irclog2html for #asterisk on 20050306

00:00.28bkw_just passed 211th street
00:00.30filecool I can hit the Send key on this phone and get a dialtone
00:00.35terrapenbkw: i'm sorry :)
00:00.39bkw_haha
00:00.45terrapen<--- texas boy
00:00.46bkw_terrapen, almost out of glenpool now
00:00.49terrapenare you on the road?
00:00.53bkw_CLEAR
00:00.53bkw_yep
00:00.54terrapenwhat are you using for commo?
00:01.00filehe's using GPRS baby!
00:01.02*** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net)
00:01.02bkw_Sapulpa Exit 1 Mile
00:01.03terrapenah
00:01.07terrapenSprintPCS?
00:01.11bkw_no
00:01.13bkw_T-mobile
00:01.15mikegrbbkw_: I'm from okc
00:01.16terrapenahh
00:01.20mikegrbbkw_: :D
00:01.21terrapenGPRS is fun
00:01.24bkw_mikegrb, kewl
00:01.25*** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au)
00:01.25mikegrbbkw_: but in florida now
00:01.27terrapenplease tell me you aren't driving, bkw
00:01.28bkw_McAlester, Here
00:01.30terrapeni mean
00:01.34terrapen*You* aren't driving
00:01.35bkw_no i'm not driving
00:01.37bkw_are you MAD
00:01.39terrapenok, good :)
00:01.41mikegrbbkw_: oh really? I never knew that
00:01.43mikegrbspiff
00:01.43terrapenwell, i've done it before
00:01.44bkw_i'm crazy
00:01.45bkw_but gee
00:01.48bkw_give me a break
00:01.52ozJames79hi all anyone having problems with broadvoice and incoming calls ?  i fixed up the outgoing but it appears no incoming now
00:01.53bkw_:P
00:01.57mikegrbwell, are you from there, or did you live there for 15 to 20?
00:02.05bkw_some guys  minnow bucket flew out of his boat
00:02.09bkw_an almost hit us
00:02.10sivanaozJames79: what kind of issues, errors?
00:02.17terrapenyou have to dip if you are going to drive through OK
00:02.18bkw_terrapen, ewww
00:02.20bkw_thats nasty
00:02.27ozJames79its not even hitting my CLI .. which is weird
00:02.28terrapenwhere are you headed?
00:02.32bkw_I was born here.. I hate it all
00:02.34bkw_Tulsa
00:02.38terrapenfrom?
00:02.39ManxPowerI wonder if I could do BOTH VON Europe AND AstriCon Europe?
00:02.41ozJames79when someone calls my BV number it rings busy
00:02.55bkw_welcome to BV
00:02.57terrapeni'm a hick, admittedly
00:03.05mikegrbozJames79: then call BV, #asterisk is not #bv-support
00:03.17ozJames79i was not asking for support
00:03.21terrapenbkw, whats in SJC
00:03.23sivanaozJames79: they have instructions on their site
00:03.26ozJames79i was asking if anyone else with bv was having the problem
00:03.29NirShey bkw, I heard congrats are in order
00:03.31ozJames79i am on the phone to them right now
00:03.46bkw_NirS for?
00:03.47sivanaozJames79: I don't think so, I havent' checked laately
00:04.09sivanaozJames79: they are my last hope route
00:04.20ozJames79thanks sivana :)
00:04.24sivana:)
00:04.30Goshenwhat are these called on the end of a dial string, I am looking for documentation on them... ,60,Ttm)
00:04.39fileparameters...options...
00:04.48terrapeni've never been around Tulsa
00:04.52fileshow application dial to see what options are available
00:04.53terrapenonly flew through on Southwest
00:04.54fileGoshen: that was for you
00:05.03filedomo arrigato Mr. Roboto
00:05.18Goshenat the command prompt?
00:05.24fileat the asterisk CLI
00:05.30terrapentake me back to Tulsa, I'm too young to marry, take me back to Tulsa, I'm too young to wed thee.....
00:05.52Goshenwow sweet! thanks :) didn't know it did that
00:05.52fileshow application <application name> Will give you... documentation
00:05.55tzangerdammit
00:06.01tzangerhow do I get ipv6 DOWN so I can remove the module
00:06.05tzangeruse count is like 96
00:06.27Goshen~docs
00:06.28jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
00:07.48terrapenwe always have a great big time, never do look sour / travel 'round the country, day and by the hour
00:08.06terrapen(it's Bob Wills and the Texas Playboys)
00:08.34Goshenjbot: docs is Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org show application <application name> in the Asterisk command line.
00:08.35jbot...but docs is already something else...
00:08.42Goshen:)
00:09.07ckruetze_I wish VON wouldn't be so expensive :(
00:09.09*** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk)
00:10.05bkw_woops missed the exit
00:10.15filehaha
00:10.28bkw_i'm not the one driving
00:10.31bkw_remember
00:11.04terrapenOn a clear day in Tulsa, you can see the back of your head.
00:11.19bkw_oh really?
00:11.20bkw_since when?
00:11.29terrapen:P
00:11.43terrapenactually, its supposed to be Lubbock, i just changed the joke
00:11.54terrapenLubbock is quite flat, you see...
00:13.00sivanaheh
00:13.09bkw_ok we are almost there
00:13.10bkw_bbl
00:14.14sivanawhat device are they using to be able to chat on IRC mobile?
00:14.33terrapenyou can use a GPRS-enabled phone and a USB cable
00:14.37terrapenphone acts like a modem
00:14.39sivanaI'm missing out on all the gadgets
00:14.42terrapenlaptop sees it as a modem
00:14.43sivanaGPRS?
00:15.05sivanaI see
00:15.14terrapenits networking that works over most modern GSM cellular networks
00:15.26tzangersivana: I can use my CDMA1xRTT
00:15.28tzangerbut it's slow
00:15.28terrapenits somewhat pricy
00:15.29tzangerand expensive
00:15.40terrapenpricey even
00:16.40sivanaI see
00:17.05sivanamaybe someday Canada will upgrade their tin cans to 3G or whatever is newest
00:17.07tzangerkeep it under the seat of the car with the CDMA1xRTT
00:17.09j0i'm having problems connecting to the iaxtel.com gateway.. it will register for a few moments when asterisk first loads.. but otherwise i keep getting "DEBUG[1152]: Raw Hangup 69.73.19.178:4569, src=2, dst=136"
00:17.24sivanaheh
00:17.32j0i'm behind nat, i have all data from their ip forwarded to *, as well as the appropriate iax ports
00:17.48sivanaj0: do yourself a favor, forget IAXTel
00:17.59sivanait's lagged and uselss
00:18.02j0sivana: so its not me? :)
00:18.09j0i just need something to test with right now
00:18.11ManxPowertzanger: You can use 1xRTT without a plan and it will use your plan mins.
00:18.12sivanaspend $10 and get a Nufone acct
00:18.21tzangerManxPower: not with bell canada
00:18.23tzangerit's expensive
00:18.38ManxPowertzanger: You can with Verizon.
00:18.48ManxPowerHas anyone here been to VON?  If so, please /msg me.
00:20.26sivanacan we even get Verizon in Canada?
00:21.06sivanamaybe with their unlimited roaming plan? :)
00:21.35ManxPowersivana: Yes.  The "north american unlimited plan" but I think you have to have a USA billing address.
00:21.43sivanaI see
00:22.36sivanain a bottle?... brilliant
00:22.56tzangerhahaha
00:23.53sivanahehe
00:26.30terrapenso drive across the border and get a UPS Store box
00:26.46jessterin sip.conf I want to setup SER for incoming and outgoing calls - Should I create a [ser_in] type=user and a [ser_out] type=peer    or just a [ser] type=friend ?
00:26.59BrianR___tzanger: I'm lucky enough to live in an area with CDMA EVDO.. It's like 2 megabit...
00:27.19BrianR___Unlimited access for like $80/mo.
00:27.47tzangernice
00:27.48tzangervery nice
00:27.54tzangerwhere's the area that has that
00:27.56BrianR___Fails over to 1xRTT in areas without EVDO equipment.
00:28.08BrianR___tzanger: A number of big metros in new england...
00:28.15tzangerahh
00:28.24BrianR___at least 50 miles around boston in my area.
00:28.41terrapenhow is the latency on that
00:28.43BrianR___http://www.verizonwireless.com/b2c/mobileoptions/broadband/index.jsp
00:28.59BrianR___terrapen: Usually <100ms.
00:29.10BrianR___Always much better than GPRS, etc. though.
00:29.14terrapennot bad
00:29.44terrapengood lord
00:29.50BrianR___latency varies quite a bit based on the number of retransmits required, how busy the cell site is, etc. But it's as good or better than CDMA circuit switched data and 1xRTT.
00:29.57terrapenthey serve Baltimore but they don't serve San Antonio
00:30.05terrapen10th largest city in the US, or thereabouts
00:30.09terrapen#1 most forgotten
00:30.14DaLionquick question
00:30.14DaLioni need a perl cgi to talk to asterisk to make it dial.. is that command exec dial ?
00:30.20BrianR___terrapen: Heh. It'll get there eventually.
00:31.04DaLionor i use manager ?
00:31.28DaLioncant really seem to find examples
00:31.49DaLionso i guess i got 3 choices.. AGI, manager or dump a file in queue ... rirhgt ?
00:32.06DaLionor is AGI only * to cgi
00:33.06*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
00:34.54*** join/#asterisk trelane (~trelane@lan.trelane.net)
00:34.58terrapen<PROTECTED>
00:35.05terrapenbrian, it seems that it IS available in my area
00:35.11terrapenif you put in my zip code...
00:35.27DaLionterra ?
00:35.30terrapen<PROTECTED>
00:35.34terrapenthis is really confusing
00:35.40DaLionim trying to talk to * from perl.. got AGI:asteirsk etc
00:35.48DaLionanyidea ?
00:35.51DaLioni need to dial out
00:36.29terrapenyeah, we dont have it brian
00:36.32terrapenjust austin
00:37.51trelaneanyone having issues with broadvoice?
00:38.09DaLionbah i use teliax no issues
00:38.31DaLionu can make special corp arangements
00:38.50Bruns!@#$%
00:38.53Brunsexcuse my cursing
00:38.59Brunsbroadvoice changed their setup
00:39.02DaLionBruns yeah tell me about it
00:39.05DaLionlol
00:39.05Brunsthey are sending me details
00:39.58jessteranyone run into auth problems having phones register with SER and then have ser forward calls to Asterisk? Asterisk is giving me auth errors
00:41.21Brunsapparently, theres an unusually high amount of asterisk users calling them
00:41.55*** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net)
00:42.38Damin_MobileYo yo yo!
00:43.11trelaneBruns, umm what?
00:44.45*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
00:45.12shmaltzandrew?
00:45.43ManxPowerHas anyone here been to VON?  If so, please /msg me.
00:46.08Brunshttp://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
00:46.27Brunsthey changed their setup slightly, ever so slightly
00:48.15*** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com)
00:53.44Damin_MobileManx: Been to? Or going to?
00:57.56Brunsnote to self, must kill broadvoice
00:58.12*** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es)
01:00.17tzangerNugget needs to use nufone like all the cool people do
01:00.36NuggetI use them too, but I still have this accursed DID from voicepulse that I hope will actually work some day.
01:00.42tzangerahh
01:05.02shmaltzanybody here using Cisco?
01:05.09shmaltzI mean cisco phones
01:05.12Nuggetsure
01:06.36Damin_MobileI use cisco
01:06.55Damin_MobileI love cisco
01:07.24r0d3nt|mcisco or death.
01:09.19*** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63)
01:14.58JohnnyCcan I forward a call to an extension based on the number called ?
01:15.25JohnnyCI have a BRI with 10 numbers
01:15.27*** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net)
01:15.36JohnnyCnumber X would go to extension 20 directly
01:15.47JohnnyCor awnsered by music , anything
01:16.02*** join/#asterisk GrimStone (~Pkunkage@203.187.245.49)
01:16.31shmaltzJohnnyC, why not?
01:16.40JohnnyChehe
01:16.45JohnnyChow ! :)
01:17.01bjohnson_quick question.  What advantage is there to running * on a wrt54g compared to just using sip phones?
01:17.01shmaltzdo you know how to configure Asterisk?
01:17.05JohnnyChow do you name this ? for me to look in docs and mailing ?
01:17.13*** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk)
01:17.22JohnnyCyes sure
01:17.29tzangerdamn all you and your fancy schmanzy domains
01:17.34blitzrageanyone use the SMS() app? I'm looking for an example, but I'm not in the UK, so I can't test.
01:17.46shmaltztzanger, whats wrong?
01:17.49expressfone1hi
01:17.57JohnnyCDDI ?
01:18.08tzanger20:24 -!- blitzrage [~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk] has joined #asterisk
01:18.10blitzragetzanger: get back to work.
01:18.17tzangerblitzrage: eat me :-)
01:18.21blitzragetzanger: yah I know... I was looking for someone specifically, but he isn't here.
01:18.30j0any thoughts on www.simpletelecom.com .. nufone seems expensive compared to them
01:18.42blitzragefile: !
01:18.47filehi hi
01:18.52blitzragej0: I use them, work fine (I don't pay them... but I use them :))
01:18.54tzangerj0: expense is in the eye of the beer-payer
01:19.05shmaltzJohnnyC, can you explaing again what you are trying to do?
01:19.05tzangerj0: not having ANY issues is worth a lot to me
01:19.08blitzragebut NuFone's network is supposedly pristine.
01:19.17filehow are you?
01:19.18JohnnyCDirect number to an Extension
01:19.31tzangerI won't say pristine, but in the past year and a bit I've used it I have neve rhad issue that was their problem
01:19.36blitzragefile: pretty good, j00? I'm working on DOCS!
01:19.44fileblitzrage: I had surmised you were
01:19.49fileblitzrage: still on track?
01:19.53blitzragefile: mostly.
01:19.54j0blitzrage: do you just use their free service?
01:20.02filesilly mostly
01:20.07blitzragej0: I signed up for a beta a long time ago, and never got cut off :)
01:20.18j0thats what i'm using now to test.. its my first attempt
01:20.22*** join/#asterisk Rick_Hunter (~rhunter@adsl-69-209-173-100.dsl.sfldmi.ameritech.net)
01:20.26blitzragetzanger: SEE!  Already sucked in
01:20.31*** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
01:20.40blitzragefile: chat with me on another channel... I'm out of this window.
01:20.41*** part/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk)
01:20.55tzangerhahaha
01:23.22shmaltzJohnnyC, you mean did?
01:23.25shmaltzDID
01:23.32JohnnyCDDI or DID ?
01:23.38JohnnyCDirect Dial in ?
01:24.34GrimStoneanyone have problems with broadvoice outgoing ?
01:25.22GrimStoneeven after setting the userid , secret, authid etc. like the email on asterisk-users says , asterisk doesn't seem to be able to "hook up" outgoing calls
01:26.07GrimStonei can see voice data coming in from broadvoice on port 5060 .. but asterisk just seems to ignore it
01:26.38JohnnyCshmaltz: I want to ring directly an extension or voicemail to diferent numbers
01:27.30GrimStonethis was working great until today , when bvoice did thier "upgrades"
01:27.56JohnnyChave to go
01:28.21shmaltzJhonnyC,
01:28.23shmaltzwait
01:29.16shmaltzJohnnyC, look up DID on the wiki, all you have to do is find out how many digits your carried sends with the bri, in extensions.conf create extensions that match the digits received from you carriedr as DID.
01:31.07GrimStoneso anyone here use Broadvoice ?
01:31.32GrimStonecause right nows it badly broken
01:36.53*** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc)
01:37.31GrimStonewhy would asterisk just ignore the voice data coming in on port 5060 ?
01:41.48*** join/#asterisk mischko (~Scott@p29-25-150.vcr.centurytel.net)
01:46.51*** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
01:48.15*** join/#asterisk trimi` (~Pharrell@62.162.232.118)
01:53.06MikeJ[Jayden]wow, it's quiet... what, is it saturday night or somthin
01:55.09GrimStonewhy would asterisk just ignore the voice data coming in on port 5060 from broadvoice ?
02:00.18jessterI have SIP phones => SER(register) => Asterisk -PSTN. For some reason my phones will only dial through if a certain UserID is set on the phones, otherwise if the UserID is not the correct one, I get a 407 Proxy Auth Required / 403 Forbidden (Im doing auth with AuthID) anyone have suggestions how to fix?
02:01.21*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
02:05.20*** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl)
02:05.29*** join/#asterisk JohnJar (~JohnJar@85-250-71-159.bb.netvision.net.il)
02:06.15JohnJarHi, where can i get more info about Asterisk and about PBX's? thanks
02:06.50mikegrbhttp://www.google.com/
02:07.15*** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net)
02:07.28mikegrbda-manFL: I disagree.
02:08.20da-manFLwhy?
02:09.51mikegrbI am da-man in florida, this state is not big enough for both of us
02:12.40tuxinator_linux~docs
02:12.41jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
02:13.21j0what was that * project that let u share your phoneline with other people around the world to make ld calls?
02:13.27j0was a sort-of p2p system based on credits
02:14.01cbachmanj0 bellster/fwd-out ?
02:16.11*** part/#asterisk mischko (~Scott@p29-25-150.vcr.centurytel.net)
02:18.15trimi`<j0>
02:18.22trimi`<j0>  it was something with out
02:18.24*** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net)
02:18.28trimi`the name i dont remember
02:18.33trimi`like s****oute
02:18.40trimi`like s****out
02:18.54trimi`let me see if i still got it in my favorite links
02:19.34trimi`yeap i found it
02:19.35trimi`http://www.fwdout.net/web/ToSignup
02:19.37mikegrbtrimi`: the question has been answered, no need for your pollution
02:20.06trimi`sorry didnt see it
02:21.46trimi`is there any bootable version of asterisk with calling card platform installet or which include ASTCC or any simmilar aplication ?
02:23.00mikegrbno
02:23.34trimi`:(
02:23.44shepherdjeeze harsh :)
02:23.56*** join/#asterisk file (~file@mctn1-142166194173.nb.aliant.net)
02:24.08mikegrbshepherd: it's the answer
02:24.21j0cbachman: yeah thanks :)
02:24.26*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
02:26.59*** join/#asterisk PhoneBooth (Phonebooth@208-25-55-247.stk.inreach.net)
02:28.37jessterI have SIP phones => SER(register) => Asterisk -PSTN. For some reason my phones will only dial through if a certain UserID is set on the phones, otherwise if the UserID is not the correct one, I get a 407 Proxy Auth Required / 403 Forbidden (Im doing auth with AuthID) anyone have suggestions how to fix?
02:28.39Chujidamn broadvoice. Why do I still use them?
02:28.54ChujiI should just go to pay per use
02:29.46filejesster: your question makes no sense, if the authid isn't the correct one... shouldn't it send back the 407 or 403?
02:30.54jessterfile - I cannot find where this perticular UserID is being approved, like on Asterisk side. The AuthID is fine..
02:32.15jessterfile - ie: PhoneA has UserID 325 and another has UserID joe  -- the phone that calls through with joe is approved and the call with 325 is given 407 / 403
02:32.35filejesster: okay, is joe a valid username in sip.conf?
02:33.48jessterfile - ah crap. I just realized that i had a matching 325 account .. that was it (my sip.conf file is huge and I overlooked it)
02:34.05jessterfile: ie a local 325 account with different credentials
02:39.08MikeJ[Jayden]wassup all
02:40.16*** part/#asterisk JohnJar (~JohnJar@85-250-71-159.bb.netvision.net.il)
02:46.39GrimStonewhy would asterisk just ignore the voice data coming in on port 5060 from broadvoice ?
02:47.01GrimStoneeven after setting the userid , secret, authid etc. like the email on asterisk-users says , asterisk doesn't seem to be able to "hook up" outgoing calls
02:59.04*** part/#asterisk trelane (~trelane@lan.trelane.net)
03:04.10Mocis there some disk quota problem again ?
03:04.36*** join/#asterisk obelisque (~samifruit@Ottawa-HSE-ppp4015539.sympatico.ca)
03:04.48obelisquehello guys! IAX2 or SIP?
03:04.59Nuggeteither's great.
03:05.10Nuggetit's not like you have to choose.
03:05.24obelisqueI heard that IAX2 was better
03:05.32obelisquebecause it was consuming less bandwith
03:05.32Nuggetthey each have their appeal.
03:05.41Nuggetfor a single channel the difference is immeasurable.
03:05.54Mociax2 support 2 thing, no NAT problems, and trunking make you consume alot less bandwidth on multiple concurent connections(calls)
03:06.11obelisquesound quality stills the same?
03:06.18Nuggetsure, that's a matter of the codec.
03:06.18Mocobelisque, IAX aint a codec
03:06.22obelisqueok ok
03:06.27Mocjust a transport tunnel
03:06.28obelisquesorry im a little noob
03:06.33obelisqueyeah I understand
03:07.02obelisquedoes X-Lite supports IAX2?
03:07.05Mocthe office I setup the other day could do 10 g726 call, once I activated trunking, I could go up to 20
03:07.05Nuggetno
03:07.06rikstano
03:07.32obelisqueany free softphone support it?
03:07.34Mocactually 23 after I resync the DSL ;)
03:07.35Nuggetyes.
03:07.59*** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net)
03:08.02obelisqueok...
03:08.18obelisqueand...
03:08.27obelisquewhat are they?
03:08.31Nuggetgoogle for "iax softphone"
03:08.37obelisqueroger that!
03:08.39Mocobelisque, check voip-info
03:09.09obelisquedamn...grandstream phones dont do IAX2
03:09.57Nuggetthe differences between IAX and SIP rarely matter when you're looking at hardware phones.
03:09.59Moconly vaporphone support IAX2
03:11.09obelisquecause grandstream phones are cheap...
03:11.14*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
03:11.23Nuggetseriously.  you will never care one way or ther other what protocol your phone is using.
03:11.31Nuggetit will have zero impact on your life.
03:11.55Mocobelisque, get the sipura phone instead, probably better
03:11.57obelisqueI need IAX2 protocol...for consuming less bandwith!
03:12.08NuggetIAX2 does not consume less bandwidth.
03:12.23harryvvits in the codec ;)
03:12.30Nuggetyou will not be trunking calls between your server and your desk phone
03:12.46obelisquewhy not!
03:12.53Nuggetbecause you only have one mouth.
03:13.02Mochehe
03:13.08Ron-NaI am looking for  a softphone for my HP5555 PDA, has anybody experience with (a good) one?
03:13.14Mocthat a good one Nugget ;)
03:13.17obelisquewe are using sjphone
03:13.21Luhiwuit's a bug in the human design, why two ears and just one mouth? :)
03:13.26*** join/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net)
03:13.29CoderCRhello all
03:13.40CoderCRi am trying to get some help on an issue i am having with a channel bank
03:14.12harryvvobekusque how has it been as a wifi phone?
03:14.22obelisquewhat IAX softphone you would recommand me?
03:14.22harryvvand what model
03:14.39obelisqueim using sjphone on my ipaq 5835
03:15.12harryvvhow is the clearity and range to a wifi hotspot?
03:16.02CoderCRI cannot get my channel bank to dial out. I see ztmonitor send data and then the channel bank seems to hang up the line. But if i have a phone on the line off the hook, then channel bank dials out.
03:16.14tuxinator_linuxMy cat just laid a really awful poo
03:16.22tuxinator_linuxneed fresh air
03:16.40obelisquearound 5 meters
03:18.03harryvvobelisque,  your kidding right? Usually wifi devices are limited to 100 meters and some times alot more.
03:18.50PatrickDKheh, I have a good 1/4 mile with my wifi phone
03:19.20harryvvPatrick thats great is that in downtown with alot of buildings or in the open?
03:19.36harryvvWhat model do you use and is it built in wifi?
03:19.40rikstaanyone here been using ADM ?
03:19.43PatrickDKthat is at the house
03:19.48PatrickDKoutside is easy
03:19.52PatrickDKinside the house is alot harder
03:20.01PatrickDKso much brick/concrete
03:20.03obelisquewell...voip with my pda sucks
03:20.19obelisquethe microphone gets all the ambient sounds
03:20.41*** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com)
03:20.43obelisqueand its seems to take a lot of cpu...
03:21.03obelisquei suggest you to get a good pda for doing voip
03:21.16harryvvob, who cares about the cpu cycles as long as you can have a decent conversation on it.
03:21.43obelisqueyeah but conversation is not descent
03:21.44harryvvPatrickDK, what model do you use and does it include the wifi or what card does yours have?
03:21.46MocI should try my pocket PC + MY bluetooth headset
03:21.48obelisqueit always cut...
03:21.57PatrickDKharry, wisip
03:21.58obelisqueblue tooth head set?
03:22.16harryvvpatrick, what are you using to do wivi voip
03:22.23obelisquenow thats interesting...
03:22.35obelisquehow much does a bluetooth head set costs?
03:22.36PatrickDKharry, wisip
03:22.40harryvvob, it means you dont have to worry about a coard.
03:22.42Nuggetdo not buy a wisip.
03:22.45harryvv300 dollars
03:22.51Nuggetthe pulver/zyxel wireless phones SUCK.
03:22.59PatrickDKheh, wisip works fine for me
03:23.03harryvvI have seen blue tooth earpieces at 300 dollars.
03:23.07PatrickDKdoes everything it's suppost to do
03:23.10Nuggetthe hitachi one seems be getting favorible reviews.
03:23.10fileLONG LIVE THE HITACHI WIP-5000!
03:23.22Nuggetbut DO NOT buy that pulver piece of shit.
03:23.24harryvvOkay PatrickDK  thats fine, What make and model are you using.
03:23.36PatrickDKnugget, was the hitachi one being sold over a year ago?
03:23.42NuggetI don't know.
03:23.45NuggetI don't think so
03:23.46PatrickDKharry, hmm? they have make and models?
03:23.51obelisquebluetooth earpieces for 300$ ?
03:23.51Mocfile, SEND ME ONE ;)
03:24.00DaminMoc: I just got a Samsung I700 PocketPC w/ an 802.11b Wifi card..
03:24.15MocI got the iPAQ 4150
03:24.20fileMoc: time little grasshopper
03:24.27Mochehe
03:24.28obelisquehey moc...do you have an blue tooth head set with that?
03:24.36harryvvPatrick.. third time...What make and model of pda are you using to so wifi voip? you never said which one you are using.
03:24.41fileMoc: I can have it shipped direct, I *think* it comes customs free due to the NAFTA paperwork
03:24.42Nuggetthe pulver/zyxel wisip can't hop access points, it's got earsplitting ringtones, it's too slow, it's a royal pain to configure, and has flaky firmware.
03:24.55obelisque320$ for wip-5000!!
03:24.57DaminMoc: We need a good IAX2 VoIP client for the PocketPC.
03:25.00PatrickDKharry, I never said I was using pda :) I was I was using a WISIP PHONE
03:25.07DaminMoc: Like FireFly PPC version.. :)
03:25.08PatrickDKwisip phone has built in wifi
03:25.12harryvvokay thats fine. which one
03:25.18Nuggetthe WISIP phone is which one.
03:25.26obelisqueyou guys are rich
03:25.32Mocyea hehe
03:25.45Nuggetthe WISIP is the one you don't want to buy.
03:25.51Nuggetunless you want to buy mine.  :)
03:25.59Nuggetbecause lord knows I don't use it.
03:26.00Mocobelisque, we just dont have girlfriends..
03:26.10obelisqueLOL
03:26.20harryvvPatrick, where have you used the Pulver wisip?
03:26.34PatrickDKwhere? hmm, at HOME
03:26.35Nuggetharryvv seems determined not to take my advice.  :)
03:26.36Moclet me tell ya when I get one, it bye bye telephoneS!! :'(
03:27.00harryvvnugget, yea I have heard some bad things about them. I saw its design...to simple.
03:27.15obelisquehey guys...I need something NOT EXPENSIVE wireless for VOIP (not a PDA)...any suggestion?
03:27.15PatrickDKI setup my own wireless mesh, for in the house, and have a good 12db omni with 500mw power for outdoor access
03:27.16Nuggetthe WISIP phone is quite possibly the worst piece of hardware I've ever owned.
03:27.19Nuggetit's *terrible*
03:27.45PatrickDKobelisque, use what my wife uses, cordless phone + sipura2000
03:27.49drumkillaNugget: I had my hands on a demo version at one point ... for some reason, they accidently left out the SIP config menu
03:27.52MocNugget worst than the barbietone ?
03:28.00drumkillaso you couldn't change any of its provisioning ...
03:28.03harryvvNugget, possibly its just a lemon ? What is a good sip phone or pda combination that works well?
03:28.09obelisquesipura2000?
03:28.18obelisquesounds nice...
03:28.19Nuggetno.  the problems are the firmware.  and the underpowered hardware.
03:28.20DaminFucking Verizon....
03:28.24obelisqueill ebay that
03:28.25Damin"We never stop working for you.."
03:28.28NuggetI have no reason to believe it isn't behaving exactly as they intend it to.
03:28.31DaminWhat a bunch of bullshit..
03:28.40DaminThey just stopped working for me.. My connection dropped..
03:28.44*** join/#asterisk erwinism (~pogz@210.213.143.73)
03:28.45filedrumkilla: VON!
03:28.46harryvvthe pulver phone is made in china ?
03:28.47harryvv:)
03:29.00NuggetI suggest the hitachi phone, but not from personal experience.  Several others in the channel have the hitachi and are happy with it.
03:29.13obelisqueWow!! sipura rules
03:29.22fileI *love* my Hitachi phone
03:29.23MocNugget, I got one on the way.... Well I hope I do ;)
03:29.23fileit's sexy
03:29.38Mocand you got the damn extended battery bas.. ;)
03:29.39drumkillafor the sake of testing!
03:29.40erwinismhello... i am planning to put my own asterisk server in my office. what phones should i use to connect to the server?
03:29.43harryvvbtw, shaw and rogers cable is going to or alredy has started pushing there voip service. I wonder how thats going to affect smaller voip services.
03:29.52fileMoc: I think it comes standard where I get it from
03:30.02Mocthat extremely cool..
03:30.08Daminerwinism: Any phone that you like.
03:30.11harryvvnugget, that is the 300 dollar phone? Ive seen it on voipsupply
03:30.18MocI'll have to get a wifi bridge at the office
03:30.23obelisqueHey! where can I get cheap sipura-2000
03:30.28Daminerwinism: Althought, you might want to stay away from rotary dial phones.. ;)
03:30.29fileI should just take orders
03:30.38harryvvob, on ebay. But be leary of the seller.
03:30.48obelisqueuh ok
03:31.01obelisqueis 70$ too much&
03:31.04DaminHey.. does anyone know if VON will have public WiFi access points?
03:31.10fileDamin: I was wondering that too
03:31.14harryvvob, Ive seen them for under 100
03:31.19brc__wouldn't count on it
03:31.33drumkillaDamin: at the last VON, the trade show floor was so saturated with wireless that we coudln't get it working with stuff right next to each other
03:31.33erwinismDamin do you know a diagram to use asterisk?
03:31.40harryvvDamin would not suprise me if thay do.
03:31.48filedrumkilla: ohhhhh I wonder how my wifi phone will handle it
03:32.05Daminerwinism: No.
03:32.07obelisqueebay is great....
03:32.09harryvvdrumkilla :) not enough bandwith in the area?
03:32.12obelisquecan we sell girls on ebay&
03:32.36*** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net)
03:32.37fileI'm excited about tomorrow!
03:32.44harryvvfile, lucky you
03:32.46erwinismDamin lets say i already setup my asterisk server at my office. how can i connect all the telephone lines from all department?
03:32.46mitchelocfile: you going to von?
03:32.47Moclol, dont forget to charge it ;)
03:32.51fileyessssssss I'm going!
03:32.58Mocmy damn collegue got to go there ..
03:33.03mitchelocfile: nice, i was going to go, but i have finals the week after this
03:33.04fileMoc: 'da phone is charging right now
03:33.08Mocnice hehe
03:33.10mitchelocit's so close, come down and say hi =)
03:33.14drumkillamitcheloc: I'm missing a week of school  :(
03:33.38Daminerwinism: My first suggestion would be to read the documentation.
03:33.51erwinismDamin ok thank you
03:33.56filedrumkilla is VERY silly
03:33.57mitchelocwell i'll extend an invitation to any asterisk people, if you want to drive down to diamond bar (1hr south of LA), I can put you up for a night or two! (PART @ MY HOUSE) =)
03:34.07DaminYou know.. I'm getting an incredible amount of battery life on this Samsung PDA!
03:34.26erwinismDamin how does it lasts?
03:34.27drumkillamitcheloc: haha ... I think we're going to San Fran on Friday
03:34.30harryvvdamin, how long when its on?
03:34.50DaminIt's lasted all day..
03:35.06erwinismDamin without playing on it
03:35.09obelisqueAm I invited?
03:35.18*** part/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net)
03:35.22erwinismok got to read docs
03:35.34DaminNo.. Using it the entire day..
03:35.37mitchelocyea, anyone who knows what asterisk is is invited heh
03:35.51DaminHell.. I just got it yesterday, so hell YEAH I am going to use it all day! :)
03:36.02fileugh I'm hungry
03:36.08drumkillaI got a PDA a couple months ago ... and it doesn't turn on anymore :(
03:36.40drumkillathe last thing I did was type "apm -s"
03:36.40DaminI got a Double Cheese, Double Pepperoni Pizza delivered...
03:36.40erwinismdrumkilla needs a truck battery i think heheh
03:37.24mitchelochey, can anyone answer this for me, i neede a second number (via broadvoice) and they say the way i can tell the difference is via distinctive rings....
03:37.34obelisqueciao guys
03:37.37mitchelocnow can asterisk take out the header in the packet initiating the connection?
