IRC log for #asterisk on 20090713

00:03.47[TK]D-Fenderkmem: looks like
00:05.23kmemgracias
00:09.40*** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au)
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01:02.11dshapat a given moment in a call, i'd like to determine if there is any audio on the channel and see if it continues for at least 1 second and then do something.  is there a dialplan app that will serve this purpose?
01:03.42drfreezeHello
01:03.44dshaphi
01:04.20drfreezeI'm building a phone server with a channel-12 and using a TE12(1|2) card.
01:04.40[TK]D-Fenderdshap: "any given moment" as in "triggered to detect, or always checking?
01:04.40drfreezeAnyone have a recommendation for a small form factor box/cpu to use?
01:04.56dshap[TK]D-Fender: triggered to detect
01:05.21[TK]D-Fenderdshap: Chanspy & silencedetect
01:05.24dshapi know exactly at which point during the call (seconds into the call) that i want to determine if there is talking on the channel
01:05.33[TK]D-Fenderbackgrounddetect or whatever its called
01:06.05dshapbackgrounddetect will detect sound and then go to an extension if it gets silence
01:06.12dshapafter the sound
01:06.20efortpastebin.ca/1493202 - how can I get incomming calls to route to my sip phone registered on extension 1000?  (havn't used * in a while and I'm a bit stuck...)
01:06.23dshapbut in my case i'm trying to detect sound that isn't going to be silenced anytime soon
01:06.48dshapdo u see what im saying
01:07.43dshapi'm doing this for my project where i'm trying to sendDTMF to an IVR
01:07.59dshapsometimes, for whatever reason, a "connection" isn't established and the IVR can't see my DTMF
01:08.05dshapso it keeps talking instead of moving to the desired menu
01:08.15dshapif that's the case, i want to detect that it is talking (right after i send the DTMF)
01:08.21dshapand then hangup and place another call
01:08.26dshaphoping that the "connection" will work
01:08.39dshapit works like 75% of the time for some reason but i need it to be more robust than that
01:10.24dshapand i've pretty much determined that the DTMF connection thing is something i can't solve
01:10.37dshapbecause when i call the system with Skype it only works 75% of the time as well
01:10.52[TK]D-Fenderefort: context=incomming_calls <--- start with SPELL CHECKING your configs for clear errors
01:10.52dshap25% of the time i'm pressing touch tones on skype and the IVR system is acting as if it can't hear me
01:11.20*** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net)
01:13.36efortthanks, been banging my head on that for a couple hours
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01:32.43*** join/#asterisk Loki (i=loki@unaffiliated/loki)
01:33.04LokiHey, is there any way to setup a SIP user/extention to be able mutiple ports?
01:33.48Loki5060&20051 and 5060,5060 in the port section didn't work
01:35.34[TK]D-FenderLoki: No,
01:35.55LokiJust add two diff extenisons?
01:35.55[TK]D-FenderLoki: You can set up multiple PEER entries however and dial them simultaneously
01:37.44Loki[TK]D-Fender: I mean, allow a user to use a port for incoming SIP clients, ie ekiga, and other soft phones
01:38.43[TK]D-FenderLoki: A call out to 2 destinations is just that.  2 separate calls
01:39.42LokiI am not looking for two seperate calls, I am looking to allow a user to connect via port 5060, or port 20051 when on 127.0.0.1 on a softphone
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01:40.29Lokifor a local user, not a trunk
01:40.36[TK]D-FenderLoki: the source port of a call FROM a device is irrelevant.  The port only matter when calling out to the deice
01:42.00LokiRegistering with the server, and maintaining a connection is on the allowed port in the extenison though? If the port doesn't match, then you can't connect on it for authorizeation, and call passing right?
01:42.55LokiI can't run a softphone on the same server as my PBX unless it is on a differnt port, but I want to be able to use the same exention on desktops to be able to connect to my PBX
01:42.58[TK]D-FenderLoki: There is no such thing as a "maintained connection".  * is SIP UDP and is stateless
01:43.14LokiBut, I mean for sending of user/pass
01:43.18Lokiand dialing of calls
01:43.21Lokiand passing of calls
01:43.30[TK]D-FenderLoki: Run your desktop phone on the same different port then.
01:43.56Lokii am asking for a way to allow mutiple ports in the config so I can allow more then one port
01:44.14[TK]D-FenderLoki: What part of "NO" was misunderstood?
01:44.44[TK]D-FenderLoki: Plenty of ways to work around this, but no direct way to do it
01:45.14LokiOkay. I was just making sure that we were on the same page because I know I can go round and round some times
01:45.30[TK]D-Fender[21:35]<[TK]D-Fender>Loki: No,
01:45.34[TK]D-FenderLoki: That said it all.
01:46.04LokiOkay, Guess I will just have to have mutiple users
01:46.05[TK]D-FenderLoki: So go make multiple peer entries and dial simultaneously
01:47.09LokiThanks for the info [TK]D-Fender sorry if I annoyed ya
01:48.05[TK]D-FenderLoki: Well when you ask a direct question and I give a direct answer you might want to take it that I meant what I said :)
01:48.28Loki[TK]D-Fender: JUst wanted to make swure you understood, I can be confusing at times
01:48.48Loki:p
01:49.13[TK]D-FenderLoki: Congratulations on beating the curve on this one then.
01:49.23LokiJust wanted to make sure*
02:06.05*** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au)
02:06.10Rob3Rtguys
02:06.26Rob3Rtive attached soft pbx running on port 5061, to aster on 5060
02:07.14Rob3Rtthe extensions to aster are 1000 and 2000, 100 is a softphone on the 5061 pbx. 1000 and 2000 are registered with aster.
02:08.37Rob3RtI can call into aster, which calls 1000 at the 5061 port, then goes to the softphone
02:09.01Rob3Rtthe problem is i cant call out, from 100, to 1000, to aster, to voip service.
02:09.19Rob3RtIs there a basic routing plan someone could enlighten me about ?>
02:13.23[TK]D-FenderRob3Rt: Can't call out?  What have you done?
02:14.05WindowsUserpastebin the asterisk dialplan
02:14.12Rob3RtI don't know, k brb.
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02:18.34Rob3RtDial Plan and Sip settings, http://pastebin.com/m78b54ca3
02:18.49Rob3Rtits nowhere near right, surely :p
02:18.51phunyguyhi
02:20.44[TK]D-FenderrobYou have extens to dial out already... your phones simply don't have ACCESS to the context those extren's are in
02:21.58*** join/#asterisk CRCinAU (n=CRCinAU@203.56.246.91.crc.id.au)
02:22.07CRCinAUwondering if I can pick some peoples brains here.
02:22.24CRCinAUthe error message that gets sent to an incoming number where no extension matches
02:22.27[TK]D-FenderCRCinAU: Brains are int he second bucket to the right of the door
02:22.29CRCinAUI believe it's a 404.
02:22.44CRCinAUhow can I progmatically return a 404 for selected calls? :)
02:23.04[TK]D-FenderCRCinAU: You can't.  404 is only if there is no match in your dialplan.
02:23.20CRCinAUso I can't intercept it in extensions.conf?
02:23.37[TK]D-FenderCRCinAU: Huh?
02:23.45[TK]D-FenderCRCinAU: Yous aid SEND a 404 as an answer back.
02:23.50CRCinAUsorry - probably not explaining myself 100%
02:23.52[TK]D-FenderCRCinAU: where does DETECT come in?
02:24.18CRCinAUa call comes in via VoIP. Hits the associated extension in extensions.conf in my [incoming] section
02:24.27CRCinAUI want to run it through an AGI script to check the CID
02:24.38[TK]D-FenderCRCinAU: "NOT POSSIBLE"
02:24.40CRCinAUthen if I don't like that person, pass back a 404 piss off.
02:24.45CRCinAUbugger.
02:24.47CRCinAUthat sucks :(
02:25.33CRCinAUthat would allow much better call controll :)
02:25.44CRCinAUie no caller ID = not connected error to the caller ;)
02:26.42CRCinAUso does that mean that by the time the call hits step 1 in extensions.conf, there is already a reply for RINGING sent back to the caller?
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02:30.22Rob3Rt[TK]D-Fender, thanks for the reply :)
02:30.37Rob3Rt[TK]D-Fender, what *should* I be setting this to?
02:35.33CRCinAU[TK]D-Fender: so other than writing a SIP control proxy to filter stuff out, is there any other way to do what I'm thinking?
02:35.35Rob3Rtmy phones don't have access to those contexts... HMM.
02:37.42[TK]D-FenderCRCinAU: pretty much "no"
02:37.56[TK]D-FenderRob3Rt: Go move the extens around
02:38.16[TK]D-FenderRob3Rt: Or learn how to use INCLUDE <-  these are dialplan basics
02:38.22Rob3Rtahh
02:38.26Rob3Rtbeen looking at that .
02:38.33Rob3RtThank you.
02:44.41ricko73[TK]D-Fender: does libpri depend in any way on zaptel or dahdi?
02:45.16[TK]D-Fenderricko73: no
02:45.38ricko73Just confirmed that with ldd
02:45.48ricko73should have done that first.  sorry for the hassle
02:51.04*** join/#asterisk juanIMP (n=juan@201.244.248.183)
03:06.18kb3ienIs there any such thing as a truly global variable in asterisk's scripting language?
03:07.53[TK]D-Fenderkb3ien: When is a global not a global?
03:08.18[TK]D-Fenderkb3ien: If you're worried about persistence between * restarts use AstDB <-
03:09.54Rob3Rt[TK]D-Fender, i included the other dial plans, now im getting "wrong password for INVITE" when I try to call out from my DID, via asterisk
03:10.03Rob3RtThanks for the previous help too :)
03:23.13Rob3Rt[TK]D-Fender, could you perhaps show me a demonstration of how my dial plans should be, i have included others, i must have syntax wrong .
03:24.49[TK]D-FenderRob3Rt: "Wrong password" has nothing to do with your dialplan.  It has to do with your SIP PEERS
03:26.07Rob3Rt[TK]D-Fender, the peers are registered, does this mean i should change the registration passwords to suit my outgoing voip provider ?
03:26.33[TK]D-FenderRob3Rt: Regsitration has nothing to do with authing calls
03:31.17xpot-mobileQuestion:  is it possible to break out of queue to voicemail (or other dialplan option) if 0 (or other digit) is pressed?
03:32.05Rob3Rt[TK]D-Fender, so I gather
03:32.14Rob3Rthow am i supposed to auth the call :p
03:32.32Rob3Rtive been googling but hard to know what too look for unless someone tells me
03:34.02[TK]D-FenderxpotYes.  Go read the sample config.  It spells it out blatantly
03:34.23[TK]D-FenderRob3Rt: Your SIP peer info is wrong.  authname, passs, etc
03:37.52Rob3Rt[TK]D-Fender, wait, the peer is registered to asterisk using diff creds to my outgoing voip provider, should I use the same creds as my outgoing provider for the extensions?
03:38.28[TK]D-FenderRob3Rt: Your peer has NOTHING to do with the act of registration.
03:38.47Rob3Rtby peer you mean extension right ?
03:38.54[TK]D-FenderNO
03:39.03[TK]D-Fender[sip.confsectionhere]
03:39.11Rob3Rtheh
03:39.16[TK]D-Fenderextension = extensions.conf
03:39.20Rob3Rtyep
03:39.26[TK]D-Fendersip peer = section in sip.conf
03:39.34Rob3Rtgot it
03:39.34Rob3Rtmy bad, yeah
03:39.35Rob3Rtsip.conf
03:39.42Rob3Rtso my extensions 1000 is failing to auth an invite.
03:39.59Rob3Rtthat extension is authed and registered to my asterisk.
03:40.12Rob3Rtso i just dont know why its not authenticating
03:40.15Rob3Rtthats all
03:40.27xpot-mobile[TK]D-Fender: I read the sample config, it mentions the "context=", I will verify that I already did that correctly...
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03:43.21WindowsUserRob3Rt: whats failing? calling 1000@asteriskbox or 1000@thepeer
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03:47.01a2_hello
03:48.05a2_I want to create my own phone system, i have a pretty big house. I want to do it all manually with asterisk and make it VOIP
03:49.42xpot-mobile[TK]D-Fender: I verified that in queues.conf my queue contains 'context=breakout' --> [breakout] 0,1,<action> --> when I press '0' while on hold in queue the cli doesn't even acknowledge a key press.  Perhaps I misunderstand the context= in queues.conf ??
