IRC log for #asterisk on 20090414

00:03.52*** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net)
00:07.52telnettechok need some help
00:08.09telnettechI have most of the dialplan done but need help with the system command
00:08.23telnettechevening jaytee
00:09.15telnettechI need to create a call file that is then moved to the /var/spool/asterisk/outgoing directory
00:09.22*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
00:09.39telnettechbut i need to create it based on the call to 911
00:09.59*** join/#asterisk culther0 (i=Admentus@c-75-73-54-42.hsd1.mn.comcast.net)
00:10.00telnettechanybody help me with the system command for my dialplan
00:10.08baliktadyour calls to 911 are going to depend on a System call to create a .call file?!?
00:10.10culther0Howdy everyone.
00:10.38_ShrikEtelnettech: System(echo blah >> /tmp/myfile)
00:10.52telnettechbaliktad: correct.....when someone calls 911, I need to notify another extension of the call
00:11.01_ShrikESystem (mv /tmp/myfile /var/spool/asterisk/outgoing)
00:11.12telnettechbut allow the 911 call to go thru with no interruptions
00:11.50baliktadI'm not sure I understand, are you going to be doing this .call file moving around before or after your Dial command to 911
00:12.06*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
00:12.17telnettechwell to complete the call and make it happen, I need to do it before the Dial app
00:12.29baliktadwhich is exactly my point
00:12.42telnettechso I need it to be the priority just before Dial happens
00:13.06*** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb)
00:13.07*** mode/#asterisk [+o russellb_] by ChanServ
00:13.13baliktadI'm a little confused and a little horrified at what you're trying to do
00:14.14telnettechi need to notify another extension that a 911 call has been made but the person that is making the 911 needs to not be interrupted....i have most of the dialplan for the call and the notification except the system command part
00:14.55telnettechso if ext 7103 calls 911, I want to play a recording that tells ext 2100 that 911 was called by 7103
00:15.19baliktaddo you understand that everything in your dialplan before the Dial() to 911 has the potential to fail and potentially prohibit the 911 call from completing?
00:15.21telnettechand i want 7103 to continue thru to 911 undisturbed
00:15.41[TK]D-Fendertelnettech: You mean... you've done everything (nothing else needed), except the *1 LINE* of dialplan you need to call before progressing to dialing your 911? :)
00:16.02telnettechi dont know how to use the system command.....I admit that
00:16.21baliktadyour problem is not with how to use the System command
00:16.28[TK]D-Fendertelnettech: I hand fed you the bloody line :p
00:16.37baliktadyour problem is with prioritizing this notification over the user's actual 911 call
00:16.48[TK]D-Fendertelnettech: and System() it doesn't have parameters!
00:16.58[TK]D-Fenderbaliktad: Hardly.
00:17.10culther0I've been googling my butt off, but I haven't found any detailed information or maybe a start from 0 point with hardware / software to get going on Asterisk.  Currently we've got 4 lines with Comcast; and want to find what equipment( whether comcast is voip to whatever) or maybe kick me in the right way
00:17.11[TK]D-Fenderbaliktad: Call System, Dial out.  Big friggen deal.
00:17.15[TK]D-FenderWhy is this so hard?
00:17.22culther0anyone able to point me in the right direction?
00:17.46telnettechTK
00:17.47jayteewhat's a phone?
00:17.53culther0^ pretty much
00:17.54[TK]D-Fendercutlass: Of you have 4 "lines" now, I'll presume regular analog.  What do you WANT to do?
00:18.23jaytee4 lines with Comcast = proprietary VOIP/ATA
00:18.23[TK]D-Fenderclobbers jaytee with a Bell rotary dial phone.
00:18.27telnettechTK: remember, I am learning as I go.....I only have this forum and the book to really learn from.....I havent used the system command before
00:18.33culther0We have a mix of analog and digital phones, would like an auto attendant to be able to select where to go and the ability to intercom / conference, may buy new phones
00:18.40[TK]D-Fendertelnettech: Go try it
00:19.00[TK]D-Fenderculther0: what is this "digital phone" of which you speak?
00:19.11culther0eh, it's at AT&T 1070
00:19.12telnettechthats the problem.......I dont know how to write the system command line
00:19.12[TK]D-Fenderculther0: what is it plugged into?
00:19.27culther0It's plugged into Cat 5 that's only using 2 wires, so basically standard phone
00:19.31telnettechthat is the only thing i lack before i can test
00:19.47_ShrikEexten => s,1,System(echo I have no hope)
00:19.47[TK]D-Fenderculther0: this? http://telephones.att.com/telephones_ui/phone_store/dsp_product.cfm?itemID=3609&parent=514
00:20.10culther0that's it
00:20.24[TK]D-Fenderculther0: Seems to be a largely dumb analog phone
00:20.33telnettechbaliktad: If the Dial application is invoked before the call file is written, when the call is hung up, the call file will never be generated
00:20.48culther0There's a mishmash of phones, and the owner of the company is open to buying new hardware; but would like some sort of ability to use analog phones for extensions
00:20.55[TK]D-Fenderculther0: that they can be perhaps daisy-chained to each other though
00:21.00culther0and yeah it's possible I was tasked with "figuring out what to do"
00:21.03jaytee[TK]D-Fender, think he'd be good with an 8 port analog card, 4 fxo's for the Comcast interconnect and 4 fxs for the phones?
00:21.23[TK]D-Fenderculther0: Do you like the price you pay for your analog service?
00:21.38culther0well it's a contract
00:21.41culther0with comcast
00:21.53[TK]D-Fenderculther0: Ok, backed into a corner.  Check
00:21.54culther0It's like 120 / mo
00:22.01[TK]D-Fenderculther0: Sounds crappy
00:22.03culther0D-Fender: Yeah
00:22.17[TK]D-Fenderculther0: Ok, life sucks, but you got in behind instead...
00:22.22[TK]D-Fenderculther0: So moving on...
00:22.52culther0D-Fender: The situation is grim and the owner acknowledges it, he wants to keep his phone numbers otherwise he'd be using skype ^_^
00:23.18[TK]D-Fenderculther0: Yup, you'll want a 4-port FXS card to take in your lines.  for that : http://www.telephonydepot.com/Catalog/Sangoma-B-Series/B600D-Analog-Voice-Card
00:23.38[TK]D-Fenderculther0: Now how many PHONES do you want to support?
00:23.50culther0no more then 20, currently only like 6 or 7
00:25.09[TK]D-Fenderculther0: For your phones : http://www.voipsupply.com/linksys-spa8000-g1
00:25.19[TK]D-Fenderculther0: that + a PC to run * on.
00:25.24[TK]D-Fenderculther0: DONE
00:25.29culther0interesting
00:25.57[TK]D-Fenderculther0: cost > $750 + PC
00:26.00culther0D-Fender: we do have a punch down block, is that even really worth it?
00:26.13[TK]D-Fenderculther0: Use if you have, but you don't require
00:26.50[TK]D-Fenderculther0: the SPA unit supports RJ21 & RJ11
00:26.59jayteecan I ask a stupid question about this?
00:27.03culther0^ yeah
00:27.11jaytee[TK]D-Fender?
00:27.12[TK]D-Fenderjaytee: Would saying "no" stop you? ;)
00:28.14[TK]D-Fenderjaytee: You know smart questions are always welcome :p
00:28.15*** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db8730b.pool.einsundeins.de)
00:28.33jayteeyou said he needed 4 port FXS card? to take the Comcast lines? if he's using them as analog trunk lines then shouldn't it be FXO ports in the * box?
00:28.50[TK]D-Fenderjaytee: Typo silly!  the product is still right!
00:29.00[TK]D-Fenderjaytee: tHAT SHOULDN'T EVEN BE A QUESTION!
00:29.03[TK]D-FenderDarn caps
00:29.17*** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio)
00:29.18jayteeso a 4 port FXO card then, ok.
00:29.19[TK]D-Fenderwishes his keyboards all had the key itself light up
00:29.29[TK]D-Fenderjaytee: "duh" :)
00:29.42jayteelong day, huh?
00:29.43[TK]D-Fenderculther0: So yes, 4-port FXO card.
00:29.53Wired_Lifehello how i make it to dial and if the person click ok jump to next in dial plan?
00:29.57[TK]D-Fenderis pretty sure he doesn't even truly know the terms anyway...
00:30.12[TK]D-FenderWired_Life: where on earth do you click "ok"?
00:30.27[TK]D-FenderWired_Life: "it" dial?  what is "it"?
00:30.35[TK]D-FenderWired_Life: Cold you be perhaps a little more vague?
00:30.47[TK]D-Fendercould*
00:31.03culther0D-Fender: Rawr, I'm at home doing some off-time research to impress the boss, the cable that comes out of the internet telephony box is a black brick looking cable that plugs into the punch down block.
00:31.39culther0from there, the punchdown block has a mishmash of wire and electrical tape and labodomized cat 5
00:31.48Wired_Lifei have this rule: exten = r,1,Dial(SIP/oli,20) and i search a option to call next thing in dial plan if oli answer the call
00:31.52[TK]D-Fenderculther0: Yeah, they are feeing you analog challes delivered over your internet connection.
00:32.19[TK]D-FenderWired_Life: if they answer, then you are TALKING with them
00:32.28[TK]D-FenderWired_Life: Typically that is the end of things.
00:32.31jayteeit would be awesome if Comcast would smell the coffee and sell cheap SIP trunks
00:32.37[TK]D-FenderWired_Life: what is there to do "next"?
00:32.52jayteebut then I ain't waiting for my hair to grow back either
00:33.13telnettechhere comes the dumb question
00:33.29[TK]D-Fenderjaytee: They are supporting CPE while getting to deliver a single circuit.  All your savings are belong to them!
00:33.42[TK]D-Fenderjaytee: ch-ch-ch-chia!
00:34.02culther0D-Fender: Makes sense; would it be silly to call comcast and ask them if they support this kind of thing or are they just going to tell me to spend 3g's on a proprietary PBX; but since this is all analogue will Asterisk route information such as caller ID and whatnot?
00:34.08Wired_Lifei will make a rule if oli answers (its like a: you are ok) then answer and playback soundfile
00:34.09jayteethe bit with Joe Pesci about the "drive thru" in Lethal Weapon 2 comes to mind
00:34.40[TK]D-Fenderculther0: Your connection to them is dumb analog. The fact that it walks in the door under IP is transparent to you.  No, you do not need their crap.
00:35.05Wired_Lifemy problem is if oli answers he has the person on the phone
00:35.12[TK]D-FenderWired_Life: what "rule"?  Where the hell is this call coming from?
00:35.44[TK]D-FenderWired_Life: and if you want him to hear an audio message before the call is bridged, thats what the A() option is for
00:36.38Wired_Lifeif a call comes from anonymous i want to call oli and if he say is ok then answer the line and playback sound
00:37.17[TK]D-FenderWired_Life: then you want the "M()" option instead and you'll need to make a macro to prompt for the "OK" and return the appropriate MACRO_RESULT
00:37.43[TK]D-FenderWired_Life: "core show application dial" <- go read the instructions and come back when you've failed a pile of tests and changes
00:38.30[TK]D-Fenderculther0: You aalready ahve all the analog phones you figure you'll need for this?
00:38.43[TK]D-Fenderculther0: And jsut want to recycle those into your new setup, correct?
00:42.17culther0So: Comcast Phone Lines go into FXO card on PC, (4 ports 4 lines?), Line from computer goes to Linksys analog "gateway", gateway has lines to each one of the phone areas.. amirite?
00:42.24culther0Yeah correct
00:42.29culther0sorry I was reading a site
00:42.49[TK]D-Fenderculther0: Yes, you seem to follow
00:43.03[TK]D-Fenderthe Linksys device connects to your LAN and talks SIP to *
00:43.07[TK]D-Fender~SIP
00:43.08infobotsip is probably http://www.cs.columbia.edu/sip/  X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/  Session Initiation Protocol (see RFC 3261)
00:43.43culther0I see, and I would have the option to add additional phones via ethernet if the boss decides to buy more things, each line would have an extension, etc?
00:44.18[TK]D-Fenderculther0: You can expand in any way you choose.  I picked this SPA unit because of your immediate intention to use existing analog phone
00:44.33[TK]D-Fendersulotherwise I highly recommend getting SIP hard phones.
00:44.34*** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb)
00:44.34*** mode/#asterisk [+o russellb_] by ChanServ
00:44.36culther0Sweet, Alright this sounds like a workable solution
00:44.55[TK]D-Fenderculther0: And it is highly cost-effective
00:44.57culther0Asterisk isn't too crazy to get a basic configuration setup is there?
00:45.00culther0*nod*
00:45.20baliktadok TK, one of my customers has a request that I don't really know how to solve
00:45.23[TK]D-Fenderculther0: grab the book, and read the guide for some "inspiration" on how simple a setup could be :
00:45.25[TK]D-Fender~book
00:45.25infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
00:45.27[TK]D-Fender~jerjerguide
00:45.28infobot[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * :  http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/
00:45.44[TK]D-Fenderculther0: There is indeed a learning curve to *, but it pays off.
00:46.13baliktadcustomer has one site with 6x Linksys SPA-942's, 1 Asterisk box, and 2 DID's
00:46.21telnettechculther: I agree with the payoff....i am starting to see it but still need some help
00:46.58culther0Alright; D-Fender your help is mucho appreciated
00:47.20culther0Yeah I will take a look at other pre-fabbed packages or the possibility of finding a local vendor who knows how to configure
00:47.21baliktadright now the incoming DID's just Dial() everyone (which is fine), but they want to be able to put calls on hold at one station and pick them up at another (ie, call parking without the parking)
00:47.50[TK]D-Fenderbaliktad: How do you "have" a DID? :)
00:48.11[TK]D-Fendertelnettech: Ask your consultant if training may be right for you!
00:48.16culther0I'd kinda at some point like to come up with a encompasing tutorial cause this stuff isn't readily apparent from what people google
00:48.17[TK]D-Fender</psa>
00:48.26baliktad2 phone numbers routed to their * box provide incoming voice services
00:48.34telnettechwe dont have a consultant
00:48.57[TK]D-Fenderculther0: I haven't seen an encompassing doc that I could jsut trow at someone yet.
00:49.05telnettechand i have gone to training.....i think i have learned quite abit over the last 6 months starting from scratch
00:49.17telnettech* toots his own horn
00:49.28[TK]D-Fendertelnettech: get a better trainer young Jedi!
00:49.41telnettechjsmith was my trainer....lol
00:49.41[TK]D-Fendertelnettech: Not in here, this is a family show!
00:49.49jayteeoh, god! not that tune again. Brian, you need to work on your repetoire
00:50.06baliktadproblem now is that each phone is independent, if one person answers, it's not easy for someone else to join or finish the call from another phone
00:50.12[TK]D-Fendertelnettech: module unload chan_brokenrecord.so !