03:37.38harryvvdrumkilla, I dont think its ever been discussed but has anyone though of putting a wifi hotspot in a dense residential area and selling wifi ata settop boxes to residents in the area?
03:37.43mitcheloci'd like to ring a different phone when that number comes in
03:37.44DaminHmmm...
03:38.17mitchelocharryvv: the range on a wifi box would make the effort impossible
03:38.21DaminI just realized that the music for the movie "Backdraft" is also used as the music for the Food Channel series "Iron Chef"
03:39.23shmaltzdoes slackware have an AMD 64 arch port?
03:39.26*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfi1r.dialup.mindspring.com)
03:39.45Ron-Naobelisque: how is sjphone on the ipaq?  I tried Xten, but it makes my ipaq a phone - cannot use for anything else, ... that is nonsense, if I want that I buy a WiFi phone!!!
03:39.45cbachmanDamin, they've discussed that before on the iron chef newsgroup
03:40.24mitchelocanyone on that distinctive ring packet?
03:40.45obelisquesjphone on my ipaq 3850 is unusable
03:41.00*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
03:41.59harryvvmitch, range is a function of db or wattage out. But with some properly orinated yagis it may work. There has been some discussion on popular science to place a heliostat in the ionospere with wifi repeater placed on it and blanketing a area of the city. My guess is the pm who is putting this together may have the initial wifi power out waived to a higher rating since it will be away from the general population. The barrier thay will
03:41.59harryvvhave to overcome is to fine a way to keep it afloat during night hours.
03:42.04obelisqueits very slow and when I get the stream, discussion is always interrupted, lets say each 1second I loose 500ms of voice...
03:42.46GrimStoneanyone have problems with outbound calls on broadvoice ?
03:43.03mitchelocobelisque: are you setting up 600 phones?
03:43.28GrimStoneeven after setting the userid , secret, authid etc. like the email on asterisk-users says , asterisk doesn't seem to be able to "hook up" outgoing calls . it just ignores the voice packets on port 5060
03:43.36obelisquemmm
03:43.39obelisquenot yet...
03:43.46obelisquewhy?
03:44.04harryvvI figure that I would get some feed back on this.
03:44.26GrimStone..... no one here uses broadvoice ?
03:44.34harryvvno
03:44.51mitchelocjust wondering what you were talking about earlier?
03:44.53jsolaresi currently use voipjet and nufone
03:45.01MikeJ[Jayden]I have an account w/ broadvoice, but I rarely use it
03:45.01MikeJ[Jayden]sorry
03:45.15jsolaresmitcheloc: why not a second account? that would definetely be easier
03:45.27*** join/#asterisk TheEmperor (TheEmperor@218.111.50.173)
03:46.04mitchelocjsolares: heh cause i only want to pay $2 a month for this...it's actually for an access system so i can get into my house =)
03:46.06jsolaresatleast beacuse i have no ide how to use distinctive rings yet
03:46.15jsolaresidea*
03:46.18mitcheloci'm thinking theres a way to read the sip packet properly
03:46.27obelisqueyeah...
03:46.32mitchelocjust wondering weather the distinctive packet is on the first request
03:46.37MikeJ[Jayden]alert-info
03:46.52obelisqueIts a project...I have to talk about this to my boss first... Cause the optical fiber will go in soon
03:46.56MikeJ[Jayden]=distinctive rings... it will be on the wiki
03:46.59rikstawould anyone be interested in helping me in the development of ADM?
03:47.06mitchelocmike: looking it up
03:47.45mitchelocadm = ?
03:47.53rikstauh http://adm.hamnett.org
03:48.50brc__what is adm
03:48.54mitchelocMikeJ[Jayden]: the distinctive ring i'm talking about is slightly different, i want to catch it on the incoming, not the outgoing
03:48.56brc__ah
03:49.18drumkillariksta: where are you a student?
03:49.30rikstaManchester University, UK
03:49.48drumkillacool
03:50.14MikeJ[Jayden]sane thing on sip, just getting instead of setting
03:50.24rikstaam looking for some help with the project
03:51.04drumkillariksta: have you tried the -dev list?
03:51.17MikeJ[Jayden]riska- are you writing adm?
03:51.24rikstayes mikegrb
03:51.37rikstadrumkilla: i posted a while ago, didn't get much of a response, was a bit disappointing
03:51.47rikstayes MikeJ[Jayden]
03:51.49drumkillaI totally understand.
03:52.09rikstadidn't want to post again too soon, might piss people off
03:52.11*** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net)
03:52.31rikstaactually, i posted to -users
03:52.43MikeJ[Jayden]especially that flemming guy.. I hear he needs anger management therapy he is so bad
03:53.20rikstahehe
03:55.04*** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net)
03:57.36harryvvmikej, who is this flemming guy
03:57.48MikeJ[Jayden]kpflemming
03:57.54harryvvon this channel?
03:57.57MikeJ[Jayden]you don't read the dev list
03:58.01*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
03:58.02brc__heh
03:58.06brc__harryvv, what about him?
03:58.07JunK-Ylo guys
03:58.09MikeJ[Jayden]yeah
03:58.11harryvvno i dont. But I would like to meet him in person.
03:58.20brc__are you in phx?
03:58.21MikeJ[Jayden]go to von
03:58.24drumkillahe'll be at VON :)
03:58.28brc__yeah
03:58.43harryvvI deal with people like that all the time. All the time since the millitary.
03:58.44harryvv:)
03:58.59harryvvI will miss von I am up here in vancouver.
03:59.13harryvvBut may give Toronto a shot in april
03:59.30MikeJ[Jayden]you are closer to san jose than toronto
03:59.52shmaltzAndrew?
04:00.01*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
04:00.14harryvvYea that is probebly true. But then, I have not seen eastern canada :)
04:01.47shmaltz~weather ketl
04:02.04*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
04:03.59MikeJ[Jayden]~weather ksjc
04:04.36MikeJ[Jayden]yeah.. all I have to say is I have 6" of snow on the ground and it sucks here...
04:04.54drumkillaI rarely see any snow
04:04.58drumkilla:p
04:05.52MikeJ[Jayden]yeah, forget the south...you guys never have to deal w/ real weather :)
04:05.52shmaltzMikeJ where you located?
04:05.54drumkillaI get hurricanes
04:05.55shmaltz~location
04:05.56jbotACTION is omnipresent
04:06.02shmaltz~sex
04:06.03jbotI'm gay
04:06.25shmaltz~pregnant
04:06.26jbotYes, shmaltz, and it's your child.
04:06.26MikeJ[Jayden]hehe
04:06.27MikeJ[Jayden]detroit
04:06.34shmaltz~lol
04:06.35jbotlol is, like, stands for Laughing Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead.
04:07.14shmaltz~sex
04:07.15jbotI'm female
04:07.19shmaltz~play
04:07.21jbotand now, for something completely different
04:07.30shmaltz~chat
04:07.48shmaltz~wtf
04:08.09drumkillashmaltz: you can /msg jbot, too, you know ...  :p
04:08.39MikeJ[Jayden]hey, let him be.. they are having a very nice conversation :)
04:08.45shmaltzdrumkilla, thanks, I know, but if the channel is quiet why not give some ppl a laugh, thanks again
04:09.02shmaltzask jbot about me
04:09.04drumkillajust playin'
04:09.22shmaltz~shmaltz
04:09.22jbotyou are, like, annoying the channel by playing with jbot
04:09.39filedrumkilla: staying up all night is bad, mmmk?
04:09.50shmaltz~sleep
04:09.51jbothmm... sleep is overrated, and a poor substitute for caffeine
04:09.56shmaltz~wife
04:09.56jbothmm... wife is the Wide Interface File Engine
04:10.06shmaltzbye gtg
04:10.08shmaltz~bye
04:10.10jbotl8tr
04:10.14MikeJ[Jayden]~wtf shmaltz
04:10.36brc__~data
04:10.37jbotDon't Ask To Ask. Just ASK
04:10.43MikeJ[Jayden]me either :)
04:10.45shmaltz~quit
04:10.46jbotACTION what bananas eat
04:10.54shmaltzoh i meant /quit
04:11.54brc__muhaha HAHAHAH!
04:12.09MikeJ[Jayden]EVIL
04:12.47MikeJ[Jayden]drunkilla, what year are you in school?
04:12.53drumkillaJunior
04:13.08drumkillaundergraduate
04:13.23MikeJ[Jayden]I miss that...
04:13.32MikeJ[Jayden]I need to go back to school
04:13.34MikeJ[Jayden]again
04:13.43drumkillait's tough ... I have a lot on my shoulders right now
04:13.56MikeJ[Jayden]why do you keep it there?
04:13.57drumkillaI really don't have much time to work on Asterisk while I'm in school full-time :(
04:14.03drumkillait makes me sad
04:14.12MikeJ[Jayden]y
04:14.42MikeJ[Jayden]just write a grant proposal to pay you for 6 months and then have some real fun
04:15.02drumkillanot a bad idea!
04:15.11drumkillabut I want to graduate asap
04:15.21drumkillaso I can get to working on Asterisk full-time :)
04:15.44harryvvdrum, how long you been coding in c
04:15.52drumkillanot very long at all
04:16.03MikeJ[Jayden]I would love too.. need to get my other work done first at the moment
04:16.31MikeJ[Jayden]sigh.. need to just get enough clients and quit my job is what I need to do
04:17.58harryvvsigh, how long you been at it
04:18.25MikeJ[Jayden]huh?
04:18.42harryvvI would say stay with it at least a year and if the income exceeds your existing one.
04:18.57harryvvThats from a common biz experiance.
04:19.48drumkillaharryvv: would you believe me if I said a year ago, I didn't know any C?
04:20.20MikeJ[Jayden]y
04:20.53MikeJ[Jayden]I have some stuff lined up at the moment.. unfortunately I had an office move and a dozen t1 hotcuts this weekend
04:21.24MikeJ[Jayden]then, after I get the lines up, I have wiring guys going behind me and unhooking things...sigh
04:21.37harryvvThats good. But the problem is your still in collage. What customers really want is confidence in the product and the service. Graduate and I am sure your base will increase. I will not to this day deply a asterisk system untill I am absolutly sure it will not crash or at least give me some warnings it may crash. Basicly the asterisk system really needs to be hardened.
04:21.49MikeJ[Jayden]me?
04:21.55MikeJ[Jayden]please!
04:22.03drumkillaharryvv: well, I don't mean consulting
04:22.04MikeJ[Jayden]I am maried with kid
04:22.14harryvvdrum, you mean dev right?
04:22.25erwinismhello, just wanna ask.. channel bank is a hardware?
04:22.27drumkillayeah
04:22.33harryvvdrum thats cool.
04:22.34MikeJ[Jayden]yes
04:22.51harryvvjust hope india does not beat you :)
04:22.55drumkillaha
04:23.08harryvvmy supervisor at M$ was east indian.
04:23.09drumkillawell, if I weren't in school right now, I'd be in Huntsville working for Digium
04:23.23harryvvick alabama?
04:23.27harryvv:)
04:23.30drumkillam hm
04:23.37harryvvdont thay allow remote employment?
04:23.44MikeJ[Jayden]M3718
04:23.52drumkillait's a time issue, really
04:25.00erwinismif i use asterisk, how much cost would i save from buying any chap pabx hardware on the market?
04:25.04harryvvI had a friend who moved from Atlanta and gave up everything to move to Bellingham. Everyone there protested and he had a big income drop as a result but he is alot happier. Bellingham has the cleanest air in the country.
04:25.26harryvver, depends what you do with it;
04:25.27MikeJ[Jayden]drumlilla, you wanna post a response to that one.. cuz mine is gunna be, "do you want fries with that" and that probably isn't that helpful
04:25.40Mocerwinism, it depend on how you want to implement it
04:25.53MikeJ[Jayden]erwinism? huh?
04:26.09drumkillaI've got to go to bed ...
04:26.10Mocyou could make it extremely cheap if you got PC and can use softphone with 15$ headset
04:26.17drumkillaI have an early flight in the morning
04:26.46drumkillag'night everyone *waves*
04:26.48erwinismMoc, im because i want to have a PBX on my office. i want to know the advantages of asterisk and pbx hardware.
04:26.53MikeJ[Jayden]niht
04:26.58erwinismMoc, ..because i want to have a PBX on my office. i want to know the advantages of asterisk and pbx hardware.
04:27.12MikeJ[Jayden]asterisk is a pbx
04:27.36Mocerwinism, well * can do it
04:27.45Mocand it will save you alot on liscencing / upgrade
04:27.48MikeJ[Jayden]and probably cheaper
04:27.52MikeJ[Jayden]much cheaper
04:27.56Mocyea
04:28.14MikeJ[Jayden]for example, you can pay 500 ish for a pri card for asterisk
04:28.26erwinismok, i will present that on the board.
04:28.32erwinismthanks moc and mike
04:28.44MikeJ[Jayden]or 3-10 times that for one on a propriatary phone systemm
04:28.45MocI just installed * the other day to someone who bought a Avaya PBX, for 8k$, but he didnt had the little liscence card, so PBX is useless, it resold it on ebay and got * With Polycom phone ..
04:29.20MikeJ[Jayden]hey moc, what phones do you like...
04:29.32Mocerwinism, most important thing to do is, make a list of the feature you want from your PBX
04:29.44MikeJ[Jayden]for office enviornment, high functionality need
04:29.45Mocthen it will help you check the price of all the PBX on the market
04:29.47*** join/#asterisk Landrocker (~landrocke@port-222-152-54-115.fastadsl.net.nz)
04:29.53MocMikeJ[Jayden], Polycom
04:30.04MocPolycom IP 500 or better the IP 600
04:30.05MikeJ[Jayden]ip500?
04:30.10MikeJ[Jayden]k
04:30.26erwinismE100P is 600$
04:31.02Mocerwinism, your in the UK ?
04:31.12MikeJ[Jayden]need to start specing phones for a job and I have only had experience with the cisco phones, and while they are nice, I have not been overwhelmed by them and I think it is time for somthing new
04:31.47mitchelocanyone need a t100p? i've got one, make me an offer =)
04:32.05erwinismmoc im in philippines
04:32.13mitcheloci need to make my carpayment on my clk320, so be generous!
04:32.41MocMikeJ[Jayden], samething for me, they are nice (except of a couple of flaws), but Polycom is just so much flexible ..
04:32.50mitchelocactually jk, i wish i had one of those cars...they are nice
04:32.56MikeJ[Jayden]are they backlit?
04:33.02harryvvahh the PI :)
04:33.04MocMikeJ[Jayden], no
04:33.14mitchelocmm the cisco backlit, that would be awesome
04:33.23MikeJ[Jayden]maybe I will order one to play with
04:33.57MocMikeJ[Jayden], 1 thing you need to understand, get the DEFAULT config, and only change the minimum, then once it work, you can play with all the options ;)
04:34.13MikeJ[Jayden]k
04:34.14MocMikeJ[Jayden], check at the admin manual : http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf
04:34.27Mocyou can change nearly EVERYTHING
04:35.03MikeJ[Jayden]and change is good :)
04:35.11MikeJ[Jayden]spread the word
04:35.19MikeJ[Jayden]you going out west moc/
04:35.20MikeJ[Jayden]?
04:35.35MikeJ[Jayden]low battery...
04:35.37Mocwhy should I go west ?
04:35.38MikeJ[Jayden]time to sleep
04:35.42MikeJ[Jayden]von?
04:35.56Mocha nope, I nearly go, but was too late to ask for tickets
04:36.14MikeJ[Jayden]ok.. come to cluecon in august..
04:36.22Landrockeranyone know if it's possible to use a keypad button instead off hook-flash to use call parking, etc? (the '#' key for instance)
04:36.23MikeJ[Jayden]battery dying
04:36.26MikeJ[Jayden]gotta go..
04:36.27*** join/#asterisk threeo (~threeofiv@adsl-146-114-25.mia.bellsouth.net)
04:36.29MikeJ[Jayden]goodnight
04:37.57*** part/#asterisk Bruns (bruns@pool-141-153-151-58.nwrk.east.verizon.net)
04:37.58hardwireyay
04:38.04Grooby??
04:38.06hardwire48 voice channels over 802.11a backbone
04:38.12hardwiregsm
04:38.15hardwireiax trunking
04:38.18Groobynice
04:38.31hardwireI did super extension transfering
04:38.32hardwireheh
04:38.41Groobyand i have no idea what that means
04:38.41hardwiredial remote.. dials local exten + 1
04:38.41Landrockerhardwire, awesome - I'm setting up something similar over the next month or so
04:38.42Grooby:-D
04:38.43hardwirewhich dials back
04:38.46hardwireI hope thats adequite
04:38.52hardwireLandrocker: I bought some wrap boards
04:38.58Landrocker...?
04:39.00hardwireand a wrap outdoor enclosure
04:39.05hardwireand some 30dbi dishes
04:39.10hardwiregeode 233s
04:39.13Landrockerah nice
04:39.18hardwirew/ madwifi supported mini-pci cards
04:39.26hardwireit took a while to get it linked at 3 miles
04:39.31Landrockerwe have the network in place - I just need to get VOIP working over the top of it
04:39.36Landrockerwhat's the latency like?
04:39.42hardwire8 times faster than a t1 so far
04:39.55hardwiredoing a b/w test
04:39.59hardwiregive me a sec
04:39.59*** join/#asterisk Newbie___ (some@218.111.157.90)
04:40.03Landrockerk
04:41.23hardwire<PROTECTED>
04:41.24hardwiresent 36 bytes  received 238556657 bytes  875437.41 bytes/sec
04:41.24hardwiretotal size is 515579904  speedup is 2.16
04:41.29hardwirecompressed hard drive image over the backbone
04:41.36hardwireapparently almost 2/1 compression
04:41.41hardwireso.. maybe not a good test :)
04:41.41Landrockeron that note - does anyone have any tips on taking latency down? using the echo test it sounds like I'm getting about 500ms round-trip, but that's just over 100mbit ethernet so it should be faster, right?
04:41.48Landrockerlol
04:41.55hardwireLandrocker: dude
04:42.03hardwireis your machine fast enough?
04:42.18hardwireI am using 233 mhz machines to pass atleast 4 channels at a time
04:42.25eraser`Landrocker: how long is the cable?
04:42.34erwinismmoc if i but E100P, what else hardware should i need to setup a PBX on my office? my setup is one pot line to connect to office from outside. and dial 9 to make outside calls.
04:42.44erwinismbut = buy
04:42.47Landrockermy test was just using my laptop which is a cel-m at 1.4ghz
04:43.01hardwire64 bytes from 10.0.9.5: icmp_seq=1 ttl=64 time=1.73 ms
04:43.01hardwire64 bytes from 10.0.9.5: icmp_seq=2 ttl=64 time=0.351 ms
04:43.01hardwire64 bytes from 10.0.9.5: icmp_seq=3 ttl=64 time=0.316 ms
04:43.03hardwire64 bytes from 10.0.9.5: icmp_seq=4 ttl=64 time=1.00 ms
04:43.06hardwireits going to be just fine
04:43.13hardwireafter going back and forth 48 times.. its noticable
04:43.13harryvvI wonder how many sip connection I could get with my opteron 244 with gigabit :)
04:43.14hardwirehowever..
04:43.24Landrockerthere's probably about 15m of cable in between the two machines I was using to test
04:43.29mitchelocgrrr, mirc is beeping me speakers, but i dunno why
04:43.30hardwirewe are going over sat once it goes to the internet
04:43.30hardwireso
04:43.30hardwirehah
04:43.31Landrockerplugged into the same switch
04:43.35hardwireit doesn't matter how slow my wireless link is
04:43.41hardwireI am slapping 600ms on top of it
04:43.48hardwirewith a shringing jitter buffer
04:43.49Landrockerhmm
04:43.57hardwirethat starts at around 700
04:44.02Mocharryvv, I got 20 call withg 800kbits sec
04:44.09Mocwith g726
04:44.10Landrockereraser`, any ideas?
04:44.17erwinismmoc
04:44.31Mocyes ?
04:44.42erwinismmoc if i buy E100P, what else hardware should i need to setup a PBX on my office? my setup is one pot line to connect to office from outside. and dial 9 to make outside calls.
04:44.55Mocyou got a E1 ?
04:44.56eraser`so to make sure I understand, you're pinging a machine that you're on the same switch with and both are configured as fastethernet and you're getting ~500ms
04:44.57jsolareslots of sip/iax phones
04:45.12mitcheloce1 for one phone?
04:45.13erwinismMoc. no
04:45.15Landrockerno, ping shows about 1ms latency
04:45.21Mocerwinism, no need for the E100P then
04:45.22eraser`ohh
04:45.26Landrockerbut call latency is definately higher than that
04:45.31mitchelocwhat country are you in earser?
04:45.35erwinismi Only got One ordinary telephone line
04:45.40eraser`US
04:45.45jsolaresyou need an fxo device erwinism
04:45.46Mocerwinism, then you only need a FXO card/device
04:46.15Mocerwinism, E100P is a card for Digital Line called E1, that support up to 23 digital channel
04:46.22Moc+ 1 Data channel
04:46.24eraser`Landrocker: sorry but I don't know asterisk well
04:46.30erwinismMoc, what else hardware should i need if i got FXO
04:46.38Landrockernp, thanks anyway :)
04:46.44jsolaresdo you want to stick to analog phones or move to voip phones?
04:46.50jsolaresthat's what you should be asking yourself next
04:46.56Mocerwinism, well if you were in the US/Canada, a X100P, or a TDM400P + FXO module, or SPA-3000 would do the trick
04:46.57eraser`I'd assume it may have something to do with the other machine, you said it was 200MHz?
04:47.02mitcheloceraser: e1 is for europe
04:47.06erwinismjsandnes yes hhehe
04:47.10Landrockernope, they're both above 1ghz
04:47.13MocI donno if those device will work where you are located
04:47.16eraser`ah
04:47.26eraser`something is seriously defunct
04:47.31Landrockerhmm
04:47.33*** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net)
04:47.48rikstacan you use an intel voice modem as a cheap type of fxo
04:47.58erwinismjsolares i want to stick to analog..
04:48.00Mocriksta, you can try, but dont ask us !!!
04:48.10*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
04:48.21rikstaMoc: i'm not asking how...i'm just asking if it's poss :)
04:48.28jsolareserwinism: then you'll need fxs devices, btw sticking to analog phones could be more expensive depending on what voip phone you end up with
04:48.39Inv_arpriksta: google says it is
04:48.41mitchelocriksta: yes you can
04:48.53mitchelocriksta: if you want the model number i've got it, but can't get it till monday
04:48.53Mocriksta, when I say that is, everything is posible.. just might be supported
04:48.57Mocmight NOT
04:49.05jsolaresalthough i've seen good prices for fxs devices
04:49.15MocI havent seen 1 person in over 1 year play with other than x100p card
04:49.30mitchelocmoc, count me as the first then ;)
04:49.46harryvvMoc what do you mean by play with other?
04:49.54Mocok.. I mean considering getting a standards Voice modem to work with *
04:50.16erwinismjsolares: i have 20 deparments in the office. so that means i need 20 FXS cards?
04:50.26mitchelocwhich should work!
04:50.31harryvvokay I see what you are saying. Are there other voice modems other then the x100p that will work with asterisk?
04:50.32Mocerwinism, not exactly
04:50.45mitchelocerwinism: nope, thats not possible unless you have multiple servers
04:50.46Mocerwinism, you say you want to be analog only ? no IP phone ?
04:51.02mitchelocdigium has cards with 4 ports each, but choose if you want voip phones first though (probably the better route)
04:51.02erwinismMoc yes, analog only
04:51.12jsolaresyeah, not exactly there are fxs devices called ata's like digiums iaxy, or sipuras or grandtream, or even linksys pap2 (or something)
04:51.26Mocerwinism, ish, then you need to get standards Home Analog phone ;) + a channel bank + the T1 Card ;)
04:51.27mitchelocso 5 digium cards, or 1 t100p card and a nice little adtran box to break it out to a punch block
04:51.43jsolaresif you're lucky with ebay you could score a t1 channel bank for cheap
04:51.57mitchelocjsolares: i got one for free =)
04:52.04jsolares:p
04:52.11mitchelocjust go bug someone to install asterisk for them, use a t100p to replace their adtran, then jack the one on the wall
04:52.34Mocerwinism, why you dont want to go IP ?
04:52.54erwinismMoc, because my phones here are analog..
04:53.11mitchelocreplace them?
04:53.13Mocso you want to reuse your current analog telephone ?
04:53.22jsolaresyou have to take the channel bank + t1 card into account, and see if it wouldnt be better to replace the analog phones
04:53.30Mocyea
04:53.33tuxinator_linuxI got a dialtone on my TDM400P !
04:53.38erwinismMoc how much would cost a digital phone?
04:53.42erwinismip phone i mean
04:53.43erwinismhehehe
04:53.48Mocerwinism, depend, there is different type
04:53.49jsolares80$ and up
04:54.14Mocthe recommanded model cost 180$
04:54.14erwinismim making some plans here ")
04:54.23jsolaresa channel bank + t1 card could be 100$ per channel
04:54.55erwinismMoc what if i will switch to IP phone? what hardware should i need?
04:54.58Mocor for lower, you drop down to 84$ for a cheap buisness phone
04:55.12Mocerwinism, ethernet card ;)
04:55.13jsolaresnothing more than the asterisk box with the fxo device
04:55.19jsolaresyeah with an ethernet card hehe
04:55.27rikstaand a switch ;)
04:55.41jsolaresyou could also hook it up to a voip provider to have long distance calls for cheap
04:55.46mitchelocor $12/channel if you do what i said to get a free channel bank =)
04:55.48Mocerwinism, you could also use SoftPhone
04:55.55erwinismMoc, Ethernet card, one FXO for outside calls and one FSX for Imcoming calls?
04:56.09Mocerwinism, FXO for connecting to your POT line
04:56.22mitcheloc** actually i got my t100p for $280 so thats not right, it's around $20/channel
04:56.23jsolaresyeah, it takes care for calling and receiving
04:56.25erwinismwow thats cheap
04:56.30Mocno need of FXS (unless you want to plug the fax on it)
04:56.44erwinismok thanks i got it
04:56.49erwinismthis is what i need
04:56.57Mocerwinism, softphone cost make the cost at about 20$ per phone too
04:57.00jsolaresit might also be cheaper to go with ata's that go from network to analog phone like the iaxy
04:57.10Moclol softphone is free + headset = 20$
04:57.25erwinismhehe
04:57.31jsolaresnot cheaper than sofpthone or stolen channel bank tho ;p
04:57.52mitchelocstolen channel bank, and 1/2 stolen t100p cards (someone sold me one here for $280) hehe
04:58.16Mocerwinism, what make * powerfull too is, you can modify it as your need, and you can interconnect all those type of device
04:58.42erwinismokay
04:58.45erwinismi got it
04:58.53jsolaresindeed it is, i have 2 voip providers for redundancy to make cheap calls to the us :)
04:59.06Mocyou can have FXO/FXS/ISDN/VoIP(IAX/SIP/MGCP/SKINNY/H323..)
04:59.11jsolaresboth are cheaper than making a local call :|
04:59.33jsolaresonly if you're a masochist moc
04:59.40jsolares:p
04:59.53mitchelochey btw THIS IS A PSA!!!! please SPREAD THE KNOWLEDGE >>>>> I learned today that, most phone companyes (99%) leave the 911 service live on the phone lines running into your house...so you can have real 911 service without paying for a phoneline @ your local telco. <<< (yes some of you may know this, but many do not)
04:59.55Mocerwinism, over here, I got no Analog line, it all done over Internet and VoIP provider, my incoming and outgoing
05:00.28erwinismwow thats great
05:00.36erwinismi will plan for this.
05:00.48mitchelocso buy an fxo card and hook it up to your house system, even if theres no dial tone, the 911 service might and probably will work
05:00.52jsolaresyeah, depending on your provider it might be cheaper than having an anolog line
05:01.19Mocmitcheloc, even if no dialtone, that cool
05:01.27MocI never tryed that
05:02.03erwinismmoc, FXO will handle the outgoing and incoming calls ?
05:02.04mitchelocyep, it's some liability they didn't want to be responsible for (i.e. you did not pay your bill, then it'd be like them saying screw you mr. dying man!)
05:02.22Landrockerthey're required to by law in the states iirc
05:02.23Mocrecently our local provider keep the dialtone, and you can dial, but get a prompt that if we want the service, we need to call the 611 ;)
05:02.30Mocerwinism, yes
05:02.36erwinismok got it..
05:02.43mitchelocbut your 911 service will work, i was about to buy a real phone line for this
05:02.56mitchelocbut now i don't have to =)
05:04.15erwinismok jsolares and moc i have to go... reading asterisk  manuals
05:04.20jsolaresgluck
05:04.20erwinismthanks
05:04.40Landrockeranywho - anyone got any idea on the flash-hook thing?
05:05.11Landrockerxlite can't flash-hook and my voip hardphone probably won't get here for a week or two
05:05.44Landrockeralternatively, are there any softphones that do have flash?
05:07.59harryvvmit, buy a real phone for 911? thats a good idea. BTW, a federal building where I have worked has a seperate phone for calling the fire department. I never asked but is it my assumption that is a strait analog phone to the CO that does bypass the pbx in the event it fails?
05:08.01ManxPower~docs
05:08.02jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
05:08.02rikstais voipuser.org down?
05:08.45*** part/#asterisk oej (~oej@64-151-42-78-dhcp-kc.everestkc.net)
05:08.53Inv_arphhmm can i set for ex.. #7  to perform an action during a conversation with *?
05:14.16*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
05:15.25Inv_arpcome on i thought that was an ez one
05:16.10brc__eh
05:16.12brc__you can't
05:16.29brc__well
05:16.30brc__depends
05:16.52brc__you can set a key to start recording while two channels are bridged
05:17.01brc__I dunn remember how
05:17.18brc__so if you can dive into the code it's possible
05:17.54jsolaresyou could use my new noob super app that's the inverse of waitforsilence that waits on sound, and start doing something only when the dsp thinks there's no silence :XD
05:20.33Inv_arpheh k
05:21.22jsolaresit's great for monitoring extensions on an avaya definity with service observing
05:24.43Inv_arpheh trying to come up with a agi/php script to handle agents and implement with sqlite...   trying to figure out how i can prgram keys to say "agent on break" or "ACW"
05:24.59rikstaInv_arp: i wanted something like that too
05:25.20rikstakinda like the avaya phones, where you can press keys to log into lunch or ACW etc
05:26.12Inv_arpriksta: exactly... hmm i might have to implement a "middle man" app to do something like that for me
05:26.53rikstai was thinking the same thing, maybe write an app that they have running on the computer that they can press buttons on, that integrates with asterisk manager
05:27.23Inv_arpriksta: hmm im thinking python/twisted  combination
05:27.33jsolareswhy not make an agi that's on an extension, and when the agent calls that extensions it executes the agi putting them on break
05:27.55rikstatrue
05:27.57ta[i]ntedlooks like broadvoice is down for asterisk users
05:28.04rikstavoipuser.org is down too
05:28.06rikstaannoying
05:28.14ta[i]ntedreally
05:28.17eraser`how reliable is broadvoice usually?
05:28.28ta[i]ntedsome tech guy just told me its some kind of SIP INVITE auth issue
05:28.33Inv_arpjsolares: but at work i ca put myself on flashing break during a call, then when call ends im on it
05:28.33loudreally ? down ? let me check
05:28.44jsolaresah ic
05:28.45ta[i]ntederaser` i'd say it's got problems.. international calling is ass
05:28.46loudoh damn
05:29.14Inv_arpBV is down? incoming werks fine
05:29.25jsolareswell if you have two line phones you can dial the break extension on the other line :p
05:29.34brc__O M G! http://www.rasterwerks.com/dev.public/phosphor_alpha_4_248.htm
05:29.44brc__check it out!
05:29.49brc__play with me
05:30.06ta[i]ntedInv_arp really everything is down for me
05:30.15loudsame here.
05:30.18*** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.res.rr.com)
05:30.24D1ng0anyone alive ?
05:30.26Inv_arpta[i]nted: on a call right nno on BV (incoming)
05:30.31Inv_arperr now
05:30.38D1ng0Inv_arp, yeah for me too
05:30.45ta[i]ntedInv_arp using asterisk or ip phone
05:30.51D1ng0asterisk here
05:30.55ta[i]ntedInv_arp BV tech said it is specific to asterisk
05:30.56Inv_arpta[i]nted: asterisk
05:31.10D1ng0they are full of it
05:31.14eraser`any suggestions in terms of a provider then, New York/New England area
05:31.16Inv_arpta[i]nted: asterisk -> HT486
05:31.21D1ng0asterisk worked fine until they made changes today
05:31.34Inv_arperaser`: outgoing/incoming?
05:31.38loudyou think ? becaose of the invite thing ?
05:31.41ta[i]ntedlooks like they changed SIP auth
05:31.42rikstawhats the key combo to reset a 79xx phone
05:31.43D1ng0yes
05:31.44eraser`both
05:31.45loudbacause rather
05:31.54loud**#
05:32.04rikstanop
05:32.20Inv_arpwhats wrong with BV? cant recieve incoming? outgoing?