03:49.48a2_can someone please point me to some resources to help me?
03:50.39*** part/#asterisk juanIMP (n=juan@201.244.248.183)
03:50.54WindowsUserlike
03:50.56WindowsUser~book
03:50.57infobot[~book] 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://astbook.asteriskdocs.org or see ~buybook
03:51.15[TK]D-Fenderxpot-mobile: PASTEBIN <-
03:51.22a2_thanks windowsuser
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03:51.51a2_i know im going to need some special hardware, asterisk, but im not sure what else, i will read the book, i hope it can answer my questions
03:53.12[TK]D-Fendera2_: Not so much.  Describe the actual phones you intend to use and what you intend to use of PSTN conenctivity
03:54.25a2_[TK]D-Fender: I want to have 5-10 phones total, regular phones nothing fancy, and be able to call outside with each one at the same time, im not sure about connectivity
03:55.09[TK]D-Fendera2_: "with each other"  Meanin?
03:55.18WindowsUserhahaha
03:55.36WindowsUsercall from one phone to another
03:55.44[TK]D-Fendera2_: As in just like a regular home circuit where everyone can join in and not act independantly?
03:55.44a2_no i meant
03:56.00a2_No, i want each phone to be independent
03:56.14a2_and if possible have its own #
03:56.30WindowsUser10 different dids for a house?
03:56.34a2_yes
03:56.36a2_Would i have to buy those #'s?
03:56.45WindowsUser...yes
03:56.57a2_How much would that cost?
03:57.00WindowsUseror have a did and then have it ask for an extension
03:57.11[TK]D-Fendera2_: http://www.voipsupply.com/linksys-spa8000-g1
03:57.21xpot-mobile[TK]D-Fender: http://pastebin.com/d743a9564
03:57.31[TK]D-Fendera2_: Price for PSTN service depends on the plan and the provider
03:57.34WindowsUsera2_: depends on how you get service
03:57.46WindowsUserasterisk works with PRI, analog connections, and SIP
03:58.39[TK]D-Fenderxpot-mobile: Looks fine.  make sure your changes have applied, call into the queue and note that on several * versions DTMF will interrupts a queue message, but not break out.  Press repeatedly to verify yours
03:58.56[TK]D-FenderWindowsUser: And apparently a whole lot more
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03:59.49WindowsUserwell if i say you theres trees and girls outside, it doesn't mean there are no icecream trucks or terrorists etc etc etc
04:00.54WindowsUserbut i get what you mean
04:01.40xpot-mobile[TK]D-Fender: running 1.4.25.1 ... I press '0' multiple times and the cli acts as though nothing happened, it doesn't seem to even try my defined context.
04:01.41[TK]D-FenderWindowsUser: No, but I could believe there is a point buried deep in there somewhere ;)
04:02.07[TK]D-Fenderxpot-mobile: make sure your config is applied
04:02.20[TK]D-Fenderxpot-mobile: because the option works..
04:02.25a2_Ok guys, i kind of lied. I have a specific ceneraio i cant describe, but what i need is a shitload of disposable phone numbers, im a programmer so the sky is the limit, how do i accomplish this?
04:02.58[TK]D-Fendera2_: Shop around
04:03.06[TK]D-Fender~itsplist-us
04:03.07infoboti heard itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
04:03.09[TK]D-Fender^^^^^^
04:03.23a2_Thank you :)
04:04.13a2_You guys will see me around
04:04.14a2_peace
04:05.12WindowsUser"the sky is the limit" :-/
04:06.23xpot-mobile[TK]D-Fender: cli> reload (all reloads) --> same problem.  Any other suggestions? (FYI: my defined periodic-announce doesn't work either, my thoughts are that its not related)
04:07.01[TK]D-Fenderxpot-mobile: pastebin your failed attempt
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04:17.51xpot-mobile[TK]D-Fender: http://pastebin.com/d37160254  I attempted to press '0' after "-- Started music on hold, class 'default', on IAX2/voip-5006"
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04:24.30[TK]D-Fenderxpot-mobile: How long did you sit in queue for?
04:25.32WindowsUserdont you have to press 0 before moh?
04:25.52[TK]D-FenderNo
04:27.39kb3iensorry got called away. was there such a thing as a global variable in asterisk that spans all channels?
04:27.54xpot-mobile[TK]D-Fender: about 1 min
04:29.33xpot-mobilekb3ien: use astdb
04:29.35[TK]D-Fenderxpot-mobile: Try from another device
04:29.49[TK]D-Fenderkb3ien: Globals DO span all channels.
04:29.57[TK]D-Fenderkb3ien: and So does AstDB
04:31.25kb3ienAH, basically i want to ensure that no 2 threads are accessing the same resource, a traditional bakery algorithm implies enumerated threads, i was looking for something to use i lieu of that. myresource=nextfreeresource++ kinda thing.
04:34.46xpot-mobile[TK]D-Fender: my tries have been from a PSTN calling into the server and calling from an X-Lite phone to the queue extension with same results.
04:35.11kb3iennot sure how threads work, is it possible that 2 threads can execute the same command concurently?
04:35.42[TK]D-Fenderxpot-mobile: Verify that the dialplan is indeed there and your queue config is as well
04:36.20kb3ienthis probably means i'll wind up using some AGI perl stuff, eh?
04:38.32xpot-mobilekb3ien: I had to do something similar in the past and ended up using a .net app that connected to asterisk via manager, I am sure there are many other options
04:41.03kb3ienhrm. its a bit beyond the scope of most pbx's. But i'm dealing with faxes that are better send to iaxmodems and transited accross less-stable links as serial data...
04:46.00[TK]D-Fenderkb3ien: AstDB has locking mechanisms against concurrency issues
04:46.36[TK]D-Fenderkb3ien: As do standard global vars
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04:51.02xpot-mobile[TK]D-Fender: everything looks good.  I am going to build out a cleaner development environment and test it again.  thanks for your help.
04:51.57*** join/#asterisk riyadc (n=riyadc@test20.citech-bd.com)
04:52.08riyadchello everyone
04:52.56riyadcis there a huge difference between asterisk gui and freepbx?
04:56.45[TK]D-Fenderriyadc: Yes.  FreePBX covers just about all the different aspects of configuring your system but isn't as flexible, yet is much more stable & mature
04:57.15riyadcits a bad idea to run both right?
04:57.29riyadci just setup a asterisk box using asterisknow
04:57.37riyadcand trying to figure out that gui i wanna use
05:03.20[TK]D-Fenderriyadc: Completely incompatible
05:14.34[TK]D-Fendercheckout time, later all
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05:26.59riyadchey does asterisknow come with freepbx?
05:27.11riyadcby default?
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05:40.28ramindiariyadc:  have you looked at this "http://www.freepbx.org/trac/wiki/AsteriskNow"
05:41.26riyadcthanks
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06:21.54trnzmetago go exetel, shit all isp assholes
06:25.42R0b3Rthuh?
06:26.15R0b3Rti reset my wireless dude
06:26.18R0b3Rt;\
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06:35.43R0b3Rtlooks like ill have to phase out asterisk
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06:44.50riyadcanyone here use astbill or a2billing?
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06:47.35R0b3Rthelp - or the asterisk bunney gets it! http://pastebin.com/m78b54ca3
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07:06.03milouxdundi lookups are cached even if you do module reload pbx_dundi.se ?
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07:14.44Rob3Rtback - did i miss an answer
07:16.55kaldemarcontext=exetel-oubound
07:17.15kaldemarlooks like a typo. what's the question btw?
07:20.19kaldemaryour configs and comments don't make much sense.
07:21.46Rob3Rtkaldemar, thanks for coming to the rescue
07:21.49Rob3Rtwell,
07:22.20Rob3Rtim running 3cx server on port 5061, and aster on 5060
07:22.35Rob3Rt5061 pbx registers to aster with extensions 1000 and 2000
07:22.50Rob3Rtaster registers my exetel voip DID so i can make calls
07:22.55Rob3Rtwhat im trying to do
07:23.23Rob3Rtis make a call, from a 3cx extension, through to asterisk via 1000, and out into the world
07:23.30kaldemarregistering has little to do with making calls.
07:23.33Rob3Rtbut I cant seem to do it for 2 days
07:23.37Rob3Rtok.
07:24.05Rob3Rti can dial the number from a sipphone on 3cx, it gets to asterisk via 1000
07:24.26Rob3Rtit then has issues getting the call out to the world via my exetel voip service
07:24.30Rob3Rtthats the issue man.
07:25.24Rob3Rtif u can sort it, ill buy you a beer over paypal :p
07:27.03rob0What is 3cx?
07:27.03Rob3Rtwww.3cx.com
07:27.06Rob3Rtproprietary pbx
07:27.46kaldemarshow a cli output of a failed call, with sip debug
07:28.10Rob3Rtwin32 doesnt allow copying of the window, and fails to log the debug
07:29.41rob0Why not simplify, take one or the other PBX out of the problem?
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07:34.04Rob3Rtnot the overall goal here
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07:47.25salviaHello, i'm having one way audio issues when initiating calls, when i receive calls everything is fine. I'm using sip.
07:47.29salviacan anyone assist?
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08:18.55Rob3Rt$10 bucks US going for the dude who can sort my stupid PBX routing issue.
08:19.07Rob3Rtconfigs posted earlier, info on the scroll down
08:26.24salviaehe
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08:30.08mbrevdaRob3Rt: you want outbound calls to hit an extension??
08:48.16kaldemarRob3Rt: you're still not showing a problem to solve.
08:49.14kaldemarRob3Rt: i don't care about the $10, but you won't get help if you don't tell what the issue is.
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09:10.00tzafrir_laptopRob3Rt, what do you mean by "win32"? do you run Asterisk on windows?
09:10.18tzafrir_laptophmm, /me had some backlog
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09:44.16Rob3Rthello
09:44.27Rob3Rthi guys sorry, off with the little ones there for a while
09:44.45Rob3Rtumm, yeah the outbound calls just need to make it to the actual target
09:45.01Rob3Rti cnat show logs guys because win32 aster sucks and wont log sip debug
09:45.56tzafrir_laptopRob3Rt, what version of Asteirsk is it? 1.2?
09:46.13Rob3Rtum
09:46.33Rob3Rt0.60 apprently
09:46.34tzafrir_laptopIIRC you should be able to build later versions of Asterisk on win32 with cygwin
09:46.50tzafrir_laptopno. That's the version of astwin32
09:47.41Rob3Rtah
09:48.02Rob3Rtthis version seems fine, just my aster-skills suck like a black-hole
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09:55.55Rob3Rtno go guys ?
09:56.32Rob3RtI cant seem to figure out how to call out from, 100->1000->aster->exetel
09:56.40Rob3Rtand then switch the call to 2000
09:56.44Rob3Rt:(
09:56.55Rob3Rt2 days, its long enough to consider pulling down asterisk
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10:01.20tzafrir_laptopwell, you use a version that is practically unmaintained
10:01.51tzafrir_laptopThere have been quite a few fixes even in the 1.2 branch since that one was built, I believe
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10:23.11Anyhi
10:25.37AnyI have a small problem with asterisk and isdn (pri). how can I find out, if a call to a specific number got redirected (for example a call to a mobile phone and then redirected to a voicemailbox)? I tried looking at the reason code and at different channel variables, but without any luck
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10:30.57angryuserAny, have you tried to look at dialplan ?
10:31.46AnyI'm using the asterisk manager api to dial out via CAPI/ISDNx/number
10:31.57angryuserand this " how can I find out, if a call to a specific number got redirected" is all but not english
10:32.18Any:)
10:32.22angryuserTry to explain better
10:33.51AnyI call a mobile phone number via CAPI (over a isdn pri line) and I want to detect if the call gets redirected to the voicemailbox of the mobile phone user (i.e. if he turned off his phone)
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12:01.28lftsyHello all, do you know when Asterisk 1.4.26 will be released? Thanks
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12:07.45a2_wheres the best place to buy BLOCKS of desposable DID's?