00:50.28jayteeI can't picture Jared without a Mango Smoothie in his hand
00:50.41telnettechi am trying guys....
00:51.07[TK]D-Fendertelnettech: thats what makes this mildly entertaining.  Now dance monkey, DANCE!'
00:51.09*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
00:51.23[TK]D-Fenderdraws his 6-shooter
00:51.27baliktadboss basically wants his SIP phones to function as if they are all sharing the same 2 lines
00:51.41mmlj4excuse me while I whip this out
00:51.57[TK]D-Fendermmlj4: EWWWWWWWWWW!!!!!!!!!!
00:52.09russellb_baliktad: Asterisk SLA support will do that
00:52.17[TK]D-Fender....
00:52.25mmlj4I was expecting laugher not ewwness
00:52.30[TK]D-Fenderrussellb_: * SLA makes baby Jesus cry...
00:52.37jayteegoes off to investigate if the lyrics to "You've gotta be cruel to be kind" were written by someone who's first name starts with A :-)
00:53.06baliktadI've been looking at SLA, although I'm not sure if it will
00:53.18mmlj4[TK]D-Fender: surely you know what movie that's from?
00:53.24jayteewhen it comes to either herpes or SLA, most people choose herpes!
00:53.26[TK]D-Fenderbaliktad: It can if you have the speed-dials to support it, but its... FUGLY
00:53.30mmlj4sla?
00:53.50russellb_sighs
00:53.53[TK]D-Fendermmlj4: its like "blah", but compiled into an * module ;)
00:54.03mmlj4ok
00:54.04baliktadif I use SLA, one extension will ring and the others just "show" ringing on the BLF, without showing the incoming caller ID
00:54.04[TK]D-Fender~sla
00:54.05infobotsla is, like, service level agreement, or shared line appearances
00:54.05mmlj4blah?
00:54.06telnettechi will say that you have taught me alot TK.....and i do look for the answers BEFORE i come into the chat.....there just isnt anything made for dummies when it comes to the system command
00:54.33mmlj4ok, I misparsed... of course I know what an SLA is
00:54.42jayteeshoots consoling, sympathetic looks in russellb's direction
00:54.44russellb_[TK]D-Fender: Your attitude got old with me many months ago
00:54.56[TK]D-Fendermmlj4: You know "bleh", the sound projectile makes if you don't build up enough pressure first ;)
00:55.23baliktadrussellb_ are you the russell that wrote the SLA pdf I've been poring over for the last month?
00:55.31mmlj4but how did key-functionality get morphed into an SLA?
00:55.38[TK]D-Fenderrussellb_: Sorry if I rag on this one feature, I know you were steered towards this, I don't want to lay blame or anything
00:55.56[TK]D-Fender*sigh*
00:56.10telnettechand i dont have much experience with linux, networking and computers....i am straight  from the old world of telephony
00:56.12jayteehe quit or at least his underscore nick did and the other is grayed out marking away status
00:56.15[TK]D-Fendermmlj4: synonymous.
00:56.44jayteetelnettech, do you know how to get to Carnegie Hall?
00:57.01telnettechtake a left at albuquerque
00:57.05telnettechsilly rabbit
00:57.11[TK]D-Fenderjaytee: I'm sure http://maps.google.com will know!
00:57.12jayteepractice my boy, practice!
00:57.46*** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net)
00:58.01jayteeand if the answers don't come fast enough then read until your eyes bleed
00:58.08[TK]D-Fendertelnettech: You need some Specialized High Intensity Training!  I dole out about as much of it at my office as I can muster!
00:58.22jayteeSHIT
00:58.25telnettechare you hiring.....:)
00:58.38knarflydoes asterisk-1.6.09 need zaptel to run with an X101P analog card?
00:59.09[TK]D-Fenderknarfly: No, * 1.6 does not support Zaptel
00:59.11jayteenope, because 1.6 won't run zaptel
00:59.28*** join/#asterisk JuStIcIa_ (i=john@cbl-sd-74-96.aster.com.do)
00:59.49knarflywill my x101P work with *-1.6.0.9
01:00.01[TK]D-Fender~dahdi
01:00.02infobot[~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav
01:00.03[TK]D-Fender^^^^^^^^^^
01:00.34knarflyand dahdi installs with the asterisk taeball?
01:00.45knarflytarball
01:00.47[TK]D-Fenderknarfly: No, it is a separate package
01:00.49jayteenope, dahdi installs with dahdi
01:01.09culther0ah ffs now I can't seem to save this chat on IRC
01:01.10culther0>_<
01:01.18jayteeget the tarball that is both the drivers and the tools
01:01.22[TK]D-Fenderculther0: copy / paste
01:01.32knarflywhere do I dahdi
01:01.33jayteeXchat logging FTW!
01:01.59culther0it won't copy paste!
01:02.00culther0>_<
01:02.47knarflywhere do I get the dahdi tarball
01:02.47jayteeknarfly, http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
01:02.54culther0*stabs own eyes out*
01:02.57[TK]D-Fenderknarfly: www.asterisk.org
01:03.06jayteeor there
01:03.13telnettechTK: See they moved all the guys that knows asterisk for my company from US to Singapore.....and the rest of us were basically told to sink or swim.....i am doggie paddling but barely
01:03.43jayteedoggie paddling all the way to Singapore? wow, that's a rough swim!
01:04.16telnettechTK: and I am the best of the rest so far....the others dont even try cause there is no one to enforce it cause even the supervisor is scared of Asterisk
01:06.35[TK]D-Fendertelnettech: Then they should welcome your becoming better trained by someone who can show you how to seriously get things done.
01:07.21knarflywhen installing do i need to install dahdi first?
01:07.27jayteeyes
01:07.30[TK]D-Fendertelnettech: There is "general" and "theory" training, and there is "here's what you need to know, here are the real gotchas & opportunities, and here's some tricks to get the most out of the basics"
01:07.37telnettechright now the voice part is in limbo with the merger we are going thru......so it is not being looked t....they are eventually going to require us to do hi speed internet and VOD thru IPTV
01:07.43jayteeknarfly, what distro are you installing it on ?
01:09.03telnettechi mean i thought i should go to the beginners from digium and was told by supervisor that i need to go to advance....i didnt do too bad but it just opened up more learning that is needed....i agree
01:09.22telnettechi didnt pass my DCAP but know why
01:10.00jayteeme neither, passed the written but bombed on the lab cuz I ran out of time. If I'd passed on installing the T1 card I probably would have finished and passed.
01:10.04telnettechi dont mind book smart but i want real world experience too
01:10.11knarflyjaytee: fedora 10
01:10.20telnettechi failed both parts
01:10.34jayteeugh, wrong platform for *, better to use CentOS
01:10.57[TK]D-Fendertelnettech: Book smart would require more "real" books :)
01:11.13knarflyCentOS sux the big red one
01:11.27telnettechI have read most of ATOF and i am also reading at same time SIP Demystified
01:11.37[TK]D-Fendertelnettech: TFOT is a kinda loose grab at newbs, and doesn't show either deep detail, or too much practical minimal use
01:11.58[TK]D-Fendertelnettech: just a bit too "spares", but it covers the wider range so IRC, etc can fill in the gaps.
01:11.59telnettechi also try to get onto voip-info to read as well
01:12.19[TK]D-Fendertelnettech: the WIKI is a seriously mixed bag
01:12.34[TK]D-Fenderknarfly: I use it all the time. jsut fine
01:12.48telnettechso thats why i come on here and ask questions
01:13.14telnettechat least I am not as bad a Wired_Life
01:13.43telnettechi don know how to read the dialplan for the most part :)
01:13.56[TK]D-Fendertelnettech: comparative self-deprecation.  SUCKcess!
01:14.01*** part/#asterisk culther0 (i=Admentus@c-75-73-54-42.hsd1.mn.comcast.net)
01:14.22[TK]D-Fendertelnettech: 11 steps to go!
01:14.41*** join/#asterisk culther0 (i=Admentus@c-75-73-54-42.hsd1.mn.comcast.net)
01:15.00telnettecheven though i know that I am going to get criticized for my question by you, I still ask them so that i can get better
01:15.11jayteeknarfly, the only difference between Fedora and CentOS is that Fedora is more bleeding edge. they both draw the same code base from Red Hat.
01:15.22[TK]D-Fendertelnettech: Think-skinned, now 10 to go!
01:15.48jayteeand it's the bleeding edge that will give you major headaches making all the bells and whistles in * work properly. Especially the whistles!
01:15.57telnettechI just dont have a test system to just play with all the time as I am required to do things on live customer systems
01:16.04[TK]D-Fendergoes to hone his katana...
01:16.18jayteeknow how to tell if it's sharp?
01:16.20culther0Lets see if this is logging now
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01:16.33telnettechso i am trying to be cautious and not make more trouble than I can handle
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01:19.25telnettechso if all my learning sources are flawed, per your opinion, then how else can i gain knowledge except real world
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01:23.02jayteeI never said your learning sources are flawed
01:23.37telnettechim talking about TK
01:23.57jayteewell, that's a whole nuther subject :-)
01:24.20[TK]D-Fendertelnettech: I didn't say "flawed", they just haven't been "right for you".
01:24.21knarflyasterisk-1.6.0.9 installed and running...will setup X101P card tomorrow....good night Mrs. Calabash, wherever you are!
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01:24.42jayteefriggen Fedora, ugh
01:25.03[TK]D-Fendertelnettech: You seem the type to best profit from directed training as opposed to lose forms like TFOT.
01:25.28telnettechi learn better in a lab as i am instructed, you are correct
01:25.45[TK]D-FenderDear God.... "Mrs. Calabash"... like that doesn't (carbon) date you.....
01:26.02telnettechi like the hands on, I dont read too well and take what i read and put it in practice....never had
01:26.08jayteebrian, do you have an older PC you can setup at home with a couple cheap X100 cards?
01:26.32[TK]D-Fenderwaits for knarfly's return so he can toss him into a gas chromatograph for testing...
01:26.38jayteeI can mail ya a couple of X100 FXO cards to use on analog lines
01:26.52telnettechi am trying to get one setup yes....i just dont have the time.....it seems like i am always working on a customer....i am even working now as we speak
01:27.07jayteeand you've got a SIP phone from class.
01:27.13telnettechyep
01:27.27jayteetime? who ever has time. we don't "have" time, we "make" time.
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01:27.46[TK]D-Fenderjaytee: Cards are the real issue, dialplan is.  Frankly everyon who thought a card will help them learn * quickly finds out that devies are a dozen lines of setup and dialplan is EVERYTHING
01:27.56telnettechand i dont like to have a problem that i cant figure out
01:28.13jayteeand since according to relativity, space and time are intertwined then maybe you just need to find the space first and the time will be there already interwoven.
01:28.17[TK]D-Fenderdarn, typing skills degrading...
01:28.45telnettechI have a system already jaytee....i have red hat loaded on it but i havent loaded the asterisk and all yet
01:28.50[TK]D-Fenderjaytee: Einstein flunked the "practical of relativity" ;)
01:28.52jayteedialplan is pretty much at least 98.7 percent of Asterisk.
01:29.31jayteeEinstein divided by zero in the proofs for one of his early papers. Plus he got a D in Algebra in High School
01:30.32telnettechand i think i do well with basic stuff in the dialplan....i just have problem with intermediate and advance programming in it....like chan variables, critical thinking Goto statements, and etc
01:30.59telnettechif it gets to complex it gets confusing to me right now
01:31.25jayteethat's a skill that takes time to develop and is more programming logic than what most telecom techs get exposed to
01:31.45telnettechlike this system command.....i have sat here tonight and taken what TK told me earlier and struggled thru most of the setup....except the most important part
01:34.44[TK]D-Fendertelnettech: System(/usr/sbin/asterisk -rx "originate SIP/operatortypeperson Local/${CALLERID(num)}@omfg911")
01:35.54telnettechthank you TK....can you explain what it does a little...so that i can decipher for future reference
01:36.10[TK]D-Fendertelnettech: [omfg911] exten => _X.,1, Playback(thefollowingtwidialed911) exten => _X.,2, SayDigits(${EXTEN})
01:36.51telnettechi just needed the system part....i have the dialplan setup
01:36.53[TK]D-Fendertelnettech: makes a call to * to have it trigger an out-call to your operator and when they answer they get this message played to them
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01:37.20[TK]D-Fendertelnettech: actually, drop the "Local/" off.  I reversed it
01:37.57DavidBerGood evening - I was wondering if someone could answer some questions regarding a 1.4.24.1 setup with extensions being busy on inbound calls.
01:38.12Qwellare they busy?
01:38.18DavidBerhehe - nah :)
01:38.37DavidBercalls come in - and asterisk is saying that the extension is busy
01:39.04Qwellpastebin a log
01:39.05telnettechthanks TK
01:39.15DavidBerI have a few dids from Vitelity.  One goes to an IVR and one goes to an extension - the extension goes busy and I get a message from Vitelity.  The IVR goes to the IVR and then when to the extension goes busy
01:39.20telnettechi will study it so that i know what it does
01:39.45[TK]D-Fendertelnettech: "help originate"
01:39.55Qwellyour extension likely isn't registered/setup properly
01:40.08DavidBersip peer shows them online
01:40.15DavidBerstupid question - how do I do a pastebin?
01:40.19Qwell~pb
01:40.20infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
01:40.23telnettechoh.....i have been looking at system all this time
01:40.45[TK]D-Fendertelnettech: As I said, System doesn't have parameters.
01:41.01[TK]D-Fendertelnettech: you jsut call * to fire off an independant call so you can resume dialing out.
01:41.11telnettechcorrect
01:41.12[TK]D-Fendertelnettech:  this spawned call is not attached to the other in any way
01:41.18DavidBerQwell - what do you want pasted?
01:41.46telnettechis there a way to log this call so that they cant say that it was never made? like some type of CDR?
01:42.37telnettechjust asking
01:45.54k-mani have a linksys spa2102, if i don't want to use the router component (as i already have a router) do i plug the ethernet port or wan port into my network?
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01:46.52otomotohow can we log to a file certina from a context using "exten => ....."
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01:52.47DavidBerQwell - here ya go - http://pastebin.com/d68ddabd9
01:54.15[TK]D-Fendertelnettech: exten => _X.,3,System(echo "dumbfuck ${EXTEN} called 911 on ${TIMESTAMP}")
01:54.43[TK]D-Fendertelnettech: If CDR does not make you feel warm & fuzzy as it is
01:54.55telnettechok thanks
01:59.44telnettechTK D: does the originate need a sound card in the server?