05:32.25loudtype reset through CLI, ill reset everything :)
05:32.26D1ng0outgoing works
05:32.35D1ng0you need to add a few lines to asterisk for it
05:32.42D1ng0incoming is still broken
05:32.49loudwhich likes D1ng0
05:32.51Inv_arpi just called myself  on BV thru my cell
05:32.56*** join/#asterisk krilloz (majestic@220-253-7-238.VIC.netspace.net.au)
05:32.58*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
05:33.08krillozhey
05:33.13D1ng0see http://lists.digium.com/pipermail/asterisk-users/2005-March/092953.html
05:33.19Inv_arpcell->BV->*->HT486
05:33.29D1ng0it fixes outbound, but not inbound
05:34.59Inv_arphmm my inbound still works.....  knock on wood
05:35.12*** join/#asterisk DyOS (~me@ip68-2-153-157.ph.ph.cox.net)
05:35.13D1ng0Hrmmm mines been broekn for hours
05:35.39Inv_arpD1ng0: lemme show u my sip.conf for BV to compare
05:35.39DyOSif anyone is interested in making a few bucks msg me i need help setting up a script in asterisk that won't work for me
05:35.46BoRiS<in an Australlian voice> "The Dingo ate your baby!"
05:35.46DyOSi will pay via paypal if anyone is interested
05:35.55D1ng0Inv_arp, /msg it to me
05:36.02rikstayou know on a cisco 7940, where you have the buttons on the right for each line, how do you set the name that is displayed, can you have alphanumeric chars instead of just the line's number, eg "line 1" rather than 1000 ?
05:36.11D1ng0BoRiS, LOL
05:36.18BoRiS:)
05:36.52krillozI was attacked savagely by a Dingo once, in the outback..
05:36.59D1ng0damn i cant even get BV customer support on the phone
05:37.05*** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
05:37.09D1ng0krilloz, no way, wasnt me :)
05:37.10loudriksta, yes. line1_shortname: "something"
05:37.13BoRiSlol
05:37.14rikstathanks :)
05:37.21rikstai had line1_displayname :)
05:37.27krillozas you can see I'm from .au , so I must be telling the truth!
05:37.40D1ng0krilloz, im Aussie also hence the nickname
05:37.46D1ng0i just like in the USA
05:37.51krillozahh
05:37.53D1ng0ehh live in the USA
05:38.02krillozI see
05:38.15krillozusing asterisk to call back home?
05:38.20D1ng0though ive been debating going back home to brisbane
05:38.39krillozahh, brisbane is a bit of a backwater though isnt it? ;)
05:38.44D1ng0krilloz, yeah thru BV which is now very broken
05:38.49D1ng0heh nahhhhh
05:41.18*** part/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
05:44.10loudwell
05:44.24loudinternational calls work .. just tried.
05:44.33jsolaresahh fun to see all of the angry bv costumers in asterisk-users
05:44.51loudhah, the wife one.
05:45.03loud"i can't stand my wife when she can not make calls".
05:45.04jsolaresi'm not sure what's worse, bv not notifying customers or the customers acting like spoiled children
05:45.42jsolaresyeah that one was funny
05:45.59D1ng0well it would be nice if BV inbound worked, i got outbound working
05:46.30DyOSI'm having a few problems setting up some things in asterisk for my small business i'm wondering if anyone is intersted in helping me out I will pay 20/hour for help if anyone is interested email me at irc@DynamicOnsiteSolutions.com
05:47.09GrimStoneD1ng0: i added those lines but even outbound doesn't work yet for me
05:47.41GrimStonelike the voice data comes in on port 5060 from BV , but asterisk seems to just ignore it
05:47.44jsolaresare you getting registered at bv server?
05:48.02GrimStoneyeah i'm registered , and it accepts my INVITE password too
05:48.04D1ng0yes
05:48.07D1ng0me too
05:48.16Shidoboink
05:48.17D1ng0my outbound works now, but not inbound
05:48.25GrimStonecan you hear any sound when you do outbound ?
05:48.40GrimStonecos asterisk just ignores the incoming voice packets for me
05:48.51Shidovoice packets?
05:48.52jsolaresatleast you havent gone mad on us like the users on the list saying how bv screwed your life since incoming/outbound was not working :)
05:48.53Shidoare you nat'd ?
05:49.24GrimStonenope , nat=no
05:49.33Shidono, I mean
05:49.33GrimStoneworked fine until yesterday
05:49.35Shidoare you behind a router
05:49.45GrimStoneno . got external IP
05:49.47Shidook
05:49.50Shidowhere is the * box
05:49.51Shido?
05:50.43GrimStoneShido: well iconnecthere works fine , running on a DSL line
05:50.49jsolareslisten to greg, he's good with the asterisk voodoo, he'll shake his * voodoo doll and have you rsystem working in no time :)
05:51.25Shidoppl know me real name now , scary
05:51.41GrimStonewhen i make a call with BV .. asterisk sends the INVITE , with password and BV accepts it
05:51.56jsolaresit IS on your hostmask, and i CAN call you on your phone ;p with the 31337 extension
05:52.26GrimStoneand then i get a lot of data on port 5060 from BV , but asterisk just never seems to ignore it and act as if it got nothing
05:52.50jsolaresand IT is thanks to you that i've gotten along fast with asterisk :)
05:53.44GrimStoneD1ng0: do you have nat=yes ?
05:56.09*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
05:56.19D1ng0yes
05:56.37GrimStonehmm .. strange cos with nat=no , nothing works , heh
05:56.48Shidoif the box is on a public ip
05:56.54Shidoand your sip phone is on a public ip
05:56.58Shidoyou dont need to set nat to anything
05:57.02Shidoleave it alone ;)
05:57.23GrimStonewell would it hurt setting it to no ?
05:57.44jsolaresthe way i remember nat working yes it hurts setting it to no
05:58.43GrimStonewhat is the default nat= setting then ?
05:59.19D1ng0how can i set the refresh rate on a registration ?
05:59.49Shidoset that in your phone D1ng0
06:00.02D1ng0no i mean for a rergistration
06:00.19Shidoregistering your phone to asterisk? is set in the phone
06:00.23Shidowhat do you mean?
06:00.32rikstaShido: he means to a voip service i guess
06:00.38rikstalike FWD etc
06:00.46GrimStoneLMAO ..
06:00.53GrimStonei just set nat=yes and it works now
06:01.07Shidoheh
06:02.37GrimStonewell thanks for the clue .. strange thing is nat=no worked fine until yesterday cos i have external ip with the right ports un-firewalled
06:02.48*** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com)
06:02.53D1ng0sip show registry shows sip.broadvoice.com:5060         3219892181@s      2528 Registered
06:03.09D1ng0look at the refresh of 2528
06:03.19GrimStoneuntil yesterday BV had a 10 second timeout
06:03.34D1ng0well my setup worked fine until today
06:03.41D1ng0and their dumb changes
06:03.48GrimStoneeven if you try to force a 10 sec. timeout it will have 2528
06:04.04GrimStoneor 2000+
06:04.19D1ng0well outbound works, inbound is still broken
06:04.52GrimStonehmm .. do you have it defined as a peer or friend ?
06:05.08D1ng0what broadvoice ??
06:05.13GrimStoneyeah
06:05.17Shidohehe
06:05.20Shidoppl are listening!
06:05.26Shidofriends are evil
06:05.43D1ng0[incoming]
06:05.43D1ng0username=3219892181
06:05.43D1ng0type=user
06:05.43D1ng0secret=password
06:05.44D1ng0host=sip.broadvoice.com
06:05.44GrimStoneand setting sip.broadvoice.com in your /etc/hosts doesn't work well too today
06:05.44D1ng0fromuser=3219892181
06:05.46D1ng0fromdomain=sip.broadvoice.com
06:05.48D1ng0context=from-pstn
06:05.49jsolaresack!
06:05.50D1ng0canreinvite=no
06:05.52D1ng0authuser=3219892181
06:05.53jsolares~pastebin
06:05.54jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
06:06.00D1ng0ouch damn sorry i know better
06:06.13mitchelocding: you having bv problems? =)
06:06.14D1ng0i went to paste it in another window
06:06.20GrimStoneyou have a seperate context for incoming ?
06:06.24D1ng0yes
06:06.38mitchelocanyone know whats wrong here?
06:06.39mitcheloc<PROTECTED>
06:06.43mitchelocMar  5 22:03:39 WARNING[5256]: ast_expr.y:483 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input:
06:06.43mitcheloc<PROTECTED>
06:06.44GrimStoneShido: they're fine as long as you have the right permits
06:07.24GrimStoneD1ng0: that isn't a good idea .. try commenting out the incoming context and put it all in one context
06:07.37D1ng0i have an incoming context, and a sip.broadvoice.com context and until today it all WORKED fine
06:07.47GrimStoneand use permit=147.135.0.0/16
06:08.16D1ng0GrimStone, even if it was working ?
06:09.22mitchelochow can i escape this to use it in the dialplan? "<http://127.0.0.1/Bellcore-dr3>"
06:09.57Shidoerrr
06:10.06Shidodid you put a register line in your sip.conf for FWD?
06:10.11mitchelocit seems escaped to me (or doesn't need to be), but asterisk complains about it
06:11.59*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
06:13.31D1ng0GrimStone, nope still broken incoming
06:17.01*** join/#asterisk erwinism (~pogz@210.213.143.73)
06:17.23erwinismhello, what port does asterisk uses?
06:18.02rikstaerwinism: depends what protocols you use
06:18.10Inv_arpD1ng0 and I have similair setup  they only diff is my refresh in sip show registry is  15 and his is 2823 and my incoming BV works
06:18.18erwinismriksta i use the default
06:18.26rikstaerwinism: what!?
06:18.27Inv_arpmy setup http://pastebin.ca/6916
06:18.29SedoroxIAX, IAX2, SIP, MG... something
06:18.34Sedoroxall use different ports
06:19.00rikstaerwinism: SIP is 5060, with RTP ports of default 10000-20000
06:19.33erwinismriksta how can i enable those in my firewall?
06:19.39rikstaread the manual
06:19.58D1ng0Inv_arp, how did you set refresh
06:22.16Inv_arpD1ng0:  never did  think its in chan_sip.c   but im no C coder
06:24.09*** join/#asterisk Gronker (~Gronker2@adsl-220-79-68.ags.bellsouth.net)
06:24.31erwinismriksta i got it.. my next question is, do i have to add users to asterisk server?
06:25.12rikstawell..what do you think?
06:25.47Sedoroxerwinism: The Answer you seek.... is 42...
06:26.20GrimStonehmm. BV works now for outgoing but it doesn't seem to hangup calls right
06:27.05GrimStonewhen i stop a call , the CDR says call had NO ANSWER and billsec=0
06:27.26D1ng0my outgoing has been fine, its my incoming thats broken
06:27.46D1ng0Inv_arp, so both your inboind and outbound are working ?
06:27.58erwinismriksta how can i add users to my asterisk server?
06:28.20*** join/#asterisk andrew` (~andrew@adsl-67-119-26-16.dsl.snfc21.pacbell.net)
06:28.46D1ng0so is there another good VOIP service that works with asterisk, BV is going to loose me as a customer
06:28.48Inv_arpD1ng0: i only use BV inbound    yes
06:29.01GrimStoneInv_arp: does CDR work fine for you ? asterisk doesn't seem to hangup calls right now
06:29.03D1ng0Inv_arp, well your config didnt work for me at all
06:29.13GrimStoneand it shows the call as NO ANSWER
06:29.23*** join/#asterisk roamer323 (~sing@67.71.60.238)
06:29.37Inv_arphmm  what version ya use?
06:32.11GrimStone2/15 cvs
06:35.07*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
06:38.04erwinismwhat is a good softphone to use?
06:38.27rikstax-lite
06:39.00PTG123eyebeam :)
06:42.05GrimStoneis there any page on wiki with a description of the "register => " command ?
06:44.24*** join/#asterisk Damin_Mobile (~pocketirc@213.sub-166-155-119.myvzw.com)
06:44.48MuppetMasterAlthough I can not get Eyebeam video to work with Asterisk at the moment.  A patch is needed for H.263+ to work, and attempting to sort that out.
06:45.21BoRiSLook at the latest CVS HEAD. Should have it.
06:45.46*** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au)
06:46.01MuppetMasterBoRiS:  Latest CVS HEAD should have what?
06:47.31Damin_MobileKram just committed a h263+ patch to cvs head.
06:47.34PTG123how do i make asterisk play a ring
06:47.36PTG123when its calling
06:48.19BoRiSadd r in dial
06:48.35PTG123add ,r where?
06:48.46MocPTG123, after the timeout
06:50.23*** join/#asterisk Newbie___ (some@60.48.53.154)
06:50.41Newbie___AMP rocks !
06:50.57BoRiSMoc!!!!!
06:51.14PTG123hey what does the ,t do? have a timeout?
06:51.41BoRiStransfer
06:51.55PTG123so i just want DIAL(IAX2/BLAH,30,r)
06:51.55PTG123?
06:52.03BoRiSMoc: Whats up?
06:52.11Mochi, nothing
06:52.19Mocdoing some virtual cleanup
06:52.33BoRiSoh yeah?
06:52.50Mocyep
06:53.08MocI got like 5 different linux machine just for my personal usage..
06:53.17Mocalittle over kill...
06:53.23BoRiSUhhhhhhh........ Yeah!
06:53.25BoRiShehe
06:53.41*** join/#asterisk The_Ball (~alex@dialup-211.43.194.203.acc02-wick-bne.comindico.com.au)
06:53.42MocIm trying to bring the number to 2..
06:53.54The_Ballis it possible to use a TDM card as a modem?
06:54.19PTG123i now use my notebook running winxp with a vmware freebsd windows.. and hummingbird exceed for my xterm
06:54.21PTG123it works great :)
06:54.29PTG123my linux and frebsd boxes collect dust now
06:55.24ShidoPTG123, thats right
06:55.25Mocoh you got exceed ? is it nice ?
06:55.34Shidoif u dont want to use the transfer
06:55.36PTG123awesome
06:55.38PTG123i have used it for years
06:55.47MocI never found it hehe
06:55.57PTG123hehe never found it warezed you mean? :)
06:56.15Mocheu.. yea ;
06:56.23PTG123i think the eDonkey network probably has it
06:56.24Mocwell I might just found it hehe
06:56.39Mocpowersuit 10 SP5 should be ok ?
06:56.55PTG123let me see
06:58.16PTG123i have no idea
06:58.18PTG123i use 9 :)
06:58.21PTG123just make sure it has exceed in it
06:58.35MocExceed PowerSuite™ 10
06:58.51Mocwill see sone ennuf
06:58.53PTG123ah yah
06:58.54PTG123that is it
06:58.58PTG123just has nfs with it
06:59.06Mocha..
06:59.07PTG123http://connectivity.hummingbird.com/products/nc/exceed/index.html
06:59.20PTG123i think 10 got better mutiple monitor support
06:59.26tzafrirgood morning
06:59.52Mocmorning
07:00.02tzafrirnow why would you use an illegal warez exceed?
07:00.36Mocto try it
07:00.37tzafrirTake a look at cygwin and tell me what is missing from there
07:00.43Mocbefore buyin
07:00.57PTG123try before you buy
07:01.02PTG123and cygwin is nothing like exceed
07:01.23tzafrirwhat do you need?
07:01.53The_Ballwhat do i press/dial to en a Record()?
07:03.34tzafrirThe_Ball, you need it in your dialplan
07:04.10tzafrirAn besides, what's this on-topic talk all of a sudden? You're interrupting our conversation! ;-)
07:04.15The_Balltzafrir, the docs doesn't say anything about thathttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x958.html
07:04.36The_Ballhehe
07:05.17tzafrirThe_Ball, you can configure Asterisk to do many things. The dialplan (extensions.conf, basically) tells Asterisk how to react to whatever is dialed.
07:05.34tzafrirIf you put in your dialplan something like:
07:07.01tzafrirexten => 12345,1,Record('/var/lib/asterisk/my.sound') (or whatever, I don't remember the syntax now) it will Record when dial 12345. Assuming you put it in a sensible context
07:15.27tzafrirslightly OT: for those of you who think that the most intruders can do is steal calls:
07:15.38tzafrirhttp://www.theregister.co.uk/2005/03/03/skype_broadreach_voip_calls/ IP over VoIP
07:16.07tzafrirskype as a tunneling protocol?
07:18.12PTG123anyoen here using g729 on bsd?
07:20.13The_Balltzafrir, that's fine, i have asterisk recording, but when i try to end the record by presing hash or star i get the buzy tone and Mar  6 16:52:52 WARNING[9196]: pbx.c:1923 ast_pbx_run: Invalid extension '#', but no rule 'i' in context 'setup' in the console
07:21.16GrimStonechan_sip.c:7943 handle_request: That's odd...  Got  a response on a call we dont know about.
07:21.40D1ng0Broadvoice SUCKS
07:21.50GrimStoneif i use nat=no with broadvoice i get this and no sound .. with nat=yes i get sound and same message
07:22.09GrimStoneD1ng0: depends .. was great while it worked
07:22.16D1ng0they really screwed alot of people today
07:22.23tzafrirThe_Ball, pressing '#' should get you to the "next priority". Put a Hangup there
07:22.42tzafrirAlternatively, terminate the recording by hanging up
07:22.52hardwirehmm
07:22.57hardwiretzafrir: you just gave me an idea
07:23.01GrimStoneit seems to be an issue with asterisk not recognizing that a call has started
07:23.02D1ng0whats a good us VOIP DID provider
07:23.09hardwireD1ng0: where?
07:23.14D1ng0florida
07:23.20hardwirevoiceconduits.com
07:23.40D1ng0ive gotta dump Broadvoice since they dont know anything about customer notifications
07:24.01PTG123teliax.com
07:24.02PTG123try them
07:24.47The_Balltzafrir, will you look at my dialplan: http://channels.debian.net/paste/316
07:25.28hardwireI need to up my prices for broadband in a few rural communities
07:25.56hardwireI think if I up them and offer 1000 minutes VoIP/LD that will help pay for more dedicated bandwidth
07:26.00hardwireas well as use it up :)
07:26.34tzafrirThe_Ball, say, do you use the Debian packages?
07:26.46The_Balltzafrir, no, this is on a gentoo box
07:27.26tzafriranyway, one thing you can do is set verbosity to something high enough (e.g: 3) and look at the CLI while the call is running
07:27.41tzafrir'set verbose 3' in the asterisk cli
07:28.49tzafriranyway, I odn't think you need Answer if this is not an incoming call
07:29.30The_Balltzafrir, http://channels.debian.net/paste/paste
07:29.41hardwirepastey
07:29.46The_Balltzafrir, oups, -> http://channels.debian.net/paste/317
07:30.12D1ng0hrmmm
07:32.10tzafrirnot that it would matter here, but it is generally sensible to use two different sounds
07:34.09tzafrirIt seems that the error is from Playback. Playback returned -1 ?
07:34.22tzafrirDid you try to remove the Answer ?
07:35.13The_Balltzafrir, that didn't work
07:36.01tzafrirThe_Ball, meaning? What happened?
07:36.24The_Balli think i have found a error, show dialplan does not show the record()
07:42.32D1ng0okay so how do i tell what my teliax phone number is after i sign up ?
07:43.55*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
07:46.53Shidook
07:46.55Shidoback
07:49.05D1ng0PTG123,  okay so how do i tell what my teliax phone number is after i sign up ?
07:50.14Shidotzafrir whats wrong?
07:51.16tzafrirShido, what's right?
07:52.47GrimStoneman .. it sucks that i can get everything to work except the CDR records are screwed
07:53.22Shidotzafrir you are recording tzafrir_home ?
07:53.38*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
07:53.51GrimStonewith Broadvoice , i can hear the sound with nat=yes , but asterisk keeps saying "got a response on a call i don't know about"
07:53.52tzafrirShido,, no I'm recording tzafrir
07:55.58tzafrirShido, last time I talked from tzafrir_home I said something related to licensing. Do you ask about that?
08:01.06*** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net)
08:01.15ta[i]ntedis BV still having problems?
08:01.35ta[i]ntedi'm getting a "the device you are using is not registered to place calls on the network.. please contact your administartor"
08:02.27D1ng0ta[i]nted, YUPP my outbounad works but no inbound
08:03.23ta[i]ntedhow did u get outbound working?
08:03.32ta[i]ntedneither of mine works now
08:05.10Shidobbl
08:05.52WilliamKthey haven't been able to port #s and keep saying "Soon.."
08:05.52ta[i]ntedWilliamK are u able to place calls?
08:05.54WilliamKI called em up and said, ya'll said soon was last summer too, so which is it
08:06.09WilliamKtainted, can't be a cust if I can't get a # ported
08:06.10WilliamK=)
08:06.22PTG123heh
08:06.34PTG123williamk: i'll port your # for you :)
08:06.54WilliamKPTG, I'm hoping we've taken care of that problem =)
08:07.07PTG123heh
08:07.29PTG123man i need to go to bed
08:07.30WilliamKdo I "DARE" upgrade to the later cvs?
08:07.37PTG123i wouldn't :)
08:08.05PTG123whats wrong with your 729?
08:08.09WilliamKthe last time I tried, it ate it for lunch
08:08.11PTG123i got mine working tonight on freebsd baby
08:08.27WilliamKbad enough I gotta re-reg it
08:08.59*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
08:12.25jontowi just did a successful test on an outbound call using Jeff Rizzo's ported zaptel drivers, using an FXO clone card
08:12.33jontow.. on NetBSD -CURRENT :)
08:15.44*** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net)
08:17.14D1ng0WilliamK, i got outbound BV working no inbound
08:17.46GrimStoneD1ng0: do you get messages like "thats odd. got a response on a call ..." ?
08:18.45PTG123hey can some people ping sip1.way2fast.com
08:18.55PTG123i wanna know the latency and the location of your cablem modem :)
08:19.21rikstaim in the uk, i get rtt min/avg/max/mdev = 169.847/316.353/475.396/90.573 ms
08:19.25jontow--- sip1.way2fast.com ping statistics ---
08:19.25jontow5 packets transmitted, 5 packets received, 0% packet loss
08:19.25jontowround-trip min/avg/max/stddev = 105.564/108.220/110.406/2.185 ms
08:19.25jontow[03:13] > i just did a successful test on an outbound call using Jeff Rizzo's ported zaptel drivers, using
08:19.25jontow+an FXO clone card
08:19.25jontow[03:13] > .. on NetBSD -CURRENT :)
08:19.27jontow[03:14] QUIT: BoRiS : "Ping timeout: 360 seconds"
08:19.29jontow[03:14] QUIT: Luhiwu : Read error: 60 (Operation timed out)
08:19.31*** join/#asterisk denon (denon@synapse.subneural.net)
08:19.31jontow[03:16] JOIN: dersteer to #asterisk
08:19.31*** mode/#asterisk [+o denon] by ChanServ
08:19.34jontow[03:18] <D1ng0> WilliamK, i got outbound BV working no inbound
08:19.35jontow[03:18] <GrimStone> D1ng0: do you get messages like "thats odd. got a response on a call ..." ?
08:19.37jontow[03:19] <PTG123> hey can some people ping sip1.way2fast.com
08:19.40jontoweh, woah
08:19.49jontowfucking table fell apart while i was pasting
08:19.57jontownever been able to claim that before
08:19.58rikstaLOL
08:20.11jontow(..shit)
08:20.23jontownonetheless, yeah, 108ms avg.. upstate NY
08:20.40PTG123jontow: where are you located?
08:20.48jontowupstate NY :)
08:20.53PTG123hmm
08:20.57PTG123can you traceroute
08:20.58PTG123that seems high
08:21.15jontowsure.. looking for number of hops or highest-latency node or other?
08:21.28PTG123i wanna see where the extra latency comes in
08:21.39PTG123we are going across the country so its not horrible
08:21.41PTG123but it shoul dbe better
08:21.47jontow13  ewr-core-01.inet.qwest.net (205.171.17.125)  29.202 ms  27.723 ms  27.052 ms
08:21.47jontow14  tmp-core-02.inet.qwest.net (205.171.205.85)  108.552 ms  108.275 ms  107.821 ms
08:21.53jontowtheres your bad node :)
08:22.07PTG123thats probably our cross country link
08:22.11PTG123why the 30ms before?
08:22.19PTG123paste the whole thing i wanna see how much it bounces around
08:22.30rikstadont paste it all here
08:22.58PTG123riskta: 10 lines of text gonna knock you off or something? :)
08:23.12rikstayes :)
08:23.16jontowhttp://mno.bsd.st/~jontow/tr.txt
08:23.55PTG123what do you think tmp stands for?
08:23.59PTG123and ewr
08:24.03jontowtempe, arizona?
08:24.07*** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au)
08:24.24jontow(or temporary, as in man-we-fucked-up-that-router,-lets-put-this-one-in-instead-and-modify-the-bgp-tables)
08:24.33PTG123and ewr?
08:24.53jontoweast-west-route?
08:24.57PTG123heh
08:25.02jontow(blatant guesses)
08:25.07PTG123we need one of those visual traceroutes with a map
08:25.23jontowbut you know what
08:25.28jontowi don't think that its too far off
08:25.38PTG123which?
08:26.07jontowit goes from time warner up here, to their syracuse POP, to NY's Level 3 major hop
08:26.22jontowthen it hits ewr-border, then ewr-core
08:26.59PTG123its probably worthwhile for me to colocate some stuff on the east coast
08:27.05dersteer64 bytes from ip-66-235-234-131.sterlingnetwork.net (66.235.234.131): icmp_seq=1 ttl=48 time=788 ms
08:27.06dersteer64 bytes from ip-66-235-234-131.sterlingnetwork.net (66.235.234.131): icmp_seq=2 ttl=48 time=756 ms
08:27.06dersteer64 bytes from ip-66-235-234-131.sterlingnetwork.net (66.235.234.131): icmp_seq=3 ttl=48 time=831 ms
08:27.12jontowdersteer; OUCH.
08:27.24PTG123ok now thats bad
08:27.29PTG123dersteer: where are you?
08:27.33dersteermichigan
08:27.39PTG123which hop is killing us?
08:27.39jontow(...really ouch.)
08:28.04PTG123i need to know so ic an report that
08:28.28jontowlemme try from a few other places ;)
08:28.53dersteersorry PTG123 its me
08:28.59dersteerI got someone hitting my ftp hard
08:29.01PTG123ah ok
08:29.02PTG123good
08:29.03PTG123heh
08:29.06PTG123had me nervous
08:29.37dersteerfirst hop was 819.717 ms
08:29.37jontow--- sip1.way2fast.com ping statistics ---
08:29.37jontow5 packets transmitted, 5 packets received, 0% packet loss
08:29.37jontowround-trip min/avg/max/stddev = 52.975/55.365/62.967/3.816 ms
08:29.39dersteer:p
08:29.40dersteerlol
08:29.47jontowthats from a machine of mine in arizona
08:29.56PTG123jontow: thats horrible too
08:29.59PTG123paste me the traceroute
08:30.01PTG123:)
08:30.11jontow.. its a 384k business DSL line thats heavily used
08:30.13jontowthats pretty good :P
08:30.28PTG123hah
08:30.31PTG123considering i am in arizona
08:30.34PTG123we shoul dbe like 20ms
08:30.36jontow;)
08:30.51dersteerPTG123: what kind of connection u got?
08:31.24jontowwell eh.. its gotta hit san jose first
08:31.31jontow(Level 3 San Jose)
08:31.43PTG123multihomed, OC192 with cox (for cable modems who peer with all cable proviers) OC192 to qwest(for business DSL) and an OC48 from some other tier1 provider who has rthe best routes to everywhere else
08:31.55PTG123yah i think i may bring in a l3 link
08:31.58PTG123although
08:32.05PTG123cox is suppose to be direct on l3 backbone
08:32.42jontow--- sip1.way2fast.com ping statistics ---
08:32.43jontow6 packets transmitted, 6 packets received, 0% packet loss
08:32.43jontowround-trip min/avg/max/stddev = 86.716/88.657/93.570/2.400 ms
08:32.57jontow<PROTECTED>
08:32.59PTG123jontow: where is that from?
08:33.01PTG123ah yah
08:33.05PTG123see thats awesome for ny
08:33.09jontowthe ISP I work for
08:33.17jontowits from my desk at work actually :)
08:33.20PTG123a voip call on that one would be pretty good
08:33.28jontowno
08:33.31jontow.. no it isn't :)
08:33.35jontowonly at 3am ;)
08:33.51jontowkeep in mind, this is the least utilized time for EVERYTHING i'm giving you ;)
08:33.55PTG123hah
08:34.00PTG123well ping times shouldn't change
08:34.02PTG123with traffic
08:34.06PTG123unless your network is way oversaturated
08:34.08dersteerwhat kinda ping time u need for good voip ?
08:34.18PTG123probably under 150ms
08:34.28jontowyeah, it is way oversaturated
08:34.37jontow4 T1s, all 80-90% drained all day
08:34.39jontowuntil about 1am ;)
08:34.54PTG123wow
08:34.56jontowbetter lately though, since they upgraded to a cisco 7200 or whatever :)
08:34.59PTG123voip would suck through that connection
08:35.03jontowyes.. yes it does
08:35.07dersteerwow jontow
08:35.11jontowits scratchy as shit and breaks up all the time :(
08:35.12dersteerwhat kind of place u work for?
08:35.40PTG123a place thats too cheap to buy a decient internet connection
08:35.41PTG123heh
08:35.41jontowthey're oversold as far as im concerned.. and working on a 10mbit DANC fiber connection, as well as multi-homing with verizon fiber as well
08:36.05jontowso its gonna get better; but the problem is that the bandwidth just isn't available to smaller companies in this area
08:36.18jontowit isn't their fault.. but they haven't exactly been doing enough to correct it, either..
08:36.30PTG123why not get a ds3
08:36.48jontowsingle connection isn't a good idea up here
08:36.52jontowunreliable routing :)
08:37.00jontowvery rural area
08:37.07PTG123you connect to a datacenter
08:37.10PTG123then multihome the bw yourself
08:37.16PTG123private loop ds3 you control
08:37.20jontowif you don't have multiple uplinks, you're fucked up here..
08:37.40jontowonly way you could do that here is with verizon, and they're so unresponsive that its unreal..
08:37.41jontow:o
08:37.59PTG123makes me appreciate living in phoenix :)
08:38.16*** join/#asterisk coppice (~chatzilla@96.196.17.210.dyn.pacific.net.hk)
08:38.17jontowwe had a cross-county circuit drop on a friday.. a week later, on the next friday, the trouble ticket was finally processed by verizon and they called us back
08:38.21jontow.. we cancelled the circuit
08:39.35dersteerI'm hoping to have good luck picking a carrier.... I'm working on starting a wireless ISP here
08:39.50dersteerI'm in a remote rural area too
08:40.08coppicewho here uses a variety of different ATA boxes?
08:41.32jontowwe're in the foothills of the adirondack mountains, long-haul wireless is a giant pain in the ass with so many obstructions :/
08:42.40PTG123i talked to a guy with a 80 mil wireless network
08:42.46PTG123all based on cheap wireless hardware
08:43.13dersteerthats what I'm working on
08:43.19dersteerthe equipment really is cheap :)
08:43.23PTG123well
08:43.30PTG123you can even use wrt54gs
08:43.37PTG123if you keep them less then 5 miles appart
08:43.45PTG123he uses a used laptop and cheap wireless cards
08:43.49PTG123and he gets 8mile links
08:44.01jontowdamn yeah :D
08:44.03PTG123his antennas costed him $35
08:44.11PTG123i was impressed :)
08:44.14PTG123running openwrt
08:44.33jontownice.
08:45.06PTG123i guess its time for bed for me
08:45.16jontowhey
08:45.20jontowhttp://mno.bsd.st/~jontow/tr.txt
08:45.22jontowhit that one more time
08:45.24dersteerwhere can I find a wrt54gs?
08:45.27jontowits a paste of all the traceroutes i've done
08:45.30dersteerone that is the right version?
08:45.59dersteerI've had a hard time finding the good wrt54g versions
08:46.47dersteerI'm going to be using devices called SBC
08:46.57dersteersingle board computer
08:47.06dersteerfor the AP
08:47.12jontowhey
08:47.15jontowhttp://www.soekris.com
08:47.16dersteerdon't know what I wanna use for customers
08:47.17jontow:)
08:47.32dersteerI want something cheap for the customers
08:47.46jontowSBCs for about $250 each, with case and power supply ;)
08:47.58dersteerouch
08:48.11dersteerhttp://www.wisp-router.com/index.php?cPath=39&osCsid=d1e9e3e5daa2c298d98923242c978c71
08:48.22jontowno, no.. they're heavy on the features
08:48.28jontowyou can get them down to about $150 each i think
08:48.46jontowvarying features, guaranteed compatible with netbsd and freebsd and linux and such :)
08:49.14dersteercool
08:49.30dersteerI'm more worried about something for the client side
08:49.34*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
08:49.39dersteerwould like to use something like the wrt54g
08:49.42jontowAP or just router?
08:50.54dersteerI want it to act as a client... taking in the wireless... then routing it to the customers machines via ethernet
08:53.57[cc]smartdersteer: afaik, openwrt has a ser router package
08:54.19dersteerhmmm
09:12.37*** join/#asterisk meshugga (philip@loeblich.linuxteam.at)
09:12.41meshuggahi
09:12.54cjkapparently some peopole got * working on openwrt
09:13.04meshuggayesterday i solved the one-way audio problem with chan_bluetooth
09:13.17meshugganow i have the problem that asterisk doesnt take the call when my phone is ringing
09:13.42meshuggaanybody here who has clue about how asterisk expects a module to route a call before taking the call?
09:13.56dersteerreally cjk?
09:14.07coppicemeshugga: do tell. what did you do to get the audio to work?
09:14.21meshuggacoppice: hehe, i knew somebody would be interested ;P
09:14.27meshuggaactually it is just a three line patch
09:14.45meshuggaget_buffer always returns 0, so asterisk thinks it gets null frames
09:15.01meshuggathe trick was to teach get_buffer to return the actual len of the frame
09:15.24meshuggarecompile module -> works
09:16.00coppiceOK, so the 3 lines are.......