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12:15.08Dovida2_: depends on what u need them for ;)
12:15.30a2_a2_: Why would that matter?
12:15.46Dovidic an give numbers but I don't want "funny things" happening with them ;)
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12:16.00a2_please explain "funny things"
12:16.14a2_Pm me if you would like
12:16.18Dovidwhen you say blocks - want say 305-XXX-0100-305-XXX-0200 or just a bunch of numbers
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12:37.59kombino such command "misdn show" <- that allways used to be there, what might it be?
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12:45.50falzmorning. trying to figure out if there's a way to set up a special extention where a user can dial it to grab a call waitining in a queue without actually being a member of that queue
12:46.29voipheroeskombi, try load misnd
12:46.48voipheroesI mean reload the module
12:48.06kombivoipheroes: from cli?
12:48.21voipheroesyea
12:48.30voipheroesi had similar problem once with a ISDN card
12:48.33kombiwhat's the syntax again...?
12:48.46voipheroesbut updating asterisk fixed the problem
12:48.50voipheroesi don't remember exactly
12:48.55voipheroesunload <modname>
12:49.02voipheroesthen load (or reload) misnd
12:49.07voipheroesmisdn*
12:49.35voipheroescan be chan_misdn even
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12:50.47kombivoipheroes: load chan_misdn! thanks! now it says Unable to initialize mISDN though...
12:50.58[TK]D-Fenderfalz: nope.
12:51.49voipheroesKombi: then this is now another error. Verify your config file, if the line is connected, etc
12:52.04voipheroesmaybe you even can turn A* debuging to an higher level to have more info
12:52.11voipheroesasterisk -rvvvvvvvvvvvvvvvvvvvvvv
12:52.27kombivoipheroes: right..
12:52.33falz[TK]D-Fender: hmm. the closest I can find is a dynamic queue with sql backend where through some other method the user would temporarily make themselves a queue member, but that's not quite what they want either
12:53.43[TK]D-Fenderfalz: AgentLogin sounds a lot easier.
12:55.49kombivoipheroes: not more output.. weired.. It's a digium card and it worked flawlessly so far
12:55.51falz[TK]D-Fender: they alwso want to streamline that to somehow have a 1 button auto login. if that gets working, then yes that way would be simpler
12:56.48[TK]D-Fenderfalz: Easily doable
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12:58.56falz[TK]D-Fender: depending on phone
12:59.09falzIve never touched polycom before, that's where these guys want it done
12:59.17[TK]D-Fenderfalz: Any phone with a "speed dial" type button will do
12:59.30[TK]D-Fenderfalz: Excellent choice
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13:02.40falz[TK]D-Fender: ah, docs for this phone do seem to list macros, as well as associating them with soft keys
13:02.40Rob3Rtwelcome back
13:02.52voipheroeskombi, are you using dahdi ?
13:02.57Rob3Rtis there a way i can debug to a file using asterisk win32?
13:03.25[TK]D-Fenderfalz: Leave a line-key unallocated to a registration and your Directory will spill over onto it in "Speed Dial" # order
13:03.48[TK]D-FenderRob3Rt: Expect little to no support for it anywhere, and certainly none here.
13:04.08[TK]D-FenderRob3Rt: mere mention of it typicaly induces fits of laughter
13:04.15Rob3Rtits the same thing, just recompiled for windows lol
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13:05.27kombiis there a digium hardware centered irc channel?
13:06.16stixCan anyone tell me where I can find a how-to on creating a php-page which can dial a number via the asterisk server?
13:06.24tzafrir_laptopRob3Rt, if so, just rebuild a newer version
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13:07.12[TK]D-Fenderkombi: Nope.
13:07.24[TK]D-Fenderkombi: This is pretty much it.
13:07.48[TK]D-Fenderstix: Go read : THE BOOK
13:07.49[TK]D-Fender~book
13:07.50infobot[~book] 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://astbook.asteriskdocs.org or see ~buybook
13:07.51[TK]D-Fender^^^^^
13:08.09[TK]D-Fenderstix: Pertinent sections are on "call files" and "AMI Originate"
13:08.26stixcool, thank you
13:08.36stixI even have a hard copy of it here :)
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13:10.49lftsyrtautoclear=yes in sip.conf is purging SIP Realtime peer on default timeout, is there a way to remove sip users in the same time?
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13:12.12falz[TK]D-Fender: that's a function of the polycom phones?
13:12.23falz[TK]D-Fender: and the speed dial can be anything, with pauses if needed?
13:13.01zeeeshcan i assign multiple host name in a single peer? like host=dynamic, host=dash.dash.com, host=daash.daash.com ?
13:13.07[TK]D-Fenderfalz: No, single # to dial.  the rest can be done in dialplan.
13:13.23[TK]D-Fenderfalz: This does not require a special phone.
13:13.37[TK]D-Fenderzeeesh: No
13:13.41falzyea, I wasnt sure if there was something special about a polycom that would present a menu and let them choose 0-9 to auto do something, from the screen
13:13.54falzinstead of listening to prompts, which is far less streamlined and what they're trying to avoid
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13:14.31stix[TK]D-Fender: I cannot find those sections in table of contents. Do you know where they are more specifically?
13:14.44[TK]D-Fenderstix: "Find"
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13:15.14stixyes it works great on the hardcopy
13:15.17mrbob009can i ask a question about pbxes.org ?
13:15.36mrbob009the virtual asterisk host..
13:15.38[TK]D-Fendermrbob009: Questions are free, answers are $4.95/min
13:15.40[TK]D-Fender:p
13:16.13mrbob009compared to 1euro/pm official support, expensive :P
13:16.21eppigyTRABAJO
13:17.07mrbob009i registered my sip (voipbuster) on the site, added a thrunk - and created a outbound-route (only, i have no inbound number did for this voipbuster)
13:17.30mrbob009i can do the echo test, by calling *43, that works fine - but calls are never established
13:17.48mrbob009i head some english mumbling, and then '... plz try again later'
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13:18.03jayteeNO TRABAJO!
13:18.11eppigyD:
13:18.24mrbob009i tried formatting the nr im calling differently (im dutch ..) tried 0031<netcode><nr> +31 etc etc
13:18.27mrbob009any ideas?
13:18.31jaytee:D
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13:18.35[TK]D-Fendermrbob009: they are managing their entire service w/ their own GUI and crap.  It is NOT supported here.
13:18.42[TK]D-Fendermrbob009: Go ask them for support
13:18.59mrbob009[TK]D-Fender: hmmmmmm, well
13:19.12mrbob009i was hoping people here would know a bit more about it...
13:19.28mrbob009i might be making a config error
13:19.44[TK]D-Fendermrbob009: It isn't * up-front, or a bolt-on taht is even freely released
13:19.52[TK]D-Fendermrbob009: It is a dead-end service
13:20.18rbdhi guys...with meetme...what does "admin mode" (a) do? I see some parts of MeetMeAdmin() can function differently for admin users..but besides that...
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13:29.12mrbob009[TK]D-Fender: not everything in this world can be free
13:29.28mrbob009as in, free to use (gratis=nomoney)
13:30.05stix[TK]D-Fender: I cannot find it, even when searching. Do you know where it is in the book=
13:30.37[TK]D-Fendermrbob009: You only get to play witht he GUI they offer you.  It isn't RELEASED.  I didn't even imply FREE.  Nobody HAS IT!  So the only users are their customers.  GOOD LUCK with that.  You're going to need it.
13:31.05[TK]D-Fenderstix: I gave you keywords.  If you can't handle searching for them well... lets just not go there...
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13:31.34KattyGOOD MORNING MY PRETTIES>
13:31.38*** join/#asterisk dominic1 (n=Miranda@213.221.82.242)
13:31.50[TK]D-Fenderstix: And it is in the index.
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13:38.23[TK]D-FenderKatty: Mew.
13:43.23kombihmm, misdn-init says mISDN modules can not be found, absolutely nothing changed on that box...
13:44.34[TK]D-Fenderkombi: And falling back on that statement isn't going to get you anywhere.  So go reinit the modules and test
13:44.56kombitrue..
13:45.19kombiby reinit you mean re-compile?
13:46.57leifmadsenhas returned from vacation!
13:47.26kombi'cause I did module unload / module load chan_misdn a couple of times..
13:47.37Kattyleifmadsen: HI!
13:47.40Kattypounces leifmadsen
13:47.51[TK]D-Fenderkombi: .... the stuff BEFORE * loads.  Kernel modules, DAHDI, etc
13:48.07leifmadsenKatty: hi! :)
13:48.20[TK]D-Fenderkombi: * channel drivers are the TOP level.  You are missing something in the lower leve so get out of *.
13:48.37eppigyDONDE?
13:48.42KattyESTA
13:49.33kombiFender: got you, misdn-init is the script in /etc/init.d that modprobes the modules and that claims it cannot find them..
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13:50.05jayteemorning Katty
13:50.08Kattyeppigy: i can't get my water open :<
13:50.12Kattyhugs jaytee g'morning
13:50.21jayteehugs Katty back
13:50.34eppigy:[
13:50.42eppigybang it on something
13:50.50[TK]D-Fenderkombi: Well i guess you'd better go fix that
13:51.23kombiright..
13:51.37Katty:>
13:51.49Kattythe blade on my punch down tool worked.
13:58.55eppigynice
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14:07.19Rob3Rthi all
14:07.37Rob3Rtive got incoming and outgoing calls, but only one end can hear
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14:11.24[TK]D-FenderRob3Rt: Read :
14:11.26[TK]D-Fender~sipnat
14:11.27infobot[~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
14:11.28[TK]D-Fender^^^^^^6
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14:21.49timeshellHappy Onesday!!
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14:23.11[TK]D-Fendertimeshell: According to whose calendar?
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14:23.31[TK]D-Fendertimeshell: Mine has Sunday on the far left.  That'd be "one", wouldn't it?
14:23.40timeshellIt's part of the transition to the new metric calendar
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14:24.11[TK]D-Fendertimeshell: Which leaves only the US & Myanmar to deal with :p
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14:38.55Kattyponders what to make for lunch
14:39.22Kattyman
14:39.28Kattyi am cranky this morning
14:39.38Kattygrade A bitchy
14:40.25*** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net)
14:40.29KavanSheh yeah, I woke up on the wrong side of hte bed
14:40.55Kattysomebody get us some perky pills.
14:43.02eppigyi am groggy
14:43.03eppigyish
14:43.36KavanSyeah I'm a little groggy/irritated today....I think it's just that i have a full week of work even before I start the day :P
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14:58.15KattyKavanS: good possibility.
14:58.21KattyKavanS: i came in with 3 servers down this morning.
14:58.32KattyKavanS: probably why i'm cranky
15:00.25*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
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15:04.17KavanSKatty, high-availability! :)
15:04.30KavanSKatty, will keep ya a little more sane when those servers crash...
15:05.19Kattyi'm not paid to be highly available
15:05.36Kattynor am i paid overtime. so yeah.
15:05.37Kattyno.
15:05.44KavanSheh...just hardening your servers is all
15:05.50KavanSso you can have crashes....
15:05.54KavanSand not have to get out of bed
15:06.10Kattywell the power was out for nearly a day here
15:06.14Kattyso..i don't think that will matter much
15:06.17KavanSlol yep
15:06.19KavanSthat'll do it! :)
15:06.28KattyI AM CRANKY!
15:06.32KattyRAWERLKJSdfklmarermergle
15:06.33*** join/#asterisk Jumpie (n=zz@c-76-100-241-4.hsd1.md.comcast.net)
15:06.35Jumpiehey guys
15:07.03Jumpieanybody have any idea why my aastra 55i with expansion module has an issue with its BLF not working after i do network maintenance unless i reboot the phone
15:07.11Jumpieis there some seperate polling mechanism for the module itself?
15:07.19Jumpieor is it still global phone related thing
15:08.31Kattybuy it flowers.
15:08.47Jumpieah...
15:09.03Kattysorry. i'm cranky today.
15:09.53KavanSheh, not sure Jumpie
15:09.59KavanSwait around a little longer for some of the regulars to hop in
15:11.42voipheroesJumpie, what kind of BLF error do you have with your aastra 55i ?
15:11.42awk_rtzafrir_laptop: ping
15:11.47voipheroesAnd what is the firmware ?