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02:06.15[TK]D-FenderGRRR
02:06.31[TK]D-Fenderfsck-ing Flash crash
02:07.32[TK]D-Fenderkicks Adobe/M$ in the nads
02:09.50jayteethey are nadless nerds, it's an exercise in futility. might as well tilt at windmills, Quixote
02:10.27telnettechgood night huys
02:10.30telnettechguys
02:10.38jayteenite brian
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02:14.39[TK]D-Fendergrabs his lance and mounts his steed.
02:15.13[TK]D-Fenderthinks that sounds entirely too dangerously ambiguous in these parts for his own good
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02:18.00jayteehehehe
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02:33.32OctothorpeHey, somehow AsteriskNOW 1.5.0 seems to be able to run Apache as apache:apache instead of asterisk:asterisk even though /var/www/html is owned by asterisk:asterisk. What's the score? Didn't they have to be the same owner with FreePBX?
02:33.49OctothorpeAh, #asterisknow exists, I'm taking it there.
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03:02.14Wired_Life1i hate macros
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03:10.51[TK]D-FenderWired_Life1: That's OK, they have no feeling for you to hurt :)
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03:14.49Wired_Life1i have a macro test and when im click answer call the call get answers but i dont want a answer call
03:15.36[TK]D-FenderWired_Life1: "click"?  What "click"?  You are talking crazy-talk....
03:16.14jayteegibbledy bibbledy
03:16.21[TK]D-FenderWired_Life1: And any YEAR now you might learn to start pastebin-ing your backup when you have problems.... unless of course you jsut want to complain about them and not try to solve them
03:16.25jayteeargle bargle whoosh!
03:16.41[TK]D-Fenderjaytee: Bah weep grah nah weep nini bong?
03:17.05jayteey'know, it's pretty damn scary when I understand that!
03:17.20[TK]D-Fenderjaytee: :D
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03:18.19[TK]D-Fenderknows too much and the ninjas have probably already been dispatched to dispose of him
03:18.24Wired_Life1http://pastebin.com/d72d2cc0b
03:19.33[TK]D-FenderWired_Life1: Where did you get this variable "ISOK" from?
03:20.21[TK]D-FenderWired_Life1: Wired_Life1 and why do you think it means anything?
03:20.38jaytee"ISOK" "AIN'TOK"
03:21.04[TK]D-FenderWired_Life1: "core show application dial" <- read the instructions.
03:21.13drmessanoDid you set IS_NOT_MADE_UP=no in unicorns.conf?
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03:21.41Wired_Life1i have read the fucking instructions
03:21.50Wired_Life1its all english
03:21.58Wired_Life1and very short i want more
03:22.04[TK]D-FenderI set MILLIONDOLLARSINMYBANKACCOUNTNOW=true , so WHERE'S MY FUCKING MONEY!??!?!?
03:22.08Wired_Life1its not helpful to me
03:23.00[TK]D-FenderWired_Life1: The instructions tell you what variable has to be set and what its values mean.  You INVENTED one out of thin air
03:23.35[TK]D-FenderWired_Life1: Now go follow the instruction.  You aren't prompting your callee for any input and you aren't determing your result based on it.
03:25.45Wired_Life1all with the macro and so on i dont want
03:26.01Wired_Life1i dont want to answer a call before check
03:26.18Wired_Life1i must check if a person is on phone
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03:27.35[TK]D-FenderWired_Life1: What you asked for is to check if they WANT to take the call
03:28.11Wired_Life1im so silly
03:28.21Wired_Life1english is not my language
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03:30.36termas_Would anyone like to help me with [what seems to be] an obscure Realtime problem?
03:31.02[TK]D-Fendertermas_: Just provider your details and see who bites
03:31.46termas_loki2301: I have a psql DB, extconf.conf points to it for sip users.
03:31.53Wired_Life1its now 5:31 in germany and i cant sleep without solving this problem
03:32.15termas_When a phone comes online, it registers with the details from the DB fine.
03:32.34termas_and can recieve calls, place calls, anything. It all works fine.
03:32.43[TK]D-FenderWired_Life1: And you can't solve much of anything with no sleep.  visciouscircle=true
03:32.46termas_and then, at some point, it stops, and nothing will work until the phone is rebooted.
03:33.21[TK]D-Fendertermas_: tahts really vague.  clarify this "nothing will work" bit... a LOT
03:33.26termas_"everyone is congested at this time"
03:33.43termas_One second, I'll get the exact error.
03:34.03termas_[Apr 14 11:32:03] WARNING[9006]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
03:34.04termas_<PROTECTED>
03:34.08Wired_Life1i need more sites with examples and so on and i think the rule i want is already online but i dont know the words to search for @ google
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03:35.18termas_I don't know what else to provide, is the DB schema useful?
03:38.47Wired_Life1i want it so: if a anonymous call comes i want to check if a person is available behind the phone and then i will answer call and play some playbacks and record voice sample for the person because the caller is anonymous
03:39.13Wired_Life1you know what i mean?
03:39.25s14ckhello
03:39.28Wired_Life1but i dont know the words to search for
03:39.40Wired_Life1im sure there is already a rule @ google
03:39.48termas_Wired_Life1: do you want to test for the status of the end point?
03:40.14termas_and then, play to the end point, a sound, when the start point is anon?
03:42.51Wired_Life1no
03:43.53termas_How do your people log into the phone?
03:44.04termas_Agents?
03:44.27Wired_Life1if a anonymous call comes in i want to check if a person is available behind the phone (make a call to check or so) and then i will answer the call from anonymous and play some playbacks and record voice sample for the person @ end point because the caller sends no number
03:45.24Wired_Life1now i search only the rule to check if end point is there
03:45.36termas_So to the caller it would be like "Hi, you have no caller id, say your name." then when it reached the end point, the recording of the name is played to them?
03:46.00[TK]D-Fendertermas_: SIP peer dumps, DB dumps, firewall checks, etc. and you are getting CLI output, there's something there as well
03:46.17jayteenite all
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03:46.45Wired_Life1termas_ yes the voice recording stuff i have already
03:47.12s14ckhow can I get one id call in progress and make relationship with id users at my db?
03:48.13termas_[TK]D-Fender I can rule out firewall I think, since it works when the phone is freshly rebooted.
03:48.14Wired_Life1i search only for the part to check if a person is available behind the phone
03:48.18[TK]D-FenderterI would think that perhaps you failed a SIP qualify or something and the peer's remote NAT (if applicable) timed out and closed the port behind it, etc.
03:48.25[TK]D-Fendertermas_: I would think that perhaps you failed a SIP qualify or something and the peer's remote NAT (if applicable) timed out and closed the port behind it, etc.
03:48.35termas_Wired_Life1, what's wrong with Dial() ? If they are there, they will pick it up?
03:49.31termas_[TK]D-Fender There is no NAT, they are on the same subnet. What do you mean by SIP qualify?
03:50.23[TK]D-Fendertermas_: a keep-alive signal.  if the host fails to respond * will flag them as uncontactable until they check in again.  This of course happens when you restart the phone which is why I suspect it may be the case
03:51.18termas_[TK]D-Fender The [DEBUG] stuff in the CLI says things like
03:51.57termas_[Apr 14 11:39:52] DEBUG[28140]: res_config_pgsql.c:170 realtime_pgsql: 1Postgresql RealTime: Result=0xb74ffc78 Query: SELECT * FROM sip_friends WHERE name = 'ross.paine' AND host = 'dynamic'
03:51.57termas_[Apr 14 11:39:52] DEBUG[28140]: res_config_pgsql.c:178 realtime_pgsql: Postgresql RealTime: Found 1 rows.
03:51.58termas_[Apr 14 11:39:52] DEBUG[28140]: res_config_pgsql.c:790 pgsql_reconnect: Postgresql RealTime: Everything is fine.
03:52.41[TK]D-Fendertermas_: enable SIP DEBUG and watch what's actually happening.  when a call fails, dump the DB, check your peer at CLI, etc
03:52.50Wired_Life1http://pastebin.com/dc114b59
03:53.06termas_ok
03:54.50Wired_Life1my problem is i dont want to forward the anonymous call to oli before he said ok
03:55.24Wired_Life1but only when oli answer it go next
03:55.30[TK]D-FenderWired_Life1: exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no) <-- extra spaces = BAD
03:56.32Wired_Life1i know the rules are buggy they are from http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
03:56.40[TK]D-FenderWired_Life1: that macro should prompt him asking if he wants to accept it or not.
03:57.21termas_http://pastebin.com/d6cf10fa7
03:58.23Wired_Life1i know i know im programming visual basic, php and so on but i dont know this fucking dial plans
03:58.55termas_extensions.conf is ugly, and hard :(
03:58.59[TK]D-FenderWired_Life1: Go read the book and play around.  its just 1 app after another....
03:59.10[TK]D-Fendertermas_: Ugly yes, hard, no.
03:59.41[TK]D-Fendertermas_: I've writtem more complex languages.  Not written IN, WRITTEN.
04:01.12Wired_Life1[TK]D-Fender look @ http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and you know my problem... this rule is answering directly a call... i want to check if anyone is there to make no costs for the caller
04:01.40[TK]D-FenderWired_Life1: the CALLER is not #^&%$ing answered, the CALLEE IS.
04:02.29[TK]D-FenderWired_Life1: the caller is not answered until it comes back "accepted" by your macro
04:02.43termas_It's not 'hard' as in it's difficult, it's hard compared to the alternatives.
04:03.01Wired_Life1[TK]D-Fender are you 100% sure?
04:03.11termas_the amount of work you have to do in extensions.conf is massive, to achieve little.
04:03.13[TK]D-Fendertermas_: its very easy when you realize its every bit as "dumb" and the 1-page instructions for each app really entails.
04:03.57[TK]D-Fendertermas_: 1 silly step after another.  Set a variable, do a comparison.  Jump somewhere if X=Y, Dial Z, whatever
04:04.09[TK]D-Fendertermas_: this is "hello world" grade stuff
04:04.27[TK]D-Fendertermas_: there is no "concurrency", its all just linear
04:04.38termas_Sure.. Then look up an extension in the db, relate that to a user, log that somewhere else, etc etc.. to hard. :p
04:04.41termas_toO^
04:05.01kc8pxytermas_: compared to?
04:05.10kc8pxytermas_: something else that's free?
04:05.34termas_I'm going to have to admit at this point, I'm using adhearsion for my dial plan logic..
04:06.01Wired_Life1[TK]D-Fender the call is get answered there is a recording and a recording can only make if a caller is there
04:06.02drmessanoheh
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04:06.53termas_But, just because I use adhearsion, doesn't make me less of a man...
04:07.02drmessanoYes it does
04:07.07[TK]D-Fenderter"relate to a user"?  No such thing.  Call come in.  Who is it auth'd as?  Go to their context.  everything else is dialplan.  loggin... bleh.. no impact on what you DO
04:07.09drmessanoEveryone says that
04:07.14drmessano"Its shrinkage"
04:07.17drmessano"No really"
04:07.27drmessanoSorry, you're less of a man
04:07.32termas_:(
04:07.48[TK]D-Fender00:05]<termas_>I'm going to have to admit at this point, I'm using adhearsion for my dial plan logic.. <- so what you're really trying to say is "No.. I don't get it either, I've jsut found my cop-out already!" :p
04:07.59termas_[TK]D-Fender The numbered extensions relate to different users on different days, in my dial plan.
04:08.27[TK]D-Fendertermas_: if you say so...
04:08.30termas_[TK]D-Fender: Well, no, I've done extensive work in extensions.conf,
04:08.57kc8pxytermas_:  callcenter?
04:09.15termas_Government,
04:09.22termas_small government,
04:10.37kc8pxytermas_:  diferent users use the same phone, but different mailboxen?
04:10.53termas_In my dial plan?
04:11.33termas_No, the users all use the same phone every day. Captain Hammer, for example, always uses the same username/phone.
04:11.55termas_But on wednesday, he's the janitor, every other day he fights crime, so he has a different phone number on wednesday.
04:11.56termas_etc.
04:12.28kc8pxy< termas_> [TK]D-Fender The numbered extensions relate to different users  on different days, in my dial plan.
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04:12.35kc8pxyhmmm
04:13.19Wired_Life1i go sleep thx [TK]D-Fender and termas_ ill be back
04:13.40kc8pxytermas_:  i thougth i just read how you can do that in the extensions.conf.
04:13.41termas_Night,
04:14.42termas_kc8pxy You can do that in extensions.conf, you can ODBC, or natively connect to the database, pull the relivant username and dial it.
04:16.33[TK]D-Fenderyup
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04:18.34Wired_Life1http://pastebin.com/d67196c2d one and last
04:19.49[TK]D-FenderWired_Life1: You keep showing us dialplan and not telling us what probelm if any you are actually HAVING with it
04:19.57[TK]D-FenderWired_Life1: Are we supposed to GUESS?
04:20.34[TK]D-FenderWired_Life1: here's one : exten => s,n(yes),SetVar(MACRO_RESULT=CONTINUE) <- in 1.4+ SetVar is permanently gone and replaced by "Set".  Another good reason to think twice before copy/paste-ing code from the WIKI
04:20.43[TK]D-FenderWired_Life1: Its often outdated crap
04:22.36[TK]D-FenderWired_Life1:  exten = r,1,Dial(SIP/oli,20,M(test)) exten = r,2,GotoIf($["${ISOK}" = "TRUE"]?10:20)<- and this is still crap.  You dial him once with a broken macro and expect to call him AGAIN after
04:23.04[TK]D-FenderWired_Life1: r,1 vs r,13
04:26.02Wired_Life1exten = s,1,Set(ISOK=TRUE) ; this is only set if oli answers and this making costs for anonymous i dont want
04:26.33[TK]D-FenderWired_Life1: it is set and the call IS COMPLETELY answered.  Your 2nd dial is WORTHLESS
04:26.41*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
04:26.50Wired_Life1thats my problem
04:26.55[TK]D-FenderWired_Life1: and the call is bridged immediately.  "oli" is never asked to confirm to accept the call or not.
04:27.07*** join/#asterisk frk2 (n=frk2@zivios/member/fkhan)
04:27.28[TK]D-FenderWired_Life1: the are 2 DIAL's in your "r" exten.  WHY?  You left broken crap in there and just pasted from the WIKI after.
04:28.00[TK]D-FenderWired_Life1: You aren't testing anything with your first dial macro, so you might as well not have even done it
04:28.27Wired_Life1i want to check if oli is there before calling he @ r,13
04:29.19[TK]D-FenderR.13 is the dial that DOES THE *#&$ING CHECK
04:29.29[TK]D-FenderWired_Life1: r,1 checks NOTHING
04:30.23Wired_Life1r,13 do the check with answering my question is how make a check without making costs
04:30.34[TK]D-FenderWired_Life1: Wired_Life1 and you can't if someone is there BEFORE CALLING.