09:16.13meshuggabut yet i dont know how to post that to the ml, im not getting my subscription
09:16.34coppicejust post it here. its only 3 lines
09:17.23meshuggaget_buffer(char * dst, char * ring, int ring_size, int * head, int to_copy)
09:17.23meshugga{
09:17.23meshugga<PROTECTED>
09:17.23meshugga<PROTECTED>
09:17.31meshugga<PROTECTED>
09:17.31meshugga<PROTECTED>
09:17.32meshugga<PROTECTED>
09:17.32meshugga<PROTECTED>
09:17.36meshuggaehm
09:17.37meshuggafuck
09:17.40meshuggawait
09:18.20meshuggahttp://rafb.net/paste/results/3B1XWl66.html
09:18.29meshuggathis is how the function is supposed to look like
09:18.33meshuggaas said, just three more lines
09:19.15meshuggayou can kick that ast_log of coz, thats quite noisy
09:19.55meshuggaso now anybody here who can point me some documentation about asterisk module coding?
09:20.09meshuggabecause
09:20.22meshuggaactually, that chan_bluetooth is a bit messy altogether
09:20.26meshugganice functionality though
09:21.35meshuggacoppice: you will tell me if it worked for you? and if your asterisk takes the call?
09:21.46meshuggai wonder how this is supposed to work anyway
09:21.49*** join/#asterisk ibr (~ibr@dsl-082-083-197-172.arcor-ip.net)
09:21.53coppicei don't know of any documentation. the usual ercommendation is to take a simple channel and follow it. the channel I wrote still has a couple of quirks because I haven't probed enough to figure out what it should do.
09:22.14ibrHi!
09:22.21meshuggacoppice: which chan did you write?
09:22.47coppicechan_unicall, which interfaces my unicall protocol modules like MFC/R2 to *
09:23.07meshuggaah, ok, dont know them
09:27.03coppiceif you were in a country that needs MFC/R2 you would :-)
09:27.21meshuggais that like BRI?
09:27.32coppicenot even remotely :-)
09:27.37meshuggahehe
09:28.04coppiceits a weird old tone signalling system used over E1s in many countries
09:28.19meshuggaah
09:28.23meshuggain line signalling?
09:28.28meshuggathis ccitt-5 thing?
09:28.33coppiceno
09:28.54coppiceits a mix of in band tones and separate line signals
09:29.52meshuggai c
09:30.02meshuggai accept that "weird" ;)
09:30.35meshuggabtw, what is that abbreveation "CIND" supposed to mean?
09:35.41coppicemeshugga: Control INDicator
09:35.44*** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au)
09:35.51meshuggadoes what?
09:36.00*** join/#asterisk jhiver (~jhiver@ABesancon-102-1-2-7.w80-11.abo.wanadoo.fr)
09:36.09jhiverhi everybody
09:36.17coppiceit reports thing like signal strength and battery state on GSM phones
09:36.52meshuggaah, i c
09:36.54meshuggathanks#
09:39.51cjkanyone here who has more detials on the disposition fields in cdr records? disposition=4 means what exactly?
09:40.28coppicei think disposition is "grumpy" :-)
09:41.04cjkgrumpy, well it can be important. i have a special case on my systems.
09:41.23GrimStonewell disposition is a plain text field
09:41.44cjkGrimStone, yeah, but if you use odbc it just logs an integer
09:42.07GrimStoneopen the odbc source and see then
09:42.22GrimStoneusing pgsql it logs in plain text
09:43.33*** join/#asterisk skraps (~mike@c-24-0-190-239.client.comcast.net)
09:44.09skrapsanyone here?
09:44.48skrapswow, 261 in the channel and nobody is here.  that must be a new record.
09:45.13cjkGrimStone, ok whats the disposition you see the most in your records? that must be 4 in mycase
09:46.03*** part/#asterisk skraps (~mike@c-24-0-190-239.client.comcast.net)
09:48.09tzafrir<PROTECTED>
09:50.03*** join/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net)
09:50.05jjghi
09:50.31jjghas anyone been successful in using SIP to establish video calls through *?
09:51.09ibrnot even for audio yet :) .
09:51.18jjgheh, hang in there
09:51.35ibrhave you done that part?
09:51.35jjgibr / what's the issue?
09:51.52ibrI defined two users in sip.conf.
09:52.05ibrThey can register on the server.
09:52.30ibrBut when they call each other, the server responds with 404.
09:52.40ibrI enabled sip debug.
09:52.44jjgwhat's 4040, don't remember
09:52.46jjg404
09:52.54ibrUser not found.
09:53.18jjghave you made a context extension to dial the sip user?
09:53.24jjgi think that is necessary
09:53.43ibrI'm not sure.
09:53.47ibrEverything I did was:
09:54.14GrimStoneis the current CVS in a working state ?
09:54.26ibr[user1] type=friend host=dynamic
09:54.32ibrThe same for the second one.
09:54.42ibrHow do I define a context extension?
09:55.08jjgexten => 1000,1,Dial(sip/username,20,tm)
09:55.26ibrWhere does it go?
09:55.34jjgexten => 1000,2,Hangup
09:55.43jjgthen if yo ucall 1000, it rings username for 20 seconds
09:56.04ibrextensions.conf?
09:56.06jjgyah
09:56.20jjgso you should create an extension for each sip user
09:56.25ibrAha. Let me try...
09:56.52jjgi'm not sure if it's absolutely necessary but that is how i do it...and my server isn't online for me to look right now
09:57.40GrimStoneis the current CVS at 1.1.0 level , or at 1.0.6 ?
09:58.02jhiverhey guys
09:58.26jhiverdo you know some URL where there is a Asterisk + SER tutorial?
09:58.51ibrjjg: Hmm, as I understand, 1000 is the "called number".
09:59.08ibrjjg: But I have two kphone clients, user1 and user2.
09:59.21ibrjjg: Do I still need extension definitions?
09:59.26GrimStonejhiver: try the voip-info wiki ?
10:01.04ibrjjg: Or should I write "exten => user1,1,Dial(sip/user1,20,tm)"?
10:02.40ibrBTW, how do I complete a user name ("jjg: ...")? I'm using ircII.
10:03.23jjgibr : i think extensions have to be numberic
10:03.26jjgnumeric
10:04.09ibrjjg: Ok, I'll try that first. I should call 1000@myserver, right?
10:04.29FaithfulAnyone the zaphfc thing???
10:04.35ibrjjg: Do you also use two software clients, or phones?
10:06.30*** join/#asterisk jeofrey (~jeofrey@202.160.45.29)
10:06.35jeofreyhi
10:06.58jeofreyanyon can help me please in MySQL problem in Fedora Core
10:07.17jeofreyERROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)
10:07.33jeofreyhow to solve that error
10:08.24Faithfuljeofrey: how are you trying to connect?
10:08.42Faithfulmight be that you are not listening on TCP socket
10:08.54jeofreyso what do i have to do
10:09.07jeofreybecause i am first time user of myqsl
10:09.10jeofreymysql
10:09.24jeofreyi just want to use in asterisk
10:09.30jeofreyto record our calls
10:09.59Faithfuluncommnet "port            = 3306" in my.cnf
10:10.39jeofreywhere i can get the my.cnf
10:11.10FaithfulIf you are running linux... where all your cnf type files are
10:11.19jeofreyahhh ok
10:11.21ibrjjg: Yes, this worked! Thank you :) !
10:11.22jeofreyi will
10:11.38ibrjjg: But the client immediately died :( .
10:11.52ibrSee you all!
10:12.18jeofreytry to locate
10:12.45jeofreyim not sure also the location..... new user of asterisk
10:14.51ta[i]ntedi have a question
10:15.06ta[i]ntedwhen a user dials an invalid number using Dial()
10:15.17ta[i]ntedand the provider says 'invalid number blah blah'
10:15.24jeofreyusr/share/mysql/
10:15.30jeofreyi cannot find also inside there
10:15.32ta[i]ntedget_Variable(DIALSTATUS) still returns 'ANSWERED'
10:15.51ta[i]ntedhow can i get the true disposition of the call?
10:16.53ta[i]ntedi guess technically the provider 'answered the call'.. but the call shouldn't register as a successful call
10:17.27Faithfuljeofrey: /etc/mysql/my.cnf
10:17.46Faithfuljeofrey:  locate my.cnf
10:18.17jeofreyokkk Faithful
10:20.01coppicemeshugga: hey! chan_bluetooth seems to be working!
10:20.17meshuggaof course
10:20.26meshuggai wonder why this bug was such a stupid one
10:21.41meshuggaand /now/ i'd like to have incoming calls working
10:21.46meshuggabut i still dont know where to start hehe
10:22.04coppiceI just called my headset and talked both ways
10:22.18meshuggayeah, that works
10:22.30meshuggaand dialling out on a S55 works too
10:23.09coppiceso what doesn't work?
10:23.19meshuggabut there seems to be a timing problem left, with both, headset and phone, sometimes there is just noise on the line
10:23.30meshuggacoppice: incoming calls on the phone arent taken by asterisk
10:24.10coppicei don't follow you. you just agreed that calls to the headset work
10:24.17meshuggaand i dont find the point in the module where this is supposed to happen
10:24.34meshuggacoppice: yes, but you can also use your mobilephone with that module
10:24.39meshuggaand thus dial out, right?
10:24.54meshuggathis works, too
10:25.05coppiceer, no. the dialing out thing is phone dependant
10:25.11meshuggabut what doesnt work is, when the phone has an /incoming/ call, from the GSM provider
10:25.21meshuggacoppice: well, i told you, it works with my S55
10:25.33meshuggaand it also worked with an ericsson t630
10:25.43coppiceso, it works with that one. it doesn't work generally
10:25.51*** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-236.arcor-ip.net)
10:26.02meshuggawell, it should work generally if the phone has bluetooth headset support
10:26.04meshuggabut anyway
10:26.15meshuggain my setup, dialling out works
10:26.24Blackveldo I have to use this line: ;deny=216.207.245.47/255.255.255.255 for iax tel incoming calls?
10:26.29coppiceyou mean you expect an incoming GSM call to get routed to *?
10:26.33meshuggabut chan_bluetooth doesnt know what to do with the incoming call
10:26.37meshuggacoppice: yes
10:27.09meshuggathis should work too, the "RING" is coming, also "+CLIP", but chan_bluetooth somehow messes it up to set the line to "ringing"
10:27.20coppicemeshugga: bluetooth doesn't seem to standardise digits from a headset, or to a headset for hat matter
10:27.42meshugga?
10:28.14jeofreyFatihful i search inside of root/etc but i cannot find the mysql directory inside...
10:28.38jeofreyis it inside of that directory??
10:28.40meshuggacoppice: -v?
10:28.56coppicebut you are right that if chan_bluetooth acts as a headset it should be able to answer and handle  GSM calls.
10:29.55meshuggaas far as i understood some postings on the mailinglist, there /is/ this functionality already implemented in chan_bluetooth
10:30.07meshuggaand it should route the call to extension "s" in context [bluetooth]
10:30.17meshuggabut woe is me, i cant find that ;)
10:30.50meshuggaAH
10:30.51meshuggai have
10:30.55meshuggafound it
10:30.57coppicei thought so, but i never tried. i only tried to use a headset before
10:31.56meshuggait would rock to be possible to dialin
10:32.18meshuggathen it would be possible to do some nice least-cost-routing setups with the cost of some handsets
10:32.20coppiceI tried another bluetooth headset, and that also works now.
10:32.28jerliquehas anyone got any flash operator panel screen dumps, with lots of extensions??
10:32.30meshuggacoppice: can you give me the models which work?
10:32.57meshuggathe one which works here is a logitech hs01, jabra bt200 doesnt work though :'/
10:33.12coppiceA Sony Ericsscon HBH-660 and an Omiz OM8055
10:33.17meshuggathanks
10:33.34GrimStoneanyone have "Got a response on a call we dont know about" errors with Broadvoice ?
10:33.46Faithfulmeshugga:  :-( I have a bt200
10:34.05The_Ballis there any demo FWD or IAX number?
10:34.18The_Ballto test the iaxtel setup
10:34.33FaithfulYou can try me
10:34.47Faithfulbut that's no guarentee
10:35.06GrimStonethe call starts but asterisk doesn't generate proper CDR's or hangup the call when its done
10:35.09meshuggaFaithful: i will let you know when i made it work
10:35.13coppicemeshugga: a number of people are looking for this fix. can you post it on the mailing list?
10:35.17The_Ballwhat's your number
10:35.29meshuggai would like to have the jabra working too, since it is more comfortable to carry
10:35.40meshuggacoppice: i didnt get the ack from the ml for my subscription
10:35.42Blackvelanyone knows how to configure sarp.sourceforge.net with asterisk on the same server? feel free to msg me. Do I have to use different SIP/RTP ports? what do I have to do with the router port forwarding? twice? how do I configure asterisk to use SARP?
10:35.42meshugga:/
10:35.59coppicewould you like me to post it?
10:36.22Faithful17005672555
10:36.23meshuggawell, yes why not
10:36.25coppice"meshugga says... " :-)
10:36.28meshuggalet me generate a patch
10:36.45Faithfulmeshugga:  so are you using a BT dongle on the PC?
10:38.20coppiceFaithful: I am. A CSR based one
10:38.27meshuggaFaithful: USB, yes
10:38.37meshuggasome noname shit
10:38.49*** join/#asterisk ranliv (~ranliv@210.5.85.249)
10:38.50meshuggacoppice: but actually, i'd like to put some docu also in it
10:39.03meshuggae.g. the s55 needs "sdptool add hf"
10:39.10meshuggaand then there is some timing problem
10:39.50meshuggaso just give me time until tomorrow, then i will try to post it
10:40.03meshuggapeople have been waiting months now, they can wait until tomorrow ;P
10:41.06The_BallFaithful, anything happening?
10:42.13coppicemeghugga: fixing this bug makes you a good guy. documenting things makes anyone a superhero :-)
10:42.46meshuggalol
10:43.45Faithfulmeshugga:  so you have blue tooth headsets working with softphones in linux???
10:46.03FaithfulNope
10:46.20The_BallFaithful, did you get anything inbound?
10:48.15Blackvelwhat is the best service you can get for the moment, for a germany/UK/US flatrate?
10:48.29FaithfulThe_Ball:  Try again
10:48.35The_BallFaithful, i get: -- Executing Dial("Zap/1-1", "IAX2/user:password@iaxtel.com/17005672555@iaxtel") in new stack    -- Called user:password@iaxtel.com/17005672555@iaxtel    -- Hungup 'IAX2/69.73.19.178:4569/3'  == No one is available to answer at this time
10:49.18The_Ballhmmm, this time i got: Mar  6 20:42:41 WARNING[19867]: chan_iax2.c:5546 socket_read: Call rejected by 69.73.19.178: No authority found
10:49.45*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
10:50.03FaithfulThe_Ball CODEC?
10:50.14*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
10:50.21The_BallFaithful, allow=all in iax.conf
10:50.39The_Balland im using a TDM card and a analog phone
10:51.58ckruetzegood morning
10:52.01meshuggaFaithful: no, with asterisk
10:52.16meshuggabut i guess bt headsets can work with the softphones if you use that alsa-bluez stack
10:53.00meshuggaalso, i found some other code which is called "handsfree", this put my softphone successfully on the stereo / my soundcard
10:53.07Faithfulmeshugga:  so how do you dial... or um? what does it do?
10:53.14coppicemeghugga: as you said, things are still not reliable. sometimes I get good sound in the bluetooth headset, but the other end gets an awful noise.
10:53.21meshuggaFaithful: Dial(BLT/nameoftheheadset)
10:53.39meshuggacoppice: i told you, /sometimes/ there is a timing problem
10:53.43meshuggabut it is not always the case
10:53.48Faithfuloh cool so you can answer
10:53.55meshuggaif you repeat, the noise will go away
10:54.03meshuggai experienced noice in 1/3 of all cases
10:54.13meshuggasame when using a handset
10:54.23Faithfulhow do I specify the codec on outgoing iax2 calls
10:54.29*** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net)
10:54.58coppicemeshugga: it seems to be less for me. I was starting to think I had something reliable :-) , but no. I get the same result as you. is the noise always in the same direction?
10:55.02*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
10:55.32meshuggacoppice: appaerently
10:55.53meshuggai guess this has to do with the starting of sco_thread
10:55.59*** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
10:56.23meshuggaif you find a way to reproduce the problem reliantly, pls msg me or write to philip@linuxteam.at
10:56.34meshuggaif i can reproduce it reliably, i can debug it too ;)
10:56.36Faithfulmeshugga: how do you pair the device?
10:56.41coppicemeshugga. i don't think so. I found i can call and get silence. when I start to speak it seems like the very first sound I make gets repeated forever
10:56.45meshuggaFaithful: rtfm ;)
10:56.57Faithfulwhich asterisk?
10:57.07meshuggaFaithful: this is not ready to deploy yet
10:57.10coppiceFaithful: the documentation on bluez sucks badly, but there is some around
10:57.22meshuggaif you want to play with it, you need to get in the docs on the net
10:57.49Faithfulit would be so cool for here at work
10:58.05FaithfulI neet to buy a dongle at last I guess
10:58.07*** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53)
10:59.43The_BallFaithful, still nothing? i get no error now, i fixed the username/password
10:59.50meshuggaim off, going hiking
11:01.28coppicemeshugga: something * locks up completely
11:02.34meshuggacoppice: thats the nature of bluetooth ;P
11:03.07FaithfulThe_Ball:  do you have a 1700 number?
11:03.26The_BallFaithful, yes, but im behind a bad firewall
11:03.29meshuggawell anyway, i am away now, pls query me with anything
11:04.00*** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net)
11:04.37*** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
11:05.36The_BallFaithful, but i found on voip-info.org Echo test: 17009999613
11:08.47The_BallFaithful, it't probably that damn xp box that is sharing the dial-up connection which is making trouble. when i get "broadband" in a couple of days and use the linux machine it should work
11:09.18*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
11:09.24ranlivguys why is that when i type zap show channels i get No such command 'zap' (type 'help' for help)
11:09.46ranlivwhen i type help zap command cannot be sen
11:10.27djinare the zap channels running?
11:10.45djinztcfg -vvv
11:10.47ranlivyes doing ztcfg -vv
11:11.10djinwhat is the bottom line of that output?
11:11.13ranliv4 channels configure
11:12.11djindid you restart asterisk already
11:13.22FaithfulAm I dreaming??? is it possible to use my Nokia 6310i as a telephone extension?
11:13.58coppiceFaithful: if chan_bluetooth works properly it should be possible
11:14.01ranlivyes 2x
11:14.23nextimeand if chan_bt don't work you can try also "miax"
11:14.52FaithfulThat is just so cool... so I will have to buy one of those 100 meter bt dongles !
11:15.01ranlivalso reinstalled zaptel many times
11:15.25*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
11:17.34Zeeekdo voicemail boxes have to be numeric?
11:18.13*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
11:18.32djinZeeek, I wouldn't know why they should be.
11:19.45Zeeekonly because you couldn't easily dial them from a phone,
11:20.02Zeeekas  in "mailbox" ?
11:20.06Zeeekprompt
11:20.06*** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl)
11:20.17MuppetMasterHello.
11:20.23Zeeekhi djin and mup
11:20.34MuppetMasterDoes anyone know how to get h.263 working from behind NAT (ie - endpoints behind NAT, Asterisk on public IP)?
11:22.05ranlivI get this warning doing make in zaptel" *** Warning: "zt_register" [/root/zaptel/ztdynamic.ko] has no CRC!"
11:22.11ranlivis this just ok
11:22.24ranlivor will i be concerned
11:24.02coppicenexttime: miax looks interesting. I hadn't heard of that before
11:24.42*** join/#asterisk in (int@stackhack.net)
11:25.00MuppetMasterAnyone on h.263?
11:25.26MuppetMasterI was able to do FWD -> FWD with two endpoints behind the same NAT, but using the FWD nat proxy.
11:26.00MuppetMasterI have enabled verify=yes and nat=yes in sip.conf, but to no avail.  The g729/g711 work fine, but then when I try to launch the video from within Eyebeam nothing happens.
11:26.47fileranliv: did you recompile asterisk afterwards so it compiles the zaptel stuff?
11:27.53*** join/#asterisk mh- (~mh@202.5.145.13)
11:28.26coppicei wonder why miax has chosen to use the world's worst DTMF detector ;-)
11:28.52*** join/#asterisk darby_t (mua@dmx189.neoplus.adsl.tpnet.pl)
11:31.45ranlivno? i do make for pwlib, openh323, asterisk, then make install for asterisk-oh323, asterisk.. after that on zaptel I make radfw.h , make linux26 then make install
11:32.11fileyou have to recompile asterisk so it knows to compile the zaptel stuff now
11:32.32ranlivafter compi;ing zaptel do i need to recomplie asterisk?
11:32.53fileyes
11:35.47*** part/#asterisk darby_t (mua@dmx189.neoplus.adsl.tpnet.pl)
11:47.36*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:49.18*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
11:51.22*** join/#asterisk hemant (hemant@220.226.24.114)
11:53.10The_Ballanyone her get gnophone to work with asterisk?
11:57.35The_Ballhm, do I have to enable something to make asterisk listen for iax connections? Asterisk is not listening to port 5036 at the moment
11:57.56fileIAX2 uses port 4569
11:58.25filethe old old IAX which you must never use, is 5036
11:58.38The_Ballfile, but in iax.conf it says port=5036
11:58.44The_Balloh
11:59.15The_Ballshould i make a config file called iax2.conf ?
11:59.33fileno
11:59.39fileyou can't change the port on IAX2
11:59.47filethat entry in iax.conf is for the old IAX
11:59.57The_Ballso i should just remove that part
12:00.03filejust leave it be... it'll do no harm
12:01.52The_Ballfile, is iax2 backwards compatible? gnophone says it's made for iax not iax2
12:02.02filenope
12:02.06filedon't use gnophone, it's old...
12:02.19The_Ballis there any other good ones?
12:02.34fileGoogle is your best hope
12:06.27ranlivguys why do i get this "ZT_CHANCONFIG failed on channel 1: No such device or address (6)"
12:06.54ranlivbefore it was not there but when i rebooted my pc there it was
12:07.17ranlivis it hardware conflict?
12:07.21filethe devices are not present in /dev?
12:07.56ckruetzefile Have fun at VON, I wish it would be cheaper.
12:08.57ranlivhow do i go about this recompile zaptel? asterisk?
12:09.40fileranliv: find out if your distribution is using udev, if so you need to read the instructions on it
12:09.44fileckruetze: thx
12:09.54file'nor get dressed
12:11.23ckruetzefile: If you find a cool, new VoIP phone, bring one for me, that is if nobody would be missing it.
12:11.25ranlivi'm using fedora core 2
12:11.38fileckruetze: haha
12:12.23fileranliv: dunno
12:12.46fileranliv: Google.
12:13.05*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
12:13.09yaboohi all getting this error under asterisk with both soft phones being xlite
12:13.11yabooMar  6 23:12:10 WARNING[6443]: dsp.c:1469 ast_dsp_process: Inband DTMF is not supported on codec gsm. Use RFC2833
12:13.24fileyaboo: use rfc2833, not inband
12:13.35yaboothanks file
12:13.57filedid you actually read the warning? <- note the word, warning
12:14.14fileI should take a poll
12:15.55*** join/#asterisk zotz (~zotz@24.231.32.191)
12:19.10fileI *must* not freak out
12:20.47filePeter Packet!
12:21.13yaboofile sounds sounds so crap thou
12:21.43coppiceas in "Peter Packet picked a peck of pickled pepper"?
12:22.42filesure!
12:23.40filePenny, and Byte?
12:25.20ZeeekWhy do I get this message?
12:25.20ZeeekApr  1 13:45:10 WARNING[6466]: phones.c:bt100 ast_phone_process: Get a life! Your Grandstream can't do RFC2833! Use INFO, idiot!
12:25.35Zeeekwhat does it mean?
12:25.54fileI should force myself to get out of bed
12:26.05coppiceI think its a phoney message :-)
12:26.06Zeeektoo much information
12:26.22Zeeekcoppice when can I fax you with my bad fax?
12:26.52Zeeekit's a problem during initial negotiation
12:26.58Zeeekloops in the dialogue
12:27.02coppiceI don't need any bad faxes. I have some since I started T.38 development :-)
12:27.13Zeeekisn't this an odd problem tho?
12:27.19Zeeekor is it common?
12:27.47*** join/#asterisk memic (~memic@dsl-084-056-106-237.arcor-ip.net)
12:28.12coppicemany people send and receive large volumes with no trouble. almost everything that goes wrong these days is something specific to the installation
12:28.23*** join/#asterisk TheEmperor (TheEmperor@218.111.50.135)
12:28.58Zeeekwell, this fax software has never had a problem sending to any other machine. I thought it would be of interest to see why it loops when calling spandsp
12:29.35coppiceI am.
12:29.36ZeeekI think I mentioned that j2.com sends me a beautiful zero distortion fax
12:29.37MuppetMasterHas anyone used XTunnels with Asterisk?
12:30.04Zeeekso at least that cool!
12:30.12Zeeek(zero distortion 3 pages of fax)
12:30.40Zeeekspandsp must work with some fax software, because spam faxes seem to get through no problem
12:30.48*** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.res.rr.com)
12:31.54Zeeekmy Sunday cappucino is ready. thanks for all the phish
12:32.56MuppetMasterDoes anyone know how to use ICE (http://www.voip-info.org/wiki-ICE) in conjunction with Asterisk?
12:33.43fileasterisk doesn't support it
12:35.13mh-we're going to setup asterisk shortly for a small demo -- implementing it for a call center in 4 months or so; thought this would be a good spot to sit and listen
12:35.31mh-so far so good :)
12:37.00mh-does anyone have it running on sles 9? a resident asterisk guy we've hired for some reason had issues with it -- got it up and running on rh advanced server
12:42.58Zeeeksles9?
12:44.56yabooanyone using usb-audio under linux
12:47.27mh-Zeeek, suse enterprise server 9
12:47.37Zeeekok
12:48.03Zeeeknever heard of anyone using suse here, but there must be somewhere
12:49.31Zeeekso mh- how's it working?
12:50.02*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
12:51.56mh-Zeeek we haven't got it up as yet -- put it up for an in-house demo on redhat advanced server with no problems
12:52.24Zeeekwhat does "advanced server"  mean in this context?
12:52.57RoyKwindows nt advanced server :P
12:53.09Zeeek~lart RoyK
12:53.13mh-yeah, exactly that :)
12:53.23Zeeekthere wasn't any such thing!
12:53.38ZeeekWindows 200 Advanced server, yes
12:54.19RoyKZeeek: there was a 4.0 NTAS
12:54.32RoyKI was an MCSE at that time
12:55.17Zeeekbut they kicked you out?
13:10.24*** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net)
13:12.10*** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp)
13:12.23Blackveldo I have to configure asterisk for the sip proxy SaRP?
13:12.41Blackvelis there any sip module which transparently rewrites the SIP header?
13:12.56BlackvelI do not want (can't) configure the asterisk server for anything
13:13.49Zeeekwhat is the problem you want to solve, Blackvel?
13:14.34BlackvelNAT
13:14.42Blackvelasterisk sends out the wrong ips, even with externip
13:14.55BlackvelI can not upgrade to cvs head where I could use the syntax variable externhost
13:15.12*** join/#asterisk file2 (~jcolp@mctn1-142166194173.nb.aliant.net)
13:15.15Blackvelso I need some application which rewrites the SIP UDP 5060 packets
13:15.16file2okay I'm packed
13:15.35file2if anybody wants to hear me at the airport, I'll be on from there in the conference in the URL
13:15.39ZeeekBlackvel I use asterisk with NAT on both sides
13:16.46RaYmAn-BxBlackvel: I've heard of it not sending out the externip ip unless localnet is also setup
13:16.59Blackvelfile2: how do you do this?
13:17.09file2there's free wifi at the airport, and I have a wifi phone
13:17.12file2plus I have a cellphone
13:17.22BlackvelRaYmAn-Bx: that is interesting. I have now configured localnet too
13:17.32Blackveli mean okay, even FWD works with my asterisk behind NAT setup
13:17.37Blackvelbut german GMX seems not
13:17.52Blackveli think I'll try to debug sip again
13:17.54Blackveland have a luck
13:17.59ZeeekBlackvel there's no reason why all those SIP things won't work
13:18.06file2I think I have everything...
13:18.15Blackvelfile2: wifi phone thats cool
13:18.22RaYmAn-BxNAT can provide some weird errors SIP...I had huge issues getting sipgate to work with my (highly broken) NAT router
13:18.25Blackvelany good quality? how much does that wifi cost at the airport?
13:18.37Blackvelsipgate (with port forwarding) works quite well
13:18.41file2the wifi at the airport is free
13:18.44file2and the phone itself is great
13:18.58Blackvelsipgate, nikotel, fwd, iaxtel (before) worked great
13:19.00Blackvelbut not gmx
13:19.02Blackveldunno why
13:19.05Blackveli have one way audio
13:19.13Blackvelfile2: what wifi phone?
13:19.24bpointI'm having some weird problems with asterisk's handling of rtp... anyone up for a few questions?  :)
13:19.28file2Hitachi WIP-5000
13:19.42BlackvelIAX2/guest@66.250.68.194/996
13:19.45Blackvelyou are online at this url?
13:19.54file2not right now, I can be though
13:20.46Blackvelhow do you handle all this different extensions with your callphone (or maybe even the WEP/WPA different keys or public hotspot authentication)?
13:20.51Blackvelwifiphone
13:21.12file2it'll automatically associate to the nearest signal
13:21.22file2and if it needs WEP it'll tell me ask me to enter the key or whatever
13:22.46file2I leave in 2 hours 23 minutes
13:24.32Blackvelcan you configure several fixed extensions and locations?
13:24.47*** join/#asterisk mesi (~raoul@dsl-082-083-143-041.arcor-ip.net)
13:24.49file2what do you mean? it just communicates with my asterisk machine...
13:24.59Blackvellike you if can create multiple extensions (short dailing numbers) and different profiles
13:25.02Blackveloh it does
13:25.04Blackvelhow clever
13:25.10file2it's a regular SIP phone
13:25.13file2except it uses wifi for the signal
13:25.22Blackveldo you configure every extension in extensions.conf?
13:25.28file2yes
13:25.36Blackveli am fed up adding all this direct ip calls to extensions.conf :)
13:25.40file[atVON]bbl
13:26.28file[atVON]I shall be on the conference when I can be btw
13:26.37mesiWhen somebody calls asterisk and immediately hangs up, without a connection being established at all, can asterisk call back the given callerid and offer a menu?
13:27.16Zeeekmesi if it gets the callerid, ya
13:27.45mesiZeeek: It gets. But I have no clue how to make it call back.
13:27.58Zeeeklook up call files on the wiki
13:28.10mesiZeeek: Ok, thanks.
13:28.30Zeeekor just use a wait() and then dial the number back
13:28.46bpointanyone have any idea why I might be getting a continuous stream of "NOTICE[7086]: rtp.c:451 ast_rtp_read: RTP: Received packet with bad UDP checksum" notices from asterisk?
13:29.00ZeeekDial(Technology/${CALLERIDNUM},time, options)
13:29.17Blackvelanyone used siproxd with asterisk?
13:29.25Blackvelhow can I configure asterisk for this?
13:29.48Blackvelis anybody using broadvoice?
13:30.21GrimStonebroadvoice is presently screwed , Blackvel
13:30.33ZeeekBlackvel would this be of interest? http://lists.digium.com/pipermail/asterisk-users/2004-April/044626.html
13:30.39GrimStonethey did something yesterday which has pretty much broken asterisk
13:31.40mesiZeeek: Yes,  that might work. If it is like the context is executed even when no connection has been established.
13:31.49Blackveloh it is?
13:31.59Blackvelthat 20$ for everything in the world is so interesting
13:32.08Blackvelohoh
13:32.15Zeeekmesi if a call comes in, it drops into the context. You do whatever you want after that
13:32.18GrimStoneyeah i got 1 of those plans
13:32.21Zeeekinlcudinf NOT answering it
13:32.26GrimStoneworked great until yesterday
13:32.39Blackveli talked to someone with broadvoice (twice asterisk) two weeks ago
13:32.41Zeeekwhy does everyone want BV ? Are they way cheaper than the rest?
13:32.52Blackvelwell 20$?
13:33.00Blackvelwhat other providers do you know?
13:33.05Zeeekor maybe they have more local DID?
13:33.18Zeeekwhere are you, USA?
13:33.19GrimStoneZeeek: they got real unlimited
13:33.23Blackvelno
13:33.24Blackvelme germany
13:33.32Zeeekyou want to call USA?
13:33.33Blackveland I am thinking to get a german flatrate
13:33.48Zeeekare most of your calls to US?
13:33.49Blackvelbut if I would get a flat for the same price, and I could also call UK/USA
13:33.52Blackvelthat would be even better
13:33.57Blackvelno germany
13:34.00ZeeekI use several providers
13:34.07Blackvelbut I want to change things
13:34.21Zeeekhere in France we now have a national unmimited SIP for 8eu/mo
13:34.32ZeeekI guess Germany should too
13:34.35BlackvelI don't really want to have 3 providers that would be something like 3x20$ :)
13:34.53ZeeekI only pay 8eu§mo for ten providers
13:35.03Zeeekall the rest are pay as you go
13:35.08Blackvelflat?
13:35.12Zeeek(not unmimited)
13:35.15Blackvelah
13:35.15Zeeekno
13:35.18Blackvelper mintue
13:35.23Blackvelminute
13:35.24Zeeekbut are you sure you need unlimited?