15:11.54voipheroes(the version)
15:11.59voipheroesbecause we had some problem too
15:12.12voipheroesbecause it wasn't conform to RFC and it confused our BLF server
15:13.12sumasumahow to cancel a call originated through AMI Originate ?
15:13.33Jumpievoipheroes let me verify, client called me, not sure if its showing ? or just not lighting up
15:13.48Joelsumasuma hangup
15:14.04JoelJumpie sip show subscriptions
15:14.15sumasumaJoel: I'm not receiving the channel information until the call is answered
15:14.22tzafrir_laptopawk_r, pong
15:14.27Kattyayo. i'm tired of usin technology
15:14.52Joelsumasuma lies.
15:15.05Joelsumasuma what does the context your sending it to look like?
15:15.15*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
15:15.23sumasumaJoel: One moment
15:16.34sumasumaJoel: http://pastebin.ca/1493653
15:17.01Joelsumasuma so make queues,1,Hangup()
15:17.23awk_rtzafrir_laptop: you see my comment on M15247
15:17.43sumasumaJoel: The call will come there only after the answer
15:17.52sumasumaotherwise it will not come to the extensions
15:17.58sumasumaI want to cancel the placed call
15:17.58Joelsumasuma then dial a local channel.
15:18.05Joeland process it there
15:18.12Katty[TK]D-Fender: :<
15:18.24Katty[TK]D-Fender: guess who found me on facebook.
15:18.30JoelChannel: Local/$number@the-context-that-might-do-the-hangup
15:18.45JoelI need a mt dew
15:18.58Kattywith absolute vodka
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15:21.35beekGood morning everyone.
15:21.45QwellJoel: I won a mouse from mt dew :D
15:21.56JoelQwell lol, really?
15:22.04Qwellyeah, heh.  a sweet gaming mouse too
15:22.07beekQwell: was it in the can/bottle?
15:22.12Qwellbeek: yes
15:23.12Qwellno, some online promotion they've been doing.  it just showed up in the mail on friday
15:23.18JoelQwell make someone make these https popups go away in ie on digium.com. blech.
15:24.35Joelqwell http://www.digium.com/en/company/contact.php click on "Questions about Digium products and software?"
15:24.44QwellJoel: webmaster@
15:24.47JoelQwell seems to be the ajax trying to load the states causing it in this particular scenario
15:24.56JoelI'm too lazy to fire up outlook :)
15:25.03*** join/#asterisk bluecrow76 (n=weechat@svr1.bignetworks.com)
15:25.32Joelevery time I open up digium.com I find at least 5 issues O.o
15:26.20[TK]D-FenderKatty: no clue...
15:29.41Katty[TK]D-Fender: religious nutcase neighboor's daughter from when i was growing up
15:29.50Katty[TK]D-Fender: same religious (jehovah witness) as my mom
15:30.15*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:30.41coppiceA witness like "It was Jesus. He did it. Nail him up."
15:31.10Kattythis is gonna be a dramatic nightmare.
15:31.27Kattyif i don't accept her friend invitation, she's going to tell her mother. and her mother still lives next to my mother.
15:31.37Kattyand there will be all sorts of whisperings around their church
15:31.42Kattyand if i do accept it, it will be worse.
15:31.58Kattyshe will see my facebook page, and lots of Big Words like Liberal and Atheist
15:32.01Kattyand rainbows.
15:32.22Kattyand that i have a BOYFRIEND that i LIVE WITH whom i'm NOT MARRIED TO
15:32.22Kattyomg
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15:33.28Kattyi should accept the request.
15:33.41KavanSlol
15:33.56carrarSINNER!!!
15:34.05KavanSoh shit....yeah that's screwed
15:34.07Kattyi so am.
15:34.11KavanSwell just do what you gotta do...
15:34.17KavanSI have crazy religious family too
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15:34.20carrarShall we all cast stones upon theee
15:34.26[TK]D-FenderKatty: ... Tell 'em to "fuck off" :)
15:34.31KavanSbleh, to each his own....but you don't need to play into it
15:34.31Katty[TK]D-Fender: ;P
15:34.40Katty[TK]D-Fender: more like fuck off ya crazy religious nutjob
15:34.41KavanSand "fucking off" doesn't leave you with a family to speak to :)
15:35.22KattyKavanS: oh she's not family.
15:35.22[TK]D-FenderKavanS: They're only neighbours.  Apparently losing them isn't such a bad thing ;)
15:35.22KattyKavanS: just daughter of mother's neighbor...who goes to the same church thing
15:35.23Kattymostly i'm worried about my mother.
15:35.36Kattybecause no matter what i do, it'll make her look like a bad jehovah witness parent
15:35.49[TK]D-FenderKatty: this is 4th hand BS.  Don't even blink at it
15:35.52KavanShehe ok
15:36.00KavanSwell never hurts to keep some friends around I guess...
15:36.06Katty[TK]D-Fender: yeah i think i'm going to accept it
15:36.10Katty[TK]D-Fender: just cause.
15:36.27[TK]D-FenderKatty: IT'S A TRAP! </ackbar>
15:36.35Katty[TK]D-Fender: i hope so.
15:36.41[TK]D-FenderKatty: Like you want these people meddling in your life?
15:37.04Katty[TK]D-Fender: not particularly, but it'd be awesome to scare the living snickerdoodles out of them.
15:37.24[TK]D-FenderKatty: There's a good reason i have nothing on mine.
15:38.12*** join/#asterisk seanmh (n=johndoe@207.114.199.107)
15:38.43rob0Hey I had a giggle the other day. A big sign on the side of a church, "Try Jesus", with a cross inbetween the words. I wanted to stop and go in and tell them, no, they already DID try Him and He was found guilty!
15:39.02jamesh1[TK]D-Fender: I have another question, maybe lame.
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15:39.25Kattyrob0: i would have said...
15:39.34Kattyrob0: we can try him, rome had no records of him for 200 years.
15:39.40Kattys/can/can't/
15:39.48jamesh1In one of my queues, call comes in, agent doesn't answer, rolls to next member logged in, rings once and starts loop over again.
15:40.52Kattypray to jesus
15:40.54Kattyhe will fix it for you
15:41.10DarthPointeranyone have any recommendations for channel bank hardware?  I need to provide 24 FXS ports; zaptel 1.4.12.1?
15:43.19[TK]D-FenderDarthPointer: What kind of usage?
15:43.23Kattypray to jesus. he will deliver the channel bank, and your soul.
15:43.33Kattygiggles
15:43.42DarthPointerproviding service for fax lines, credit card machines, etc.
15:43.48Kattysorry.
15:43.52Kattybut at leas ti'm not cranky anymore.
15:44.08rob0Jesus has delivered you from the crankies!
15:44.12DarthPointerAre the Rhino's inexpensive or cheap?
15:44.17stixCan anyone tell me what to write as "extension" in this call-script: http://pastebin.com/m4a2813e2 ?
15:44.32Kattyrob0: praise be to the lord!
15:44.56stixI am using a default trixbox, but don't know if I should write the name of the trunk as the "context" ?
15:45.14[TK]D-Fenderstix: an extension in your dialplan.  As in "what you dial"
15:45.32stixhmm okay
15:45.37[TK]D-Fenderstix: And no, you probably want to use "from-internal" as the context
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15:46.00[TK]D-Fenderstix: Stuff you should be asking ELSEWHERE as their dialplans are FreePBX derived and not your own design
15:46.04[TK]D-Fender~freepbx
15:46.05infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
15:46.36[TK]D-FenderDarthPointer: Rhino's are a very good value.
15:46.59[TK]D-FenderDarthPointer: They've got a decent rep.
15:47.49DarthPointerthats good to hear, what about their T1 cards?  I'm happy w/ the digium gear, however they seem to have some value sku's including T1 cards as well
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15:48.18[TK]D-FenderDarthPointer: For T1, I wouldn't...
15:48.21Kattyjameswf <- Rhino
15:48.31Katty^- you might talk to him
15:49.20DarthPointerpreviously I've only used Adtran and now Digium for T1 cards; I'm currenlty using one of their Wildcat 2400 cards configured w/ 24 FXS and am extremely unhappy w/ it
15:49.34ifluxre
15:51.00[TK]D-FenderDarthPointer: Umm... what card?
15:51.41DarthPointerWildcard TDM2400P
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15:53.05DarthPointerusing digium external power supply; working w/ digium, they want me to reseat all of the modules, the card, upgrade to dahdi and latest asterisk; it's just been flakey; and is now throwing a new error.
15:53.42[TK]D-FenderDarthPointer: A card I would never recommend...
15:53.54DarthPointer"ProSLIC on module 19 failed to powerup within 504 ms (750 mV only)"
15:54.29*** join/#asterisk bijit (i=1000@190.241.15.48)
15:55.07coppice504ms? who sets a timeout like that? it should obviously be 504.375ms
15:55.12DarthPointerAfter the effor we've put into getting it to work, I would second that.  You've had good experiences w/ the Rhino's?  The system also has an open T1 Port on the TE420 (4rth gen)
15:55.31bijitI'm having trouble when putting moh and getting call back..with sip over vpn..anyone have any suggestion9$?
15:56.12DarthPointerAny other Channel Bank recommendations?  I just need reliable
15:56.24carrarADIT600
15:56.33*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
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15:56.55DarthPointernot cheap.. but I suppose you get what you pay for; 10 year warranty is nice
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16:01.54DarthPointercarrar: do you have a couple of these units in service?
16:02.00carraryes
16:02.13carraruse them for faxes also
16:02.25carrarPRI > * > ADIT600
16:02.56DarthPointer*> T1 Brand / card? > ADIT600?
16:03.25carrarADIT600 w/power supply module, T1 module & FXS modules
16:03.41carrar6 slots for either FXS or FXO modules
16:03.52carrarup to 2 T1's woth for channels
16:04.00carrarfor=of
16:04.20carrareach of the 6 FXS or FXO modules has 8 channels
16:04.39carrarsolid devices
16:04.45carrarmajor telcos use them
16:05.25carrarsometimes you find good deals on them on ebay
16:05.44carrarsometimes get them for free as telco's leave them behind
16:06.01carrarhttp://i15.ebayimg.com/01/i/000/a1/eb/ad1a_1.JPG
16:06.04DarthPointerit looks like a carrier grade device; thanks for the advise; I'm looking for distributors right now; force10 is available at Techdata, but I don't see the adit in there
16:06.04carrarthats a pic
16:06.20coppicethere are generally a lot of cheap channel banks on E-Bay, if you don't mind second hand
16:06.24DarthPointerhttp://www.force10networks.com/products/adit_600.asp
16:06.31carrarhas ethernet and serial also for configuring
16:06.46carrarand monitoring
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16:12.16aliveriushi! why does the RT Linux patch disactivate the mISDN system?
16:12.36*** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es)
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16:12.38casixhello
16:14.28ifluxbasically i'm wanting to tinker with the remote-party-id
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16:15.25ifluxmake it look up not just that a creep is calling but who the creep is
16:15.37ifluxoops
16:18.21casixI have a problem with an asterisk. It doesn't start at all. If you start it in background you can connect to it but a lot of modules are no loaded. If I start it with -c then it stops at the same place (after iax2) but it don't crash it only stop starting. If I execute an strace to the asterisk proces i have this: http://pastebin.com/m56fca8dd. what can be happening?
16:20.06*** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk)
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16:21.49*** part/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com)
16:22.27ThatKidKelWhat would cause Asterisk (1.4.20.1) to all of a sudden stop responding to REGISTER requests from a client?
16:23.33*** part/#asterisk stew (i=1413@freenode/staff/stew)
16:24.47uviHello. How to save call start and end times while auto dialout?
16:25.22Qwelluvi: CDRs capture that information
16:26.41uviMy asterisk heavy loaded, I want to fix times in extension and do not want parse CDR
16:27.02*** join/#asterisk ingenius (n=alektro@186.136.6.218)
16:27.04uviexten => s,2,Answer
16:27.04uviexten => s,3,TrySystem(date > /tmp/${phonenumber}.log)
16:27.04uviexten => s,4,Wait(1)
16:27.04uviexten => s,5,Playback(${play1})
16:27.04uviexten => s,6,TrySystem(date >> /tmp/${phonenumber}.log)
16:27.17uvisomething like this
16:27.39uvibut if user hangup while playing, s,6 not working
16:28.09*** join/#asterisk kn0x (n=pinochle@67.159.48.101)
16:28.32kn0xanyone have any suggestions for Monitoring asterisk call quality?