04:32.12[TK]D-FenderWired_Life1: WTF is "sip/oli"?
04:32.27Wired_Life1my softphone
04:32.32termas_SIP/oli would be the user "oli".
04:32.58[TK]D-FenderWired_Life1: If you call it and it doesn't answer, how is your caller "answered"?
04:33.11[TK]D-FenderWired_Life1: I don't see it answering if you IGNORE the call.
04:33.15[TK]D-FenderWired_Life1: So what is your problem?
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04:34.18[TK]D-FenderWired_Life1: Doesn't matter if you fail to answer.  Doesn't matter if the softphone is even contactable.  niether will "answer" the caller
04:34.28[TK]D-FenderWired_Life1: This is avery large waste of time
04:35.34Wired_Life1i want a signal @ my softphone if someone is calling anonymous and then i say ok and then the call go answered and play woman voice
04:35.52termas_Busy() and Congestion() signal the Caller,
04:36.01termas_everything signals the caller,
04:36.38Wired_Life1i want to signal the end point oli
04:37.05Wired_Life1if he said ok then the call go answered
04:37.09termas_You can't signal the end point,
04:37.47Wired_Life1i cant signal the end point without making costs? why?
04:38.07[TK]D-FenderWired_Life1: you're calling a soft-phone... at what point are you TALKING to them?
04:38.41Wired_Life1can i make only with dial?
04:38.59[TK]D-FenderWired_Life1: you're calling a soft-phone... at what point are you TALKING to them? <--------
04:38.59termas_You can dial and add headers,
04:40.39Wired_Life1plz say only if my idea should work or not
04:41.11[TK]D-FenderWired_Life1: you're calling a soft-phone... at what point are you TALKING to them? <--------
04:42.15Wired_Life1only when i make a call or?
04:42.29leifmadsenyo!
04:42.40termas_Here's what I'd do, I have snom hard phones. When a call comes in, i'd take the call, check it for callerid, if it's good, Dial(SIP/oli), if it's anon, have asterisk answer it, n,background(woman-you-are-anon)
04:43.04termas_record() then Dial them into a holding extension.
04:43.06[TK]D-FenderWired_Life1: you're making a fucking softphone RING. when are these 2 fucking people supposed to actually START TALKING TOGETHER?
04:43.38leifmadsen[TK]D-Fender: never!
04:43.38termas_the snom phone would be configured with a BLF, so a little light comes on, when oli pushes it, he gets the recording, and if it's ok, he can push 1 to bridge the call.
04:43.44*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
04:43.47[TK]D-Fendergoes off to do something productive
04:44.29Wired_Life1termas_ but in this case the call are already answered and time is running and making costs
04:44.34[TK]D-Fenderleifmadsen: Another idiot inexplicably trying to use a fork as a hammer when he really needs TOILET PAPER
04:44.55leifmadsen[TK]D-Fender: no comprende senior
04:45.02termas_You must answer the call.
04:45.06[TK]D-Fenderleifmadsen: thats the problem.
04:45.15leifmadsen[TK]D-Fender: :)
04:45.23leifmadsenI don't want to meet your mom! I just want...
04:45.27termas_Since when do you get charged for receiving a call anyway?
04:45.36[TK]D-Fenderleifmadsen: its the "lets jsut shove apps in, dial shit, and wait, WTF does this have to do with my goal?"
04:45.40leifmadsentermas_: forever.
04:45.51[TK]D-Fenderleifmadsen: Not tonight... I have a headache :p
04:46.01leifmadsen[TK]D-Fender: booooooooooooooooooooooooooooooooooo
04:46.04termas_So, anyway..
04:46.08[TK]D-Fender~!!!
04:46.11infoboti heard !!! is BANG BANG BANG at http://www.starterupsteve.com/swf/Group_X_video.html
04:46.11leifmadsencalls his girlfriend
04:47.45termas_Can anyone explain what this debug output means: http://pastebin.com/d6cf10fa7
04:48.57Wired_Life1oh man you are really sure there is no way to ask my softphone before the recording stuff?
04:49.14leifmadsenask your softphone what?
04:49.56[TK]D-FenderWired_Life1: How many times do we have to say it.  If the macro says to reject the call then it is NOT ANSWERED
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04:51.03jblackUnless he has an answer() before the dial...
04:51.33[TK]D-Fenderjblack: Which we don't see as he has not even shown a TEST CALL or the actual problem HAPPENING
04:52.00[TK]D-Fenderjblack: Just keeps PB-ing broken code he doesn't understand
04:52.01jblackwhy ARE we SHOUTING so MUCH?
04:52.01Wired_Life1you mean exten => 890,n,Record(${SCREEN_FILE}.gsm|6|25) is not answering the call from anonymous?
04:52.03jblack:)
04:52.15leifmadsenjblack: apparently you haven't met [TK]D-Fender
04:52.21[TK]D-FenderWired_Life1: Why are you asking them to record their name?
04:52.32jblackheh. Good point.
04:52.46[TK]D-Fenderjblack: Bloody hell you havent :p
04:53.20drmessanowhat THE eff
04:53.30Pan3DWired_Life1: it's like this... the call has to have been answered (as in audio flowing) for the record to do anything
04:53.37Wired_Life1because the callerid is anonymous and i cant see the number?
04:53.40drmessanoI want TO empasize THE wrong works, ok?
04:53.44drmessanowords too
04:53.46leifmadsens/THE/TEH/g
04:53.48[TK]D-FenderWired_Life1: exten = r,11,Playback(screen-record) <- THIS will answer the call.  Why are you doing it?
04:53.55jblack[TK]D-Fender: Some day, your heart is gonna explode. And if _it_ holds out, your head is gonna pop off like a cork. Sure you're not american?
04:54.14drmessanoleifmadsen: Sorry, i'm not norse
04:54.28leifmadsendrmessano: well fuck you then :)
04:54.28Wired_Life1screen-record: Please record your name press pound when finished.
04:54.40[TK]D-Fenderjblack: Yes, I'm aiming for tactical-nuclear capacity and then I'm going to pay a visit to his town....
04:54.53drmessanojblack: I think IRC is KEEPING him FROM shooting up A McDonalds.. So let it be
04:55.02Pan3Dlol
04:55.22[TK]D-FenderWired_Life1: You answered the call to ask their name.  YOU.  If you don't want to answer the call, WTF are you doing playing audio and trying to record a damn name?
04:55.36[TK]D-Fenderlooks areound for a water tower
04:55.50drmessano[TK]D-Fender: Sounds like some awful spammy project
04:55.56jblack[TK]D-Fender: That's gonna be messy.
04:55.59Wired_Life1i want this after i have said: "ok its ok"
04:56.09drmessano"If you do not wish to accept this call, give your name and hang up now"
04:56.13drmessanoTelespammer
04:56.27drmessanoIm calling shenanigans
04:56.38[TK]D-Fenderdrmessano: this appears to be inbound
04:56.52Pan3Dit's inbound judging by the pb
04:56.58drmessano[TK]D-Fender: Im calling Nihilism
04:56.59[TK]D-FenderWired_Life1: Look when you're asking.  You don't get to make this a 2-stage approval process.
04:57.25[TK]D-FenderWired_Life1: and what is Oli supposed to base his decision on concerning whether or not to take the call?
04:57.25drmessanoMaybe he just likes pain
04:57.34Pan3Dha
04:57.37jblackHeh. Like a cheap robotic secretary that always has to ask whether or not he wants to take the call.
04:57.39drmessanoNihilomasochism?
04:57.58Wired_Life1my english is not so good
04:58.15drmessanoNeither is mine
04:58.17Wired_Life1and i dont find good german support
04:58.29[TK]D-FenderNarcochism = Wanting all the pain to be about you.
04:58.33drmessanoSomeone called me a douchebag and I thought it meant "Guy with tight pants"
04:59.27[TK]D-FenderWired_Life1: You can't think in a straight line.  thats your porblem.  You can't even describe steps in order.
05:00.50*** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net)
05:00.50drmessanoONEEEEE
05:00.52drmessanoTWOOOOOO
05:00.55drmessanoFOUR
05:01.27Wired_Life1ok... if a call comes from anonymous i want to ring on my softphone if i want to accept and then i will make a record of callers voice
05:02.48Wired_Life1of course only if i said ok
05:02.56Wired_Life1you know what i mean?
05:04.18[TK]D-FenderwireWHY would you "accept"?
05:05.06Wired_Life1i can only accept if i on my pc
05:05.28[TK]D-FenderWired_Life1: Again, WHY would you "accept"?
05:06.41Wired_Life1why? because some people call me without callerid(num)
05:08.33[TK]D-FenderWired_Life1: So they have no callerid.  What makes you decide to ACCEPT TEH CALL OR NOT?
05:09.08termas_[TK]D-Fender That's the problem, he doesn't know. He wants to signal the phone, in advance
05:09.18drmessanoBased on WHAT?
05:09.25termas_ie, pass the callerid to the end point.
05:09.26[TK]D-Fendertermas_: no...
05:09.27drmessanoIm lost
05:09.40[TK]D-Fenderdrmessano: You're not alone, and you're not the first
05:09.41termas_He wants the end point to display that the call is anon.
05:10.07[TK]D-FenderBut what fucking human decision is being made on whether to accept it or not?
05:10.21[TK]D-Fender"Or its TUESDAY, why the fuck not!"
05:10.54termas_Well, the idea, (this is what I think he means) is that prior to the call being sent to Oli, the caller is recorded.
05:11.14[TK]D-Fendertermas_: No, that will ANSWER the call <-
05:11.22[TK]D-Fenderter#1 violation.
05:11.25termas_but, Yeah,
05:11.38[TK]D-Fendertermas_: No "buts"
05:11.39termas_It will answer the call, you can't record audio from a hung up channel.
05:11.47termas_but it doesn't go to end point.
05:11.49Wired_Life1i know that
05:12.04Wired_Life1oh man you still dont know what i mean
05:12.15[TK]D-FenderWired_Life1: WHY would you choose to "accept" this caller?
05:12.58Wired_Life1i will know whos behind
05:13.15[TK]D-FenderWired_Life1: How will you know?  THEY HAVE TO callerid!
05:13.18[TK]D-Fenderno*
05:13.55Wired_Life1asterisk will record a voice sample from the person?
05:14.19[TK]D-FenderWired_Life1: THAT WILL ANSWER THE FUCKING CALL!
05:14.37termas_It will answer the call, but you can drop them into hold music..
05:14.38[TK]D-FenderWired_Life1: How can you record from the caller if you haven't answered?
05:14.50Wired_Life1right i will that make asterisk answer the call after i said ok
05:14.59[TK]D-FenderWired_Life1: NO
05:15.11Wired_Life1why
05:15.18termas_Wired_Life1 by the time the call reaches the end point, it's too late.
05:15.31[TK]D-FenderWired_Life1: You said you don't want to answer until you accept.  You accept based on the recordin.  in order to record THE CALL HAS ALREADY BEEN FUCKING ANSWERED
05:16.35termas_Within 30 secs 69 calls +10 61.61 %
05:16.35termas_Within 45 secs 77 calls +8 68.75 %
05:16.35termas_Within 60 secs 84 calls +7 75.00 %
05:16.35termas_Within 75 secs 91 calls +7 81.25 %
05:16.35termas_Within 90 secs 96 calls +5 85.71 %
05:16.37termas_Within 91+ secs 112 calls +16 100.00 %
05:16.40[TK]D-FenderWired_Life1: you can't record the name and THEN answer.  its too damn late.
05:16.43termas_gack.
05:16.58Wired_Life1i know
05:17.15Wired_Life1you said the playback is answering
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05:18.49termas_So, watching [TK]D-Fender working his way to an aneruysm is fun and all, but I'm not getting any closer to my solution..
05:18.53[TK]D-FenderWired_Life1: Seriously... how retarded are you.  You can't play a prompt asking for a name, record it, and THEN think "Oh I don't want it to count as answered yet!"
05:19.45[TK]D-FenderWired_Life1: You seem to have several key problems with things like "temporal mechanics", and "linear progression"
05:20.01[TK]D-FenderWired_Life1: And "logic" and "stuff"
05:20.08Wired_Life1nononono
05:20.13Wired_Life1hear
05:20.18[TK]D-FenderOh very certinly "yes"
05:20.28Wired_Life1after i say its ok i can make costs
05:20.34Wired_Life1it
05:20.47[TK]D-FenderWired_Life1: You have no REASON to make it "ok" since you can't record without CAUSING COSTS
05:20.54[TK]D-FenderWired_Life1: You can't go BACK IN TIME
05:21.00termas_I can go back in time.
05:21.16termas_I feel like i've been going forward and back in time ever since this conversation started.
05:21.19[TK]D-Fenderknocks termas_ back into last tuesday
05:21.40Wired_Life1thats it why i have make 2 dials
05:21.40termas_1. State problem. 2. Listen to flaws with concept, 1. State problem.
05:22.19[TK]D-FenderWired_Life1: NO
05:22.20Wired_Life11 dial to me to check if ok and then the answer with record that makes costs
05:22.29[TK]D-FenderWired_Life1: You have answered the call.  You are FUCKED
05:22.53[TK]D-FenderWired_Life1: You have no REASON to accept the call.
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05:23.25[TK]D-FenderWired_Life1: Because you can';t get the name without answering.  Answering automatically costs you.  You can't chang your mind afterwards!
05:23.56[TK]D-FenderWired_Life1: And you have nothing to base a decision to answer on without taking a name or something.
05:24.03Wired_Life1i know you said serveral times
05:24.10[TK]D-FenderWired_Life1: then what don't you get?
05:24.57Wired_Life1i want to check first if iam there
05:25.20[TK]D-FenderWired_Life1: And if you are?
05:25.43Wired_Life1i click ok anywhere
05:25.53Wired_Life1and if not the call runs into timeout
05:26.15[TK]D-FenderWired_Life1: there is no fucking click.  You're on a softphone.  You can answer, you can press digits.  There is no "OK"
05:26.43[TK]D-FenderWired_Life1: If you're there are you always going to pick up the phone?
05:27.24HoosierDaddyany Broadvoice customers here?
05:28.32Wired_Life1if im on the pc i will pick up
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05:29.24[TK]D-FenderWired_Life1: Then what?
05:29.34Wired_Life1the thing is i will only make costs for the caller if i there to call
05:29.44Wired_Life1you know what i mean?
05:30.08Wired_Life1its fucking shit if i record and let the caller wait and im not there
05:32.05[TK]D-FenderWired_Life1: Now you are only doing this for anonymous calls.  So if they aren't anonymous then you are happy to immediately charge them?
05:33.59Wired_Life1if they aren't anonymous the call go directly to me
05:34.29[TK]D-FenderWired_Life1: they BOTH go to you.  How is anonymous "special"?