13:35.27Zeeekwe don't
13:35.40Blackvelno i am not
13:35.52Zeeeklike you could use pay as you go for international
13:35.54BlackvelI just know my german costs for a familay are usaually more than 30 Euros per month
13:35.57Zeeekit's very cheap
13:36.00Blackvelarcor german flatrate is 20EUR
13:36.03Blackvelbut without US/UK
13:36.18Blackvelmaybe its enough to use for UK/USA per minute billing, but who knows
13:36.19Zeeekdoes sipgate.de do unlimited or very cheap German calls?
13:36.36Zeeekwell you need to know about how many minutes you'll use per month, ya?
13:36.41Blackvelvery cheap is the word
13:36.45Blackvelyou can buy 8,90EUR
13:36.50Blackvel1000 minutes
13:36.50Zeeekand with asterisk, you can count up the seconds
13:36.51ranlivI got it working, I just transfered the card to a diffrent PCI slot
13:36.56Blackvelthats 0,89EUR cent
13:37.10BlackvelZeeek: i am not too sure
13:37.21BlackvelI think it was 15 hours or something
13:37.24BlackvelI would have to check again
13:37.35Blackvelbut my monthly costs are more than 20EUR
13:37.39Zeeeklike I said, for family use in this country we have 8.00eu§month for the whole country
13:38.05The_Ballwhat exactly happens on "Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996" ?
13:38.25Zeeeka bunch of devs talk for three hours and you listen
13:39.20Blackvelbtw
13:39.32Blackveli have the urgent request for a voip traffic calculator
13:39.38Blackveli am trying to find a good url on voipinfo
13:39.44Blackvelis tehre anything like this?
13:39.59Zeeekyou mean bandwidth vs calls?
13:40.19file[atVON]eek we aren't leaving yet
13:40.23Blackvelhe needs to know how far to come with 1 gIG for g711/g726
13:40.45Blackvelfile let me call you
13:40.51Zeeekstatr a call and measure the bw used on the asterisk box
13:40.58file[atVON]I don't have the phone on, conserving battery
13:41.03Blackvelah ok
13:41.14Zeeekfile is waiting for Mr. Right
13:41.23Blackveli dont care :)
13:41.25file[atVON]file is waiting to go to the airport
13:41.35Zeeekbetter than waiting for Godot
13:41.36Blackvelwhere are you flying to
13:41.38Blackvel?
13:41.39file[atVON]so he can board his flight to Toronto
13:41.42file[atVON]then to Chicago
13:41.44file[atVON]then to San Jose
13:41.48*** join/#asterisk Ron-Na (~ronald@203.70.36.126)
13:41.50file[atVON]so he may be at VON, http://www.von.com/
13:42.33Blackvelah
13:42.37Blackvelcool
13:42.44Blackvelhow expensive is that trip then?
13:43.04file[atVON]for normal people? expensive
13:43.09Zeeeknot to be confused with http://www.vons.com
13:45.11*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
13:45.26coppicehow much for weirdos, then? :-)
13:45.48file[atVON]and this is my *first* time flying
13:46.54coppicethe important thing to ensure is its not the pilot's
13:47.08Blackvelfile: are you normal or not? what is a normal person btw? :)
13:47.27file[atVON]a normal person is someone who doesn't work for a VoIP company, and is not an asterisk developer
13:47.46Blackveloh
13:47.49Ron-NaHas anybody SJphone working registered to an Asterisk box?
13:47.51Blackveldigium is sponsering it
13:47.52Blackvelgreat
13:47.52Blackvel:)
13:48.06*** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1)
13:48.07file[atVON]no they aren't, they have a pavilion though... the... Asterisk Pavilion!
13:48.20Blackvelhow do you chat for the moment? laptop and wifi?
13:48.28file[atVON]I'm at home on my workstation
13:48.34file[atVON]we haven't left to go to the airport yet
13:48.38Mw3goddamn i'm a normal people :(
13:48.39Blackvelahhhh
13:48.48file[atVON]I'm trying to think if there is anything I Forgot to pack
13:48.51Blackveli need someone "un-normal"
13:49.04Blackvelto back port externhost into asterisk 1.0.6
13:49.22GrimStoneUnable to create channel of type 'SIP' (cause 3)  <== what does this mean ?
13:49.22file[atVON]all my power adapters...
13:49.30*** join/#asterisk afe ([snGf3oJZj@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
13:49.35Blackvel2nd battary pack?
13:49.47Blackvelpda?
13:49.57Blackvelyour VIP pass ticket?
13:50.08Blackvelfirst call ticket?
13:50.16Blackvelclass
13:50.31*** part/#asterisk ranliv (~ranliv@210.5.85.249)
13:50.33Blackvelwho runs openwrt + siproxd + asterisk voip?
13:50.53file[atVON]well, I have my plane ticket
13:50.59file[atVON]and my pass for VON is... well, in San Jose
13:51.29Blackvelwhere do you live? how long do you have to fly to san jose?
13:51.41file[atVON]it'll take me 16 hours
13:51.50file[atVON]cause I live in Canada on the other side of the country
13:52.11Darwin35File come home we miss you
13:52.18file[atVON]ha
13:52.21Darwin35Toronto is calling you back
13:52.21file[atVON]I haven't left yet
13:52.26Darwin35hehhe
13:52.28file[atVON]and I don't live in Toronto :p
13:52.37file[atVON]thank god I don't live in Montreal
13:52.42file[atVON]I purposely booked this flight just to avoid Montreal
13:52.50Darwin35http://www.von.com/schedule_wifi5.htmlwait your a frenchie right
13:52.55Darwin35grrr
13:52.59file[atVON]nope
13:53.02file[atVON]I'm english :p
13:53.06file[atVON]muahahahaha
13:53.09Darwin35i hate cut and paste
13:53.14file[atVON]Moc is french, and Junky is french
13:53.30Darwin35<---is scottish Canuck
13:53.43Blackvelcan I configure asterisk to use siproxd?
13:54.13coppiceDarwin35: did you head to canada for warmer weather? :-)
13:54.27file[atVON]I'm heading to San Jose for warmer weather, that's for sure
13:54.30file[atVON]I can actually... wear shorts!
13:54.57Darwin35I wear shorts and kilts all year round
13:55.12file[atVON]Darwin35, get pneumonia every year too don't you
13:55.25Darwin35nope
13:55.31Darwin35healthy as a horse
13:55.34file[atVON]uh huh a likely story
13:56.06Darwin35I have 6 utility kilts  4 family kilts and 7 pairs of shorts
13:56.11coppicea horse as in the expression "flogging a dead *****"
13:56.22file[atVON]I must soon go
13:56.30file[atVON]very very soon
13:56.35file[atVON]everyone wish me luck
13:56.55Darwin35I wanted to go to Von but the company would not give me the time off or pay for it
13:57.06file[atVON]be back later, maybe from... an airport... or something
13:57.07Darwin35have fun for me
13:57.13file[atVON]maybe cafe... I dunno
13:57.33file[atVON]oh, and if bkw, twisted, paulc, drumkilla, or kram show up tell them I'm on my way
13:57.34*** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net)
13:57.54ckruetzeDarwin35: you are not the only who can't go to VON :(
13:57.59Darwin35you all must be sharing a room
13:58.14\Grooby\VON?
13:58.27Darwin35lets see if they setup a asterisk box for chat from there
13:58.35Darwin35www.von.com
13:58.38coppicecan you really wear shorts in san hose now? its north of here, and i'm bloody cold right now.
13:58.58\Grooby\ahhh ok
13:59.00Darwin35I wear them all year round
13:59.05Darwin35I have thick blood
13:59.23Darwin35but I wear flannel boxers to keep the boys warm
14:00.36Darwin35we no do regamental in the winter
14:00.50YoYook, so what's new in asterisk land?  can I send mp3 attachments for voicemail yet?
14:01.03Darwin35and in the summer in the office we have to wear a jock company rules
14:01.26Darwin35read the wiki it tells all
14:02.38YoYobah... yer no fun
14:02.57YoYoand neither is that wiki... it's overloaded with disorganized info
14:03.35Darwin35then go organize it
14:03.46Darwin35put yourself to some use
14:03.54Darwin35help klean the place
14:06.32Blackvelhm
14:06.44Blackvelsiproxd runs without cahgning asterisk configuration?
14:06.55Blackvelthere is simply no sip outgoing proxy gateway option in asterisk
14:06.57Blackvelonly chansip2
14:07.02Blackvelnot sure how smooth it runs
14:08.08mesiIs there a Dial command wich continues execution after a connection has been established?
14:09.07*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
14:09.58Blackvelwhat do you want to do?
14:11.25mesiI want to Dial out to somebody and put him into a certain context as if had dialed in to my * box
14:11.47Blackveli dont know what you mean
14:12.27mesiWell, if somebody dials in to my box, he is in a special context.
14:12.54mesiI on the other hand, want asterisk to call out and if the called party picks up the phone, it should be like he had called in to my asterisk box.
14:13.34mesiUnfortunately, Dial() will stop executing the current context. So if I use Dial() to dial out then the called party wouldn't have a context executed and I cannot offer a menu or the like.
14:13.50GrimStonewhats wrong if i Dial on a SIP channel and can hear the other side fine , but asterisk keeps saying "Got a response on a call we dont know about."
14:15.00Blackvelmesi: hm...I have never done this. maybe AGI is the way to go
14:15.02GrimStoneand the CDR records show the call as "NO ANSWER" , even though it got connected
14:15.46roamer323mesi - call out, once pickedup - transfer to an extension -> the 'outside guy' will get context of extension (for dialplan etc)
14:16.20*** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co)
14:16.49mesiBlackvel: I'll check that, thanks.
14:17.26mesiroamer: Ok, but the outside guy would have to do this transfer, right?
14:17.33mesiThat's not comfortable. ;-)
14:17.52BlackvelYES YES YES STRIKE STRIKE STRIKE
14:18.00Blackvelasterisk 1.0.6 and GMX works now
14:18.13Blackveleven without the externhost but only externip feature
14:18.40mesiBlackvel: You mean, GMX internet telephony?
14:19.00Blackvelyeah!
14:19.08Blackvelthere had been a problem in asterisk 1.0.2
14:19.18Blackveldunno what is was, something with sending out NAT ip even it shouldnt
14:19.38mesiwell, I should register with gmx, too. Do you have to pay a fee?
14:19.47Blackvelno
14:19.53Blackvelcosts are 1ct
14:20.08Blackvelyou have to register with gmx and give your landline number
14:20.25roamer323mesi - actually , you can do the transfer without the 'outside guy' knowning any diff - but it is a kludge :-)
14:20.28Blackvelthey will send you a bill each month and pay your bank
14:22.38Blackvelso what has broadvoice changed?
14:22.43Blackvelthat asterisk is not working anymore?
14:24.56mesiroamer323: So how does this kludge work? ;-)
14:25.16Blackvelhm
14:25.20Blackvelif you call the person
14:25.27Blackvelis it really clever to show up a IVR menu?
14:25.30Blackvelan
14:26.22mesiBlackvel: So I would not be able to register with GMX for free to do internet telephony?
14:26.46*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
14:26.51roamer323blackvel - I think mesi is doing a callback - and call through (LD savings from remote part of the world?)  the outcalled party can probably blind-dial
14:26.51mesiroamer: I can use M(mymacro) and call Transfer in this macro?
14:27.23roamer323mesi - yes - or the manager interface with an outside monitoring script (big kludge)
14:27.42Blackvelmesi: you can
14:27.47Blackvelbut pstn calls cost something
14:27.51Blackvelas other providers do
14:27.51mesiroamer: Yes, if I use my answering machine, I would make people angry when using their phone ;-)
14:28.16mesiroamer : so I have to make my asterisk call me back on an outside pstn.
14:28.40mesiBlackvel: ah, I see.
14:28.45*** join/#asterisk massivexb (~mirc@HSE-Toronto-ppp300289.sympatico.ca)
14:29.03massivexbhey folks what operating system will be the best to install asterisk onto?
14:29.45mesiroamer: The macro is better. It would be a one - line - macro.
14:29.56roamer323mesi - I agree :-D
14:30.01mesimassivexb: any free unix or derivate. Linux, FreeBSD or the like.
14:30.14mesiroamer: Ok :-))
14:30.35Blackvelis there some changeliste of asterisk 1.0.6?
14:30.44Blackvelchange list
14:30.55Blackvelsuddenly NAT IPs are working
14:31.14massivexbcool thanks.. was going to install on freebsd but wasnt sure of compatibility :)
14:31.26*** join/#asterisk sudhir492 (~sudhir@4.7.58.171)
14:31.34sudhir492hi all
14:31.38massivexbdoes anyone know the digium products well?
14:31.50sudhir492Anyone using a quad T1/E1 card here
14:32.04sudhir492massivexb: what do you want to know
14:32.40massivexblooking at gettign a 24 channel pri installed wanted to know what card to get to interface with it
14:32.56NukemizerDo any Readhat users know where I can get the kernel-source ? I downloaded and installed FC3 but canno find the kernel-source on the CD's or through googling
14:32.58sudhir492for US right
14:33.03massivexbcanada
14:33.08massivexbprobably same as us
14:33.12sudhir492yes
14:33.19*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
14:33.19*** mode/#asterisk [+o bkw_] by ChanServ
14:33.32bkw_yo yo yo
14:33.35bkw_guess file is on his way
14:33.58D1ng0bkw_, where ya going
14:34.02Mw3von ...
14:34.07bkw_VON
14:34.27sudhir492I have been using T1 cards for about a year. They work well
14:34.36D1ng0hey bkw_ what VOIP provider you using ?
14:34.51sudhir492newer card TE110P should work well too. I have not used them yet
14:34.59bkw_D1ng0, um
14:35.01bkw_i'm a provider
14:35.01massivexbare you running them on freebsd?
14:35.04bkw_asterlink.ccom
14:35.14bkw_er asterlink.com
14:35.15bkw_haha
14:35.17bkw_double c's
14:35.31sudhir492bkw_: do you use quad T1/E1 card at all?
14:35.41bkw_yes
14:35.57bkw_why oh why do you ask?
14:36.20D1ng0cause BroadVoice has literally screwed alot on people inbound calls are not working
14:36.35sudhir492I need to prepare a system for that which can get pretty busy sometimes
14:37.11sudhir492I want to know a good combination of hardware that can take significant load,
14:37.32bkw_well for that my friend you'll either have to pay someone
14:37.35bkw_or learn like we did
14:37.37bkw_:P
14:37.44bkw_that information is worth its weight in gold
14:38.07sudhir492bkw_: If I had gold, I wouldnt be here :-(
14:38.27bjohnson_silver?
14:38.31bjohnson_diamonds?
14:38.43sudhir492bkw_: how much do a few megabytes weigh anyway? :-)
14:39.12bkw_no clue
14:39.19sudhir492bkw_: what did you learn? :-)
14:39.25D1ng0even platinum
14:39.31bkw_what not to do
14:39.40massivexbgold pressed platinum :)
14:39.53massivexbwhat os are you unning bkw
14:40.46GrimStonewhats wrong if i Dial on a SIP channel and can hear the other side fine , but asterisk keeps saying "Got a response on a call we dont know about."
14:40.54bpointwould anyone have any idea why asterisk would be spewing "NOTICE[7210]: rtp.c:451 ast_rtp_read: RTP: Received packet with bad UDP checksum" messages at me? :/
14:41.02GrimStoneand the CDR records show the call as "NO ANSWER" , even though it got connected
14:41.22bkw_ok guys
14:41.25bkw_i'm putting my laptop away
14:41.28bkw_and waiting on my flight
14:41.30bkw_HAVE FUN
14:41.32bkw_see you in denver
14:41.34*** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
14:43.01Blackveldenver?
14:43.07Blackvelwhat does bkw do in denver? :)
14:43.56mesiWhen executing the Dial() M Macro, I get the error: Could not stop autoservice on calling channel.
14:47.21NukemizerDo any Readhat users know where I can get the kernel-source ? I downloaded and installed FC3 but cannot find the kernel-source on the CD's or through googling
14:48.26*** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz)
14:49.00dfunnellHi, anyone online who can help regarding CAPI (namely dialing out using CAPI)?
14:49.51Mw3~bluetooth
14:49.52jbotit has been said that bluetooth is at http://www.handhelds.org/z/wiki/Is%20anyone%20working%20on%20Bluetooth%20for%203870 or at http://dnsv6.iihe.ac.be/iPAQGPRSv6/BtConfig.html .. and you might wanna take a look at http://bluez.sourceforge.net/contrib/HOWTO-Mobile-Phone
14:50.09Mw3hm
14:50.15Mw3where is the bluetooth channel driver ?
14:51.19BlackvelContact: <sip:<mynumber>@213.23.34.236>
14:51.20Blackvelcool
14:51.25Blackvelthe sip header is fine with 1.0.6
14:51.39Blackvelhmm bluetooth
14:51.40Blackvelfor pda?
14:51.46Blackveldo you try to use voip?
14:52.40Mw3no, somebody said here that i can use my mobile phone as a handset with chan bluetooth and an usb dongle
14:52.46Mw3i'd like to try
14:53.23cbachmanNukemizer, I remember reading something to the effect that with one of the FCx releases they started only releasing it as a SRPM
14:54.20Nukemizercdbachman: thank you I will get those CD's ( big help thanks much)
14:54.56*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
14:56.12tzafrirNukemizer, if it's not in a separate package, get the SRPM of the kernel and only build it up to the "patch" stage (rpmbuild -bp)
14:57.59ckruetzeMw3: Try http://tribble.crazygreek.co.uk/content/chan_bluetooth
14:59.02Nukemizertzafrir, thanks i will try
14:59.26tzafrirrpm -i kernel-something.srpm
14:59.48tzafrirrpmbuild -bp ~/RPM/SOURCES/kernel.spec
15:00.00tzafrir(you should build RPMs as non-root)
15:00.12*** join/#asterisk Tarox (someone@pD9E7B5D4.dip.t-dialin.net)
15:02.17Nukemizertzafrir, thank you for guidance :) this helps
15:05.27dfunnellHi all - trying to dial-out via CAPI, but keep getting message 'didn't find capi device with outgoing msn = xxx', where 'xxx' is the MSN number I'm trying to use.  Have spent two days trying every possible combination of MSN numbers, local telco (Telecom NZ) provided MSN gives same error.  Can anyone help?
15:07.39*** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp)
15:09.49*** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net)
15:10.15Mneumonichey anyone know why on my sipura 841 it takes like 10 seconds after i dial an internal extension to connect to it?
15:11.37sudhir492I have a Polycom phone behind a firewall that registers fine with the Asterisk and make calls too. However, I am not able to reach it from outside. Anyone has a suggestion for that? I do have nat=yes and canreinvite=no in sip.conf
15:11.49bjohnson_anyone understand traceroute?  trying to track down a slow ping problem and traceroute keeps giving me asterisks instead of data
15:12.01RaYmAn-BxMneumonic: assuming the sipura 841 works anything like Sipura SPA-2000, it means you are lacking a dialplan entry for the numbers you dial.. (i.e. if internal numbers are 6 digits you need a setting that tells it to accept 6 digit numbers..)
15:12.36*** part/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp)
15:12.49sudhir492Mneumonic: also try entering '#' after the digits
15:12.53MneumonicRaymAn - you mean the dialplan on the phone itself?
15:12.56RaYmAn-Bxbjohnson_: that means it doesn't get a reply usually..
15:13.01RaYmAn-BxMneumonic: yes
15:13.25RaYmAn-Bxand as sudhir492 says it will prolly do it immediately if you type # after the number
15:13.45*** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp)
15:13.56Mneumonicyea, it does...
15:14.02Mneumonicthis is the phone's dialplan
15:14.04Mneumonic(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
15:14.14Mneumonici have 3 digit extension
15:14.45Mneumonicso the first batch of x's should only be 3?
15:16.02bjohnson_does a nat router port need to be forwarded to get traceroute to work?
15:16.11RaYmAn-Bxonly add to it..but yeah, you just need to add an xxx part
15:17.09RaYmAn-Bxbjohnson_: not afaik, but it requires a few things...Generally traceroute uses UDP, so it relies on getting certain replies about the attempted UDP send...it could either mean that the servers refuse to send these or that they simple doesn't reach you
15:17.32*** join/#asterisk jsolares (~jsolares@200.12.33.64)
15:17.40bpointtraceroute uses icmp, not udp
15:17.44RaYmAn-Bxtry with traceroute -I (depending on the version of traceroute you have) and reach the manpage for more information
15:17.44sudhir492Mneumonic: there is an ambiguity for 3 digit extensions in the dialplan, hence your best bet is to dial a '#' indicating end of dialing. Another approach can be to have 3 digit extensions *xxx that way you will have dial a star in the beginning but it will resolve the ambiguity
15:18.23bpointmost servers filter icmp packets for security reasons
15:18.48bjohnson_-I did the trick
15:18.56RaYmAn-Bxbpoint: it depends on the implementation. The traceroute on my server uses udp. (unless you tell it to use ICMP with -I)
15:19.11bjohnson_what do the three ms times mean?
15:19.12Mneumonicsudhit - thanks :)
15:19.22RaYmAn-Bxthere are also implemenations using TCP
15:19.24Mneumonicerr sudhir
15:19.37bpointrayman: that's interesting... first for me :)
15:21.38RaYmAn-Bxbpoint: I believe it sends a UDP packet to a port nothing is expected to listen on..it sends it with a ttl so low that each router on the way sends back an ICMP TIME EXCEEDED reply, hence getting the addresses and response times for each router on the way
15:21.47sudhir492anyone here an idea why I can only call from a phone not receive calls there? They register with Asterisk fine.
15:21.50RaYmAn-Bx(as far I can understand..Not sure it's entirely accurate)
15:23.02jsolaressudhir492: is it behind nat?
15:23.05bpointrayman: hmm.. sounds like it would work the same as icmp
15:23.22bpointI've just never personally seen a traceroute use udp tho :)
15:24.43RaYmAn-Bxbpoint: the difference is that routers happily send out ICMP errors :) Sure, they might block ICMP ECHO's (ping), but it would be silly to block ICMP errors
15:26.53*** join/#asterisk Ron-Na (~ronald@203.70.36.126)
15:27.28sudhir492jsolares: yes it is behind nat and I have nat=yes and canreinvite=no in sip.conf
15:27.48jsolarestry with qualify=yes as well
15:28.03Ron-NaHas anybody setup a softphone on a PDA? I tried SJphone, but it does not register, ... any hints?
15:28.12sudhir492thx. I will try. what does qualify=yes mean?
15:28.13jsolarescan you atleast receive calls like a second after it registers? (it should)
15:28.38jsolaressudhir492: that'll it'll poke the device from time to time to keep the connection open and up
15:30.11sudhir492jsolares: maybe. Because, once, and only once the phone rang when called. :-(
15:30.36jsolareshope and pray that your phone isnt one of the ones that die if poked via qualify=yes then
15:30.55jsolaresit will work, that's how i have the phone in my house behind a nat connecting to the * box in the office
15:34.16sudhir492jsolares: I set qualify=yes, and on reload I see the message that the extension is no longer reachable
15:34.39bpointis there anyone out there that has to disable rtp checksums?
15:35.03sudhir492jsolares: It is a Polycom 500, hence I presume that the phone should be stable.
15:35.38sudhir492Unfourtunately, I cannot physically verify that as the phone is 40 miles away from my home right, with no one in the office
15:35.57RaYmAn-Bxbpoint: I ended up disabling it because I got a fair amount of checksum errors, without any problems with sound or anything
15:36.43bpointrayman: after ~0.5sec into a call, asterisk is just spewing out udp checksum errors... :/
15:37.01Zeeeksudhir492 some phones (and some providers) can't qualify
15:37.20Zeeekdon't ask me why
15:37.21Blackvelwho knows freenet and gmx?
15:37.35bpointit only happens when dialing out remotely or when a call comes in.  local extensions have no problems (weird)
15:37.54RaYmAn-Bxbpoint: do you have any sound problems or anything?
15:38.47bpointonce the udp checksum errors start, the downstream audio gets dropped (internet -> asterisk)
15:39.19bpointI don't think this is an asterisk problem, though...
15:39.25RaYmAn-BxHow is it with rtp checksums off?
15:39.38RaYmAn-BxNo, It sounds like a bad connection (or a dumb router that breaks things)
15:39.43bpointno different :)  it still shows the errors too
15:40.00*** join/#asterisk Voip_Help_Me (TheJudge@196.46.66.98)
15:40.00bpointit could very well be this crappy router
15:40.10Voip_Help_Mehello all
15:40.18sudhir492jsolares: it seems that qualify=yes did the trick. I had to restart asterisk though. Now when I call this extension, it goes to voicemail after 20 seconds, whereas another extension without qualify=yes goes to voicemail right away !
15:40.19vaewynok... so who's gonna be in San Jose tomorrow?   :}
15:40.24psywarIs it normal for the SPA-2000 to gen UDP packets with bad checksums?
15:40.33psywarthey're cutting off my calls.
15:40.33bpointsince it works fine on the inside of the network.... hmm
15:40.44Zeeekthis seems to be a bad checksum day!
15:40.49Voip_Help_Medoes any one know how to priopritse voip packets ? usign sip phones and have cisco infrastructure ?
15:40.53bpointpsywar: bad checksums for you too?
15:40.56bpoint:)
15:40.58psywaryeah
15:41.08bpointI've been poking through rtp.c, but there's nothing out of place
15:41.26bpointthe kernel would have to be returning EAGAIN from the recvfrom() call anyway
15:41.29Zeeeksomeone put a backdoor in the checksum code and now they're all sabotaged muhahaha
15:41.29bpoint*sigh*
15:41.37psywarI replaced a homemade cat5 cable with a professional one, haven't had another yet.
15:41.45psywarBut I havne't been on it that much
15:41.49bpointzeeek: the kernel is doing the checksums, afaik
15:41.54RaYmAn-Bxrtp checksums shouldn't be the same as udp checksums though?
15:41.59bpointyeah
15:42.12bpointer?  they are, aren't they?  :)
15:42.28Zeeeksomething just occurred to me: I saw some of those yesterday; I just switched to 1.0.6 - what vers are you guys using?
15:42.29bpointrtp is over udp
15:42.33RaYmAn-BxWhat's the point in naming them entirely different then?
15:42.47RaYmAn-Bxyeah, but isn't it simple udp checksums then? If that's all it is
15:42.50bpointzeeek: cvs head as of friday :)
15:42.59ZeeekI got 1.0.6 stable friday
15:43.01bpointrayman: that's what it should be
15:43.06*** join/#asterisk MikeJ[Jayden] (~ircatjerr@adsl-69-212-48-186.dsl.sfldmi.ameritech.net)
15:43.28psywarI'm on 1.0.5
15:43.41bpoint1.0.6 didn't compile for me
15:43.53bpointso I had to get cvs head... :/
15:43.59Ron-NaIs anybody using SJphone to connect to *  ?
15:44.05psywarUDP suggests either the SPA-2000 is messed up or something  is mangling packets
15:44.05vaewynouch... bus error on my linux sparc box... on a simple IAX2 auth
15:44.21psywar~docs
15:44.22jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:44.30Voip_Help_Meanyone know sip priopirty for a cisco router ?
15:44.32*** join/#asterisk MikeJ[Jayden] (~ircatjerr@adsl-69-212-48-186.dsl.sfldmi.ameritech.net)
15:44.58dfunnellHi all - trying to dial-out via CAPI, but keep getting message 'didn't find capi device with outgoing msn = xxx', where 'xxx' is the MSN number I'm trying to use.  Have spent two days trying every possible combination of MSN numbers, local telco (Telecom NZ) provided MSN gives same error.  Can anyone help?
15:48.29psywarI just realized I could make a digit on my "local extension" context go to my "dial-in welcome" context, so I can stop calling my own number on the cell now.
15:48.32psywarfor debugging
15:49.00*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
15:49.23psywarDoes anyone have any theories or guides on mapping out your contexts?
15:49.46ManxPowerKeep them simple enough to easily understand.
15:50.20psywarMine are like spaghetti code
15:50.29ManxPowerpsywar: Then simplify them.
15:50.35bpointI gotta get to bed... I'm gonna poke with this rtp stuff later *sigh*
15:50.49*** join/#asterisk MikeJ[Jayden] (~ircatjerr@adsl-69-212-48-186.dsl.sfldmi.ameritech.net)
15:50.50sudhir492jsolares: qualify=yes makes the phone more reachable. However it is still not foolproof
15:51.21sudhir492is there some kind of keepalive that need to be set on the phone?
15:51.23ManxPowerFor example all incoming calls from untrusted sources come into the [incoming] context.  From that context I include [extensions] and that's about it.
15:51.28ariel_Morning all
15:51.29psywarwhat are the different valid states?  I saw an error recently about not having an "h" state.  What is that?  Never heard of "h", just "s" and digits.
15:51.42vaewynh is hangup
15:52.04RaYmAn-Bxsudhir492: if it's only a single phone (or a single asterisk) behind the NAT and you have access to it, forwarding the needed ports would prolly help
15:52.05ManxPowerpsywar: You must have been running in debug modde.  "h" is called when the far(?) end hangs up.
15:52.08Zeeekpsywar /dial
15:52.12ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
15:52.21psywarcool ty
15:52.46sudhir492RaYmAn-Bx: Actually there are multiple phones
15:54.01RaYmAn-Bxin that case you could set them up to use different ports for the phones (i.e. 5061 for second phone, 5062 for third, etc) and do static mapping to them...It's not exactly a plug and play solution (and it doesn't really work with dynamic ips)
15:54.08*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr)
15:55.08PoWeRKiLLHi
15:55.33PoWeRKiLLI have a strange sip error the same peer suddenly got a 401 unauthorized
15:56.03sudhir492RaYmAn-Bx: Yes, in this case making a static mapping like that creates more problems that solves.
15:56.12sudhir492than it solves
15:56.37PoWeRKiLLafter day and day of good connection I trace sip packet and see the 401 error but password are correct on both side
15:57.24Blackvelanyone has some workaround for this broadvoice / asterisk problem yet?
15:57.24RaYmAn-Bxsudhir492: in that case try and increase the nat timeout on the router
15:57.27sudhir492any idea what causes this kind of behavior? I suspect either the router in the middle or the phone is dropping the NAT connection
15:57.54Blackveli mean how can it be that the call goes to voicemail instead of to asterisk?
15:58.05Darwin35yes my fbsd-asterisk -cf drive works
15:58.19Darwin35512 meg
15:58.48Darwin35just no festival of sphinx
15:58.51sudhir492RaYmAn-Bx: thx for confiriming my hunch.  Is there a way to set time for qualify in asterisk?
15:58.56Darwin35thats next
15:59.22Darwin35is kram around
15:59.31sudhir492If that is possible, I can tweak that, i.e. make that smaller than routers timeout
16:00.34Zeeekexit
16:00.39Zeeeknot.
16:01.11ManxPowerIt's Caffiene Awareness Month?
16:01.28Zeeekwhat about sleling awareness?
16:01.55ManxPowerOrganized my anti-caffiene people, I'm sure.  Damn tree huggers.
16:02.01tzafrirManxPower, you mean "a month in which we really need caffeine to be aware of things"?
16:02.06sudhir492RaYmAn-Bx: I found in the wiki. qualify=xxx in milisecs
16:02.18vaewynHey guys...  on SIP stuff... are the RTP ports initiated by the server or the phone?  ie will UDP session tracking be enough to let them out a firewall or do I have to open up all those ports to any incoming as well?
16:02.28Mocqualify is being BANNED from all my config file (except IAX peer)
16:02.30ManxPowervaewyn: both.
16:03.01vaewynManxPower: so i need to leave a big gapping hole in the firewall?
16:03.10ManxPowervaewyn: Yup!
16:03.16ManxPowervaewyn: Sucks, doesn't it?
16:03.21Zeeekwhat are they gonna do with 1000-10100?
16:03.24vaewynohh geese... and I thought the NAT part was bad enough
16:03.24Mocthere is a big problem in the SIP message managements with qualify
16:03.45ManxPowervaewyn: BUT if Asterisk is on a public ip, then the SIP client can be behind NAT and things are usually just fine.
16:03.56sudhir492Moc: thx for the warning
16:04.13vaewynThe * is on a public IP...  but is also behind a firewall so...
16:04.18Zeeekvaewyn I open 10000-10100 because we dion't have a lot simul calls
16:04.24Mocsudhir492, np, I keep having phone get 2500ms+ so it get unrechable, but it local...
16:04.35*** join/#asterisk nctk (~nctk@lsne-catv-dhcp-29-238.urbanet.ch)
16:04.48vaewynZeeek: how many RTP ports does it use per call?
16:04.51Mocbest way to see the problem is set qualify, and check what is the MS with your peer on a local network
16:04.52roamer323vaewyn: double-NAT ... you're playing with explosive
16:05.01ZeeekI would think 2-4
16:05.03vaewynroamer323: is NOT NAT...
16:05.09vaewynis justa  firewall
16:05.13MocNEVER NAT your *..
16:05.21sudhir492Moc: what is the solution? The clients router is probably unconfigurable for me. I dont know the password and nor do they :-(
16:05.30Mocunless your dealing with IAX
16:05.52*** part/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
16:05.52ZeeekMoc I have double NAT
16:05.52ManxPowervaewyn is one of those people that realize that NAT != firewall, even though a lot of people (myself included) use them to mean the same thing.
16:05.56MocZeeek, your in for a bunch of problems...
16:06.04Zeeeknone yet after a year
16:06.26Zeeekmy asteriks anniversary is coming up soon
16:06.30Mocsudhir492 ?? I used to lower the register on the phone to 60 second
16:06.37ManxPowerZeeek is one of those people that the Universe Smiles Upon and does not seem to have NAT related issues, even in the most macabe configurations.