16:29.19kn0xperhaps analaysis of recorded files to determine a MOS score?
16:29.54[TK]D-Fenderuvi: prio "6" should be an "h" Asterisk Standard Extension
16:32.21uvi[Jul 13 20:30:56] VERBOSE[3071] logger.c:     -- Executing [s@outboundmsg1:3] TrySystem("SIP/100-09b64800", "date > /tmp/123456789.log") in new
16:32.21uvi[Jul 13 20:30:56] VERBOSE[3071] logger.c:     -- Executing [s@outboundmsg1:4] Wait("SIP/100-09b64800", "1") in new stack
16:32.22uvi[Jul 13 20:30:57] VERBOSE[3071] logger.c:     -- Executing [s@outboundmsg1:5] Playback("SIP/100-09b64800", "ktv_2_1&200&30&roubles&30&5&copecks
16:32.22uvi[Jul 13 20:30:57] VERBOSE[3071] logger.c:     -- <SIP/100-09b64800> Playing 'test_2_1' (language 'er')
16:32.23uvi[Jul 13 20:30:59] VERBOSE[3071] logger.c:   == Spawn extension (outboundmsg1, s, 5) exited non-zero on 'SIP/100-09b64800'
16:32.26uvi[Jul 13 20:30:59] NOTICE[3071] pbx_spool.c: Call completed to SIP/100
16:32.38uviexten => h,6,TrySystem(date >> /tmp/${phonenumber}.log)
16:32.42uvinot working
16:33.20kn0xor even analysis of jitter and latency?
16:33.34rob0~pb
16:33.35infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
16:33.43rob0"date > /tmp/123456789.log"
16:33.57voipheroesuvi : it's s,h,TrySystem, not h,6
16:34.01voipheroesand use pastebin please
16:34.15voipheroesh mean "hangup"
16:34.25coppicewhy do people love bogus MOS scores so much? :-\
16:34.54rob0Also, what was done was create/overwrite ">" redirection, not append ">>" redirection.
16:36.07[TK]D-Fender[12:33]<voipheroes>uvi : it's s,h,TrySystem, not h,6 <- wrong
16:36.32[TK]D-Fender[12:32]<uvi>exten => h,6,TrySystem(date >> /tmp/${phonenumber}.log) <- Also wrong
16:36.54[TK]D-Fenderuvi: And PASTEBIN, do not spam in here.
16:36.55voipheroesshow us the right way
16:37.07uvisorry for spamming
16:37.37[TK]D-Fenderexten => h,1,TrySystem(date >> /tmp/${phonenumber}.log) <---- Learn to start your priorities at ONE
16:37.45casixI have a problem with an asterisk. It doesn't start at all. If you start it in background you can connect to it but a lot of modules are no loaded. If I start it with -c then it stops at the same place (after iax2) but it don't crash it only stop starting. If I execute an strace to the asterisk proces i have this: http://pastebin.com/m56fca8dd. what can be happening?
16:39.21*** join/#asterisk jgoo (n=r3rman@athedsl-4537080.home.otenet.gr)
16:39.27jgooI have a netmod that isn't sending the DID - halpz
16:39.45jgoointracom - I have 3 netmods, 2 send the other 4 numbers fine, this one has one number, and doesn't send it
16:39.59[TK]D-Fenderjgoo: ..... what si a "netmod"?
16:40.10[TK]D-Fenderjgoo: And PASTEBIN output from your problem.
16:40.12[TK]D-Fender~pb
16:40.13infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
16:40.14[TK]D-Fender^^^^^
16:40.15uvithanks, last work for me
16:40.20[TK]D-Fenderjgoo: You aren't showing us anything
16:40.34*** join/#asterisk nkohh (n=justin@unaffiliated/kohh)
16:40.48nkohhhello! anyone have any experience with a tool called "voipspear"??
16:44.15kn0xcoppice: its managemnt
16:44.35kn0xi know that MOS is bogus, but they want to see something automated about call quality
16:45.01kn0xperhaps MOS isnt the best approach, but what baout Call jitter / packet loss /latency
16:45.09coppiceMOS isn't bogus. these crooked numbers aren't MOS, though
16:45.56jgoo[TK]D-Fender, from-DID=s
16:46.09jgooinstead of from-DID=SUPERAWESOMENUMBER
16:46.16jgooi.e. my DID on one channel is 's'
16:46.31[TK]D-Fenderjgoo: Because you didn't specify the EXTENSION to dial in your REGISTER statement
16:46.41jgoo<[TK]D-Fender> jgoo: You aren't showing us anything << hey, join the queue
16:46.59kn0xcoppice: well i dont believe in a machine telling me what sounds good
16:47.15[TK]D-Fenderi don't join queues.... I'm the DOOR MAN.
16:47.27jgoo[TK]D-Fender, ... I have 3 netmods, and erm, I didn't make a register statement - I have DIDs working for 2 of them (4 numbers) and this netmod has one number and
16:47.29[TK]D-Fendergoes all enforcer-like
16:47.32kn0xanyway, where could I access this kind of information in asterisk?
16:47.34jgoo[TK]D-Fender, in that case, you get to see for free
16:47.40coppicethat's the point. MOS is a lot of people over many many hours telling you what is good
16:47.49[TK]D-Fenderjgoo: what the hell is a "netmod"?
16:48.19*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
16:48.22*** join/#asterisk propellerhead (n=yogurt2u@200.55.16.21)
16:49.42*** part/#asterisk ThatKidKel (n=Kelvin@c-24-98-147-214.hsd1.ga.comcast.net)
16:49.47jgoo[TK]D-Fender, ??? you serious?
16:49.50kn0xcoppice: right. obviously the best case..
16:50.01kn0xbut I still need something automated
16:50.21*** join/#asterisk exvito (n=exvito@80.172.25.71)
16:50.23jgoonetmod is an isdn terminator, right?
16:50.52jgoo[TK]D-Fender, am I in the wrong channel?
16:52.01jgoomaybe ISDN terminal adapter
16:52.50kombihow do I make a call from cli for testing?
16:53.58*** join/#asterisk napster` (n=kurir@118.96.132.189)
16:54.29*** join/#asterisk sjobeck (n=Adium@137.118.193.9)
16:54.40*** part/#asterisk sjobeck (n=Adium@137.118.193.9)
16:55.20exvitokombi: it depends... but check the originate command (help originate at the CLI)
16:56.00exvitokombi: simple example: CLI> originate SIP/something application musiconhold
16:56.07jgoo[TK]D-Fender, ok - I set the DID to match s... and it works... wtf who cares
16:56.18jgoonow. problem - aaargh. sound file doesn't actually play. fffffffffff-
16:56.49jgooIVR works
16:57.23jgoono soundfile plays - daaaamn, is there a cute online flash audio recording output as correct format wav file? I am on mac, and don't have a clue of a decent sound app to make sure, or can someone dump the sox command?
16:57.26*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
16:59.15kombiexvito: thanks! what would it be like to call an outside line?
17:01.26*** join/#asterisk micols (n=mio@rlogin.dk)
17:01.52exvitokombi: maybe something along the lines of > originate local/number@outbound-context application musiconhold ? (it would callout an external line and play music)
17:02.53jgoo[TK]D-Fender, hrm - ok, I setup a feature code to chanhe the recording but DTMF isn't coming out of the grandstream... i'll play and send the DTMF in other ways other than audio.
17:03.34kombithanks!
17:09.58jgooyey worked lol
17:13.17*** join/#asterisk everplays (n=behrooz@89.165.22.146)
17:16.17*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
17:16.42*** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve)
17:16.47*** join/#asterisk hi365 (n=hi365@94.159.178.57)
17:16.59[TK]D-Fender\o/ - back up after about 15 mins w/o power due to gail-force winds, 45 degree (angle) hard rain & hail.  If the walls start bleeding I'm outta here....
17:17.26*** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
17:19.45*** join/#asterisk xpot-mobile (n=james@144.35.21.74)
17:22.55*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
17:23.50beek[TK]D-Fender: Sounds like you're having fun up there.
17:24.08[TK]D-Fenderbeek: Diet fun.  Just like real fun, only half the fun....
17:24.38beeklooks at the nice sunny day outside his window.
17:25.33DarthPointer[TK]D-Fender: out of curiosity, is it your job to provide support on this channel?  are you @ digium or something?  (you've helped me a couple of times now and I really appreciate it)
17:26.00[TK]D-FenderDarthPointer: Nope, I do it out of a sense of pure masochism...
17:26.14beekDarthPointer: if it was his job he wouldn't be half as good.
17:26.25DarthPointeris that like diet support?
17:26.32DarthPointer1/2 the intelligence?
17:26.40DarthPointer;)
17:26.48*** join/#asterisk apdg (n=grey@ash.stopavoiding.us)
17:27.11DarthPointerbtw, you're one sick, but patient, puppy
17:28.38Kattysprawls
17:35.14*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
17:35.43*** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net)
17:36.14*** join/#asterisk tokozedg (n=toka@95.104.37.29)
17:37.20Kattyhmm
17:37.22Kattytime to do nails.
17:37.38beekYour's or your dogs?
17:37.41Zuchmir2how can i transfer the MWI from one * server to another (i.e. i have a type=friend, and it doesn't look like i'm getting the MWI from the other server)
17:38.03Kattyriddick's a male.
17:38.07Kattyi, am not.
17:38.19Kattyergo. mine.
17:38.33*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
17:38.36beekBut doesn't Riddick like a good pedicure every now and again?  (To at least save your flooring some abuse?)
17:38.58*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
17:39.09[TK]D-FenderZuchmir2: You don't.  * doesn't take inbound MWI
17:39.27Kattyuhmm well.
17:39.31Kattyhe has nail clippers.
17:39.48[TK]D-Fenderbeek: that'd be a peticure ;)
17:39.56beek[TK]D-Fender: as seen on TV
17:39.56Kattyand i have assorted shirts for him.
17:40.04*** join/#asterisk Skyp (n=sky@cag.cronon.org)
17:40.21*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
17:42.02Kattythe dog wears a larger shirt than i do.
17:42.52Kattyone says I had a nightmare.... I was a Chihuahua
17:43.04Zuchmir2[TK]D-Fender: any work around to get MWI?
17:43.09kombiHow should I troubleshoot: http://pastebin.se/198494
17:43.13Kattyhe has a light blue one that says Bed Hog on it
17:43.30Kattyand a black hoodie type one that says Security on it
17:43.31[TK]D-FenderZuchmir2: There is nowhere to pass it to.
17:44.13[TK]D-Fenderkombi: How about actually looking at your DAHDI status?
17:44.23kombiwait..
17:45.00Zuchmir2looks like when it comes in, * responds "SIP/2.0 489 Bad event"
17:45.19[TK]D-FenderZuchmir2: Like I said... "not supported".
17:45.47kombifender: says ok.. http://pastebin.se/198495
17:45.52Zuchmir2you mean, * doesn't know which "phone" to pass it to?
17:47.40*** join/#asterisk lanning (n=lanning@nat/yahoo/x-661a5fb6699dd4a8)
17:48.44*** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
17:50.17[TK]D-FenderZuchmir2: Correct.
17:51.24[TK]D-Fenderkombi: Pay attention to your FIRST line.
17:52.04kombido you think the wildcard board interferes?
17:52.21[TK]D-Fenderkombi: I think you should apy attention to what order your modules are initialized...
17:52.26[TK]D-Fenderpay*
17:52.56kombiI'm paying..;) Guess I just rip that wildcard one out now
17:53.32*** join/#asterisk alrs (n=lars@dsl211-216-141.lax1.dsl.speakeasy.net)
17:55.04*** join/#asterisk jitter (i=jitter@188.132.92.111)
17:55.47*** join/#asterisk mykhyggz (n=mykhyggz@evolone.org)
17:57.08*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
17:58.02Zuchmir2[TK]D-Fender: just found this: http://asterisk.mdaniel.net/?p=14
17:59.03*** join/#asterisk lasko (n=lasko@70.102.15.210)
17:59.42laskoDoes anyone know if schedule=yes inside of meetme.conf actually works in the latest trunk of 1.6?