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05:35.45Wired_Life1anonymous can record voice after a trigger from me so i can play the record and know who is there
05:36.17Wired_Life1are you still dont know what i will make?
05:37.23[TK]D-FenderWired_Life1: Why aren't you checking if you're there for NON anonymous callers?
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05:37.52[TK]D-FenderWired_Life1: Why are only the anonymous ones getting the chance to be ignored by you at no cost?
05:38.19Wired_Life1^^ you are funny
05:38.33Wired_Life1because it is so ^^
05:39.15kc8pxy[TK]D-Fender: because he only wants to have it bother with anonymous "spamers"  or bill collectors :)
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05:39.45[TK]D-Fenderkc8pxy: you are missing the fact they get a BENIFIT from this for no good reason
05:40.22[TK]D-FenderAnyways, enough bacwards logic and broken descptions for one (lifetime) day.
05:40.23[TK]D-FenderI'm off.
05:40.29kc8pxy[TK]D-Fender: mebbe I'm only reading part of the conversation.
05:41.42Wired_Life1i know what he mean
05:42.02Wired_Life1its not possible to check who is there without making costs
05:42.09Wired_Life1lol
05:44.59*** part/#asterisk Tanker1 (n=lromano@216.115.201.131)
05:52.17kc8pxyi apologize if I'm even more noob that i seem,  but i tried to get my 1.6.0.9 server up, with a single sip channel definition, and i can't seem to get ekiga to register to it.  from all the howto's I've read I'm doing things right.
05:52.34kc8pxybut my asterisk does not seem to be listening..  anyone willing to help?
05:53.22*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
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06:05.01kc8pxyadmittedly,  this is my very first attempt at makign asterisk work.  anyone?
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06:05.42HoosierDaddyI'm no expert either...do you now how to look for sip registration errors?
06:06.35kc8pxyHoosierDaddy:  not really..   asterisk -vvv only complains that i have no modules.conf or features.conf
06:07.37kc8pxyHoosierDaddy:  event_log is empty
06:08.34HoosierDaddyasterisk -rvvvvvv
06:08.47HoosierDaddysip show peers
06:09.37HoosierDaddyyour asterisk box is on your local subnet I presume too
06:09.39termas_kc8pxy iptables?
06:10.34kc8pxyNo such command 'sip show peers' (type 'help sip show' for other possible commands)
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06:11.13kc8pxytermas_: don't think either client or server have any rules, and ther is just a switch between them.
06:12.05wierdokc8pxy, If modules.conf is not loaded and chan_sip, you cannot sip show peers
06:12.06termas_1.6.0.9 is much newer than mine, sip show peers works for me..
06:12.47HoosierDaddyfrom * CLI, "sip show ?" doesn't list peers as an option?
06:13.04kc8pxywierdo: so i need load chan_sip at a minimum?
06:13.57HoosierDaddypeers
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06:14.19wierdokc8pxy, yes.  module load chan_sip.so
06:14.35wierdonot sure if this is the correct syntax in 1.6.x
06:15.30orangeserviceHi all, I have two SIP trunks to two different gateways... I am trying to make it so that any inbound call on either of them gets through to the same set of extensions on my gateway... anybody have experience with this?
06:16.15wierdokc8pxy,  ls /usr/lib/asterisk/modules/
06:16.21wierdois anybody there :)
06:16.39wierdoor just no modules.conf, or empty, check permissions
06:16.43*** join/#asterisk keebler (n=Christop@h36.247.20.98.dynamic.ip.windstream.net)
06:16.57keeblerAnyone try using Gizmo5 Business with Asterisk?
06:17.40termas_orangeservice set both the initial contexts to the same context?
06:17.56termas_orangeservice, or set the initial context of one, to goto the other..
06:18.41kc8pxygot it
06:18.57orangeservicetermas_: I tried that but strangely enough having them both sent to same context doesnt work... the context works for one provider, but not the other (that being said, I am quite new to asterisk and telephony in general)
06:19.06kc8pxysip show peers now shows my sip peer, but it's nto connected..  trying now.
06:19.26kc8pxyYES!
06:20.07kc8pxyat least i have a registered peer :)    from here,  everything is just how to fold the origami,  yes?
06:20.13termas_can you do anything with the trunk that doesn't work?
06:20.51HoosierDaddygood deal kc8pxy
06:20.54termas_yes kc8pxy, but you should probably look into your iptables.. no rules is odd.
06:21.27orangeserviceI can make outbound calls via it - it does have to be said though, the trunks are quite different in the way they authenticate and so on
06:21.38drmessanokeebler:  I have.. works fine
06:21.51kc8pxytermas_: nothing needed on the lan,  it's my house. i have a beefy gateway with paranoid rules.
06:21.55keeblerdrmessano: How well does extension calling work?
06:22.01keeblerdrmessano: From a mobile phone?
06:22.14drmessanoFrom a Blackberry, shitty
06:22.17termas_kc8pxy oh, my mistake.
06:22.23keeblerdrmessano: Damn
06:22.24drmessanoNot sure about Windows Mobile
06:22.32keeblerI only have blackberries
06:22.40drmessanoIts not a SIP client
06:22.40keebler8110
06:22.48keeblerNot true sip?
06:22.50keeblergrr
06:23.03drmessanoIt just initiates call bridging
06:23.07keeblerRIM BB_MVS is too fucking expensive.
06:23.11drmessanoThere is no SIP client for the BB
06:23.14keeblerYes
06:23.15keeblerthere is
06:23.16keebler:/
06:23.25keeblerBut its only through RIM
06:23.41drmessanoThats not a SIP client.. thats native OS support
06:23.45keeblerAnd they say it won't work with Asterisk.
06:23.54drmessanoYou cannot download and install a client
06:24.10keeblerYeah, thats what I'm discovering.
06:24.11kc8pxykeebler: intentionally, or ommisively?
06:25.09keeblerkc8pxy: Said they tried testing it, but the  configurations can't be standardized for every client, and there are too many factors that would cause it to work improperly, and they said they couldn't get the ATA;s to register properly for extension calling
06:25.16keeblerkc8pxy: Talked to their project lead today.
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06:25.33kc8pxy_X. matches to "press anything, right?
06:25.59kc8pxyrather, "pick a number, any number!"
06:26.01keeblerafaik
06:26.29keeblerI haven't worked on my asterisk server in 2 months.
06:26.35keeblerAlready started to forget crap.
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06:27.17keeblerdrmessano: Can you recommend any GSM/GPRS Phone that properly supports SIP?
06:27.23keeblerWe have to use ATT.
06:27.40keeblerI can change from BB if I have to.
06:28.06drmessanoNot sure
06:28.18keeblerThe clients just need to be able to dial an extension from the phone properly.
06:28.26keeblerwith no diminished call quality.
06:28.33keeblerAnd still maintain its own extension
06:29.12keeblerhttp://na.blackberry.com/eng/services/blackberry_mvs
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06:29.44keeblerIts its only stand alone software os that works in conjunction with the BES and PBX
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06:29.59keeblerSo theoretically you'd need three servers.
06:30.05keeblerBES/MVS/PBX
06:30.30keeblerAnd they charge you $8K JUST to install the system.
06:30.37keeblerWon't even let you manage it on your own.
06:30.57aesiamunhi, is it possible to get Asterisk to dial two numbers at the same time but set the callerid differently for each one?  I'm using the system to ring internally to sip phones and to my cell phone.  The problem is that i can't control callerid text on my cell.  So I would like to use a coding system in my address book.
06:31.36kc8pxyhttp://rafb.net/p/bj2Qr310.html
06:31.54kc8pxywill that be the nice test i hope it will be?
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06:35.32aesiamuni'd be interested in knowing if anyone has been able to do that before.
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06:38.51kc8pxyWHEE!!!  a split!
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06:39.42k-mananyone here use pennytel?
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06:50.57k-mananyone know where i might find documentation on how to write diaplans for a linksys spa2102?
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06:59.21kc8pxyk-man: wish i knew..  I'm just trying to get my softphone to get noises played at it.
06:59.52kc8pxyfor Playback(), do i need any specific modules?
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07:07.31k-mankc8pxy: not afaik, do you get any sound or error messages when you try and play a sound?
07:07.47k-mankc8pxy: its very probably permissions. thats the problem i had when i was trying to do it
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07:14.12k-manis there some tool i can use to ping a host every 30 minutes to see if its reachable?
07:16.36kc8pxyk-man: cron?
07:18.31k-manoh - i meant on my windows mahcine
07:18.33k-manmachine
07:18.45kc8pxyk-man: all i see on teh sip sotfphone is dialing sip:1 , and then i get it disconnected.    asterisk -rvvvvvv shows the following, every time i try to dial that.
07:19.04kc8pxy<PROTECTED>
07:19.41k-mankc8pxy: sorry, no idea
07:20.09k-mankc8pxy: you'll have to post your extensions.conf and sip.conf and a log of what happens when you try and dial
07:20.19k-mankc8pxy: but I haev to go now so I can't help you
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07:29.25wierdok-man, you can use dude: http://www.mikrotik.com/thedude.php
07:29.26kc8pxyok,  so now i have beeping going on.
07:29.55kc8pxybut tt-mokeys.gsm should not sound like beeping,  should it?
07:30.05kc8pxymonkeys.gsm
07:30.09wierdofree tool, does some advanced monitorin,can ran as service, does ping at some period if configured, pretty simple and cool
07:32.13kc8pxywierdo:  ideas?
07:34.14wierdokc8pxy, let me read some posts up
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07:35.57wierdokc8pxy, sorry, could you repeat the problem
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07:39.37kc8pxywierdo:  i can get my sip softphone to connect, and i can dial a number, but I'm not getting what I'm expecting.    it seems to get hung up on before i hear monkeys.  repost of extensions.conf comming.
07:40.28wierdokc8pxy, incopatible codecs ?
07:40.57kc8pxywierdo:  I'm too noob to know.
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07:42.19wierdokc8pxy,  pastebin sip.conf for softphone, what softphone are you using
07:43.16kc8pxywierdo:  ekiga
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07:44.01kc8pxyhttp://rafb.net/p/afGrlO88.html is my extensions
07:44.54orangeservicehi, having a strange one - I am using WaitExten() to grab user input... when asterisk tries to place the call to the exten, ties extention@inbound-context, not extention@local-ext-context like I want it to.... how would I tell WaitExten to use the numbers in a local context?
07:45.14kc8pxywierdo: http://rafb.net/p/AxiAw511.html is sip.conf
07:47.39kc8pxywierdo:  that help?
07:48.15wierdoallow=alaw
07:48.15wierdoallow=ulaw
07:48.49mostyorangeservice, pastebin your dialplan and sip.conf (assuming your local extensions are sip)
07:49.13kc8pxyDOH
07:49.20wierdokc8pxy, add these to sip.conf
07:49.34kc8pxyi remove the path,  and just say tt-monkeys, and it works :-/
07:49.45kc8pxyfrom the playback
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07:50.54kc8pxyyay me :0
07:50.56kc8pxy:)
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08:00.58wierdokc8pxy, i have configured it working ekiga, asterisk, no problem
08:01.21wierdokc8pxy, ulaw, alaw are configured by default on the softphone
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08:04.36wierdokc8pxy, in configuration druid what type of connection you've specified
08:06.31wierdokc8pxy, maybe if setup as default 56k, ekiga switch to some lossy codec which is not allowed in sip.conf
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08:32.19jgooIs it possible to setup a call group so that people have to dial the extension to take the call (and just picking up won't accept the call)
08:32.24jgooor is that the definition of another feature?
08:33.19jgooI was thinking to use a queue for this, does that sounds more applicable?
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08:46.39mostywith a queue, you could dial an extension to join the queue
08:47.19mostyyou would then be able to receive calls from the queue
08:47.53jgoomosty: I would like that behaviour but without having to dial an extension to join
08:48.08Chris-NBhi
08:48.09jgoohowever, I want the extension to ring, and let people 'pickup' the extenion.
08:48.27mostyjgoo, you can make sip extensions static members of a queue
08:48.39jgooaaah. right. so queue is the right way to do that then?
08:48.49jgooand they will see the queue extension flashing on their phone, press it to take another call?
08:49.41Chris-NBanyone using the 'd' option in the dial application?
08:49.44jgooI suppose I would also want one actual phone to ring, but just one, then after 20 seconds, pass to another queue or call group that rings all.
08:49.52jgooany ideas on what the setup would be?
08:50.13Chris-NBto dial a 1 digit extension during a call is beeing established,  to exit to that extension?
08:51.06mostyjgoo, find an asterisk queue tutorial/guide- perhaps there is one in the book
08:51.08mosty~thebook
08:51.08infobotfrom memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org
08:52.09Chris-NBI've configured the EXITCONTEXT variable and dtmf is transmitted correctly to the server
08:52.27Chris-NBbut the server ignores my key press during an call establishment
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09:55.27orangeservicemosty - sorry for the delay, manage to solve it... now left with another pickle. WaitExten is only registering the 1st digit of extentions dialed.... booo, anybody seen thins?
09:56.44mostylike i said before, pastebin
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10:10.53orangeserviceabbreviated sip.conf - http://pastebin.com/d6b8c42aa
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10:13.13orangeservice@ mosty: and extensions.conf http://pastebin.com/d6cca6c24
10:16.33orangeserviceand the faily: http://pastebin.com/d24f48571
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10:44.12mostyorangeservice, what extension are you trying to dial?
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11:08.25timater123can anybody help me, i have an issue with dahdi
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11:10.15timater123i setup an asterisk exchange on centos 5.3, the problem is that i have to run dahdi_genconf each time i boot my asterisk server otherwise my analog phones wont work
11:11.55defsdoortimater123: sounds like you need some modules loading
11:13.31timater123defsdoor: i commented out every module except wctdm because i have a tdm400p card
11:14.11defsdoortimater123: reboot - do lsmod > ~/modlist
11:14.20defsdoorrun dahdi-genconf
11:14.28defsdoordo lsmod > ~/modlist2
11:14.32defsdoordiff modlist modlist2
11:14.52defsdoorI'm assuming dahdi-genconf is loading some modules
11:15.02defsdoorseems most likely reason
11:16.04timater123diff returned no output
11:17.09defsdoordahdi_genconf just writes a config file (I'm looking at it now)
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11:17.21defsdoorso unless your config file is being clobbered - I'm clueless
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11:23.14timater123defsdoor: are you talking about /etc/dahdi/system.conf
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11:30.04timater123<PROTECTED>
11:30.49timater123defsdoor: modlist file gives a difference when i restart asterisk after dahdi_genconf
11:31.02defsdoorwhat modules ?