16:06.42Mocit get anoying within *, but it working fine
16:06.49sudhir492Moc: My asterisk box is on a public IP, no nat at all. its the darn phone
16:07.04roamer323ManxPower - NAT==firewall for all home folks ... anyone who says NAT!=firewall is prob talking about equipment at work :-)
16:07.05Mocsudhir492, what is your problem exactly ? ;)
16:07.05ZeeekI don't know what I did, but since 80% of the American people believe in the existence of angels...
16:07.06*** join/#asterisk nctk (~nctk@lsne-catv-dhcp-29-238.urbanet.ch)
16:07.08sudhir492Moc: Thx. I had completely forgotten about that part .
16:07.34ManxPowerZeeek: Pretty sad huh?
16:07.55ZeeekManxPower, there was one router that wouldn't do astrisk no matter what, but even in bridge mode with no firewall, it still didn't work
16:08.02sudhir492Moc: Actually in the wiki, about Polycom 500, I myself had made a note to decrease the register timeout.
16:08.09Zeeekmaybe I have an angel in my asterisk box?
16:08.18coppiceZeeek: and 100% of CEOs believe in angles :-)
16:08.27ZeeekBusiness Angels
16:08.31roamer323Zeeek - never run fdisk on that * box!!
16:08.33MocZeeek, hope he doesnt get her wing stuck in the fans
16:08.42[cc]smartye, they are called powerpoints in business
16:08.44ManxPowerZeeek: You were prolly someone like Ghandi in a previous live and you are just getting Karma Carryover.
16:08.50sudhir492Moc: Never mind. And thanks for waking me up :-)
16:08.55Zeeekthere was a burnt feather smell down there yesterday
16:08.59Mocsudhir492, ha Polycom, my favorite phones ..
16:09.14vaewynpolycom rocks!
16:09.18Mocoh yea
16:09.28sudhir492Moc: Mine too. Though I hate the company for not supporting asterisk!
16:09.30vaewynI only like my hitachi cable wireless one better :P
16:09.33ZeeekThis may be a foolish question but can't you just use STUN on most home NAT routers? (I don't!)
16:09.42Zeeeker phones I mean
16:09.44[cc]smartkarma is under extreme inflation, forget about investing in that
16:09.52Zeeekthe GS has a STUN field
16:09.53roamer323Zeeek - that's one karma point per NAT tranversal with no glitches... better top up those karma point soon :-)
16:09.54sudhir492Moc: At $170 cannot find a better value !
16:10.22Mocsudhir492, same as cisco, but atless polycom doesnt freekout on firmware avability,  and thought, polycom have alot of feature that * dont support
16:10.27Zeeekso if I call myself with FWD I'd need at least 3 points?
16:10.34PoWeRKiLLwhat that     -- Got SIP response 415 "" back from 81.218.111.79
16:10.35Mocpolycom will never support * until we do support them
16:10.52sudhir492Moc: Yes. which version are you using?
16:10.56Mocsudhir492, at 170 yes it nice, I get it at 185$
16:10.57Zeeekme ->NAT->NAT-->ast-->NAT-->FWD-->NAT-->asterisk-->NAT-->me...
16:11.00Moc1.4.1 I think
16:11.24Zeeekthat can't be right because it still works
16:11.29ManxPowerMy eyes!  My eyes!
16:11.41Zeeekit is Sunday
16:11.56ManxPowerZeeek: You realize you're going to hell for twisting a network into that config.
16:11.58ZeeekI got yer hairpin right here
16:12.05vaewynmy polycoms and * seem to get along great... only wish they would release the devel info for playing with the menus on the phones
16:12.28sudhir492Moc:  tritechoa.com gives me at $175, no shipping if I order 5 at a time. (talk to steven or scott) You can tell him my name
16:12.30roamer323Zeeek - with your configuration... it looks like you objective is to connect your NATs through your *'s :-)
16:12.31ZeeekI think you guys have talked me into an ip500 when I go to the US
16:12.53vaewynZeeek: they really do rock
16:12.59Zeeek(why not buy one here you ask? - 20% VAT)
16:13.14vaewynhehehe
16:13.20sudhir492Moc: another friend of mine gets it from a wholeseller in MD for $165 a piece and gives that to me at $170.
16:13.30Zeeekwho has the lowest single unit pricing on Polycom?
16:13.48vaewynZeeek: voipsupply.com had it when I looked about 3 weeks ago
16:13.50Zeeekand how much is that in 2005 dollars?
16:13.56ZeeekI'll look now*
16:14.11Mocsudhir492, how much for the IP 600 ?
16:14.17ManxPowerThe problem with Polycom is that people that sell polycom supported PBXs get really good discounts on polycom phones.  The rest of them have to pay a much higher price.
16:14.25sudhir492Moc: I never inquired.
16:14.37sudhir492Zeeek: where do you live? How many do you need?
16:14.38Zeeekso we need to find someone that works at a polycom pbx place but is hip
16:14.49Zeeekone
16:15.00sudhir492Zeeek: where do you live?
16:15.00Zeeekare you a reseller?
16:15.04ZeeekFrance
16:15.23Zeeekbut I'll be in the USSR^H^HA in May
16:15.27roamer323they VAT on the declared value?  declare it at $30
16:15.43ManxPowerLOL!
16:15.57Zeeeklet me explain - FedEx bless them, is obliged to open everything now, anti-terror ya know - and they seem to be in bed with customs
16:16.01sudhir492no I am not a reseller, but would not mind helping a fellow *user
16:16.15Zeeeksud where are you?
16:16.31sudhir492I do provide solution to businesses though. I am a reseller in that sense.
16:16.43sudhir492Near D.C in northern VA
16:17.13Zeeekah, well I'll have to see what the cost situation is on the various offers
16:17.49Mochttp://www.voipsupply.com/product_info.php?cPath=94_161&products_id=455
16:17.51Mocthat cool
16:17.54Mocbut expensive
16:18.21Mocwish it were 1000$ ;)
16:18.21Zeeeksughir you have a bunch new as in 'in the unopened box"?
16:18.26*** join/#asterisk jeofrey (~jeofrey@202.160.45.29)
16:19.01jeofreyRROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)
16:19.13jeofreyanyon can help me please with this error in mysql
16:19.14ManxPowerPolycoms support PoE and have 2 ethernet ports.  There are phones that are much cheaper, but still pretty good for much less.  However, they either don't support PoE or only have 1 switch port.
16:19.20vaewynMoc: wow... that is cool
16:19.42jeofreyim using fedora core core.....
16:20.05Zeeekgod there's 20 times more SIP phones <$200 since last time I looked!
16:20.24jeofreybut i cannot run the mysql .... i want to use it for recording our calls
16:21.14jeofreyeverytime i run mysql alway come out this error RROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)
16:21.20Zeeekanyone tell me what ground shipping is on the Polycom? About $10?
16:21.25sudhir492Zeeek: Yes I do. Usually keep around 10 in stock. When I get an inquiry from a business, I usually like to complete the sale the same day. Hence have to keep some Polycom and PAP2-NA (for cheaper clients) in stock
16:21.28Mocjeofrey, you need to start mysql server first
16:21.52Mochow much is the PAP2-NA ?
16:21.52jeofreyMoc i do it for so many times already
16:21.56sudhir492Zeeek: In priority mail, probably a few bucks less
16:22.25Darwin35grrr kram where are you
16:22.40Zeeeksudhir492 msg me your email and I'll get back to you before I leave (late April)
16:22.40sudhir492Darwin35: what do you need from kram?
16:22.52Darwin35work on the freebsd g729
16:22.53vaewynDarwin35: probably getting ready for San Jose
16:23.08Zeeekor a URL is you have one
16:23.29jeofreyMoc after i restart the mysql server this error comeout ERROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)
16:23.43Darwin35the freebsd g729 is having issues it is not always loading
16:23.50Mocjoe, check the logs for errors
16:24.04sudhir492sudhir@cequip.com
16:24.23jeofreyMoc where can i get the log..... sorry im new to this.....
16:24.26vaewynAnyone else gonna be in San Jose with us this week?
16:24.50Mocjeofrey, dont recall for mysql, but check in /var/log or /var/lib/mysql
16:25.01ZeeekManxPower are all the firmware updates and docs (if exist(docs)) available somewhere for download? Or do yiou have to cisco them for money?
16:25.12Darwin35i WISH i WAS BUT WORK WOULD NOT PAY FOR IT
16:25.15Darwin35sorry
16:25.16vaewynhehehe
16:25.30vaewynThey are barely paying for me to hit the exhibits only
16:25.45Darwin35ahh  that bites
16:25.47vaewynbut it is worth it... got some contacts to hang out with that will be there
16:25.49jeofreyok moc
16:26.13Zeeeksudhir492 ? Not hearing from you
16:26.15Darwin35well I wanted to be there because snom stole my mini pbx box
16:26.26Darwin35and I wanted to confront them in public
16:26.37vaewynHope jerJer is gonna be there cause I want the telco guy in my group to meet him
16:26.38Darwin35thier box looks just like mine
16:26.46vaewynDarwin35: hehehe
16:26.59Zeeekvaewyn, what group, you play in a band?
16:27.05ZeeekLike Oasis?
16:27.16vaewynZeeek: hahaha... nah... our department group :P
16:27.32vaewynnothing quite that fun :P
16:27.40Zeeekwho can answer my Polycom quest: do you have to pay for firmware?
16:27.45DaminNo, I'm the one in Oasis. He played in Journey.
16:27.53jeofreyMoc here is the log http://www.pastebin.com/250075
16:27.54vaewynZeeek: nope... is free
16:28.03Zeeekand I sang the SttarTrek Enterprise theme song
16:28.16Zeeek"It's been a long unload..."
16:28.29vaewynWill be fun... can do some long distance geocaching while I am there also :P
16:28.31Zeeek"but I think I can dial now..."
16:28.33Darwin35StarTrek Enterprise with Scott Beckula is dead
16:28.38Darwin35put it to rest
16:28.48ZeeekI thouhgt he landed in Nazi America?
16:28.49Darwin35like all his other shows he never finished
16:28.52vaewynOr at least make him leap again  :P
16:29.07ZeeekThe last episode was just on here
16:29.19Zeeekdue to alien help, Germany won the war
16:29.45ZeeekI could have consoled T'pen or whatever her nampe is with the tight purple velvet suit
16:30.00Darwin35Tapal
16:30.02Zeeekshe comes on here sometimes with the name Katty
16:30.08Zeeekbe sure to be nice to her
16:30.20ZeeekT'Pal not Tapal
16:30.31ZeeekTapal is a dishwashing detergent on Mars
16:30.39Darwin35heheh
16:30.58ZeeekRegistered to '69.73.19.178', who sees us as T'Pal
16:31.09Zeeekprobe me
16:31.37Zeeeknot big enough
16:31.42Zeeekor long enough
16:32.07vaewynqualify=yes
16:32.08vaewyn:P
16:32.21ZeeekI think kram should sing that song at the next AstCon: "It's been a looooong road...."
16:33.38Zeeekshit T'Pal is now UNREACHABLE
16:34.15MocZeeek, remove qualify=yes, it a bug ;)
16:34.21nctkhi, everybody, after seeking for a while, I still can't find the answer of may question
16:34.32ZeeekI love qualify - I live for it
16:34.47Zeeeknctk ask and yee shall be told
16:35.08Mocjoe: install TaoLinux (RH Enterprise recompiled/free version)
16:35.14nctkis there a way of getting the tax information from the isdn level using zaphfc
16:35.57Darwin35so those of use not going will hoold a online lunch gathering and discuss asterisk and other thigns
16:36.00Zeeekwhat information do you want?
16:36.18Darwin35nsal
16:36.39Darwin35n/s/a/l
16:36.39Zeeekasterisk and other thighs... I like that
16:36.52Darwin35things
16:37.00Darwin35fingers typing to fast
16:37.00Zeeekless interesting
16:37.05nctkat least in switzerland, but in germany to, you get messages on the D channel saying you just paid 10 cents for example
16:37.05Zeeekthighs
16:37.05Darwin35true
16:37.43Zeeeknctk I don't know if that happens in USA
16:38.28nctkzeek you said you're in france, no? Isn't there something like that with "FT Numeris"?
16:39.14Blackvelhow do i handle 0911 and 49 in the dailplan?
16:39.38Blackvelif EXTEN starts with 09 do this, if extension does not start with 0 do this?
16:41.27ZeeekNumeris is ISDN but I don't use it anymore so I can't answer your question
16:41.44ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
16:47.09*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfl9t.dialup.mindspring.com)
16:48.18nctkI know Numeris is ISDN, I continue to dig, hopefully I will find something ;)
16:48.21sudhir492Zeeek: I went out for a while. what do you want to hear from me?
16:48.35*** join/#asterisk SuPrSluG (~SuPrSluG@pool-141-149-253-154.buff.east.verizon.net)
16:48.42ZeeekI was saying you could msg me an email or URL for contact later
16:48.48Darwin35I need to find more features to add to my dial plan I need dial100  and callrecord
16:48.55ZeeekI'll be in the uS for only 4 days
16:49.09Darwin35zeek where you from
16:49.25ZeeekMinneapolis,MN,USofA
16:49.31sudhir492Zeeek: my email is sudhir@cequip.com
16:49.38Zeeekok
16:50.03vaewynZeeek: heh... I'm flying through there tomorrow on way to San jose.... is a nice place
16:50.35*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
16:50.40vaewyngot 2 friends that live there also
16:50.41Zeeekno way!
16:50.44PatrickDKanyone know what size powersupply supira2000 needs?
16:50.55Zeeeksudhir492 I wrote you with my addr
16:51.34Zeeekby the way I see mention of 12v power supply but not input voltage. Would it be 100-250 by any chance?
16:52.28PatrickDKI just need output of the transformer, or the input of the 2000
16:52.32PatrickDKvoltage and amps
16:53.07ZeeekSorry I was talking to sudhir on a similar subject
16:53.17PatrickDKah :)
16:53.33Zeeekdoesn't sipura have a site with a datasheet?
16:53.39Blackvelanyone has some workaround for this broadvoice / asterisk problem yet and why broadvoice sends you to their voicemail?
16:53.44PatrickDKI'm looking around, but haven't found anything
16:53.53PatrickDKit doesn't even say on the device, normally they do
16:53.59Darwin35there is a fix in 1.0.5
16:54.04djinI read about a wifiaxy, does anyone know what that is?
16:54.15Blackvelno
16:54.19Blackveli tried with 1.0.6 today
16:54.21Blackvelnot working
16:54.25shepherddjin: it's a joke
16:54.39*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
16:54.44Darwin35I am building 1.0.6 now i will let you know
16:54.47Blackvelthere is a BV fix in 1.0.4
16:54.53djinok, don't get it though :)
16:54.58shepherddjin: basically it was a ton of batteries + wifi card + iaxy
16:55.04shepherdand it only lasted 30 mins
16:55.05Blackvelah you mean there is an additional fix for BV in 1.0.5?
16:55.10Blackvelthat would explain it
16:55.14Darwin35yes
16:55.25djina sheperd, but was it meant as a joke?
16:55.27Blackveland it only got commited to 1.0.5?
16:55.28Blackvelweird
16:55.32PatrickDKah, found it
16:55.35Blackvelshouldnt the lastest be fixed
16:55.36Blackvel?
16:55.50Darwin35should I am building now
16:55.59Darwin35let me check and test
16:56.00*** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net)
16:56.02Blackvelyou have to be very patient with me darwin :)
16:56.07shepherddjin: yeah.. .they embeded it into an old rotary phone, and were putting it in elevators with no cords :)
16:56.10Darwin35its ok
16:56.13Blackvelmy english is sometimes werid
16:56.15Blackvelweird
16:56.18ZeeekIAXy needs almost 1A power
16:56.19Blackveli dont understand too fast
16:56.22Blackvelsorry
16:56.27Darwin35my human is some times strange
16:56.31djincool :0
16:56.42shepherdzeek: 1500 mA
16:56.47Blackvelis it the cvs of 1.0.5 or the .gz files of 1.0.5?
16:56.56Zeeekactually I think it's 1200ma
16:56.59ZeeekI have one here
16:57.14Zeeekbut they may have recommended a 1500ma supply
16:57.14shepherdhaha.. you could be right about that
16:57.15Darwin35I know its in the tar.gz and should be in the cvs
16:57.20Darwin35brb restroom
16:57.46NukemizerI am trying to test my TE110P card with Redhat, since I could not get PRI to work correctly with Mandrake. I am trying to successfully test PRI in loopbackmde but I always get alarms. Is there a switch for letting card run in LoopBack mode ?
16:57.52djindid anyone look into this dCAP certification?
16:58.48djinor signup for it?
16:58.51BlackvelDarwin35: you are a killer :)
16:59.19*** join/#asterisk jtar (~john@cpc2-mapp3-4-1-cust214.nott.cable.ntl.com)
17:01.08*** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org)
17:04.03*** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz)
17:04.17NukemizerI may have found my problem... To get a Digium PRI to work do you need the hisax module for ISDN ?
17:04.41djinnope
17:04.49djinwhy?
17:05.13Nukemizerlooking for diff between RH and Drake.. to get PRI up..
17:05.41NukemizerI still am trying to find a way to test in Loop Back mde without card complaining
17:05.58djinDon't know much about Mandrake, but RH shouldn't have that much obstacles.
17:06.49*** join/#asterisk JerJer[mobile] (~jj@65.173.197.109)
17:07.14NukemizerRH still complains in Loop back mode. I amnot connected to PBX since I am testing at home today. Should be able to create a loopback test right ? where card does not complain or go into RED alarm ?
17:07.37djinehat card do you use?
17:07.48NukemizerTE110P
17:08.51NukemizerThanks to all that tolerte my questions, I am just looking for a predictable testing path for trouble shooting
17:09.21djinNot sure you can loopback a one port card. Only used it with 4 ports.
17:09.40*** join/#asterisk Tarox (~chris@pD9E7B7CA.dip.t-dialin.net)
17:09.50djinI assume you want to test an outgoing call being routed back in?
17:10.29Nukemizerperhaps the very nature of PRI and D channel communication will prevent my from just using a Loop Back plug
17:10.42memic<PROTECTED>
17:10.58memicdo i have to patch asterisk?
17:11.17memicim using asterisk from debian/testing
17:11.20Darwin35ok back
17:11.22Nukemizerno, not even to that stage..  calls will go through if card is not in the midle of a choke fest. I want to get card(s) to work just in an idle state without error
17:11.24Darwin35sorry
17:12.00Darwin35I heard there where some bad cards and a new rev of the card was comming out
17:12.16Darwin35any further info call digium
17:13.03NukemizerI got digium to send me out a second card but they both behave the same. even in different boxes. so I am not trying RH and Xorcom to see if it really is the OS
17:13.03scrubbany sip gurus around?
17:13.11Blackvelhttp://www.voip-info.org/wiki-Asterisk+settings+Broadvoice
17:13.13scrubbI think I found a bizzare bug!
17:13.19Blackvelis the last descrition useless?
17:13.27Blackvelit doesn't tell anything about the filename
17:13.33*** join/#asterisk Jayden (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
17:13.46tzafrirNukemizer, what's the problem?
17:13.54Darwin35<PROTECTED>
17:14.43scrubbtwo SIP registrations seem to always use the first information for an inbound call, inspite of the SIP invite.
17:14.53Nukemizertzafrir, I am looking for a way to loopback test my TE110P to help my locate why my card  goes into Red Alarm all the time
17:15.27scrubbno sip heros in here?
17:15.28NukemizerLoopback will work with e&m wink trunk mode but there is really no functionality like ISDN
17:15.36scrubbok, will post to BUGS and see what I get.
17:16.00Zeeekscrubb describe your network and phones
17:16.02tzafrirscrubb, better ask your question anyway. One of those heros may be away but reading logs
17:16.14scrubbok, I have TWO seperate broadvoice accounts.
17:16.18Zeeekstop
17:16.18scrubbI register each properly.
17:16.21NukemizerI was just hoping that someone might be able to help me find a way to test via loop back mode so I can fight this one step at a time.
17:16.33scrubbits not broadvoice's problem.
17:16.36scrubbits * problem.
17:16.50scrubbbroadvoice sends the right INVITE for each account.
17:17.00dfunnellHi all, trying to dial out with exten => _1.,1,Dial,CAPI/470:${EXTEN:1} but it tries to dial only second digit of EXTEN (i.e. digit after '1'), as it is dialled.  Means it always fails to dial out. Any ideas?
17:17.10Zeeekeveryone seems to be saying BV is broken as of a day or two
17:17.12scrubb* receives the inbound call as if they are the SAME (first registration) number.
17:17.17scrubbits NOT BV!
17:17.24JerJer[mobile]dfunnell: because _1. is invalid
17:17.24scrubbI packet sniffed.
17:17.37JerJer[mobile]_1X.  is valid
17:17.39scrubbBV is sending a seperate and proper invite for each call.
17:17.58scrubbif I do a SIPGetHeader on the To field it is correct.
17:18.08MikeJ[Jayden]hey JerJer... wassup.
17:18.12mikegrbJerJer[mobile]: and he has ${EXTEN:1}
17:18.20scrubb* just is interpretting the call to the second account as if it came to the first one.
17:18.46scrubbBv is working fine for me otherwise.
17:19.24scrubbok, I'll wrap up my findings and post it.
17:21.10RaYmAn-Bxscrubb: if you avoid using the /XXXX in the register => statement (i.e. to send it to a specific extension from there) and simple send it to a context, you should be able to seperate by extension and hence process the calls seperately.
17:21.59scrubbRaYmAn-Bx: hmmm, checking.
17:22.45dfunnellJerJer[mobile]:  Thanks, have changed to _1X. but now dials only first two digits (after initial '1' for outside line).  Any way I can get it to dial full number without specifying number of digits (number length will be variable).
17:22.53scrubbRaYmAn-Bx: my register looks like: register => #####:password@sip.broadvoice.com
17:23.02scrubbRaYmAn-Bx: there is no /XXXX
17:23.25dfunnellmikegrb:  Is this incorrect?  Intention is to strip first '1' from number to prevent it from dialing it externally.
17:24.03mikegrbdfunnell: that is correct
17:24.17mikegrbdfunnell: I misread statements above
17:24.57dfunnellmikegrb:  Cool, thanks.
17:25.48RaYmAn-Bxscrubb: then if both calls are being caught by the same block in sip conf, you should be able to "catch" the seperate extensions in the extension blok it directs to
17:27.45dfunnellJerJer[mobile]:  Still there?
17:27.56*** join/#asterisk Tarox (someone@pD9E7BB98.dip.t-dialin.net)
17:28.04scrubbRaYmAn-Bx: yeah, except that its not distinguishing between them.
17:29.35BlackvelDarwin35: i mean, what happens the instructions on http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice when it is not clear what filename it is? :) better to remove the code instructions, I am so confused.
17:29.45scrubbRaYmAn-Bx: asterisk show the channel name as the wrong one.
17:30.56BlackvelDarwin35: I mean, what do the instructions help on ...
17:34.39RaYmAn-Bxscrubb: hmm, weird. I've had it working vaguely ok with two registrations to same server before..I seem to remember having the same problem but I can't remember what I did to fix it..I think I just put specific exten => number,1 extensions in the extension tbh
17:34.40*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
17:43.31Blackvelhey bjohnson_
17:43.43Blackvelfeel free to ask me privately if I can be any guide for you :)
17:43.51Blackvelforget
17:47.29*** join/#asterisk fugitivo (~ajf@201.255.100.59)
17:48.44*** join/#asterisk dan2 (~beta3@dan2.active.supporter.pdpc)
17:50.21*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
17:50.50*** join/#asterisk feklee (feklee@genba.ffii.org)
17:51.31fekleeI always get this message when trying to authenticate to sipgate:  Mar  6 18:50:30 NOTICE[4831]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:9779619@sipgate.de>;tag=as6a89cf36'
17:51.41fekleeHow do I find what is going wrong?
17:51.52fekleeI already tried different configurations, but to no avail.
17:52.12fekleeIf sipgate is hard to set up, what alternatives are there Ggermany)?
17:52.59Blackvelnikotel
17:53.00Blackvelgmx
17:53.03Blackvelfreenet
17:53.10Blackvelbut sipgate is not hard to setup
17:53.19Blackvelwho told you that nonsense? :)
17:53.51Blackvelplease make sure you have set insecure=very in sip.conf [sipgate.de]
17:53.54fekleeBlackvel: Do you run a sipgate setup?
17:53.57Blackvelsure
17:54.07fekleeBlackvel: Already have insecure=... in sip.conf
17:54.15Blackvelremove auth=md5
17:54.24fekleeBlackvel: Could you make the setup available to me?
17:54.31*** join/#asterisk IceBerg (iceberg@cpe-24-166-0-83.indy.res.rr.com)
17:54.38ZeeekI don't have insecure in mine
17:54.49Blackvelpaste your config on pastebin.com
17:55.00Blackvelwithout the username + passwort :)
17:55.28*** join/#asterisk memic (skdmwnf@dsl-084-056-106-237.arcor-ip.net)
17:56.48*** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.res.rr.com)
17:56.52fekleeBlackvel: Well, I tried this config: http://www.sipgate.de/faq/index.php?aktion=artikel&rubrik=650&id=359&lang=de&highlight=asterisk
17:56.55fekleeand others
17:57.16Blackveltry to put on pastebin and I take a look
17:57.20BlackvelI'll
17:57.44dan2drumkilla: ping
17:57.47fekleeOK, I'll justclean up my config a bit before poting.
17:57.54D1ng0so has Broadvoice fixed incoming calls yet ???
17:58.09dan2Everybody who is having broadvoice troubles join #broadvoice please
17:58.18Blackvel1.0.5
17:58.20Blackveltry to get this
17:58.24Zeeekhis problem seems to be on the register
17:58.26Blackveldarwin has mentioned a fix
17:58.31memicanybody has a hfc card running with 2.6.9 ?
17:58.42Blackvelmemic: i use 2.4
17:58.45memichm
17:58.46Blackveljoin #asterisk-drinkers
17:59.05D1ng0dan2,  jeeez a whole two people in there me and you
17:59.07memicwhats that chan 4?
17:59.21scrubbRaYmAn-Bx: thanks.
17:59.25memici cant use 2.4
17:59.58IceBergHi guys, Im a noob, and Im trying to get basic asterisk up, Im running it on FreeBSD and Im following the setup on voip-info.org. I backed up all the installed configs that the BSD ports installs and put in the sample ones from the source packagea renaming them all to get rid of the .sample. edited the sip.conf so that my xlite phone is listed. But I get this error:  retrans_pkt: Maximum retries exceeded on call
18:00.30fekleeBlackvel: http://www.pastebin.com/250099
18:00.44*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
18:00.44*** mode/#asterisk [+o bkw_] by ChanServ
18:00.55psywarhow come * sometimes requires a # to terminate a single-digit extension and other times not?
18:01.04psywaris it because it's a prefix of another extension?
18:01.19bkw_i'm here in denver
18:01.22bkw_with steve and oej
18:01.30bkw_they are on the same flight with me to San Jose
18:01.39dan2bkw_: could you modify the topic for me
18:01.48psywarIceBerg: I get that every time I restart *.  Ignore it.
18:02.01Blackvelfeklee: where is your [sipgate] section? it is missing
18:02.04dan2bkw_: I'm leaving for San Jose tomorrow, but I'd like a note about #broadvoice in the topic
18:02.25*** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net)
18:02.48IceBergpsywar, ok, but also the demo call never goes through, xlite give me the reorder tone and then I get errors about not being ableto start the mp3player, and ues mpg123 is installed and working
18:03.34fekleeBlackvel: That's the config they propose here: http://www.sipgate.de/faq/index.php?aktion=artikel&rubrik=650&id=359&lang=de&highlight=asterisk
18:03.41psywarI got the mohmp3 errors a bit, make sure you have the right mpg123 version
18:03.51psywarI had to manually create the mohmp3 dir
18:03.56*** join/#asterisk FarrisG (~farris@c-67-162-181-62.client.comcast.net)
18:04.19psywarstart a console with -vv and see what errors you're getting when you try to make the demo call, whatever that is
18:04.29fekleeBlackvel: Couldn't you just post your config.  I'm not so much interested in learning * at the moment.  I need it running by tomorrow.
18:04.38IceBergpsywar, yea I had to create it, and I know the one error I get is due to no mp3's in there but shouldnt the demo call go through no matter what?
18:05.03psywaryes, I think things things are not the problem though
18:05.10FarrisGAnyone have the patience to help me get asterisk running? I'm at a new job, in charge of phones/servers... Asterisk had been running smoothly for over a month, and then somehow the machine got restarted, and now it won't start back
18:05.31psywar/etc/init.d/asterisk start
18:05.31FarrisGI can post error messages or configs, just wondering where would be the first place I ought to look
18:05.43dan2bkw_: ?
18:05.52FarrisGpsywar: I wish. It appears to have been setup differently
18:06.36psywarhave you started a console with -vv?
18:06.42psywarasterisk -gvvc
18:06.45psywarsee what it says
18:07.08bkw_flight delayed
18:07.20bkw_i had to go from gate B59 to G16
18:07.20bkw_er B16
18:07.30bkw_its like 2 miles
18:07.31IceBergwhere are the audio files by default for the demo line, are they also in the mp3moh dir?
18:07.53bkw_oej is trying to get on GPRS
18:08.34Blackvelfeklee: are you behind NAT?
18:08.40dan2bkw_: could you update the topic to make a note for #broadvoice
18:09.20fekleeBlackvel: No, and I even disabled the firewall.
18:09.37Blackvelyou run it on the firewall?
18:09.49shepherddenver is like the coolest airport though
18:09.50Blackvelor the router
18:09.52fekleeBlackvel: No I'm connected via dialup.
18:10.01Blackvelwhat do you mean?
18:10.10Blackveladsl ppp0?
18:10.22Blackvellinux? windows?
18:10.23fekleeNO, via 56k-modem.
18:10.25fekleeLINUX
18:10.47feklee(I know that this is too slow)
18:10.57FarrisG"Loading module chan_zap.so failed"
18:10.58fekleeI want to install it on a different machine later
18:11.09Blackvelwhat is not working? outgoing or incoming?
18:11.09IceBergpsywar, the version of mpg123 I have is also installed from the ports, there is no reason it should not work
18:11.20*** join/#asterisk syncoherent (~synco@c-24-98-180-64.atl.client2.attbi.com)
18:11.29tzafrirFarrisG, there should be a reason a line or so above
18:11.36fekleeBlackvel: If you could just post your config, that would be great.
18:11.54Blackvelnot really
18:12.02Blackveltoo different
18:12.07FarrisG"Unable to open D-channel 24 (No such device or address)"
18:12.28Zeeekdoes sipgate.de have test numbers?
18:12.34Blackvel10000
18:12.45tzafrirAs there are a number of existing security holes with mpg123: when I use it with a sane Asterisk configuration can I be certain that it will only play the moh, and not anything from remote users?
18:12.51BrianR___Hmm.. I wonder if anyone's working on a debian package for asterisk-1.0.6... Perhaps I should do it..
18:13.06ZeeekBlackvel for me?
18:13.11Blackvelyeah its 10000
18:13.12fekleeBlackvel: The problem is that I've no clue where the problem may be.  When I google for the error message, I get no hit.
18:13.13tzafrirBrianR___, I have a working prototype, or so
18:13.21dfunnellHi all, can anyone tell me how to get * to wait until entire number is dialled (before dialling out) without having to specify exact # of digits?  Have tried exten => _1X.,1,Dial,CAPI/470:${EXTEN:1}, but this tries to dial out with first two digits after '1'.
18:13.27BrianR___tzafrir: Based on the 1.0.5 package?
18:13.34Blackvelyou didnt paste the complete config on pastebin anyways
18:13.38Blackveland i have no time now
18:13.47dfunnellNumbers are of variable length, but I'm not sure how to handle them.
18:13.48tzafrirBrianR___, http://tzafrir.org.il/rapid/APT.html
18:13.52Blackvelpreparing for a UK conference the next minutes
18:13.59syncoherentdrunnell: you can use Disa() -- that's what i have to do for one of my phones
18:14.06tzafrirbased mostly on the pkg-vopip debs
18:14.13Blackveli would remove srvlookup=yes completely
18:14.14shepherddfunnel: you can use agi
18:14.20ZeeekBlackvel that wasn't me
18:14.24tzafrirAnyway, there are now 1.0.6 packages ready for upload
18:14.31tzafrirI don't have the URL , though
18:14.32Blackvel<Zeeek> does sipgate.de have test numbers?
18:14.39Blackvelyes the number is 10000
18:14.42Zeeekyes but the rest about pasting
18:14.44BrianR___tzafrir: Aah. Are you on that team?
18:14.48Zeeekyhx gotcha
18:14.50tzafriryes
18:14.51Blackvelno
18:14.53Blackvelsomeone else
18:14.54FarrisGpsywar: what exactly would cause that "Unable to load D-channel" error, or where should I look?
18:15.09BrianR___tzafrir: I'm a debian maintainer also - albeit a bit of a slacker these days.
18:15.16Nuggethooray for slackers.
18:15.24psywarsorry, dunno FarrisG, I'm a * noob
18:15.27psywarsounds like an ISDN thing
18:15.29shepherdslackers + asterisk #1
18:15.51FarrisGIt occurred right after it parsed zapata.conf
18:16.41BrianR___yay.. dev.asteriskdocs.org is working again. .
18:16.54fekleeBlackvel: Here's an update but it's still nto working: http://pastebin.com/250111
18:17.19Zeeeksipgate isn't letting me in
18:17.19BrianR___Hmm.. Maybe it's not..
