DovidTK: wouldnt that cause a media negotiation error ?
23:32.52isamarmoy: I am getting undefined symbol error for get_supervisory_tone_set when starting up asterisk with astunicall..
23:33.03isamarmoy: any thoughts?
23:34.39*** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net)
23:34.43jchonigIf I have abbreviations for numbers in dialplan (say 9nnnn expands to NXXNX9nnnn for a remote office ext) is there a way to tell the calling SIP phone to display the expanded number (and even update it's redial list)?
23:36.10*** join/#asterisk eldorel (n=eldorel@ip68-108-238-144.br.br.cox.net)
23:37.27eldorelHello everyone. Quick q: is System() depreciated? I can't seem to get it to do anything (asterisk 1.4.21 on ubuntu 8.10 server)
23:39.26Dovidworks for me
23:39.33Dovidwhat are you trying to do ?
23:40.16eldoreli'm recording an audio file into /tmp and then trying to copy it to the aterisk-sounds default after the user saves it.
23:40.42Dovidru sure asterisk has permission to write there ?
23:40.46[TK]D-Fendereldorel: Consider the PRIVILEGES the * user has <-------
23:40.58Dovidtry looking @ the full log and see if you see any erros there
23:41.25*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)
23:42.15eldorelyep, chowned the directory after it didn't work the first time just to be sure. and i've been watching the cli with verbose=15. it seemd to execute fine, but nothing happens....
23:43.17*** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br)
23:43.35Doviddo u have permission to write there or is the file read only ?
23:43.44Dovidmost likely a persnissions issue
23:44.17*** join/#asterisk zenfox (n=steven@c-98-213-240-12.hsd1.il.comcast.net)
23:44.18Dovidi am too tired. cant even spell correclty. night ev1
23:44.24eldorelthe /usr/share/asterisk/sounds file is currently owned by user asterisk
23:44.43eldorelnight, thanks anyway
23:48.43*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:52.42*** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar)

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