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11:32.48timater123diff modlist modlist3
11:32.48timater1235c5
11:32.48timater123< wctdm                  39884  2
11:32.48timater123---
11:32.48timater123> wctdm                  39884  4
11:32.49timater1237c7
11:32.51timater123< dahdi                 190728  9 dahdi_echocan_mg2,wctdm,wcfxo
11:32.53timater123---
11:32.57timater123> dahdi                 190728  13 dahdi_echocan_mg2,wctdm,wcfxo
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11:40.37tmjbhello my asterisk is crashing on ubuntu amd 64 asterisk[28472]: segfault at 31 ip 00007fcb7c50d1bd sp 0000000040cec9b0 error 4 in libpri.so.1.4[7fcb7c505000+2f000] ?
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11:51.18tmjbU use stable asterisk 1.6.0.8
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12:03.16leifmadsentmjb: have you tried latest 1.6.0 branch? Are you able to reproduce it? Also, make sure you have DONT_OPTIMIZE enabled in menuselect under the Compiler Flags. Once you've done all that, you can open a bug on the tracker at bugs.digium.com
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12:16.07thedonvaughn~gsm-bug
12:16.26thedonvaughngsm bug still exist with asterisk and gcc-4.2 or greater?
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12:26.42JakobsenHi there. Can anyone point me to a good guide, that describes how to create a High Availability Asterisk cluster with two servers?
12:26.59leifmadsenJakobsen: I don't know of any document for that
12:27.27Jakobsenleifmadsen, same conclusion here :)
12:27.42leifmadsenI doubt such a document exists; what do you expect something like that to have?
12:28.21[TK]D-FenderJakobsen: You'll find it filed somewhere between "Invisible flying unicorns" and Jimmy Hoffa ;)
12:29.15leifmadsenthedonvaughn: http://bugs.digium.com/view.php?id=13846
12:29.40JakobsenWell.. I'll just create a Heartbeat and DRBD cluster with Asterisk on top of it then :)
12:29.51leifmadsenJakobsen: heh, that's pretty much the way to do it
12:29.56JakobsenThat should do the trick.. Just wanted to know, if there were any easier way..
12:30.03JakobsenThank you for helping anyways.. :)
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12:41.53*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
12:42.40Dr-Linux|homeHi guys,
12:43.48Dr-Linux|homewhy my pattren doesn't work with + sign? i.e. exen => _+1NXXXXXXXXX,1,Goto(ivr,s,1)
12:43.49Dr-Linux|home?
12:44.12[TK]D-FenderDr-Linux|home: show us
12:44.38thedonvaughnis there a way to unregister a sip provider through the cli?
12:45.26Dr-Linux|home[TK]D-Fender: I did
12:45.37Dr-Linux|home<PROTECTED>
12:45.45[TK]D-FenderDr-Linux|home: No you didn't.  Show a complete dialplan and failed call attempt
12:46.07[TK]D-FenderDr-Linux|home: You aren't showing a failure, jsut a single unvalidated line that we should not take at face value
12:46.48Dr-Linux|home[TK]D-Fender: sure, let me try access again on that server and reproduce the error
12:47.14[TK]D-FenderDr-Linux|home: And you appear to be hand-typing that line as well given the fact the spelling on "exten" changed between each.
12:47.33Dr-Linux|home[TK]D-Fender: BTW, generally this line is okey? the one i shown?
12:47.49Dr-Linux|homeyeah sorry for that
12:48.46Dr-Linux|home[TK]D-Fender: actually I just wanted to confirm that _+1NXXXXXXXXX is right as it is using + sign
12:54.07mmlj4how do you dial a plus sign?
12:55.01[TK]D-FenderDr-Linux|home: You're making claims.  Back it up
12:55.35[TK]D-Fendermmlj4: Think "Outside NANPA"
12:56.00mmlj4fair enough
12:57.49*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
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13:00.32Dr-Linux|home<PROTECTED>
13:01.18Dr-Linux|home[TK]D-Fender: the problem is  _+1NXXXXXXXXX is on top but it when i call it ignored
13:02.00Dr-Linux|homeand it goes below and pick the hard coded number
13:02.06*** join/#asterisk Urthwhyte (n=urthwhyt@195.249.182.253)
13:02.12[TK]D-FenderDr-Linux|home: pastebin EVERYTHING.
13:02.28Dr-Linux|homeok
13:02.30*** join/#asterisk kaii (n=kai@ciphron.de)
13:02.38leifmadsenDr-Linux|home: uhhh.... hard coded is always going to match first
13:02.46kaiimh .. when SIP/10 dials "SIP/11&SIP/12&SIP/13" all of these three peers go into ringing state (given they are not busy or unreachable).   does anybody can point me how to obtain this information?  (via AMI, CLI, custom app, whatsoever)
13:02.46leifmadsenmost specific will get matched before a pattern match
13:02.52kaiii need to check somehow if a channel is ringing on more than 1 peer ..
13:02.56kaiiin a stupid "ringall" setup the customer now wants to put priority on transfers and internal calls .. means making internal calls/transfers possible, even if the target is in ringing state with the external ringgroup
13:03.25leifmadsenkaii: what version?
13:03.52kaiiyou dont want to know.. 1.2-bristuffed ...
13:04.06leifmadsenkaii: not sure you will have any method then
13:04.08kaiii am able to patch/backport, if necessary.
13:04.32Jas_WilliamsDr-Linux|home: If you do a show dialplan context it will show you the evaluation order
13:04.53Jas_Williamsfrom cli
13:05.15kaiileifmadsen: extstate for example can tell me if an extension is ringing or not .. but it does not resolve how many channels are actually ringing
13:05.56Dr-Linux|homeJas_Williams: nice idea
13:05.58leifmadsenkaii: you could set a variable that contained that info
13:06.57kaiileifmadsen: how?    if i Dial(SIP/11&SIP/12&SIP/13) and two of them are busy/congested ...  how do i detect this during Dial() is executed?
13:07.43*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:09.20Dr-Linux|homeJas_Williams: CLI logs are too fast due to heavy traffic :) also i've lot of self defined conf files i.e. did.conf tf.conf etc ..
13:09.27[TK]D-Fenderkaii: "core show channels concise" <- get parsing
13:09.29kaiileifmadsen: the external caller goes into Dial() which results in "n" new channels in ringing state ... and after that dial is executed its to late to set the variable ..  only app_dial could do this
13:10.29[TK]D-Fenderkaii: then don't ring your devices directly, do it vial a Local channel
13:10.48[TK]D-Fenderv/vial/via/
13:10.56[TK]D-Fenders/vial/via/
13:10.59[TK]D-Fendersdhshffdsfhbfdg
13:11.36kaiiin the local channel i can increment, dial, and decrement a global variable. ok .. but there is still a race condition for the time dial() needs to detect a busy/congested channel and jump to the next (decrementing) priority
13:11.43mmlj4v/ ?
13:12.58[TK]D-Fenderkaii: And I gave you another option already
13:13.38kaiiim still trying to find out what "show channels concise" does and if i somehow can achieve this on asterisk 1.2   ... i'm still bound to that old stuff
13:14.25[TK]D-Fenderkaii: A CLI command, it ain't Raw-Cat Science.
13:14.30kaiiok 1.2 does support consise too.
13:15.15[TK]D-Fendernever would have guessed...
13:15.30kaii[TK]D-Fender: seems this could work .. thx for pointing
13:16.26*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
13:16.32tamielhello, I need to set sip and iax users/peers in postgresql tables. I must use res_odbc or res_config_pgsql ?
13:16.43Kattydrags in
13:16.53[TK]D-FenderKatty: Mew.
13:17.12Kattyhugs [TK]D-Fender and falls asleep mid-hug
13:18.31Kattyi need sleep.
13:19.37*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
13:19.41Jas_WilliamsDr-Linux|home: how about calling just the command eg asterisk -x "core show dialplan context"
13:19.51kaiii need the missing link between channels in "show channels [concise]"
13:20.03Dr-Linux|homeJas_Williams: yes I got that
13:20.29Jas_Williamsthen will display to your shell session rather than in logs
13:20.52Dr-Linux|homeJas_Williams: it is showing my desired line at the end, so this line will be executed as last option:
13:20.52Dr-Linux|home'_+1NXXXXXXXXX' => 1. Answer()                                   [pbx_config]
13:20.52Dr-Linux|home<PROTECTED>
13:21.20kaii[TK]D-Fender: parsing show channels isnt a solution too. :-(   the link between the initiating channel and the ringing ones is missing .. you can detect "SIP/10" is ringing, but not realiably "Zap/1 is ringing on SIP/10"
13:21.25Jas_WilliamsAfter anything above that may match
13:21.26*** join/#asterisk mortsmel (n=andrew@corp-nat.kaslnet.net)
13:21.29Dr-Linux|homeJas_Williams: but i'm configured it at the top in the context ... this is a problem
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13:22.05eppigyhello
13:22.08eppigyi am dave
13:22.15Jas_WilliamsHave a read on asterisk parsing orders or put everything else in an include
13:22.18Kattybops eppigy with pillow.
13:22.21[TK]D-Fendereppigy: O S U R
13:22.36[TK]D-FenderDr-Linux|home: Where's the failed call?
13:22.47Kattyeppigy: you missed spaghetti
13:22.49Dr-Linux|home[TK]D-Fender: no fialed call
13:22.53eppigyD:
13:23.00eppigyspaghettle
13:23.06[TK]D-FenderDr-Linux|home: Show us it failing to match as its supposed to.
13:23.23Dr-Linux|homebut the line is not executing that i want .
13:23.34Kattyeppigy: yeah, guess what i had for breakfast.
13:23.37Dr-Linux|homehhm..
13:23.42Dr-Linux|homelet me try another thing
13:24.15Dr-Linux|homeJas_Williams: I've a number to way to get what i want, but just wanted to understand the trick
13:24.32eppigyKatty: hrmmmm
13:24.41eppigyKatty: eggs benedict?
13:25.54Katty<PROTECTED>
13:25.59Kattyplam?
13:26.02rob0plum
13:26.09Kattyyou're a plum.
13:26.13KattyPROFESSOR PLUM.
13:26.25Kattywhere were you last night, with the candle stick?!
13:26.30*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
13:26.31rob0I know. In the Ballroom with the candlestick.
13:27.23Jas_WilliamsDr-Linux|home: look at this what a nightmare :) http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting
13:27.50Dr-Linux|homeJas_Williams: thanks dude
13:28.09kaiihow can i find out which channel initiated another new channel?   e.g.  Zap/1 executes Dial(SIP/10)  ..  in show channels i see that Zap/1 is in Dial(..) and see that SIP/10 is in ringing state, initiated by AppDial ... instead of seeing that the channel SIP/10-xyz was initiated by AppDial, i want to go further and know that AppDial was initiated by Zap/1 and therefore know that Zap/1 is calling SIP/10
13:28.11eppigyI love eggs benedict
13:28.31Kattyeppigy: i don't think i've ever had eggs benedict.
13:28.36Kattygoogles.
13:28.38eppigyman it is so grood
13:28.55Kattyis that cheese?
13:29.14[TK]D-FenderHollandaise = if bacon doesn't kill you, this will
13:29.35[TK]D-FenderKatty: a cream & egg-yolk based sauce
13:29.43Kattyeww.
13:29.46eppigyhollondaise sauce
13:29.58eppigyman it is so good
13:29.59Kattythis might have something to do with my allergy to egg.
13:30.05eppigy:[
13:30.10eppigyyou cannot eat egg?
13:30.10leifmadsentips his cap towards Ms. Katty
13:30.13Kattyit's just a mild allergy tho.
13:30.22eppigyI love eggs so much
13:30.23Kattyeppigy: i can. it just makes me naeusous for like an hour
13:30.27eppigyD:
13:30.30Kattyhugs leifmadsen
13:30.37leifmadsenis happy he is not allergic to any foods
13:30.40eppigyI eat at cracker barrel, IHOP, and waffle house
13:30.45eppigyevery week
13:30.50Kattywe have a cracker barrel here.
13:30.54Kattybut.. never been there.
13:31.01eppigyuncle herchel's favorite
13:31.06Kattythey don't have nutrition info posted on their website )=
13:31.07eppigyherschel's?
13:31.15eppigyCOUNTRY HAM
13:31.46rob0Katty, I'd call that "can" a definite "can't". :)
13:31.54Kattyhave you had cocacola ham?
13:32.08*** join/#asterisk SilentGreen (n=chatzill@201.217.56.16)
13:32.17eppigynegative
13:32.28rob0If I wanted to be nauseous for an hour, I'd watch Faux News.
13:32.44Kattyfox news is pretty bad.
13:32.52Kattyit's something about 911 or terrorists every 5 minutes.
13:33.03eppigyTERRISTS
13:33.05Kattyeppigy: you get a ham. from the store. uncooked.
13:33.10Kattyeppigy: and 2 bottles of coke.
13:33.18Kattyeppigy: and you boil the ham in the coke for like... 4 hours.
13:33.25eppigySNAP
13:33.35Kattyeppigy: then cover it in molassas, ground mustard, and cloves. then bake.
13:33.46eppigyoh wow
13:33.50leifmadseneppigy: don't forget about the terrorist democrats!
13:33.53eppigyand it tastes good?
13:33.54rob0diet coke and Mentos
13:34.02Kattyrob0: also fun.
13:34.08eppigyleifmadsen: that is all part of the partisan elusion
13:34.16Kattyeppigy: it's a nigella bites classic
13:34.20eppigyevery 4 years we are pitted against eachother
13:34.23SilentGreeni prefer coke zero and mentos... ;o)#
13:34.29eppigyrather than identifying our true adversary
13:35.03eppigy*illusion
13:35.06eppigyjesus
13:35.32eppigyi guess elusion is corrcet as well though
13:35.33eppigyNICE
13:35.56Kattyeppigy: http://www.cookstr.com/recipes/ham-in-coca-cola
13:36.12Kattyeppigy: you will never eat ham out again.
13:37.19SilentGreenneed help with asterisk establishing connection from one asterisk to another works but voice isn't send, with a bit luck just for 2 secs
13:37.37eppigyKatty: it sounds intriguing
13:38.07SilentGreenmaybe there is some intelligent software at the telephone and provider side that anlyses the stream if it is voice or data
13:38.14SilentGreenany chance to cheat?
13:39.12SilentGreenis there a possibility to get rtp-stream tunneled? or something else?
13:39.20*** join/#asterisk juanjoc (n=jcomella@200.69.219.113)
13:39.24SilentGreenthat the provider can't block VoIP
13:39.28eppigywell you can of course tunnel it
13:39.43eppigybut that has nothing to do with asterisk
13:40.16*** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com)
13:40.27SilentGreenthat's right, it's ssh but is it possible to tunnel an rtp-stream?