18:18.39dfunnellsyncoherent: Disa() is for dialling in to *, isn't it?  Will this help me with dialling out?
18:18.52FarrisGBut there is no mention of D-channel 24 in zapata.conf
18:19.46syncoherentdfunnell. i use it to dial out as well.
18:19.47dfunnellshepherd:  I was trying to avoid using agi, as I'm not looking forward to buggering up scripts, no other (easy) way?
18:20.20shepherdi can't think of any
18:20.25shepherdbut there might be
18:20.25dfunnellsyncoherent:  Sounds interesting, will give it a try.  Any chance of sending me your exten string so I can see how you do it?
18:20.37dfunnellshepherd:  Ok, thanks.
18:20.45shepherdextensions.conf is it's own programming language :)
18:20.57fekleeZeeek: I've the same problem. Are you trying to connect to Sipgate in Germany?
18:21.03ZeeekBlackvel do you have a paid account at sipgate?
18:21.13Blackvelyes
18:21.16Zeeekyes I am registered but it doesn't accept a call to 10000
18:21.23fekleeZeeek: However, I'm quite certain that this is a configuration problem.  It worked with X-lite this morning.
18:21.24Zeeekbut I don't pay -
18:21.33Blackveldunno
18:21.38Blackvelmay be becoz of that
18:21.39Blackvelnot sure
18:21.48ZeeekI think I have an X-Lite setup - I'll try it now
18:21.54IceBergI still get the reorder tone when I try to dial the demo line even though the console says it's playing the greeting files
18:22.38ZeeekWOW sexy Greman voice say ing I'm cool!
18:22.45Zeeekso there *is* an issue
18:23.07*** join/#asterisk Corydon-w (~tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net)
18:23.15fekleeCould anyone have a look http://pastebin.com/250111 ? I still cannot connect to Sipgate and I tried different configs all afternoon (although I absolutely don't have the time for that).
18:23.18*** join/#asterisk rephorm (~rephorm@cpe-24-28-67-25.austin.res.rr.com)
18:24.39Blackveltry again end of week then :)
18:24.50Blackvelor pay someone to do it ;)
18:25.20Blackveltype=peer
18:25.23Blackveltype=friend
18:25.25Blackvelchange this
18:25.52BrianR___damn it. the 1.0.6 branch doesn't have the disconnect dtmf tone configurable in features.conf :(
18:25.58RaYmAn-Bxfeklee: try adding the authuser in the register => line..i.e. number:password:authuser@host/exten
18:26.03Blackvelexten => _X.,2,Dial(SIP/${EXTEN}@sipgate.de
18:26.06Blackvelchange this to @sipgate
18:27.38Zeeekauthuser is same as user, no?
18:27.54Blackvelyou dont need authuser
18:29.37NuggetAsterisk CVS-v1-0-03/06/05-12:24:02 built by nugget@suburbia.slacker.com on a i386 running FreeBSD
18:29.40Nuggetyay
18:29.48fekleeBlackvel: Still doesn't work.  I'd be happy to pay someone say 10 EUR for helping me.
18:30.06fekleeRaYmAn-Bx: authuser doesn't improve things.
18:30.32shepherdnugget: what all did you have to do to get that running?
18:30.34RaYmAn-Bxwhat is the exact problem again? Failing to register, unable to make calls, unable to receive calls?
18:31.03Zeeekfeklee maybe sipgate changed something because mine used to work a long time ago
18:31.06NuggetI compiled it.
18:31.12Nuggetthen I ran it.
18:31.13fekleeRaYmAn-Bx: I get this error message: NOTICE[5660]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:9779619@sipgate.de>;tag=as1670ba42'
18:31.27fekleeRaYmAn-Bx: And I cannot make calls from outside (that's what I'm interested in)
18:31.27shepherdheh... ok
18:31.42*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
18:31.42shepherdi hear it will hardlock
18:31.49shepherdat times
18:31.50Zeeekfeklee you get that without even making a call?
18:31.57Nuggetit's flaky with zaptel, but just asterisk is just fine.
18:32.32fekleeZeeek: Yes I get the error message right after strating up *
18:32.34TheBearhi all, after a long while of not using my * server or my snom200 phones I ant to get back into VoIP. However I can remeber my snom phone password and help ?
18:33.01RaYmAn-Bxfeklee: ignoring the error, do you show as online on the webpage? Can you make/receive calls at all?
18:33.59fekleeRaYmAn-Bx: I cannot receive calls.
18:34.12fekleeRaYmAn-Bx: I'm not interested in making calls, but I'll try.
18:34.13feklee...
18:34.29RaYmAn-Bxfeklee: it helps in debugging
18:34.36ZeeekI reister ok and show as online but I can't call either!
18:35.02RaYmAn-BxI can't say I've tried using sipgate for anything but free calls so I dunno whether I can call properly tbh
18:35.16*** join/#asterisk jsolares (~jsolares@200.12.33.64)
18:35.25ZeeekI used it a long time ago but he's right, I can call with X-Lite and not thru asterisk
18:35.37ZeeekI am not authenticating properly at INVITE
18:38.26RaYmAn-Bxdoesn't sipgate have a test number of some kind?
18:38.55Zeeek10000
18:39.14ZeeekI call it and it won't auth me on INVITE but it works with X-Lite
18:39.48RaYmAn-Bxhmm, it works fine for me
18:40.18ZeeekCan you pastebin your peer entry (w/o passwd)
18:40.43Zeeeknote that on sipgate page on asterisk they have the same allow= twice - they didn't proofread the page very well ;)
18:41.19*** part/#asterisk rephorm (~rephorm@cpe-24-28-67-25.austin.res.rr.com)
18:42.24RaYmAn-BxZeeek: http://skumler.net/sipgate.txt (Admittedly this is for sipgate.co.uk, but it seems to work 100% identically..I have a .de account as well)
18:42.47Zeeekthx - i'll take a look
18:44.43Zeeekthat worked!
18:44.49Zeeekbut mail voice this time :(
18:45.01ZeeekI removed tyhe register statement and restarted
18:45.35Zeeekthey must require one of the new lines (I didn't use before): fromdomain or fromuser
18:45.46Zeeekfeklee you get that without even making a call?
18:45.57Zeeekfeklee oops - you paying attention?
18:46.09RaYmAn-BxI seem to remember I had to add the third field to the register statement as well... (i.e. authuser, being the same as username)
18:46.40ZeeekI don't have that. There is some confusion tho since I never call out, never get calls from mo sipgate number and never register :)
18:47.14fekleeZeeek: I'm still not successful.  But I guess I need to learn some Asterisk basics first.
18:47.16ZeeekI'll put the register back in now and see
18:47.27RaYmAn-Bxheh. I only receive calls from PSTN on my sipgate number. And that works fine
18:47.34Zeeekfunny with the register I get the woman!
18:47.51ZeeekWell Raym somehow one of the two lines did it
18:47.54RaYmAn-BxI get a male voice as well
18:47.59Zeeekfeklee Look at his file
18:48.14fekleeZeeek: What's your config now? (for sip.conf and extensions.conf)
18:48.15RaYmAn-Bxit makes sense, they are often required
18:48.17ZeeekI got th ewoman when I registered (or maybe they alternate - politicall correct?)
18:48.38ZeeekLook at the file Mr. RaYmAn posted
18:49.15FarrisGOk, when asterisk tries to start and parses zapata.conf, it errors out with "Unable to load D-channel 24", but there is no 24 in zapata.conf. How do I determine why this is happening and fix the problem?
18:49.48ZeeekAlthough I don't really care about sipgate.de, thanks for helping - I HATE when stuff doesn't work! Even free stuff
18:50.00ZeeekRaYmAn-Bx^^^^^^^^^^^^
18:50.43RaYmAn-Bxheh, same here..It just bugs me
18:50.53Zeeek<PROTECTED>
18:51.05Zeeekthey actually have a decent interface
18:51.27*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
18:51.29RaYmAn-BxI love the fact that sipgate uk gives free geographical DID's though :) It rocks.
18:51.40Zeeekmy number is a 207 - that must be expensive to call, ya?
18:52.00Zeeeklike 8p/min
18:52.07RaYmAn-Bxa uk 207? as in 0207 or?
18:52.21Zeeek<PROTECTED>
18:52.30RaYmAn-BxI can't quite remember how it works, but most 020 numbers are just london numbers...
18:52.34RaYmAn-Bxand hence at normal cost
18:52.39*** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net)
18:52.41Zeeek3p ?
18:52.54RaYmAn-Bxbut I remember something about there being special numbers in that range...not sure though
18:53.02RaYmAn-Bxsame rate as calling any other UK number
18:53.17RaYmAn-Bx(except for 07,08,09 numbers)
18:53.23ZeeekLast time I looked itr wasn't but I think we killed this horse once already :)
18:53.49RaYmAn-Bxactually, it's only 01XX and 02XXX numbers that are definitely cheap (and of course 0800)
18:53.53RaYmAn-Bxokay
18:53.53fekleeZeeek: It seems to work now.
18:53.59RaYmAn-Bxand yeah, we did
18:54.04ZeeekGreat! I'll sleep better tionight
18:54.22Zeeekwell, gotta run maybe I'll check the conf in a while
18:54.46Zeeekno one there yet
18:54.51Zeeeklater folks
18:55.02*** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
18:55.11fekleeZeeek, RaYmAn-Bx: The reason for the problem was almost too stupid to be true: I used the wrong password (Sipgate password instead of SIP-Passwort)
18:55.19fekleeTypical beginners problem.
18:55.22FarrisGD-channel should be 11, but it keeps truing to open the default 24. How do I make certain it uses 11 instead of 24?
18:55.38shido6set it in zapata.conf FarrisG
18:55.51*** part/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de)
18:55.58*** join/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com)
18:56.27FarrisGshido6: Where is zapata.conf?
18:56.37FarrisGshido6: Sorry, I meant where IN zapata.conf\
18:57.34shido6"/etc/asterisk/zapata.conf"
19:00.03*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr)
19:00.21PoWeRKiLLGot to unlock the PAP2 ! Now working on the GP2 :)
19:00.25mitchelocFarrisG: i missed the conversation, you can probably just add it "setting=value" anywhere in zapata.conf
19:01.59FarrisGmitcheloc: I tried that, but (sorry, total n00b here) "dchannel" doesn't seem to be a valid setting. It got ignored when asterisk parsed zapata.conf
19:02.33FarrisGThe odd thing is that this was working fine yesterday, and no conf has been changed. So maybe some module or init didn't get loaded/run when the server mysteriously rebooted last night?
19:03.42*** join/#asterisk Tili (~Tili@202-133-67-103-dialup.sat.net.pk)
19:04.23*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
19:05.50*** join/#asterisk Tr0j4N (b3nz3r@pcp08761169pcs.mtlrel01.nj.comcast.net)
19:07.08Tr0j4NHello all, n00b asterisk user here. Just came by to see if anyone's talking about anything interesting.
19:07.40Tr0j4Nbtw, if any of the asterisk devs are here, just wanted to give a BIG THANKS for your work!
19:09.31jontowstick around, you'll find interesting conversation.. this is one of the most active channels im in, overall :)
19:09.40*** join/#asterisk ScythelX (Fleb@pc-24-181-176-72.sbi.ct.charter.com)
19:10.05jontowif you wish to follow any of it make sure you have a nice long scrollback buffer in your irc client ;)
19:10.59*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
19:11.39FarrisGOk, I think I may know what the problem is now, but still don't know how to fix it
19:11.43mitchelocdoes anyone in here know .net?
19:11.44FarrisGzaptel isn't being initialized
19:11.49*** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com)
19:12.05mitchelocare you modprobing your cards?
19:12.07jontowfarrisg; had that one last night, amongst other things.. did you install zaptel before compiling asterisk?
19:12.10harryvvI am sure alot of people are having a good time at von. My asterisk hd failed last night so now trying to install it on a new system with fd3 instead of debian. Having a issue finding the nessesary libraries that were not installed on this system. fun fun.
19:12.46jontowharryvv; if you have a bit of time and patience, i'd give gentoo a go
19:12.52jontow.. it really works well for me
19:13.06harryvvjontow, to late fdc3 is installed.
19:13.26jontow;) np, just giving a third party opinion
19:13.33harryvvThis is what some of the tutorial authors used So im sticking with what a majority use.
19:13.36jontowgive it a try sometime when you've got a moment
19:13.54harryvvbtw, for the most part really like fdc3 is was a smooth install.
19:14.13jontowgentoo is much more of a manual process.. but its well documented
19:14.18harryvvjust this system is a workstation/server and deserves raid :)
19:15.35PTG123suse :)
19:15.44PTG123i tried gentoo, after 8 hours of install process
19:15.45PTG123it failed
19:15.52PTG123suse works great
19:15.55PTG123and is easy to use
19:15.57jontowwow
19:16.01jontowi've never had an 8 hour install
19:16.06PTG123oh man
19:16.11PTG123it ftp'd and compiled everything
19:16.12jontowtry the universal cd image :)
19:16.13PTG123was insane
19:16.21jontowyou don't need to start from scratch unless you have particular needs
19:16.27PTG123hah
19:16.31PTG123well i'll stick with suse
19:16.32RaYmAn-Bxjust do a stage3 install and you're done quite quickly
19:16.33PTG123and a mouse click :)
19:16.39jontowit is INSANE if you start from scratch
19:16.46jontowi've yet to complete one of those.. :P
19:16.51PTG123suse detected all my hardware great, etc
19:16.52RaYmAn-BxI did a stage1 install on my server
19:16.52PTG123heh
19:17.05PTG123maybe no one ever did a stage1
19:17.08PTG123thats why it faile d;)
19:17.12jontowif you're building all binaries with optimization on purpose to squeeze the last amount of performance
19:17.20jontowthen stage1 is a good idea
19:17.40jontowbut honestly.. the machines i work with aren't the highest-end servers
19:17.42PTG123i just wanted my computer to work
19:17.50jontowand compiling shit for 3 days isn't always my favorite thing
19:17.56PTG123i am too use to freebsd, why can't it be that easy
19:18.10Nuggetfreebsd is tasty.
19:18.23jontowfreebsd is what i use everyday on so many machines
19:18.34IceBergI cant figure out why asterisk's console says it's playing the files for the demo extension, but x-lite gets a reorder signal
19:18.53jontowwhich is why im a fan of gentoo :)  if im forced to be using linux; then a system as comprehendable as gentoo works for me :)  it probably will not for everyone
19:19.06harryvviceburg do a asterisk -vvvvvvvvvvgc then if its sip do a sip debug
19:19.18shido6IceBerg
19:19.30PTG123so then why even use linux :)
19:19.39shido6show me what you have for that softphone in sip.conf at pastebin.ca, IceBerg
19:19.45harryvvits nice to see asterisk show every sigle step on the cli and debug it that way.
19:20.01IceBergshido6, ok, one sec
19:20.27*** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com)
19:21.49IceBergshido6, http://pastebin.ca/6935
19:22.20psywarhas anyone else had problems with using "toast" to convert .wav or .au to .gsm?
19:22.26psywarit makes loud static for me.
19:22.46shido6where is your context IceBerg ?
19:26.27*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
19:26.33shmaltztzanger you around?
19:26.39shido6what context do you want to use for this phone, IceBerg
19:26.39shido6?
19:27.05shmaltzanybody here using slackware?
19:27.36jontowbecause lets face it.. * doesn't run as well on freebsd yet ;)
19:28.28jontowall of my servers / desktops / laptop run freebsd or netbsd or openbsd .. not linux (excepting a webserver implemented before I got the job and my * servers)
19:29.50*** part/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com)
19:34.57jontowi do have * on free and netbsd successfully though
19:38.43*** part/#asterisk NatRH (~Nat@dargo.trilug.org)
19:39.29harryvvdoes it make a difference what version of gcc is installed when compiling asterisk
19:40.08*** join/#asterisk Tili (~Tili@202-133-65-168-dialup.sat.net.pk)
19:42.02TrepaliumIt probably matters for the zaptel kernel modules, since they have to match the kernel's compiler version.  Other than that, no idea.
19:42.21harryvvmmm okay well im getting some compile issues with make
19:43.02*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
19:44.20psywarwhat other values does Monitor accept as the 1st arg
19:44.23psywarinstead of wav
19:46.47shido6show application Monitor
19:47.04shido6harryvv what do you get?
19:47.08shido6pastebin.ca them
19:47.12vaewyninteresting... anyone else run into a SIP device that works just fine... but you can't qualify against it reliably?
19:48.21*** join/#asterisk NatRH (~Nat@dargo.trilug.org)
19:48.47*** join/#asterisk viLeR (1000@ip-33-104.telesat.com.co)
19:51.30MocI never got good qualify using SIP
19:52.02*** join/#asterisk letherglov (~robbie@8036aa59.resnet.ucsd.edu)
19:52.49vaewynThis is just freaky.. i am on the phone and seeing Peer '6103IP5000' is now UNREACHABLE!  Last qualify: 1529
19:53.06vaewynSo I'm like "the @#$@#$ it isn't!"  :}
19:53.11Nuggetheh
19:53.19letherglovis the phone locking up?
19:53.22vaewynnope
19:53.28vaewynruning just great
19:53.32letherglovdoes it become reachable when you pick up the handset?
19:53.35letherglovor do you get a fast busy?
19:53.48letherglovmabe it's just sleepy ;-)
19:53.48vaewynis a wireless... so no handset :}
19:53.58letherglovlow-power mode or something
19:54.11letherglovbut that doesn't help you so much, because it'll never take incoming cals if it's unreachable
19:54.12vaewynIt does it even when i am talking on it...
19:54.37vaewynthat's the funny part... if i turn qualify off it always takes the calls just fine... with it on * freaks out
19:55.06Mocjust dont use qualify using SIP
19:55.15Mocqualify using IAX seem ok
19:55.24vaewynwas seeing if it would help a registration drop problem I am having at one site
19:55.56Mocvaewyn, exactly, remote qualify, it what I did, no more stupid problem
19:56.03Mocremote = remove
19:56.03*** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com)
19:56.16Mocuntil someone fix qualify in SIP, it pretty useless
19:56.58Mocjust reduce the registration time
19:57.02Mocyour behind a na
19:57.03Moct
19:57.10vaewynnope... no NAT
19:57.26vaewynWAP11 -> Linux router for subnet -> * server
19:57.58vaewynI am actually thinking it may be the  WAP11 piece of junk
19:58.05Mocwell wap doesnt help
19:58.24Mocbut in normal opperation, what is your average qualify MS when you show sip show peers
19:58.32vaewynnot WAP protocol... linksys WAP11 Access point
19:58.39vaewyn34ms or less
19:58.46vaewynif it is there
19:59.07Mocand what is your real ping with the device ?
20:00.15vaewyn9 > 18ms
20:00.56Mocyour qualify should show about the samething as a ping
20:01.09vaewynwell... it isn't
20:01.12vaewyn:}
20:01.18Moclike IAX does... but SIP qualify have a problem and sometime take a while before he see the answer
20:01.40MocI get 1MS qualify with a IAX connection, but with the same system using SIP, I get 30
20:01.46Mocand sometime 2690
20:01.50vaewynheh
20:02.06Mocit broken in sip, and until someone fix it, it will stay that way
20:02.19Mocbut maybe you should post a bug with info on the tracker
20:03.14Mocthis problem exist for over a year now hehe
20:04.14*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
20:11.00*** join/#asterisk revjim (~revjim@24.32.18.59)
20:11.25shmaltzanybody here using slackware?
20:11.44NuggetI do.
20:12.09NuggetI sort of felt obligated to.  It's not too bad if you install minimally, but it has some really stupid ideas about where things belong if you do a full install.
20:12.48Nuggetyou can also expect to get asterisk compile errors if you install slackware without x11.
20:14.35shmaltzWhy is that?
20:14.39*** join/#asterisk zotz (~zotz@24.231.32.191)
20:15.47*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr)
20:16.15Nuggetbecause the slackware installer is stupid and the asterisk makefile is optimistically short-sighted.
20:16.30PoWeRKiLLlast stable cvs are broken since I upgrade last week I got one or two time a day Too many open files ! any idea ?
20:16.37TrepaliumWell, at least D.J. Bernstein didn't write slackware.  Everything might've ended up in /var or something.
20:16.37tuxinator_linuxoptimistically short-sighted, that's funny
20:16.42Nuggetif you install slackware without x11 it will still install the gtk libs.
20:17.02Nuggetand the asterisk makefile uses the presence or absence of the gtk makefiles to decide whether or not to build the gtk gui console stuff.
20:17.22Nuggetso it tries valiently to build the gtk console which fails miserably because x11 isn't present
20:18.03NuggetTrepalium: funny you should say that.  one of my big beefs with slackware is that it installs all of apache to /var/lib/apache.
20:18.07shmaltzbut why only with slackware does it have this problem?
20:18.31Nuggetbecause no other linux (that I'm aware of) is stupid enough to install gtk libs on a machine that doesn't have x11
20:18.38TrepaliumSlackware has no dependancy checking, which is how GTK could be installed without X11 libs.
20:19.14shmaltzso how do I make sure that slackware doens't install the gtk library?
20:19.15*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-140.dsl.scarlet.be)
20:19.29Nuggetremove it after you're done installing.
20:19.29Nuggetor install the x11 libs.
20:19.31shmaltzhow do I remove it?
20:19.38Nuggetwith the slackware package tools.
20:19.45shmaltzthanks
20:19.59shmaltzis there any advantage of using 2.6 with asterisk?
20:20.02TrepaliumNugget: That is a pretty strange place to put Apache.
20:20.06shmaltzor can I leave it with 2.4
20:20.10tuxinator_linuxshmaltz: I'm using it
20:20.20NuggetTrepalium: indeed.  I feel like I need a shower whenever I have to type "/var/log/apache/bin/apachectl stop"
20:20.22shmaltztuxinator_linux, what are you using?
20:20.25Nuggets/log/lib/ rather
20:20.31tuxinator_linuxI make a wiki page http://www.voip-info.org/wiki-Asterisk+CentOS-4.0+Zaptel
20:20.45Nuggetfucking linux.  I hate linux.
20:20.56shmaltztuxinator_linux, we are taling about slackware
20:21.17shmaltzNugget, is there any reason to use 2.6 instead of 2.4 for asterisk?
20:21.31Nuggetshmaltz: I have no opinion.  I'm waiting until 2.6 is stable.
20:21.37mitchelocisn't 2.6, newer, more secure, more stable?
20:21.40mitchelocor i guess not
20:21.44Nugget2.6 is not stable release yet.
20:21.51SedoroxI find it stable...
20:22.02mitchelocit's stable so long as you don't through mythtv at it heh
20:22.03TrepaliumMaybe never will be with this new 'development model' for 2.6
20:22.05tuxinator_linuxIt has been stable for me so far, but I don't do much outside the norm
20:22.08Nugget2.6.8 has some crippling networking bugs.  I've never tried 2.6.9 or newer.
20:22.15Nuggetbut it's not stable yet.  so says linus, at least.
20:22.42tuxinator_linuxand I am not using X
20:22.55Nuggettoo much flux and change in 2.6 development to use on production boxes, imho.
20:23.16SedoroxI'm running 2.6.10-cko3 right now.. with Reiser4... with X, ATI Binary.. etc.. on my personal laptop.. and everything is great
20:23.30Nuggetthe plural of anecdote is not "data".
20:23.46Nugget2.6 is not stable release yet.  any success you are having with 2.6 is both fortunate and accidental.
20:24.04`Sauron2.6 isn'
20:24.08`Sauronisn't stable?
20:24.08tuxinator_linux2.6.9 here
20:24.13`SauronHummrh.
20:24.15vaewyn2.6 is stable
20:24.17Nuggetno, 2.6 is not stable release.
20:24.21vaewynand supported by *
20:24.24Nuggetthat's straight from linus.
20:24.25Trepalium2.4.x wasn't stable until well past 2.4.14, so I guess I can't complain... much.
20:24.27*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
20:24.32Nugget2.6 is a development branch
20:24.37tuxinator_linuxdefine stable
20:24.40tuxinator_linux~stable
20:24.41jbotextra, extra, read all about it, stable is the status of a Debian release when no packages will be added or changed unless a security fix is needed, or sta-ble adj; uptime in excess of 365days, or where the horses live  The current stable version of Debian is woody (3.0).
20:25.00`SauronI don't know that linux ever won't be a development branch.
20:25.19*** join/#asterisk stepcut (~redlion@ip68-107-21-88.sd.sd.cox.net)
20:25.21Nugget`Sauron: it used to have a stable branch.  with 2.6 that all flew out the window.
20:25.22vaewynNugget: 2.6 is both... read back some of the kerneltrap archives
20:25.38`SauronNugget: I know. Sucks, if you ask me.
20:25.42NuggetI agree
20:25.56`SauronBring back the even/odd numbered version thing.
20:26.00harryvvtyping bison on the command line show result in a responce? I have it installed but getting command not found when typing bison.
20:26.00Nuggetyeah, exactly.
20:26.06Nuggetthat system was working well
20:26.16vaewynhate to say it... but they are correct... with something that huge getting every part "stable" is a joke
20:26.30tuxinator_linuxharryvv: check path
20:26.35TrepaliumIt was supposedly done to reduce the number of patches Red Hat, and similar distributors put in their own kernels, but I think it's encouraging it more than anything else now.
20:26.49tuxinator_linuxtry 'find / -name (name of prog)'
20:27.03*** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl)
20:27.10harryvvI install bison rpm on /var/lib/rpms
20:27.25Blackvelwhat are the latest features in asterisk 1.0.6?
20:27.27harryvvbut typing bison on the command line echos no command found.
20:28.37tuxinator_linuxharryvv: I'm not familiar with bison, sorry
20:29.36harryvvodd, doing  a find / -name bison echos no such file.
20:30.07TrepaliumHow about an 'rpm -qa bison'
20:31.35harryvvechos no reply
20:31.52TrepaliumThen it's not installed.  "rpm -Uvh rpmname.rpm" to install it.
20:32.33harryvvdid, command not found after executing it.
20:32.47`SauronSigh.
20:32.48harryvvim going to find another copy
20:32.53`SauronHate the new nvidia driver
20:33.01`SauronIt doesn't clear the splash screen.
20:33.26dfunnellHi all - having trouble with Chameleon VM where it doesn't seem to be registering keys pressed by the user (i.e. mailbox and password).  Any ideas?
20:34.57harryvvmeridian voicemail df
20:35.40*** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au)
20:37.33*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-140.dsl.scarlet.be)
20:37.49dfunnellharryvv: Hi, was that for me?  If so can I get you to elaborate a little? D>
20:37.49harryvvTrepalium, found the issue. The bison i downloaded was one charicter off.
20:39.08harryvvnot really you need to read the wikis. ohh take a look at tx/rx and also if its a ata look at the dialplan if its a sipura ata.
20:40.05shido6dfunnell yes
20:40.09shido6dtmfmode dfunnell
20:40.15shido6u need to set the proper dtmfmode
20:44.51dfunnellOk, thanks, looking in to it now.  dtmfmode seems to work (with external IVR), just not for VM.
20:45.01harryvvshido it could also be what i mentioned since i had the same issue. if his tx is so low that the asterisk box cannot hear it then its moot.
20:45.22harryvvdf, what asterisk client are you using
20:46.01harryvvdf, has anyone complained of "you sound far away" when dialing though the asterisk to somone else?
20:46.35Tr0j4Nwhat are all the v's for in the command "
20:46.43Tr0j4Nasterisk -vvvvvvvvcg
20:46.46Nuggetverbose logging.
20:46.46znoGverbose
20:46.50Nuggeteach v makes it more chatty
20:46.52Tr0j4Nhow many do I need?
20:46.59Tr0j4Noh, ok
20:47.01znoGall the ones that you want
20:47.05Tr0j4Nthx muchly
20:47.53Tr0j4Ncan I just take any old house phone and plug it into an fxo port?
20:48.25dfunnellshido6:  you are a star, works, thanks.  Had set it already, but noticed I had a mistake in config.
20:49.03shido6:)
20:49.22psywarhow can I convert gsm<=>wav?  toast makes garbage files.
20:49.42psywarwait this is in the FAQ isn't it
20:49.45psywarnm
20:50.19Nuggetwow.  that's the first time that's ever happened!  :)
20:50.57harryvvdf, what was the config issue
20:51.00Tr0j4Nnevermind, I got this tdm4000p card but it's only got RJ45 ports on the back and my phones are all RJ11
20:51.25*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
20:51.44*** join/#asterisk iceberg (iceberg@cpe-24-166-0-83.indy.res.rr.com)
20:56.41BrianR___Tr0j4N: You can plug any POTS phone into a FXS port...
20:57.02BrianR___Tr0j4N: If you're careful, male RJ-11 connectors will fit into a female rj45
20:57.07Tr0j4Nhmm, yes I was just reading that
20:57.35Tr0j4Nthe card has a 2 FXO modules and FXS modules on it so in theory I should be ok.
20:58.07Tr0j4Nhowever, I tried plugging the RJ11 into the port numbered 1 and I ran zapcfg but it had an error
20:58.19BrianR___the way it's usually done is to connect to the PSTN with a bunch of FXO's, a T1/PRI, or VOIP gateway service. Then you use Analog Terminal Adaptors (ATA's) to provide FXS ports for any legacy POTS devices you might have.
20:58.37BrianR___Tr0j4N: One thing that you  might find very confusing at first is that FXO ports use FXS signalling and vice versa.
20:59.04Tr0j4Nthx, it looks like I have some reading to do.
20:59.08BrianR___Tr0j4N: So if ztcfg bitches about the signalling type being wrong for the card, that's probably why.
21:00.04BrianR___Tr0j4N: The TDM400p provides very good FXS jacks for pots gear, but is a bit expensive and you can only fit so many of them into a PC. Past about 4 ports you want to use either a channel bank or ATA's, depending on your wiring situation.
21:00.17Tr0j4Nshould I see a green light on the port if something is plugged in right?
21:00.32Tr0j4NI just bought one for testing
21:00.33BrianR___Tr0j4N: I don't know the answer to that particular question.
21:00.44harryvvtroj, what do you have
21:00.57Tr0j4Na TDM400P digium card
21:01.05icebergare there any good wireless IP phones that work with asterisk?
21:01.08harryvvcant help you there
21:01.21Tr0j4Nnp, I will get it. thx for help
21:01.23BrianR___iceberg: A few.. The cisco 802.11 / SIP phones work...
21:01.46icebergBrianR___ is it B or G?
21:01.56BrianR___iceberg: B, I think.
21:02.11icebergbummer, wont work on my wifi then
21:02.15Tr0j4Nwhen I run modprobe wcfxo I get ZT_CHANCONFIG failed on channel 1: No such device or address (6)
21:02.16BrianR___You only need to be able to transmit at 7KB/sec worst case, so..
21:02.26BrianR___iceberg: You don't have dual mode AP's?
21:03.14icebergI do but I require IPSec and Authentication and a minimum connect of 24kbps
21:03.18psywarI wonder if GSM compression would affect voice stress analysis like "liar liar"
21:03.40psywar(open-source VSA)
21:03.57BrianR___iceberg: Dunno if the cisco phones do ipsec either...
21:03.59icebergpsywar yes
21:04.04psywarhrm
21:04.09psywar:-(
21:04.39psywaripsec + voip is the bomb
21:04.52icebergBrianR___ I guess for wireless I'll have to use POTS to Lan converters for now
21:04.54*** join/#asterisk Damin_Mobile (~pocketirc@110.sub-166-155-107.myvzw.com)
21:05.05BrianR___wish sips / srtp was more widespread too..
21:06.17*** join/#asterisk alegh (~ag10@OL217-17.fibertel.com.ar)
21:06.23icebergare there any good yet inexspensive hard phones that support confencing? (key here being inexspensive)
21:06.45BrianR___Could always use something like the Uniden  UIP1868 too. It's essentially a 5.8ghz cordless station with a built in ATA.
21:07.37icebergchecking that out
21:08.22iceberg5.8 would be nice, less interferance with my wifi
21:08.22aleghHi, I'm looking for IP phones. Any recomendations about Sipura, Clipcomm or ArtDio?
21:09.23Goshenalegh: if you are looking for IAX phones I know iaxtalk.com has them but they ship from China
21:09.43Gosheniceberg: grandstream?
21:09.54*** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl)
21:10.04icebergGoshen what do you mean?
21:10.14Gosheniceberg: http://voipstore.pulver.com/product_info.php?products_id=33&osCsid=c428ece0900381122735332294d7001f
21:10.36Goshenblack grandstream phone...I have one, it works great, and the speakerphone is nice and LOUD(adjustable)
21:10.40*** join/#asterisk DonX (~don@adsl-69-155-217-211.dsl.rcsntx.swbell.net)
21:10.41Goshencomes in white too
21:11.11icebergGoshen , nice thanks
21:11.26BrianR___The grandstream gxp-2000 looks like it's going to be sweet. POE, 4  line appearances, and in the ~$120 price range..
21:11.30Goshenwelcome :)
21:11.38DonXI'm having strange problems with MOH on xlite. Everything seems to be working fine but music on hold is choppy on the soft clients. My ATA-186's and 7940's are working fine
21:11.53DonXanyone seen this issue before?
21:12.11icebergUIP1868 doesnt exist on Uniden's site
21:13.42*** join/#asterisk fitzel (~flint@p3EE39AF8.dip0.t-ipconnect.de)
21:13.49fitzelHello
21:13.51BrianR___The ZyXEL 2000W has 802.11 b and g
21:14.38fitzelI am playing around with iaxclient and speex. xlite with speex sounds nice, but iaxclient sounds really like I have my head in a tincan.