17:59.55[TK]D-Fenderis getting tired of that stupid article now....
17:59.59[TK]D-FenderZuchmir2: ... AND?
18:00.08[TK]D-FenderZuchmir2: that has nothing to dow ith inbound MWI
18:05.13Zuchmir2good point
18:05.52*** join/#asterisk everplays (n=behrooz@89.165.20.141)
18:07.15*** join/#asterisk errotan (n=errotan@62.201.123.220)
18:07.48eppigyEL BANO
18:09.37[TK]D-Fendereppigy: Nope.  If being clueless alone was worthy of banning this would be a much emptier place ;)
18:09.37*** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:10.25leifmadsenwe don't ban; we ridicule
18:10.44*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:11.30[TK]D-Fenderleifmadsen: Googly-eye, then scoff, then smite, then ridicule ;)
18:11.54[TK]D-Fenderleifmadsen: Also funny how the ban-lists clear out....
18:12.05leifmadsen:)
18:12.07leifmadsenoops! :)
18:12.08rob0Let he among you who is without brainfarts, cast the first smirk!
18:12.34WindowsUserthrows a rock at nobody in particular
18:12.35[TK]D-Fendersmirks
18:13.19kombiripped out the x100, rebooted, now "Now such command dahdi"... module show like chan_ happily lists dahdi. what might it be?
18:13.20[TK]D-FenderEverybody must get stoned! </dylan>
18:13.27kombiI am!
18:14.17[TK]D-Fenderkombi: Like usual you aren't verifying anything before * starts, and not doing any diagnostic steps to share with ue.
18:14.23[TK]D-Fenderus*
18:14.29kombiyou know me....
18:15.18[TK]D-Fenderkombi: Yes... I can't afford the psychotherapy it would take to do otherwise...
18:15.34kombi...;)
18:15.39kombifair point!
18:16.47*** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:17.24kombiso here's what I did: lsmod | grep dahdi <- modules loaded. Looked in modules.conf which seemed ok. Did module show and dahdi is there. I must find a way to gain more meaningful output..
18:18.31[TK]D-Fenderkombi: What about "dahdi_cfg -vvvv"?  What about CLI status?   What about trying to reload the module manually?
18:18.32rob0sets up a table with a sign, "Psychotherapy, 5c"
18:19.20[TK]D-Fendersets up a table with a sign, "Psychotherapy, 5cc" and places a box of questionably clean unpackaged & pre-filled syringes.
18:19.53kombishudder...
18:20.08eppigy[TK]D-Fender: that means "the bathroom"
18:20.45*** join/#asterisk dajhorn (n=dajhorn@206.16.96.160)
18:21.01*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
18:21.05rob0eppigy missed the ~ over the N
18:21.13eppigy8[]
18:21.25rob0(but I figured that was what you meant)
18:21.42*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:21.49eppigywell thank you
18:22.00*** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:22.15Kattysits at rob0's table.
18:22.27rob05 cents, please.
18:22.28[TK]D-FenderKatty: C'mon... first hit's free!
18:22.40Kattygives rob0 a hug, and a dime.
18:22.49rob0:)
18:29.01*** join/#asterisk alexfranci (n=alex@host30-53-dynamic.8-79-r.retail.telecomitalia.it)
18:30.45*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
18:32.07*** join/#asterisk theHub (n=theHub@69.177.93.21)
18:32.26*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:32.39Kattyhmm.
18:32.41Kattyi like this color.
18:33.32Kattyhttp://images.revlon.com/HTML/Images//Products/P-lecSweetNothings-lg.jpg
18:34.14eppigysweet nothings
18:34.21kombisweet..
18:34.25Kattythat's the line.
18:34.32Kattyit's actually 735 Please me pink
18:35.21kombias opposed 734 get off me red
18:35.32eppigymay I whisper sweet nothings in your ear?
18:37.52Kattyhow am i supposed to answer that.
18:40.35Kattysorry i'm busy doing my nails right now.
18:40.42Kattyplease come again.
18:41.00KavanSuhm...
18:44.56Kattyeppigy: <3
18:45.22*** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
18:46.41*** join/#asterisk Tophat (n=Tophat@fpal5-a01.peop.tds.net)
18:46.50Tophathow many users is asterisk intended for?
18:47.39Kattyi love questions like that.
18:47.41[TK]D-FenderTophat: Only 5.  there are over a hundred of us here, but only 5 are really meant to run it...
18:47.56[TK]D-Fenderfeels l33t
18:48.14Kattyyour face.
18:48.16Tophathaha
18:48.41eppigyKatty: I was talking to kombi :[
18:49.01eppigyYOUR PBX'S FACE
18:49.38Katty:<
18:49.48Kattyi am now sad.
18:50.00Kattyand my pbx is also sad.
18:51.17*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
18:51.18*** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu)
18:51.30*** part/#asterisk Tophat (n=Tophat@fpal5-a01.peop.tds.net)
18:52.08Katty[TK]D-Fender: did you skeer him away.
18:52.33*** join/#asterisk tokozedg (n=toka@95.104.37.29)
18:52.35eppigyKatty: don be sad
18:52.53eppigyI am sleepy
18:53.03eppigythese new contacts are sleepifying
18:53.04WindowsUserhes probably not supposed to IRC at work?
18:53.08*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
18:53.20Kattyk
18:53.25eppigyWindowsUser: that is just silly
18:54.02WindowsUserim just guessing, i checked the whois cos I know a guy who goes by "toph"
18:54.14stixcan anyone tell me why this won't work in asterisk 1.6.x but works fine in 1.4.x: http://pastebin.com/me7b0532 ?
18:54.29eppigyWindowsUser: STALKER
18:54.34WindowsUseryup
18:55.04Kattyi could use a break
18:55.21eppigyme too
18:55.31Kattywhat shall we break first.
18:55.32eppigyi am max_tired
18:55.39eppigyhrmmm
18:55.51Kattyi hear fender has an awfully nice sword collection
18:56.00Kattymaybe we could borrow for our break.
18:56.05eppigynice
18:56.18[TK]D-Fender:D
18:56.20Katty[TK]D-Fender: require photos of your arsenal.
18:56.21eppigyone day I looked outside
18:56.25tokozedghello, anyone here ever used asterisk cdr analyzer asterisk-stat v2.0 ?
18:56.38eppigyand there was a window missing in the 10 story office building across the way
18:56.39Kattytokozedg: we use it.
18:56.42[TK]D-FenderKatty: I'll take a full-spread pic of them for you tonight
18:56.45Kattyeppigy: gosh.
18:56.46eppigyevidently someone threw a monitor through it
18:56.49Katty[TK]D-Fender: most excellent.
18:56.51[TK]D-Fender...that sounded dirty..
18:56.51eppigyfrom like the 6th story
18:56.52Kattyeppigy: hot.
18:56.55eppigyI know dude
18:57.01eppigyI am jealous to this very day
18:57.06Kattythat should've been youtubed.
18:57.11eppigy[TK]D-Fender: do you have real forged steel swords
18:57.13eppigy?
18:57.17eppigyor folded
18:57.43tokozedgKatty, can you tell me why i get this errors?  http://img403.imageshack.us/img403/5218/snapshot1i.png
18:57.46eppigyI would also like to shoot my m1a through a car
18:57.52eppigyit doesnt have to be occupied
18:58.21[TK]D-Fendereppigy: Both.  I have 2 differentially tempered non-folded forged, and 1 forge-folded.
18:58.34*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
18:58.44[TK]D-Fendereppigy: this is my Folded : http://www.roninswords.com/custom_kiku_in_tea.htm
18:59.02Kattytokozedg: i would have a look at your php and see if there is a missplaced ' or >
18:59.09[TK]D-Fendereppigy: I commissioned it a little over a year & a half ago
18:59.32tokozedgKatty, if you have tested cdr.php, can you upload it somewhere?
18:59.44tokozedgi`ll pastebin mine
18:59.44Kattyi suppose i could post that for you
18:59.49Kattyhold on
18:59.55tokozedgKatty, ok
19:00.02eppigy[TK]D-Fender: that is beautiful
19:00.07eppigythis is my sword http://i58.photobucket.com/albums/g245/eppigy/PC010371.jpg
19:00.53Kattytokozedg: http://pastebin.ca/1493857
19:01.16Kattyi uhh
19:01.18Kattyhmm.
19:01.20[TK]D-Fendereppigy: Balance is wonderful as well.  I need to re-lacquer the throat-cap as the exposed wood on the inside of the saya is a little chewed up and the habaki is too loose for my liking/safety
19:01.36[TK]D-Fendereppigy: M14... perty
19:01.41eppigywell at least you are getting use out of it then
19:01.56laskoDoes anyone know the way to setup extensions.conf for using realtime meetme? I'm having issues where I can get into the conference with both admin and user pin but there is never a marked user.
19:02.09Kattyhttp://farm4.static.flickr.com/3595/3512375990_1713a771e7.jpg?v=0 <- not my sword.
19:02.17Katty^- boyfriend sword.
19:02.27tokozedgKatty, same
19:02.32KavanSlol
19:03.05Kattytokozedg: well isn't that a bummer.
19:04.07tokozedgKatty, so, whats the problem?
19:04.40Kattythe problem is you have a problem that you don't know how to fix.
19:04.42Kattythat's the problem.
19:04.57Kattyi have no problems.
19:05.01eppigy8[]
19:05.05Kattymy nail polish didn't even smudge.
19:05.23Kattythat may have come off a bit rude. let me rephrase.
19:05.26Kattyi don't know.
19:05.31Kattymaybe some else can help you (=
19:05.54Kattysomeone pass me the prozac, zeesh
19:06.01[TK]D-Fendereppigy: Thats the std in .223 right?
19:06.16eppigy.308
19:06.26eppigy.308 TRABAJO
19:06.42tokozedgKatty, thanks anyway
19:06.47Kattywhat's the difference between a .223 and a .308
19:06.58Katty.085?
19:07.26eppigya lot of kinetic energy
19:07.31eppigyZONG
19:07.35Kattywhat's it used for.
19:07.39Kattycivil war
19:07.42Kattyfrench banana war
19:07.48eppigywell .223 is medium range antipreonnel
19:07.55stixcan anyone tell me why I get permission denied here: http://pastebin.com/m73956dcd ?
19:07.57eppigy.308 is more long range
19:08.01eppigyhigh penetration
19:08.05Kattydefine antipreonnel
19:08.07eppigymedium to high range
19:08.12eppigyantipersonnel
19:08.14eppigy:{
19:08.22Kattyself defense?
19:08.27eppigy.223 fragments when it hits a target
19:08.30Kattyouch.
19:08.35eppigyshredding vital organs
19:08.40Kattybad.
19:08.44eppigy.308 pierces armor
19:08.45Kattywhy do you have one.
19:08.55eppigyI love various guns
19:09.01eppigyI have 3 more
19:09.05Kattyso you have them because they look pretty?
19:09.08[TK]D-Fendereppigy: .308 niiiiicce
19:09.32eppigyKatty: I have them so if I need them I am not sol
19:09.38*** part/#asterisk awk_r (n=awk@nat/digium/x-843d4dc8a6dba133)
19:09.45eppigyhttp://i58.photobucket.com/albums/g245/eppigy/PC010372.jpg
19:09.53eppigyhttp://i58.photobucket.com/albums/g245/eppigy/PC010373.jpg
19:09.54Kattyjust in case that girl from the house decides to haunt you.
19:10.05eppigyhttp://i58.photobucket.com/albums/g245/eppigy/PC010375.jpg
19:10.08Kattyespecially if she's wearing armor.
19:10.22eppigyyesh
19:10.25[TK]D-Fendereppigy: Sprinfielf Armory custom 1911?  Looks like only a nickel finish & comp triffer...
19:10.33eppigyKimber 1911
19:10.34[TK]D-Fendertrigger*
19:10.44eppigycustom shop
19:10.46eppigy:D
19:10.55Kattywhat's that black one
19:11.00Kattythat kinda looks like a glock
19:11.01[TK]D-Fendereppigy: Ah.  I'm partial to SA myself.  they've got some awesome .45's
19:11.04eppigywalther p99 .40 cal
19:11.24Kattywhat is it used for
19:11.29Kattybesides James Bond stuff
19:11.41eppigydaily carry
19:11.50[TK]D-Fendereppigy: the slide grips near the front make me nervous :/
19:12.09Kattyohisee.