13:40.38eppigyyou can tunnel anything
13:40.55[TK]D-Fendereppigy: Sounds dirty :p
13:41.02eppigyi know :D
13:41.27eppigygurl i wanna tunnel that rtp stream till next july
13:41.42SilentGreenok, sorry. i am new to asterisk and voip, thats why i'm asking if it is possible to tunnel an rtp-stream...
13:41.48eppigywell I mean
13:42.04SilentGreenby the way. i tunneled my gurl last night... ;o)
13:42.04eppigyhave you checked to make sure there are no firewall rules on your end
13:42.10eppigythat could eb to blame
13:42.14eppigySilentGreen: NICE
13:43.01rob0"Tunneled", how? Define what you mean. Some means of tunnelling, like over ssh, won't work for RTP.
13:43.04eppigyI hope your key length was sufficient to maintain data integrity
13:43.13rob0oh hush boys
13:43.21eppigylol
13:44.03Kattyeww.
13:44.15Kattywe can't talk about this stuff so early.
13:44.30rob0See, now Katty has the ewws. You meanies.
13:44.36eppigyFLUFFY KITTENS
13:44.41Kattymuch better.
13:44.52*** join/#asterisk djMax (n=chatzill@66.92.91.132)
13:45.01*** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)
13:45.23djMaxDoes anyone know how trackable phone numbers (e.g. one phone number per print ad) work?  Do the telco's handle it all, or do you need a local PBX?
13:45.34*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
13:45.39SilentGreeneppigy: ok i check... let's tunnel it... ;
13:45.45Kattyhugs [intra]lanman
13:45.46rob0Tunnelling via a VPN like openvpn is as simple as using the right IP addresses with functional routing tables on both ends.
13:46.01eppigyyes I am a fan of openvpn
13:46.25SilentGreenyou mean "open" for "tunneling"??? lol
13:46.30eppigyoh boi
13:46.30Kattyeww.
13:47.00eppigycovers Katty's ears
13:47.04Kattykthx.
13:47.17SilentGreenoh sorry... my fault. i mean these nice tunnels for fluffy kittens they can run through and play...
13:47.27eppigyFLUFFY ORANEG KITTENS
13:47.30eppigyORANGE
13:47.58*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:48.06Kattyhttp://icanhascheezburger.files.wordpress.com/2008/04/funny-pictures-happy-fluffy-orange-cat.jpg
13:48.58SilentGreenCUUUUUTE!
13:49.17eppigy^______^
13:49.39Kattysoda all gone :<
13:49.54eppigymy rockstar is gone
13:50.15SilentGreenmaybe, because some kind of offtopic for this channel??? :D
13:50.33[intra]lanmanhugs katty...
13:50.41[intra]lanmanKatty: what's up girl?
13:50.55Katty[intra]lanman: not sure yet.
13:51.05Katty[intra]lanman: so far i have 1 ticket open. a samsung box is off by 5 minutes.
13:51.42[intra]lanmanoff?
13:51.43eppigyntpd
13:51.52Kattywe're talking about a samsung 7100 here
13:51.58Kattyit doesn't do ntpd
13:52.06Kattyit does manaul.
13:52.08Kattymanual.
13:52.22eppigywhat r that
13:52.32Kattywhat your mom does.
13:52.36Kattyto the clocks in her house.
13:52.45eppigyoh
13:52.50eppigy:<
13:52.52rob0omg, tmi
13:52.58[intra]lanmanrofl
13:53.08KattyTOPIC CHANGE: http://icanhascheezburger.files.wordpress.com/2008/04/funny-pictures-happy-fluffy-orange-cat.jpg
13:53.24SilentGreenmanual??? how nasty... i thought its abaout cute fluffy kittens
13:53.42SilentGreenthx katty
13:55.33*** join/#asterisk ingenius (n=alektro@69.90.72.173)
13:56.47SilentGreenok, please short back to topic ;o)
13:57.10*** join/#asterisk moy (n=chatzill@74.12.124.89)
13:57.34SilentGreenfirewall is opened for rtp and the related ports on both sides??? thats annoying...
13:58.26SilentGreenfew months ago it worked perfectly, and now it seems to be blocked...
13:58.36eppigylol
13:58.39eppigyyeah bro
13:58.56SilentGreeneppigy: what?
13:59.03eppigywell I mean did you check?
13:59.14eppigyI thought you were affirming
13:59.16eppigythat indeed
13:59.20eppigythey were not open
13:59.22eppigyon both ends
13:59.52SilentGreenno, i only checked it again...
14:00.07SilentGreenand its ok, so far.
14:00.27SilentGreenwould have been strange, because there were no changes
14:02.02*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
14:02.14SilentGreenand like i said it worked until few weeks ago
14:02.57Kattyhugs jaytee
14:03.12jayteehugs Katty
14:04.06eppigyI LIKE THAT BOOM BOOM POW
14:04.16*** part/#asterisk SilentGreen (n=chatzill@201.217.56.16)
14:05.36*** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell)
14:05.37*** mode/#asterisk [+o Qwell] by ChanServ
14:07.44Great_Anta_Bakai want to wait till a user enters 4 digits.. is this waitexten(4) ?
14:08.50eppigyor Read(variable|filename|4)
14:09.08Great_Anta_Bakaty.. will try that out
14:09.21eppigygod speed friend
14:13.42*** join/#asterisk grmartin (n=grmartin@c-76-110-3-120.hsd1.fl.comcast.net)
14:13.52*** join/#asterisk theHub (n=theHub@69.177.93.21)
14:14.16grmartinDoes anyone have an example of dynamically generating a dial in menu from a db? maybe in perl?
14:15.11Great_Anta_Bakammm eppigy i want it to block the outgoing call until i have put in all the digits of my pin code
14:15.22*** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173)
14:17.21[TK]D-FenderGreat_Anta_Baka: So have them enter a value, check it and continue if it matches.
14:18.14xrmx__anyone has a sip tapi software to reccomend?
14:18.17Great_Anta_Bakashould i do waitexten(-1) instead.. i am trying to program this in an agi script on an asterisk 1.2 box
14:19.15[TK]D-FenderGreat_Anta_Baka: No, you should be using the AGI commands for getting input.
14:19.33[TK]D-FenderGreat_Anta_Baka: Not typying to call dialplan apps directly for it
14:21.00Great_Anta_Bakai see
14:27.08pifhi, has debian stopped updating its asterisk package ?
14:28.12[TK]D-Fenderpif: What version do they list?
14:28.26pif1:1.4.21.2~dfsg-3
14:28.47[TK]D-Fenderpif: and what version does the /topic list?
14:29.13pifyour point?
14:29.35[TK]D-Fenderpif: if (1:1.4.21.2~dfsg-3 != 1.4.24.1) then omfg
14:29.43*** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db8140e.pool.einsundeins.de)
14:29.53Wired_Lifegood morning
14:30.08eppigyhi
14:30.10Wired_Lifehello again [TK]D-Fender
14:30.29keeblerWon't even let you manage it on your own.aaaaaaaaaaaaaaaasdfghjkl;'asdfghjjjjjjjjjkl;'sxdcfghjkl;/'Zsxdfghjk
14:30.40eppigycalm down guy
14:30.50piftzafrir_laptop: you there?
14:31.03[TK]D-Fenderkeebler: ?
14:31.15*** join/#asterisk ludan (n=daniele@unaffiliated/ludan)
14:31.34keeblerThe aalkhla;h; was my kids. The other was in reference to Blackberry MVS installation.
14:31.39Wired_Lifetoday i have a simple yes no question.... is its possible to check if a sip softphone is online?
14:31.48*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
14:31.50Wired_Lifevia dialplan
14:32.17[TK]D-FenderWired_Life: "core show application chanisavail"
14:32.18keebler[TK]D-Fender: Tring to find a decent SIP capable ATT phone.
14:32.36Wired_Lifethx
14:32.39[TK]D-Fenderkeebler: AT&T makes SIP phones?
14:32.55[TK]D-Fenderkeebler: and why would you limit yourself to them even if they do?
14:32.56keebler[TK]D-Fender: Not that I'm aware.
14:33.06keebler[TK]D-Fender: Our business
14:33.18*** join/#asterisk machoman48 (n=machoman@wifi-eduroam.mendelu.cz)
14:33.19keebler[TK]D-Fender: I need a dual capable phone
14:33.33[TK]D-Fenderkeebler: Is it named "WeWhoMustBuyAT&T"?
14:34.07keeblerBlackberry has a really sweet BES Suite but its too damn expensive and they won't support Asterisk.
14:34.41keebler[TK]D-Fender: My boss signed the contract before I joined. Its what everyone in this area has. Verizon doesn't work in the field.
14:34.57keeblerSo yes. For this part of Texas. ATT is a must.
14:35.04[TK]D-Fenderkeebler: What does this have to do with a phone that does SIP & analog?
14:35.21keeblerI need a cell phone that does SIP.
14:35.33[TK]D-Fenderkeebler: Oh wait, not its CELLULAR as well?
14:35.39keeblerYes.
14:35.46[TK]D-Fenderkeebler: Perhaps you could provide a new & complete description.
14:35.49keeblerATT + SIP capable phone.
14:36.00[TK]D-FenderATT is a COMPANY, not a TECHNOLOGY
14:36.01keeblerI never really consider ATT for landline anymore.
14:36.28djMaxiPhone does SIP
14:36.31djMaxish
14:36.39keeblerOkay, my clients want to be able to call the local VOIP Lan from their Cell phone, and have their phone act as a detached extension
14:37.00keeblerdjMax: So does BB and many others. But most of the implimentations are too complicated for my clients.
14:37.18djMaxthat's a bit trickier, yeah.  It would murder battery performance pretty much on all phones
14:37.29keeblerShit. Late for work. I'll return in 45 minutes.
14:37.34[TK]D-Fenderkeebler: So a cell with a VoiP client.
14:38.53*** join/#asterisk jtodd (n=jtodd@144.sub-70-214-184.myvzw.com)
14:38.53*** mode/#asterisk [+o jtodd] by ChanServ
14:39.05*** join/#asterisk curious101 (n=q@125.212.122.188)
14:40.26curious101hi! my question is: will asterisk run fine on a server with 2 IP addresses? can it be configured to listen to just one interface?
14:41.23blebleblecurious101: sure
14:42.38curious101blebleble: can you explain further please?
14:44.19*** join/#asterisk Wired_Life1 (n=Chatzill@mgdb-4db87385.pool.einsundeins.de)
14:44.59blebleblejust set the bindaddr
14:47.43*** join/#asterisk Gabriel25 (n=gabe@pool-72-68-157-205.nycmny.fios.verizon.net)
14:49.05curious101blebleble: ah, I see. Thank you.
14:51.51*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
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14:57.43Nuggetpoor dvossel
15:03.28*** join/#asterisk bakermd (n=bakermd@38.101.225.215)
15:04.12bakermdHow can I change my config so that when it sends email voicemails it connects to a different smtp server?
15:04.45rob0That would be an issue in your MTA, not in *.
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15:06.45*** join/#asterisk jmls (n=jmls@host217-36-208-155.in-addr.btopenworld.com)
15:06.48jmlshey guys
15:07.41jmlstrying to get 2 instances of asterisk running on the same box (1.6 and 1.4)
15:07.41jmlsgot everything made and installed into the appropriate separate directories
15:07.41jmlsstarted 1.4 ok
15:07.41jmlstrying to run 1.6, I get "Asterisk already running on /var/run/asterisk.ctl"
15:08.00jmlsis there an option to specify where asterisk.ctl can be placed for each instance ?
15:08.26rob0I suppose you'd want them to have different settings in asterisk.conf <=== see that
15:08.38Corydon76-digjmls: no, but there's an option to source asterisk.conf from different locations, and one of the options in asterisk.conf is where the ctl file is located
15:09.03*** join/#asterisk zapotek6 (n=edpman@mail.comelit.it)
15:09.19jmlsCorydon76-dig, that's what I was looking for. Thanks
15:09.29*** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM]
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15:16.13cutlasshello...can anyone tell me what it means to get a <ZOMBIE> channel?
15:18.56russellbcutlass: That is a very interesting question, actually.  :-)
15:19.26russellbEssentially, you're seeing what happens to channels when you do something like a transfer.
15:19.50russellbThe way it works internally to Asterisk is that a new channel gets created, all of the "guts" of the old channel are moved to the new one, and the old one becomes a Zombie.
15:20.01russellbNow, you don't really need to understand why (nor do you need to care) why this is the case ...
15:20.18russellbbut it is expected to see zombie channels with transfers, parking, redirects ...
15:21.11cutlassok...so it's not to be interpreted as an error then, right?
15:21.27*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
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15:23.47cutlasssorry...I just read your last statement which has the answer to my last question :)
15:23.59cutlassthanks Russell
15:24.16russellbyou're welcome
15:26.24*** join/#asterisk DGTL_Magician (n=boerg@siona.servers.nosco-ict.nl)
15:26.29jayteedamn zombies! they're everywhere
15:26.49jayteejust ask Woody Harrelson
15:27.00outtoluncsays brainsssssss!
15:27.17DGTL_MagicianI like zombies
15:27.34DGTL_Magicianexcept on my server :)
15:33.44*** join/#asterisk Erol_ (n=x@88.234.108.194)
15:36.32Erol_does SIP proxy mean PSTN gateway?
15:37.46[TK]D-FenderErol_: No.
15:37.55mmlj4no
15:38.58*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
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15:39.53Erol_i went to iptel.org web site
15:39.59plundraOne of my sip-providers gives me "T-Bone 2.3.3" as the User-Agent, anyone know what company that is?
15:40.16*** join/#asterisk bbryant (n=olpc@68.208.65.50)
15:40.48plundraI can't seem to find _anything_ about it :)
15:41.25Erol_and I got a bit confused
15:41.52Erol_does SER do the same thing asterisk do?
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15:44.31[TK]D-FenderErol_: Not at all
15:45.08Erol_[TK]D-Fender: uhm. could you give me a crash course, =)
15:45.26*** join/#asterisk c4t3l (n=c4t3l@c-76-31-57-251.hsd1.tx.comcast.net)
15:45.36c4t3lhello world
15:48.01[TK]D-Fender~book
15:48.02infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
15:48.04[TK]D-FenderErol_: ^^^
15:48.26*** part/#asterisk machoman48 (n=machoman@wifi-eduroam.mendelu.cz)
15:50.34Great_Anta_Bakalol.. i remember when you did the same thing to me.. although i did have a hardcopy of the 1.2 book at hand
15:50.36eppigy[TK]D-Fender: when does your lecture on TDM technologies start?
15:50.54Great_Anta_Bakaand does it go online?