21:14.45icebergWOW that baby is exspensive
21:14.57fitzelAny settings that are recomendable?
21:15.53Gosheniceberg: I like my grandstream...where there is voicemail waiting the display flashes letting you know
21:16.18Damin_MobileSitting in the airport drunking beer, waiting for my flight.
21:16.25BrianR___Goshen: Have you been able to get your hands on a GXP2000 yet?
21:16.32BrianR___Everywhere I look they're backordered...
21:17.28vaewynDamin_Mobile: going to VON?
21:17.46Damin_MobileYep...
21:18.24vaewynI've got the 6 am flight in the morning
21:18.37GoshenBrianR: I haven't heard of the GXP2000 yet, have a link?
21:18.40*** join/#asterisk SirPrize (~roshan@pc016.dcs.kcl.ac.uk)
21:18.41vaewynnothing like 6+ hours in the sky to get you bored
21:19.34aleghAny experience with clipcomm CP-100D?
21:19.43Damin_Mobilevaewyn: Email me your cell # and we can hook up at San Jose.
21:19.50SirPrizeIs there a way that I can send incoming calls to Voicemail, but in such a way that if *I* dial-in on the same number, that I can enter my PIN to listen to my voicemail instead?
21:20.11Damin_Mobiledamin@nacs.net
21:20.28vaewynDamin_Mobile: cool... will do
21:20.36GoshenSirPrize: I have been woundering that too
21:21.29RaYmAn-BxSirPrize: if you always dial in from the same number, then yes, definitely..Otherwise it might be possible...It should be at least
21:21.30SirPrizeMy only thought so far was to set up a two-second silence through background, and a timeout of 2 seconds, and on the timeout dialplan, do the "Leave Voicemail" path
21:21.41vaewynSirPrize: You can do a "super secret extension" an use background to play the greeting message instead of letting VM do it
21:22.08SirPrizeRaYmAn-Bx, Ideally, I want to dial in from multiple numbers
21:22.28vaewynOr use the '0' for operator and dump into a context with the super-secret extension
21:22.38SirPrizevaewyn: so there's no 'built-in' method at the moment ?  could then put an Authenticate on that super-secret extension too
21:23.00RaYmAn-BxSirPrize: then you might be able to make it so you can press an extension while playing the "please leave a message" message
21:23.01vaewynOnly buuilt in is the '0' operator path
21:23.39SirPrizevaewyn: Could you please explain a bit more about the '0' operator path?  I didn't know there was such an extension, or that it had special capabilities
21:23.46Damin_MobileSigning off
21:24.25SirPrizeIs that the 's' extension ?
21:24.53BrianR___Goshen: http://www.voipsupply.com/product_info.php?cPath=95_111&products_id=331
21:26.17vaewynSirPrize: pressing '0' in most points in the VM will send you to the 'o' extension in the current dialplan context
21:26.48vaewyn'o' for operator
21:27.26SirPrizecool, I'll read up on that
21:27.41SirPrizeWhere could I find a ChangeLog for 1.0.6?
21:28.51Sedoroxont he FTP Server
21:29.35SirPrizestupid me - should have looked there first.  Was wandering around the web site looking for one
21:31.22*** join/#asterisk mtmachen (~matthewma@cable-68-113-71-35.grd.al.charter.com)
21:31.42Sedorox:-p
21:33.54*** join/#asterisk SirPrize (~roshan@pc016.dcs.kcl.ac.uk)
21:34.35SirPrizeMight this be a place to ask about an issue I'm having with a Sipura SPA 3000, or could anyone suggest a better place to ask in ?
21:35.07SirPrizethe issue isn't at all related to Asterisk, so I know it would probably be ot here
21:35.14shido6this is the place
21:35.26shido6you have a user configured for the spa?
21:35.26*** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com)
21:35.34shido6and a user and peer in sip.conf?
21:35.41SirPrizeWell, in my Line1 dialplan, I've got it forwarding all calls to Asterisk
21:35.50SirPrizeSorry, start again
21:35.53shido6hehe
21:36.04SirPrizeWell, in my Line1 dialplan, I've got it forwarding all calls out via PSTN on the PSTN line
21:36.26SirPrizeif I want to make a VoIP call, that has to be accessed by dialling #9 first
21:36.44SirPrizethis works fine - I can make standard PSTN calls, as well as VoIP calls via #9
21:37.06SirPrizethe problem is with the speed-dials - they don't work in this configuration
21:37.20SirPrizeif my dialplan simply says (xx.|*xx.), then the speed dials work
21:37.36SirPrizeBut I don't want the default out being the VoIP line
21:38.00SirPrizehitting a speed dial button will give a UK BT recording of "That number or option is unavailable"
21:38.16SirPrizeany ideas why the dial-plan is affecting the speed dials?
21:38.38SirPrizehave programmed the speed dials with numbers that CAN be dialled properly from the keypad when my own custom dialplan is installed
21:40.58GoshenI need a sipura spa 2100 dialplan howto...
21:42.01SirPrizethere's a general Sipura one here: http://voxilla.com/forum-viewtopic-t-619.html
21:42.29shmaltztzanger, you around?
21:42.34shmaltz~seen tzanger
21:42.36jbottzanger is currently on #asterisk.  Has said a total of 748 messages.  Is idling for 20h 21m 41s
21:42.43SirPrizeGoshen: there's a general Sipura one here: http://voxilla.com/forum-viewtopic-t-619.html
21:42.57dfunnellDoes anyone make use of DDI numbers with BRI ISDN and *?  Trying to find best way of running it, but best I can find is DNID and wiki page (http://www.voip-info.org/wiki-DNID) doesn't make it sound very reliable.  Any ideas?
21:42.58GoshenSirPrize: great thank you
21:43.38RoyK<PROTECTED>
21:43.46RoyK~seen zoa
21:43.47jbotzoa <~zoa@ip-212-239-162-26.dsl.scarlet.be> was last seen on IRC in channel #asterisk, 2d 8m 17s ago, saying: 'brian!!!!'.
21:48.12*** part/#asterisk SirPrize (~roshan@pc016.dcs.kcl.ac.uk)
21:48.34Damin_Mobileady for takeoff
21:49.13vaewynBon voyage!
21:49.51vaewynhehehe... I wonder if we can get his plane grounded by typign in IRC 'bomb voyage'  :P
21:50.15*** join/#asterisk DarkFlib (darkflib@dialup357.ts002.bmt.esat.net)
21:50.20vaewynoops... i said the 'b' word
21:50.39*** join/#asterisk NirS (Nir@l192-117-110-178.cable.actcom.net.il)
21:52.23*** join/#asterisk ^HeLL^ (~admin@85.137.127.182)
21:52.30^HeLL^hello all
21:53.45*** part/#asterisk DonX (~don@adsl-69-155-217-211.dsl.rcsntx.swbell.net)
21:55.11Tr0j4Nwoohoo got my tdm400p working! (had to run modprobe wctdm instead of modprobe wcfxo)
21:55.26Goshencongrats :)
21:56.00Tr0j4Nthx for all your help. in a few months I should be able to help some fellow n00bs myself.
21:59.35Moc;)
22:00.25*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
22:00.42mikegrbdoubtful
22:01.55TheBearIf I have my * server and standard phone connected to the same phone line. and have a SIP Phone connected on my network.  Can I answer an incoming call on a standard phone, and then (a) redirect  (b) pick up this call on the SIP Phone ?
22:02.38TheBearAnyone know ?
22:02.55shido6yes
22:03.09shido6fxs card in the * box?
22:03.15shido6or are you connecting an fxo card
22:03.17shido6to the same line
22:03.20shido6as the analog phone
22:03.22shido6with a splitter
22:03.22shido6?
22:03.57Qwellprobably a splitter
22:04.01QwellThats what it sounds like
22:04.54Qwell* will probably pick it up right away
22:05.08TheBearshido6: using a splitter one side goes into the std phone the other into a Digium card
22:05.09QwellIf you want a normal phone, you should get an FXS port too
22:05.15shido6oh lord
22:05.17DarkFlibdepends if you have an answer before you dial the sip phone
22:05.21shido6get a fxs card
22:05.30shido6or use your softphone to answer or sip phone to answer
22:05.48Qwellfxs port is probably the best option
22:05.54shido6or IAXy
22:05.56shido6or Sipura
22:06.03Qwellwhich is also an fxs port ;]
22:06.23shido6what kind of FXO do you have in the * ?
22:06.27TheBearSIP phone is a snom 200
22:06.33shido6modular or the x10xp ?
22:07.35*** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net)
22:08.06shido6what fxo device do you have for your asterisk system?
22:08.13TheBeardigium card has 1 FXS module on the 4 port TDM card
22:08.24Qwellfxs port?  You mean fxo
22:08.38Qwellat least...I hope you mean fxo
22:08.48Qwellotherwise, its not gonna work, no matter what you do.  heh
22:09.05TheBearthe mini daughter card says " Digium Ringing FXS module"
22:09.15Qwellyeah...
22:09.16TheBearI also have the FXO.
22:09.31QwellThen you may wish to restate your question.
22:09.35shido6ok both of em on the card?
22:09.35QwellWhy are you not using the FXS port?
22:09.37*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
22:09.46shido6so u have 2 modules on the card?
22:09.47shido6one fxo
22:09.49shido6one fxs?
22:09.56shido6or one fxs on the card and a x100p?
22:09.59TheBear1 year back I had the * server connected to a phone line and a std phone connected into the * server. I could then make calls from the std phone via the * server to the phone line
22:10.04Qwellwall > fxo | fxs > analog phone | ethernet > SPA
22:10.09TheBearNo two cards
22:10.10da-manFLhi shido6
22:10.29QwellTheBear: You have two TDMs, with one module each?
22:10.45*** join/#asterisk DyOS (~me@ip68-2-153-157.ph.ph.cox.net)
22:10.55*** join/#asterisk lesouvage (~chatzilla@cc341200-a.assen1.dr.home.nl)
22:11.04*** join/#asterisk xlyz (~xl@81-208-36-176.fastres.net)
22:11.22fitzelAnybody know something about the digi datafire cards and how to make them run with linux and maybe even asterisk?
22:11.32TheBeartrying to find the module now .....
22:12.24shido6sounds like an x100p
22:12.45shido6as it has 2 ports, 1 fxo and 1 for adding another phone or whatever you like but they both cant use the line at the same time
22:13.08QwellI think he's just confused.
22:13.14shido6hehe
22:13.42xlyzhi! I've a little problem with musiconhold. the sound is awfully distorted. I tried to change mpg123 parameters but no luck. Any idea how can I fix it?
22:16.33xlyzanyone?
22:18.30shido6using xwindows on that box, too?
22:18.30fitzelxlyz, do you use the 'real' mpg123?
22:18.36xlyzyep
22:18.39shido6why?
22:18.39Qwell0.59r?
22:19.07xlyz0.59s
22:19.15Qwellbad
22:19.19Qwellget 0.59r
22:19.53xlyzthanks
22:20.06TheBearok  Card 1 has two ports RJ11 marked Line and Phone  Digium S/N BJX116986.     The other card  has four ports RJ45 with a single daughter card marked "Digium Rining FXS Module".
22:20.08Qwellit may or may not fix your problem...but...
22:20.28Qwellrj45, on a pci?
22:20.32TheBearI know I connected a std. phone RJ11 into the second card before. And had * dial the std phone
22:20.35Qwellquadspan?
22:21.26QwellThe first one sounds like an x100p
22:21.30TheBearQwell: I guess so it can take four daughter card one for each port I guess
22:21.47QwellYou're sure its rj45?
22:22.05*** join/#asterisk in (int@2001:5c0:8fff:fffe:0:0:0:1d41)
22:22.08TheBearit's the same size as rj45 it is not as small as rj11
22:22.09inhey
22:22.18QwellIs it a TDM?
22:22.24sedoroxTheBear: you have a TDM400P card I believe
22:22.27TheBearyes
22:22.28sedoroxwith one FXS module
22:22.32sedoroxfor regular phones
22:22.32QwellIs the tdm rj45?
22:22.36sedoroxyes
22:22.38Qwelloh
22:22.41sedoroxhehe
22:22.48QwellI guess that makes sense
22:23.07icebergWhat are the best docs for a noob wanting to get Asterisk working for internal VoIP only?
22:23.18SedoroxMy only guess (This confused Katty too) is that they just keep the RJ45's because of the ability to use the board for the TDM and for the T1 TDM cards...
22:23.32Qwellahh
22:23.34TheBearlookin on the Digium website I would says it's a TDM10B
22:23.35Sedoroxiceberg: voip-info.org has really good stuff on asterisk
22:23.54icebergSedorox, not really I followed the directions there and nothing works
22:24.08Sedoroxyea.. looks right
22:24.13Sedoroxiceberg: for sip.. or?
22:25.00iniceberg
22:25.02inhey man
22:25.11SedoroxI have asterisk setup.. with phones using SIP.. then IAX2 to other * boxes and other providers
22:25.15icebergSedorox, yea, when I try to dial 1000 from x-lite it waits, the console says everything is workign but then x-lite just gives me a reorder and I never get the voice info from the demo box and asterisk never closes the call
22:25.17invoip-info.org is all i have used really
22:25.31Sedoroxhmmm
22:25.33TheBearOk now having est. the cards If I split a phone line to the 1st card and a std phone.  Can I answer a call on the std phone and then transfer this call to a SIP phone (snom 200) via the * server ?
22:25.37Sedoroxcheck codecs
22:26.02SedoroxTheBear: yes
22:26.23Sedoroxiceberg: I had problems with x-lite and codecs... I always moved gsm so it was first...
22:26.30Sedoroxand make sure you have it enabled in sip.conf too
22:26.45Sedoroxeither in general.. or in the section for the login for x-ite
22:27.03TheBearSedorox: care to explain a bit how, can I just pick a SIP phone and be connected to the call or what ?
22:27.09dfunnellIs anyone familiar enough with CAPI to know why when one channel of my BRI is in use Asterisk ignores the other one when it is ringing?  Can dial-out using other channel, but just rings and rings when trying to dial in.  I have incomingmsn=* in capi.conf and it does answer first channel, just not second.
22:28.01lesouvageI have .wav files that I want to convert to sox I read the info on the wike but it doesn't work. I got a message: bad input format for file foo.wav: data size 44 is invalid. What am I doing wrong
22:28.05SedoroxI'm not sure how to make it where you can take over a call once its been answered.. but you just set it up where Asterisk would take the call,, and then you could just ring out the other card for the std phone, and ring out a sip interface for the snom200, and you can either transfer between the two, or use call parking to switch phones
22:28.35Sedoroxlesouvage: are you telling it special options for the input file?
22:29.49xlyzQwell: 0.59r is not secure http://www.mpg123.de/
22:30.25xlyzany other player / format that can be used in musiconhold?
22:30.47*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
22:30.54lesouvagesedorox: I just following the instruction from http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk  I used: sox foo-in.wav -s -w foo-out.wav (with a name of an existing file)
22:31.13Sedoroxhmmmm
22:31.38TheBearsince it has been at least a year since I last used * and the SIP phones.  Can I setup * where it will only handle calls between certain hours eg 8 - 4 otherwise it will ignore all calls ?
22:32.08SedoroxTheBear: yes.. its in the wiki somewhere.. I've seen it.. just not sure where...
22:32.21Sedoroxlesouvage: I'm not sure.. I just converted mine to pcm using that guide.. and it worked...
22:32.25marloweSedorox: Give sox <in_file.wav> -r 8000 -c 1 <out_file.gsm> a try
22:32.34Sedoroxhaven't tried playing .wav's
22:32.43Sedoroxmarlowe: its for lesouvage :-p
22:32.52marlowewoops
22:32.55TheBearSedorox: ok a wiki'ing I will go
22:32.55Sedoroxthats the way I did mine
22:32.57Sedoroxand it works
22:32.58TheBearthanks
22:33.19SedoroxTheBear:  yup... I remember seeing it.. just not sure what page.. I think if you look in the example config parts.. there is one in there that has it....
22:33.27icebergdoes asterisk require a soundcard?
22:33.53marloweiCEBrkr: Read  the FAQ... But, no.. It doesn't.
22:33.58Sedoroxiceberg: nope... unless you wanna do calls on the console... or do a PA System type thing...
22:33.58*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
22:34.00marloweOk nick completion sucks today
22:34.22Sedoroxlol
22:34.26Sedoroxmine always seems to suck :/
22:35.30*** join/#asterisk techie (gus@asterisk.horizonte.us)
22:35.43harryvvanyone ever see this when doing a zaptel compile on fc3 in the /usr/src/zaptel dir  sorry, unimplemented: 64-bit mode not compiled in make: *** [gendigits.o] Error 1
22:36.10harryvvmake linux26 is generating this error.
22:37.30TheBearIs there somewhere the changes made to * over the last year esp. regarding NAT support and development ?
22:37.57SedoroxTheBear: changes should be in the changelog on the ftp server
22:38.11harryvvgendigits.c:1: sorry, unimplemented: 64-bit mode not compiled in
22:38.12harryvvmake
22:38.19harryvvthats not good.
22:38.33elriccan I have 3 groups within zapata.conf?
22:38.36Sedoroxharryvv: you running a 64bit system?
22:38.39harryvvyes
22:38.42harryvvopteron system
22:38.55elricwith three different contexts?
22:39.01harryvvso something needs to be compiled to run un 64 bit mode.
22:39.09Sedoroxand did you install fc3 as 64bit.. or you running it in 32bit mode?
22:39.28harryvvyes I downloaded everthing to run as 64 bit.
22:39.56harryvvsed, but is there a way to know or find out?
22:39.58Sedoroxwhat version of zaptel you compiling? 1.0.6?
22:40.05SedoroxI think uname -a will tell you
22:40.17NuggetI wasn't aware that zaptel would build on 64 bit linux.
22:40.27Sedoroxthats what I'm not sure about...
22:40.30harryvv2.6.9-1.667 #1 Tue Nov 2 14:50:10 EST 2004 x86_64 x86_64 x86_64 GNU/Linux
22:40.33Sedoroxseems I wanna say no
22:40.38NuggetI'd be surprised to learn that it did.
22:40.40Sedoroxand yes.. your running 64bit...
22:40.51harryvvi know :)  its fast
22:40.53Sedoroxhehe
22:41.01Nugget64 bit doesn't make things run faster.
22:41.04Sedoroxdoesn't seem like you can do zaptel on 64...
22:41.14Nuggetin reality is just means that your cache is only half as useful.
22:41.16SedoroxNugget: have you used a 64bit system?
22:41.17*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
22:41.18Nuggetsure.
22:41.22Sedoroxok..
22:41.29Sedorox'cause I'ce noticed a difference when I used one
22:41.30Nuggetthere's no inherent benefit to 64 bit.
22:41.49Nuggetunless you have processes that need to access gobs and gobs of ram
22:41.56Sedoroxeh
22:42.02Nuggetopterons rock, but they rock with 32 bit os just as much
22:42.05lesouvageI have a set  of sounds in Dutch but it's in wave format. Is it possible that they have a kind of protection on the .wav files.
22:42.14NuggetIk ben een vliegende koe.
22:42.27harryvvopterons have the memory controler on the cpu running at cpu speed is one reasons.
22:42.31Nuggetyeah
22:42.45lesouvageNugget: hoi
22:42.49harryvvmy bottle neck is the hard drive.
22:43.03Sedoroxgo SATA2 :-p
22:43.08Nuggetgo 3ware.
22:43.09*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)
22:43.11Sedorox3gb/s transfer
22:43.14Sedorox*drools*
22:43.25Sedoroxyea.. 3ware sata2 controller
22:43.26TrepaliumBetter yet, go Ultra320 SCSI
22:43.37SedoroxTrepalium: slower then sata2
22:44.19TrepaliumNot when you can get 15,000 RPM drives.
22:44.27lesouvageIs there a list in writing of the sounds that comes with Asterisk
22:44.27harryvvsed...I lost my asterisk box last night the harddrive kept renitializing with hda: dma timmer_expirery dma status = 0x21 and did a reboot. even fsck -p could not repair it.
22:44.47Sedoroxeh.. looks like they are on par...
22:44.53Sedoroxnice :(
22:44.55harryvvyea
22:45.13harryvvI have my configs backed up on a windows box but thay are not current.
22:45.28harryvvanyway need to get this system up and running and get asterisk working again.
22:45.33Sedoroxyea
22:45.38harryvvyou said something about zaptel version?
22:45.46lesouvageSedorox: how can I dial a numer from the asterisk prompt.
22:46.02SedoroxI wanna say the dial command.. but I'm not sure.. never did dialing from the cli
22:46.13harryvvls, you cannot download xlite from xten
22:47.17Sedoroxyou can...
22:47.22Sedoroxthats whereI get it from
22:47.42TrepaliumI think there should've been a period in the middle of that statement.
22:48.34Sedoroxhmmm
22:48.47fekleeI'd like to stream an incoming phone call using Ices (interfaces with Icecast).  Has anyone ever tried that successfully?
22:49.14fekleeI've the following line in my extensions.conf: exten => 9779619,3,Ices(/home/feklee/asterisk/asterisk-ices.xml)
22:49.16Sedoroxmy guess... have it being recorded with monitor using wav.. and use the wav to stream...
22:49.28Sedoroxhmm
22:49.40fekleeBut, I always get the error message "Execute of ices failed" when a call arrives
22:50.03lesouvageNugget: Do you know about protecton messures for the .wave files from tric?
22:50.13fekleeThere's a tutorial for streaming a conference: http://www.voip-info.org/wiki-Asterisk+cmd+ICES
22:50.15Nuggetwho is tric?
22:50.25fekleeBut I just want to stream one call, not multiple ones.
22:50.44TrepaliumThe wav files may be already compressed with some other codec.
22:51.36lesouvagenugget: a dutch company who offers services around asterisk. They have dutch sounds for asterisk for download but I can't get them working.
22:52.54harryvvwho here has asterisk running on a 64 bit system
22:53.23*** join/#asterisk stifl3r (~nasty@xtreme-28-156.dyn.aci.on.ca)
22:53.35Nuggetno clue, sorry.
22:53.43*** join/#asterisk GaryH (~ghawkins@gromit.garysoft.co.uk)
22:53.53*** join/#asterisk plc5_250 (~chatzilla@pcp01103028pcs.pntiac01.mi.comcast.net)
22:54.32harryvvwell, next time * is installed its going to be on a system with raid.
22:54.44harryvvor just have two asterisk boxes side by side.
22:54.52Sedoroxlol
22:54.53NuggetI don't think I'll ever build another machine without raid.  it's just too cheap and too easy to not do it.
22:55.35SedoroxI've been pricing a computer for my parents.. pricing a Raid1 80gig SATA setup..
22:55.55NuggetI've never had a drive failure that I wouldn't have happily and instantly paid $300 or whatever to not have had to deal with at the time it happened.
22:56.07Nuggetbut you can't buy raid /after/ the drive fails.  :)
22:56.19mikegrbhehe
22:56.27harryvvnugget well this was installed on one of my systems that was cheap. now i can see that it should have had some redundency built in.
22:57.15TrepaliumJust make sure you either get a high quality RAID controller (not Adaptec), or one so stupid nothing can go wrong with it (ATA fakeraid).
22:57.46SedoroxDo Not EVER get a Promise controller
22:57.52Nuggetyes.
22:57.58Nuggetpromise controllers are awful
22:58.19sivanawhat's a good mfg then
22:58.20QwellTo be fair, they never do promise that it'll work
22:58.22NuggetI swear by those 3ware things.  cheap, rock-solid drivers, and they're supported in any os.
22:58.30Sedoroxhehe
22:59.03Nuggetand there's a nice healthy range of performance, from the $99 two channel cards to the $900 dollar hardware cache zillion channel things
22:59.19TrepaliumDon't get Adaptec ATA RAID controllers either.  Those stupid PATA ones never seem to work unless you want to use the highpoint drivers instead of Adaptec's version.
23:00.05plc5_250can someone help me with connecting via IAXTEL?
23:00.24Sedoroxif I was building a system that needed the preformance and stability to stay up through naything.. I'll go 3ware.. otherwise.. I'll stick with the onboard stuff... normally VIA
23:00.27PatrickDKheh, I like the 2120S for my scsi raids, and use 3ware for ide bulkstorage
23:00.35Sedoroxgood enough for a faimly computer running raid1
23:00.51PatrickDKhmm, they don't make onboard raid
23:00.59elricWhen I do a Background(file) it chops off a bit of the initial voice? but when I use PlayBack() it works just fine.
23:01.01PatrickDKthat is software, not hardware
23:01.07Sedoroxfor sata they do...
23:01.19Nuggetthe onboard stuff is sufficient for saving data, which is certainly the more important point.  they tend to fail miserably at keeping a machine up and running though -- for that goal there's 3ware.
23:01.25harryvvsed, i have a old promise controler that controled my iomega jaz for years.
23:01.36Sedoroxhmmm
23:01.56NuggetI hate using raid that requires cooperation from the os, though.
23:01.58Sedoroxok... Promise PATA or Promise SATA controllers :-p I do admit the old scsi ones were nice
23:02.02Sedoroxditto
23:02.06Nuggetthat's the other big upside of 3ware
23:02.09elricso Playback() is "Welcome to ABC, Press 1 for Sales ..."  Background() is " to ABC, Press 1 for Sales .... "
23:02.11xlyzmusiconhold plays with very high distortion even with mpg123 0.59r. any other idea?
23:02.21elricanyknow what might be causing this?
23:02.22xlyzQwell ?
23:02.26Qwell?
23:02.31Qwelldunno
23:02.36Qwellbut you should keep 0.59r
23:02.44TrepaliumThe volume of your MP3s might be too high.
23:02.59harryvvqwell, ever run asterisk on a amd 64 box
23:02.59vaewyndistortion usally means you have a sucky timebase
23:03.10Qwellelric: might want to do a Wait() first
23:03.12xlyzTrepalium:  how do I set it?
23:03.15Qwellharryvv: can't say that I have
23:03.28elricQwell, alright Wait(1) should work right?
23:03.33Qwellelric: probably
23:03.34harryvvi have googled my make linux26 error and have yet to see anything come up.
23:03.50QwellYou don't need to do make linux26 anymore, do you?
23:03.57QwellI think the makefile does the check on its own now
23:04.05Trepaliumharryvv: You have the x86_64 version of the kernel headers, right?
23:04.16elricok testing.
23:04.19elricthanks Qwell
23:04.27harryvvwell even with make it generates this error. gendigits.c:1: sorry, unimplemented: 64-bit mode not compiled in
23:04.27harryvvmake: *** [gendigits.o] Error 1
23:04.28ManxPowerxlyz: use quietmp3: in musiconhold.conf
23:04.51xlyzI did, but no luck
23:05.12harryvvTrep I suspect I do but what is needed to verify this. Im running fedora core 3 with opteron system.
23:05.21xlyznow I'm using manual
23:05.43harryvvdarn...company showed up now need to leave. If anyone has a answer message me.
23:06.18plc5_250is IAXTEL having problems?  It won't register me via *
23:06.33Qwellplc5_250: What error?
23:06.38QwellI'm not getting any.
23:07.28plc5_250doing an iax2 show registry takes multiple minutes to get to a "registered state"
23:07.29GaryHhi all, I have just upgraded to the 2.6.11 kernel from a 2.6.10 kernel and now the capi drivers don't seem to work any more.  I don't know whether to blame asterisk,chan_capi or the Fritz CAPI drivers for this.  I suspect not Asterisk as the same code (latest stable CVS) works fine on 2.6.10.  Anyone else seen this?
23:08.48plc5_250generally over 10 minutes.  Can't seem to receive calls via IAXTEL either.
23:10.05plc5_250Mar  6 19:24:31 WARNING[7703]: pbx.c:1945 ast_pbx_run: Timeout, but no rule 't' in context 'local'
23:10.22plc5_250I know about the no "t" rule, but it always times out when trying to connect
23:10.33plc5_250er... dial to myself
23:12.04xlyzanybody knows what value want mpg123 -g to reduce gain? I'm googling around and find nothing
23:12.51*** join/#asterisk Tarox (~chris@pD9E7B7CA.dip.t-dialin.net)
23:13.01ManxPower't' means "I didn't get enough digits to find a matching extension"
23:13.48plc5_250I thought 't' was the timeout rule
23:14.15ManxPowerplc5_250: It is.
23:14.21ManxPower't' means "I didn't get enough digits to find a matching extension so I'm giving up"
23:14.31plc5_250interesting...
23:14.37ManxPoweri.e. I didn't get enough digits to make a match before DititTimeout happened.
23:14.45plc5_250can someone try connecting to me via iaxtel?  7003432431
23:15.00ManxPowerplc5_250: IAXtel will let you call yourself.
23:15.15plc5_250that is what I am trying and it is not working
23:15.27ManxPowerplc5_250: Then it's not going to work for other people.
23:19.09*** join/#asterisk mrgoby (~mrgoby@141.211.162.97)
23:19.13plc5_250is there a test number on IAXTEL that I can use to see if my setup is correct for the outbound dialing?
23:20.07mrgobycan you use macros recursively and maintain the original macro exten var ?
23:20.28plc5_250I am not convinved that my exten string is correct for iaxtel...
23:20.34mrgobyor jump from one macro to another and get back to your original extension ?
23:20.49mrgobyvia the vars ?
23:21.45*** join/#asterisk zippp (~zip@c66.190.109.98.ts46v-01.rckprt.tx.charter.com)
23:22.35cjkmoh works great from call coming from an external source (zap, remote voip provider) but when i call someone on my network, both moh's are messed up
23:22.41cjkany one know this problem
23:23.50cjkmoh works great from call coming from an external source (zap, remote voip provider) but when i call someone on my network, both moh's are messed up
23:23.50cjkany one know this problem
23:24.07zipppcan anyone help w/ compiling iaxcli (iaxclient simpleclient) for arm using crosstool .28rc39 on linux
23:24.16*** part/#asterisk srt (~nobody@gw0-cgn.reucon.net)
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23:25.56ManxPowercjk: Are you using 1.0.6?
23:26.39cjkManxPower, no. 1.0.4
23:26.57cjkManxPower, the problem is that i define before every number (extension) the setmusiconhold of the user
23:27.19cjkwell if user1 calls user2 i have the command setmusiconhold executed twice
23:27.29cjkthis is not the case when the call comes from zap
23:27.37cjkat the moment i do not see the solution
23:28.01PatrickDKsounds like your using dial(,,m)
23:28.29ManxPowerPatrickDK: I don't think setmusiconhold twice would cause problems
23:28.36PatrickDKna
23:28.52PatrickDKhmm, you have vad turned on?
23:29.00cjkvad == ?
23:29.08PatrickDKguess that is a yes
23:29.15PatrickDKsilence suppression
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23:29.33cjkno
23:29.44cjkits not turned on
23:29.47PatrickDKwhat phone is user1 and user2?
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23:30.40OloBolahow can I save voicemail using AGI?
23:30.41cjkPatrickDK, well does not really matter. but user1 is grandsteram and user2 is fritz box
23:31.05lesouvageIs there a written list of the sounds with filename and the lines used so I can easily make a tanslation andrecord my own files. The list should be on http://www.ctitec.com/asterisk/
23:31.07lesouvagesounds.htm but isn't.
23:32.29ManxPoweryou mean like sounds.txt in the astrisk source code, or sounds-extra.txt in the asterisk-sounds root?
23:32.35cjkPatrickDK, ManxPower. as i sees it muiconhold is defined on a per channel basis and * can only handle one definition per channel not two. am i wrong here?
23:32.48PatrickDKno
23:32.52*** part/#asterisk xlyz (~xl@81-208-36-176.fastres.net)
23:32.54PatrickDKthe current one wipes out the old one
23:32.57ManxPowercjk: a "channel" is "one leg of a call"
23:33.12cjkManxPower, ok
23:33.17ManxPowerso it would make no sense to have multiple music on hold for the same leg of the same call.
23:34.04cjkManxPower, thats true.
23:34.18cjkbut i predd the hold button on user1's phone and it starts onhold of user2
23:34.25*** join/#asterisk tzanger (~tzanger@165.154.13.35) [NETSPLIT VICTIM]
23:34.33cjki guess this is somehow related to my enum lookups.....
23:34.52ManxPower*grumble*  I don't want to go to New Orleans tomorrow.
23:35.00cjk<PROTECTED>
23:35.00cjk<PROTECTED>
23:35.10cjkI guess this part is causing me the problems
23:35.27cjkthe channels are the mixed up
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23:38.51Sedoroxmmm.. wow...
23:39.21Sedoroxanyone on sipphone?
23:40.13OloBolahow can I save voicemail using AGI?
23:43.08shmaltzwhy am I getting this:
23:43.09shmaltzkernel is too big for standalone boot from floppy
23:43.11shmaltzwhen running make bzImage in /usr/src/linux on slackware 10?
23:43.20BoRiSshmaltz: Thats normal
23:43.31shmaltzMenaing? BoRiS
23:43.36BoRiSwhen you compile alot of stuff into the kernel.
23:43.36shmaltzI should ignore it?
23:43.47BoRiSyeah
23:45.20Sedoroxshmaltz: thats just saying if you wanna put it on floppy.. its too big.. you use to have to do that.. but its old...
23:45.27Sedoroxlike BoRiS said.. safe to ignore...
23:45.38shmaltzgoing for reboot will see
23:47.47Sedoroxcan anyone dial sipphone from fwd and have it go through?
23:48.58shmaltzlooks good, thanks guys
23:50.47brc__woohoo http://www.ssiworld.com/watch/computers.htm
23:54.51brc__http://www.ssiworld.com/watch/washing_machine.htm
23:57.51PTG123should it pass $EXTEN between contexts?
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