19:12.24Kattymakes note to never meet eppigy
19:12.29eppigy:<
19:12.47Kattymakes note it's okay to meet eppigy, if all pockets are searched ahead of time.
19:13.01eppigyI am like the guy you want around you at all times
19:13.09*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-158-35.lns10.mel4.internode.on.net)
19:13.38eppigy[TK]D-Fender: what do you mean?
19:13.47[TK]D-FenderKatty: Pockets?  Seriously!
19:14.12[TK]D-FenderKatty: full belt check, in & out, ankle holsters, shoulder rigs, the works!
19:14.14Katty[TK]D-Fender: you know what i mean.
19:14.15eppigyi like hot pockets
19:14.28[TK]D-Fendereppigy: Is that what the kids are calling it these days? ;)
19:14.38eppigyhaha
19:14.45Kattyi already have a 100lb body guard.
19:14.50Kattythat sheds.
19:15.03eppigyawww
19:15.19[TK]D-FenderKatty: And that you can't trust to go pee on its own ;)
19:15.46Kattyhttp://farm4.static.flickr.com/3288/2956414085_fc95ff2de8.jpg?v=0
19:16.02Kattywell, you know. the squirrels and rabbits might pose a threat.
19:16.06Kattybarking at them takes priority.
19:16.11eppigy:D
19:16.26Kattyhttp://farm4.static.flickr.com/3170/2956414389_7d11292647.jpg?v=0 <- so vicious!
19:16.32eppigyKatty: I am gong to start buyinggroceries and cooking in the new place
19:16.36eppigyi need your halp
19:16.38Katty!
19:16.45Kattyi will be there in...
19:16.46Katty14 hours.
19:16.51eppigyTHANKS
19:17.28*** part/#asterisk exvito (n=exvito@80.172.25.71)
19:17.30Kattyfor some odd reason, riddick doesn't like black people.
19:17.37Kattyhe growls at them. i'm not sure why )=
19:18.47rob0If he ever goes out at night wearing a white sheet, worry.
19:19.05Katty*hee*
19:20.01Kattyrob0: OH NOE http://farm4.static.flickr.com/3231/3061747713_aaaca8eef0.jpg?v=0
19:22.08[TK]D-Fenderrob0: 50 years ago 100 men chasing a black man across a field was the KKK.  Today its the PGA Tour...
19:22.52*** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com)
19:23.27KavanSlol
19:23.28KavanSwhoa...
19:23.30KavanS!
19:27.50DarthPointerWhat would cause the members of a Queue to become invalid: Local/xxxxxx@from-internal/n (Invalid) has taken no calls yet
19:28.08DarthPointera "reload" fixed the problem
19:28.27[TK]D-FenderDarthPointer: Probably because an exten changed along the way
19:29.32DarthPointerpoint me in a direction for further research? Guide maybe?
19:30.56[TK]D-FenderDarthPointer: extensions.conf <---
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19:31.05DarthPointerthx
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19:32.30buttons840I'm using Action: Originate in the AMI, what does the async option do? and what is it's syntax?  i'm using the reference here http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate    anyone know of a more complete reference?
19:32.50*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:35.48buttons840simple    use Async: yes
19:35.57buttons840if anyone cares
19:36.00[TK]D-Fenderbuttons840: Info on that parameter earen't too clear...
19:36.21[TK]D-Fenderbuttons840: says "for fast setup", and that it doesn't wait for X to complete whatever that means.
19:36.56*** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net)
19:37.23buttons840normally when using originate in the AMI you originate a call (create an outgoing call) and then must wait for something to happen before you can do anything else, so you could be sitting around for 45 seconds doing nothing while the phone rings
19:37.37buttons840by using async: yes you can originate calls as quickly as you would like
19:37.44buttons840this is my understadning
19:38.22buttons840[TK]D-Fender, but yes, it's not clear, where do you look for documentation and the like?
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19:41.18[TK]D-Fenderbuttons840: I googled it once you asked and came up with virtually nothing...
19:41.21[TK]D-Fender(of value)
19:41.58buttons840i've answered my original question myself, all works as i expected, but i'm still looking for more comprehensive documentation on the matter
19:42.15buttons840any suggestions [TK]D-Fender ?
19:42.44buttons840for example, i received a response of 5, but what does that mean?   the voip.org site doesn't say
19:43.16[TK]D-Fenderbuttons840: No clue... I'd suggest digging the source a bit.
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20:06.08empiricguys when i dial any no it says no exit signal
20:06.08*** join/#asterisk bwat (n=IceChat7@atlantis.kanobe.com)
20:06.11empiricwhat it means
20:08.21buttons840this if frustrating, the originateresponse event seems to produce all sorts of response id's, but no documentation anywhere as to what they mean!
20:08.39[TK]D-Fenderempiric: It means "pastebin the complete call with channel debug enabled so we have something to comment on"
20:08.41[TK]D-Fender~pb
20:08.42infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
20:08.43[TK]D-Fender^^^^^^
20:08.47digilinkcan anyone recommend a SIP or IAX provider that allows unlimited inbound/outbound calling here in the U.S. for a flat rate? I'm setting up my phone system at home and now I need a provider....
20:08.54empiricok wait
20:10.59[TK]D-Fenderdigilink: What is your expected usage?
20:11.26digilinkjust local/ld calling, don't need 800 support or DID
20:11.57[TK]D-FenderdigiIn terms of minutes...
20:12.14digilinkunlimited, if such a plan exists
20:13.01digilinkVitelity looked promising until I saw that it was 1.4c for outbound terminations :/
20:13.02[TK]D-Fenderdigilink: I said your expected USAGE
20:13.11*** join/#asterisk f2knight (n=ben@enetworkservices.net)
20:13.14digilinkIm not sure what your asking me?
20:13.40dajhorndigilink: voip.ms has flat rate packages for $8 ish.
20:15.15[TK]D-FenderWow, great Canadian rates...
20:15.33f2knightQ: Not totally sure about if this exists or not, but does anyone have a sample of how to make a webpage call an end user to i can record a message or memo?  I know packages like trixbox and druid and such do this but i dont want to redo my system to use there interface I just want to make it easier to record a few messages. Idealy someoen would type in a file name and description put in there extension number and asterisk would call them, they pick
20:15.33f2knight<PROTECTED>
20:18.13[TK]D-Fenderf2knight: Go read THE BOOK, and loko at "call files" and "AMI Originate"
20:18.14[TK]D-Fender~book
20:18.15infobot[~book] 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://astbook.asteriskdocs.org or see ~buybook
20:18.17[TK]D-Fender^^^^^^^^^^^
20:19.34rob01.4c/min termination isn't bad at all, on a small scale. On a large scale you can probably get a better deal.
20:20.03rob0I'm using Vitelity, seems okay so far.
20:20.23KavanSvitelity is pretty decent
20:20.33KavanSin my experience...
20:21.28f2knight[TK]D-Fender: thanks I looked in to the AMI Originate, but i was wondering if anyone knows of or has a web page to take the input. I know i can simply telnet in and use the commands to do the call that works fine. but an interface to do that is what I was hopping for.
20:21.28Nuggettelnet is eeeeeeevil!
20:21.58beekwishes Nugget would shove that stupid script.
20:22.20[TK]D-Fenderf2knight: All your job.
20:22.47laskoIs there any documentation on realtime meetme for 1.6? I can get into all my conferences, but for some reason the Admin pin does not set me as marked user.
20:23.07KattyNugget: telnet
20:23.12f2knight[TK]D-Fender: oh well one could hope. seems someone else would have done that already. well guess its back to the text editor.
20:23.14Kattytelnet.
20:23.16[TK]D-FenderKatty: Only triggers once /24h
20:23.20Kattyoh :<
20:23.21Kattybummer.
20:23.44buttons840how am i suppose to use ami originate when the documentation says 0,1,4,8 are the possible responses and i'm getting responses 3 and 5  arrg!
20:23.50*** join/#asterisk dwery (n=dwery@nslu2-linux/dwery)
20:23.50KavanSssh 4l!
20:23.58[TK]D-Fenderf2knight: Of course tons of people have done this already, but everyone's usage is PERSONAL.  their auth methods, their coding standards, etc
20:24.13Kattyponders taco salad for dinner.
20:24.27dwerygr0mit: hi. I will test the portech unit the next week
20:24.34Katty[TK]D-Fender: what're you having for dinner?
20:24.44f2knight[TK]D-Fender: well that I understand just didnt want to start from the ground up, thought I might find  a simple usage to expand from.
20:25.15[TK]D-FenderKatty: Not sure... might try passing by for Indian again...
20:25.30[TK]D-Fenderf2knight: FFS its 5 lines of HTML.
20:25.47f2knightFFS ???
20:25.52[TK]D-Fenderf2knight: And a few puny lines of PHP/etc for the script to execute it
20:25.54[TK]D-Fender~ffs
20:25.55infoboti heard ffs is for f**k's sake, or for fine's sake.  UCB's Fast File System
20:26.25beek[TK]D-Fender: You've had waaaay too much fun with infobot.
20:26.34*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
20:26.47Alfiohi
20:26.59Alfioi im having this error
20:27.17AlfioAsterisk ended with exit status 134 Asterisk exited on signal 6 automatically restarting asterisk mpg123: no processes killed <-------------- i know it was fixed
20:27.28[TK]D-Fenderbeek: that one wasn't mine.
20:27.34[TK]D-Fenderbeek: Clearly.
20:27.38Alfiohttps://issues.asterisk.org/view.php?id=14228
20:27.56Alfiohow i can get the fix an apply to my asterisk pbx
20:28.30*** join/#asterisk citywok (n=chatzill@67-148-102-3.dia.static.qwest.net)
20:28.36QwellAlfio: Upgrade to a release made after Jan 15th
20:29.13Alfiook you mean upgrade my asterisk install Qwell?
20:29.24*** part/#asterisk droemel (n=droemel@p548ECA07.dip.t-dialin.net)
20:29.29Qwellyes...
20:29.36[TK]D-Fendercheckout time, BBIBA
20:29.42AlfioQwell ok
20:30.14citywokI was just tasked with replacing our call centers Inter-Tel system with a full on Asterisk solution.  So far i've done a lot of work and am pretty good with a base Asterisk install in debian (have been using it for 2 years now), what i'm curious about is whether there is a better solution for such a big install with custom requirements for ACD's, automated voicemails, auto dialing, or if i...
20:30.16citywok...would be better suited doing it all by hand
20:31.26*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
20:32.25Xetrov`switchvox?
20:33.16citywokversus freepbx?
20:35.35Joelversus a million other pre-rolled solutions?
20:37.01*** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426)
20:37.02citywokYea, pretty much.  I'm talking 40-50 concurrent calls, 140 clients, failover (i really want hot failover, not a switchvox cold spare), full agent call center functionality inbound & outbound
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20:40.22Joelgo research others of the million solutions?
20:40.41Joelback to the origional question
20:40.49Joelare you going to run into issues hand rolling? yes
20:40.55Joelare you going to run into issues with closed solutions? yes
20:41.04citywokThat's the idea.  I started here to see if anybody had any suggestions on what has worked for them.
20:41.34*** join/#asterisk ingenius (n=alektro@host38.190-138-197.telecom.net.ar)
20:41.43Joelthe question is are you capable of recovering from all the millions of possible issues that could come up, or do you want a vendor to point the finger at.
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20:42.39citywokI've been told that i have until April to have it 100% done, i'm okay with doing a lot of work myself and figuring it out one little component at a time if i have to, and it takes me a few months.  I'm also happy to use a semi-rolled solution, and do quite a bit of customization.
20:43.19citywoki've worked with trixbox and i'm not super happy about it's configuration set up, it's not very easy to customize on a big scale
20:43.30f2knightcitywok: the only way i have heard of being able to do a hot failover is by using a sip proxy like openSER or better yet sipXecs provides built in support for multiple servers for failover. (and a web based gui for administration)
20:44.16f2knightcitywok: freeswitch seems to be able to handle this too.. which is why sipXecs is using it as the back end
20:44.27Joelf2knight you can hot roll t1's
20:44.31Joeland analong lines
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20:44.47citywokthe other requirement was it has to be fully redundant, no matter what the solution is.  2 identically configured boxes using a heartbeat and a VirtualIP is a valid solution in the event of a failover the standby box can take over, but obviously all existing calls get dropped (assuming they already did in event of failure)
20:44.59citywoki dont have any T1's, it will be 100% sip termination
20:45.21citywokwe already convert out analog PBX to sip using asterisk, and do all our call recording & monitoring on the asterisk side, so that part is taken care of
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20:53.55planetWayneHi all, simple question, a phone connected on skinny, is there any way to get Asterisk to reload the skinny.conf file without a restart ?
20:54.25MiccWill there be a 3rd edition of the asterisk book soon?
20:54.43MiccI don't want to buy the 2nd edition if theres a new 3rd edition coming out soon.
20:54.52beekplanetWayne: module reload chan_skinny
20:55.08beek(I'm assuming this as I don't have the driver for skinny)
20:55.23planetWayneah brill - cheers beek - been stopping and restarting - I'll give it a quick test :)
20:57.54planetWayneMmm interstingly it wont let me reload it as there is already something bound to skinny... which is a little odd as the phone works :/
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21:04.23Zuchmir2how do i set CID on Dial()
21:04.48Zuchmir2when i bridge a call, it sends the CID num, not name
21:05.34De_MonZuchmir2 you set the CID before dialing using Set()
21:06.39Zuchmir2Set(CALLERID(name) = "Name") does not send it to the phone :-(
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21:08.36*** part/#asterisk Carlos_PHX (n=carlos@ip68-3-162-244.ph.ph.cox.net)
21:12.52Zuchmir2looks like it only sends the CID num from the incomming channel to outgoing channel
21:13.45*** join/#asterisk voip_troll (n=voip_tro@96.51.229.227)
21:14.18voip_trollI'm having some trouble with config for faxing.  What do I need to to have the received faxes stored on the server?  (I can see them in /var/spool/asterisk/fax/ as they are received, then the fax-to-email script is called, and then the file is gone.  I don't see anything in fax-process.pl that is deleting the file)
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21:17.24chrisbanyone from xmission here?
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21:53.54kombican't call out with zoiper over * http://pastebin.se/198499 (this might be embarassing, another 5c for fender)
21:55.33*** join/#asterisk micols (n=mio@rlogin.dk)
21:59.53[TK]D-Fenderkombi: [Jul 13 23:42:14] NOTICE[6183]: chan_iax2.c:9265 socket_process: Rejected connect attempt from 192.168.0.50, request '01776034346@buero' does not exist <- SPELL-CHECK.
22:00.57kombi5c...
22:04.47kombithanks fender!
22:07.45*** join/#asterisk Alfio (n=Amunoz@190.94.61.34)
22:16.07chrisbanyone from xmission here?
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22:20.04*** mode/#asterisk [+o mog] by ChanServ
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22:41.17laskoWhen submitting a bug, what does it mean when status is "needs license" ??
22:43.30laskooh nevermind, figured it out
22:45.07*** part/#asterisk voip_troll (n=voip_tro@96.51.229.227)
22:49.46buttons840[TK]D-Fender, do you know when Corydon76-dig is usually available, he told me much about creating outbound calls, and about the async options.  I'm hoping he might know about the OriginateResponse values and where i can get some documentation on them?
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23:00.12Rob3Rti think im starting to hate kids today
23:00.29Rob3Rtthey get a vid on youtube at 12 years old and think theyre cool
23:02.28Rob3Rtdo i need any others ports open to hear voice?
23:02.31Rob3Rtor just 5060
23:02.40*** join/#asterisk Gnewt (n=hackerle@because.iownyour.info)
23:02.54[TK]D-Fenderbuttons840: No, I don't
23:03.05[TK]D-Fender~sipnat
23:03.06infobot[~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
23:03.10[TK]D-FenderRob3Rt: ^^^^^^^^^
23:03.14Rob3Rtta dude :)
23:03.24Gnewt[Jul 13 15:56:52] NOTICE[27568]: chan_sip.c:14035 handle_request_invite: Call from '26953928' to extension '<redacted>' rejected because extension not found.
23:03.26Rob3Rtyeah i faved that one last night
23:03.30GnewtIncoming doesn't work
23:03.35[TK]D-FenderRob3Rt: I passed that to you YESTERDAY as well...
23:03.35Gnewtit makes me a sad panda
23:03.47GnewtI have an "incoming call rule" set (from the GUI) but nothing else, what am I missing?
23:03.58[TK]D-FenderGnewt: GUI's are not supported here <-
23:04.14Gnewtokay
23:04.15[TK]D-FenderGnewt: See the /topic for related channels
23:04.27Gnewtthanks [TK]D-Fender
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23:05.58kb3ien<PROTECTED>
23:06.03*** part/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au)
23:06.04*** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au)
23:07.00Rob3Rtman i wish aster logged phone calls in win32
23:07.06Rob3Rtbrb gonna triple check
23:07.34*** join/#asterisk Micc (n=dotirc@97-126-72-14.tukw.qwest.net)
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23:10.10[TK]D-FenderRob3Rt: No, it isn't supported in there.  Only the GUI is.
23:12.15Rob3Rtrage
23:12.23Rob3Rttrying so hard to get a log sorted for you
23:12.31Rob3Rtyoud spot the issue straight away i know it.
23:14.24[TK]D-FenderRob3Rt: if you can't cut & paste from * CLI expect this to drag on for a while....
23:14.31[TK]D-Fender(longer)
23:15.25*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:16.22Rob3Rt[TK]D-Fender, best i can do man, http://pastebin.com/m7da09a84
23:16.57Rob3Rtmy provider is 58.96.1.2
23:17.28Rob3Rtmy stun is 58.96.37.87
23:17.36Rob3Rt3cx is 10.0.0.5:5061
23:17.42Rob3Rtaster is of course 5060
23:17.52Rob3Rti *can* call out to my mobile
23:17.57Rob3Rtbut i cant hear voices
23:18.14Rob3RtI WANT TO HEAR VOICES
23:18.15Rob3Rt:p
23:23.16SaiSomaanyone here with experience connecting nortel (cs1000e specifically) to asterisk over a TMDI PRI?  maybe even using NI2?
23:25.26[TK]D-FenderRob3Rt: Contact: <sip:0756416068@10.0.0.5> <------------- line #asterisk 343.  You are passing back your PRIVATE IP.  Follow the damn guide
23:25.47Rob3Rtwhat guide ?
23:26.05[TK]D-Fender[19:03]<[TK]D-Fender>~sipnat
23:26.06[TK]D-Fender[19:03]<infobot>[~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
23:26.08[TK]D-Fender[19:03]<[TK]D-Fender>Rob3Rt: ^^^^^^^^^
23:26.16[TK]D-FenderRob3Rt: The one I linked you half a dozing &^#$ing time
23:26.22[TK]D-Fenderdozen
23:26.30Rob3RtI followed it, but ill read it again :\
23:26.51Rob3Rti think that means im calling out from the wrong context
23:26.53Rob3Rtor sommit
23:26.54Rob3Rtbrb
23:26.58[TK]D-FenderRob3Rt: follow-through FAIL
23:28.23GnewtIn my incoming context, I have
23:28.24Gnewtexten = s,1,Goto(default,6000,1)
23:28.31GnewtThat should make everything go to 6000
23:28.36Rob3Rt=>
23:28.46Rob3Rtafter 1 second
23:29.03GnewtOkay, let me try that
23:29.06Rob3Rtchange 1 to 60 to allow more time for transfer (guessing)
23:30.02GnewtWell it works if I set exten = <mynumber>,1,Goto(default,6000,1)
23:30.09GnewtSo I don't think that's the problem?
23:30.28Rob3Rtcheck the definition of s again
23:30.32Rob3Rthavent got there yet
23:31.23*** join/#asterisk Faiz (n=otakucon@c-69-253-143-177.hsd1.nj.comcast.net)
23:31.31*** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com)
23:33.14GnewtOh! _! did the trick :)
23:33.23Rob3Rtahh
23:33.47*** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep)
23:33.54teknoprep[TK]D-Fender, you around ?
23:34.07teknoprep[TK]D-Fender, i know you remember how to setup Polycom phones to allow Buddy Watch
23:34.22teknoprepmy polycom phone doesn't even show the option for buddy watch when adding aline
23:34.34SaiSomateknoprep, that's a change in your tftp files
23:34.39[TK]D-Fenderteknoprep: Go read the WIKI page that shows you the sip.cfg settings to enable Presence
23:34.46[TK]D-Fenderteknoprep: It doesn't come stock
23:34.47SaiSomawhat fender said
23:34.51[TK]D-Fenderteknoprep: in *, make sure its
23:35.00teknoprepfeature feature.1.name="presence" feature.1.enabled="1"
23:35.01[TK]D-Fenderteknoprep: in *, make sure its "type=peer", "call-limit=99"
23:35.06teknoprepyeah i just couldn't remember where it was
23:35.25Rob3Rt[TK]D-Fender, my calls start out correctly didnum@provider to mobilenum@provider, my mobile rings but i hear no voice, then i see didnum@10.0.0.5 to mobilenum@10.0.0.5
23:35.38SaiSomateknoprep, you running the latest firmware?  there's another setting I had to do.  are you testing now?  if it doesn't work, yell and i'll open my vpn and look at my settings and pastebin them for you
23:35.44[TK]D-FenderRob3Rt: And what does the CONTACT header for that call out say?
23:35.47SaiSomai had to contact polycom to get the right settings
23:35.56Faizhey again, would Asterisk be able to run on a 4GB flash drive?
23:36.14SaiSomahey Faiz.  as long as you can boot to the drive, sure
23:36.15[TK]D-FenderFaiz: * doesn't care
23:36.18SaiSomaafaik
23:36.24Rob3Rtk one sec
23:36.30Faizah, great! thanks
23:36.51[TK]D-FenderFaiz: People run * on stupid Linksys routers.... 4 GB sounds "deluxe" by their standards
23:37.06Faizthe WRTs?
23:37.15[TK]D-FenderFaiz: Yes
23:37.22Faizhah, classy
23:37.42Faizthough that might be an interesting endeavor..
23:37.50Faiznonetheless, how is everyone?
23:38.24SaiSomagood.  how's your project coming Faiz?  professor like your ideas?
23:38.57FaizYes, so far all is well. the only downside is that I have to pay for all of my equipment
23:39.15Faizand ofcourse, being a professor, being frugal isn't part of his vocabulary
23:40.25SaiSomahow weird.  frugal is built into our instructor's bones:)
23:40.30Rob3Rt[TK]D-Fender, FROM: sip:DIDNUM@PROVIDER.IP TO:sip:MOBILENUM@PROVIDER.IP CONTACT: sip:DIDNUM@10.0.0.5 <---
23:40.36SaiSomapart of being a small community college i guess
23:40.38Rob3RtFAIL
23:41.04Faizhehe, that's pretty awesome though
23:43.00Faizalright, take care everyone
23:43.03Faizsee ya next time
23:43.08SaiSomaFaiz:  see you man.  have fun
23:43.13Faizthanks, likewise!
23:46.29Rob3RtThis is ridicuous.
23:47.05Rob3Rtthe calls are working correctly, its gotta be a codec issue.
23:47.18Rob3RtIm trying to pass-through for G729
23:47.24Rob3Rtthats all my provider supports.
23:47.29Rob3Rtffs. bbiab.
23:47.39teknoprepyour provider doesn't support ULAW ?
23:47.43teknoprepthats just retarded
23:48.53[TK]D-FenderRob3Rt: No, it isn't a codec issue or the call wouldn't work in EITHER direction.  You are not following the instruction
23:48.55[TK]D-Fenders
23:49.18[TK]D-FenderAnyway, I'm off to volleyball, BBIAB
23:50.06SaiSomahave fun fender
23:52.15*** join/#asterisk mechbangirc (n=mechbang@mbl-99-48-169.dsl.net.pk)
23:52.46mechbangirchi can I send hang up cause code "34" on sip channels?

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