15:52.07[TK]D-Fendereppigy: Usually following the receipt of a personalized training payment :)
15:52.19eppigyNICE
15:53.01*** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net)
15:53.16deadpigeonis it better to use an IDE or a SCSI drive on an asterisk server?
15:53.38*** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net)
15:54.12[TK]D-Fenderdeadpigeon: SCSI is typically more reliable and faster, so do the math :)
15:54.45mort_gibdeadpigeon: Sata RAID 5
15:55.34deadpigeonthanks. im just experiencing some unsatisfactory results with dahdi_test and im trying to narrow down the culprits
15:55.41[TK]D-FenderMFM RAID 0 :p
15:55.56[TK]D-Fenderdeadpigeon: and in other unrelated news...
15:56.22[TK]D-Fenderdeadpigeon: Try shoing some useful information because that last bit was a wildly aimed pot-shot
15:56.35[TK]D-Fendershowing*
15:56.59deadpigeonI think I can handle it myself for now. I was merely asking what was preferred storage devices.
15:58.01mort_gibdeadpigeon: Would dhadi_test have anything to do with HDD performance?????
15:58.01deadpigeonIt could if there was a lot of HD activity or DMA settings were wrong.
15:58.01tzafrir_laptoppif, yes
15:58.51deadpigeonAnyone know off hand if I have to recompile dahdi_tool to use watchdog for IRQ misses like we used to with zaptel?
15:59.41Erol_I understand that SER is something like an IP router. If I am willing to set up a pbx for for example 10 extensiona would I need SER?
16:02.32*** join/#asterisk captiancrash (n=jonmoore@70.159.118.70)
16:02.58[TK]D-FenderErol_: A SIP router is not a PBX.  PBX typically entails backend services like voicemail, IVR's, etc.  this is core functionality with *.  * also speaks a LOT more than just SIP.
16:03.17[TK]D-FenderErol_: If you want a PBX, and want it to speak SIP, then * could very welll be for you
16:04.59djMaxIs 2B channel transfer likely to work on an XO PRI, and/or is there an easy way to find out if it's enabled?
16:05.23Erol_[TK]D-Fender: and by the way why people like SIP more than H323?
16:05.42Erol_[TK]D-Fender: as far as I can see from the specs H323 has more features
16:05.59djMaxErol_: I don't think they're really on the same level
16:06.00[TK]D-FenderErol_: SIP is in plain-text an there are some other fine points but nothing I'm qualified to elaborate on that much.
16:06.14djMaxSIP is about "rendezvous" more than transfer
16:06.21djMaxtransport I should say
16:06.40[TK]D-FenderErol_: and SIP is a more "routed" infrastructure than H.3232 IIRC as well
16:06.54_Steve_anone want to help me with vlan config on my polycoms?
16:07.43Erol_[TK]D-Fender: a website says that best solution for a pbx would be * + SER.
16:08.01djMaxa website? for real?
16:08.13[TK]D-FenderErol_: taht is a rather unqualified statement.
16:08.13eppigyi will argue about this on a forum
16:08.42[TK]D-FenderErol_: SER can help in certain large & redundant scenarios, but not for most user's needs
16:08.46eppigyErol_: do they mean like multiple asterisk instances with SER for sip registration balancing
16:08.47eppigy?
16:09.05eppigyits uncanny
16:09.19*** join/#asterisk telnettech (i=telnette@gw.percipia.com)
16:09.19eppigyhow [TK]D-Fender and I's thought processes just mesh
16:09.21mmlj4I haven't even looked at ser
16:09.28Erol_http://www.en.voipforo.com/ser/ser_asterisk.php
16:09.30mmlj4guess I should at some point
16:09.47[TK]D-Fendermmlj4: think "basilisk" :p
16:09.54*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
16:10.34[TK]D-FenderErol_: But regardless of that you have not been at all specific about the scope of your needs.
16:10.35eppigySER by itself it not very useful but SER teamed with Asterisk is how you make Asterisk scale.
16:10.45eppigykeyword "scale"
16:11.02djMaxwhere scale I assume is defined as hundreds or thousands of connections
16:11.09djMaxbecause under that it would seem overkill
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16:11.15*** mode/#asterisk [+o russellb_] by ChanServ
16:11.23Kattywhere are we having lunch today?
16:11.47*** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com)
16:11.51[TK]D-FenderKatty: General tao.... I feel soo... white.....
16:12.01Erol_djMax: so you mean its for large solutions?
16:12.13djMaxfrom my reading of it, yes.
16:12.33[TK]D-FenderErol_: "large" is a matter of perspective
16:12.36mmlj4just what I need, scaly software
16:12.37*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
16:12.50[TK]D-Fenderhands mmlj4 a fillet knife
16:13.05_Steve_hmm, i don't even see a VLAN setting in the polycom web interface...
16:13.05Erol_[TK]D-Fender: pbx with thousand extensions amybe?
16:13.06djMaxwhich brings us back to the question of what Erol_ actually needs
16:13.13djMaxhow many active calls?
16:13.21[TK]D-FenderErol_: Yeah, maybe a thousand...
16:13.29[TK]D-FenderErol_: Also depends on what they're doing...
16:13.46Erol_[TK]D-Fender: just simple phones
16:13.54Erol_[TK]D-Fender: talking
16:13.58djMaxI suppose another way to look at it is if you'll need more than one asterisk box, you might consider SER.
16:14.00*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
16:14.05*** join/#asterisk keebler (n=Christop@adsl-99-179-135-231.dsl.rcsntx.sbcglobal.net)
16:14.13keeblerOkay, that was a bit longer than 45 minutes...
16:14.18djMax(IANASE - I am not a SER expert)
16:14.27[TK]D-FenderErol_: all at the same time>?  recording?  transcoding?  other processor intensive stuff?  ivr's with ton's of DB lookups?
16:14.35djMaxhow many PRI's do you have, or is it all VOIP?
16:15.04[TK]D-FenderPRI?  Never heard it....
16:15.16keebler[TK]D-Fender: Back to my original description. I need a GSM/GPRS Cell phone that can do SIP over both WiFi AND GPRS/UMA, and it has to work on ATT network. Any ideas?
16:15.34Erol_djMax: not so detailed
16:15.54[TK]D-Fenderkeebler: Good to see you managed to boil that down to a complete & coherent single question :)
16:15.55djMaxkeebler - "and is easy for noobs to use" right?
16:16.04keeblerdjMax: yes.
16:16.19keebler[TK]D-Fender: Yeah, I had JUST woken up this morning during that chaotic convo.
16:16.29[TK]D-Fenderkeebler: And.... no, SIP = plenty of clients... its the quality of network & integration with the phone I'm not so sure of.
16:16.33Erol_and what are communigate and sipcat? what is the difference between them and asterisk?
16:16.42eppigyKatty: I am thinking krystak
16:16.43keeblerdjMax: My clients don't even know how to reboot a router.
16:16.46eppigykrystal
16:16.55djMaxwell, that might not earn them noob status. :)
16:16.57keeblerdjMax: And they make $120k A YEAR.
16:16.59Erol_I mean I know they are for pbx solutions but do they offer anyth,ng more than *?
16:17.20eppigyno
16:17.25keeblerto them, it just "has to work".
16:17.28eppigyYou need ser
16:17.30djMaxThe iPhone sip client is good, IMHO, except no background running
16:17.38eppigyat the point where your cpu load
16:17.43djMax(unless you want to recharge every 3 hours)
16:17.43eppigyis too high
16:17.45keeblerdjMax: Is it real SIP? Or just call-through?
16:17.49eppigyand you need to scale
16:17.50djMaxreal sip
16:17.53[TK]D-FenderErol_: Do those products let you do whatever you want for any call placed?
16:17.54djMaxjailbroken phones only though
16:17.58keeblerAh
16:18.00keeblerDamn
16:18.01djMaxthere's a non-jailbroken skype client
16:18.15keeblerNot sure Skype will work. and Gizmo5 sucks ass.
16:18.27keebler(Not real SIP on the Gizmo, I know.)
16:18.36djMaxwonder if android has anything
16:18.54[TK]D-FenderErol_: Dial 911 on a tuesday night where the Raiders won their last home game and it happens to be raining and my PBX can decide to boil a pot of coffee rather than terminate your call.  Can those other products do that?
16:19.04Erol_[TK]D-Fender: i dont know, I just asked maybe you have more exp about them
16:19.36[TK]D-FenderErol_: You mean we have to know these other products that you don't as well as * AND provider you a complete comparison?  Would you like fries with that sir? :)
16:19.47djMaxandroid might be more promising and/or palm pre
16:19.54Maliuta[TK]D-Fender: but can you teach it to use my espresso machine?
16:19.59Erol_[TK]D-Fender: i just said maybe man..
16:20.05[TK]D-FenderMaliuta : MAYBE
16:20.10djMaxthough none of them can touch the iPhone for quality IMHO.
16:20.18Maliuta[TK]D-Fender: if you can show me the dialplan/AGI for that I'll be your eternal slave ;)
16:20.32*** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27)
16:20.59[TK]D-FenderdjMax: iPhone is nice hardware... and not to say I'm a FOSS-Nazi, I just refuse to have my ass owned by a stupid fruity logo'd company :)
16:21.35[TK]D-FenderMaliuta: Send me the manual for your machine and we'll takl :)
16:21.37[TK]D-Fendertalk*
16:22.03coppiceAll the iPhones here are sold unlocked, but they only unlock the network. you still need to jailbreak them to do anything useful
16:22.39[TK]D-Fendercoppice: and there are bigger processors out there and so much more that can be done... very sad this is the best we've got
16:22.59coppicewhat's wrong with the processor?
16:24.26*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
16:24.48[TK]D-Fendercoppice: Just saying there is bigger out there.  Many free OS's to base things on.  non-proprietary connectors for common stuff.
16:25.02[TK]D-Fendercoppice: jsut that everyone locks you down in some nasty way or another
16:25.56coppiceThe only 2 real pains with the iphone are the need to jailbreak, and the lack of real multitasking (and on a Unix platform as well). The other stupid stuff like no cut and paste seems to have been fixed on the latest revision
16:26.56*** join/#asterisk los415 (n=los415@sfca-office.corp.race.com)
16:27.09[TK]D-Fendercoppice: lack of stereo bluetooth, tethering restrictions (more a carrier grievance), etc
16:27.33[TK]D-Fendercoppice: proper flash support, as well
16:27.38coppiceI think the stereo bluetooth is also fixed now.
16:27.59*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
16:28.04coppicethey seem anti-flash, but on a wince machine flash really drains the battery
16:28.08los415has anyone used the fax for asterisk product yet?
16:28.47coppicethat's a strange product. why didn't they do V.34 FAX? Commetrex say they have it
16:29.05[TK]D-Fenderlos415: Nobody wants to break the Freshness Seal :)
16:29.15los415lol
16:29.49los415i want to try the t.38 termination right to asterisk and do fax to email but i dont want to beat my head on the desk for hours trying toget it to work
16:29.52coppiceI think they could have got some buzz around a V.34 FAX package, but a me too?
16:31.08los415i guess there is only one way to figure out if it works or not
16:31.40coppicedo they supply fax to email software with it?
16:32.15los415well it says with 1.6 it will create a tiff
16:32.23coppiceif you look around there are several very simple scripts to do it. doing email to FAX well is rather more involved
16:32.23los415so i can figure it out from there
16:32.38coppiceeveryone creates a TIFF :-)
16:33.14los415well yea but we already have a system that does the entire tiff to pdf to email
16:33.26los415it's just getting constant good faxes
16:33.32los415and not cut off mangeld
16:33.50coppiceand what gives you cut off mangled FAXes?
16:34.24los415then normal fax stuff that comes with trix
16:35.45los415our switch supports t.38 and i have it working perfectly going to adtran ta's series cpe's ect
16:35.59los415so if asterisk could terminate the t.38 would be nice
16:36.14coppiceapp_fax in 1.6 does that
16:36.52*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
16:38.37mazpeis there an default feature code for *67 in asterisk?
16:39.43mazpeor how can i block the caller id, for a particular call
16:40.40[TK]D-Fendermazpe: Depends what you are calling out via
16:40.55[TK]D-Fendermazpe: and no, there is no "default feature in *" for this.
16:41.10*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
16:41.28mazpe*67, gives dial tone and allows a call to be made with the caller id blocked
16:41.45[TK]D-Fendermazpe: Didin't answer my question...
16:42.01mazpevia a sip trunk
16:42.07[TK]D-Fendermazpe: You've just described common CLASS functionality without your specific circumstances
16:42.10*** join/#asterisk rafiks (n=rafiks@c-68-56-8-98.hsd1.fl.comcast.net)
16:42.24[TK]D-Fendermazpe: Depends how they want you to tell them to block CID
16:42.42piftzafrir_laptop: are you still packaging asterisk for debian?
16:42.53tzafrir_laptoppif, yes
16:43.04mazpe[TK]D-Fender: the provider allows me to manipulate the caller id. I can just make it 00000000
16:43.17piftzafrir_laptop: why are you still at 1.4.21 ?
16:43.48[TK]D-Fendermazpe: try "core show application setcallerpres"
16:44.16mazpe[TK]D-Fender: thanks
16:45.28tzafrir_laptopexperimental is at 1.6.1-rc3
16:45.43tzafrir_laptopI have testing packages of -rc4 at my repo
16:45.54tzafrir_laptopfor lenny and etch as well
16:46.16tzafrir_laptoprepos actually . the "-experimental" ones under:
16:46.30tzafrir_laptophttp://updates.xorcom.com/pkg-voip/
16:46.44pif'apt-cache show asterisk' only shows 1.4.21 ...
16:46.51tzafrir_laptope.g. deb http://updates.xorcom.com/pkg-voip/repo-i386-etch-experimental unstable main
16:46.52pifthe package name has changed?
16:46.58tzafrir_laptopno
16:47.06tzafrir_laptopread what I wrote
16:47.07pifah
16:47.15rafikspif : update ur sources.list
16:47.23pifgot that
16:47.35tzafrir_laptopthe dahdi+libpri (removing bristuff) + asteirsk transition takes time
16:47.39tzafrir_laptopLenny has 1.4.21
16:47.40pifwhy not in debian repos?
16:48.36pifgood to hear you are removing bristuff ;)
16:48.52*** join/#asterisk phl4kx (n=supervis@webmailserver.nisira.com.pe)
16:50.45*** join/#asterisk jtodd (n=jtodd@182.sub-70-214-133.myvzw.com)
16:50.45*** mode/#asterisk [+o jtodd] by ChanServ
16:54.57*** join/#asterisk ntbourey (n=ntbourey@c-76-110-3-120.hsd1.fl.comcast.net)
16:55.09ntboureyHey every one
16:55.44rafiksntbourey ; hey
16:56.01ntboureyI was wondering if someone could provide me with some assistance
16:56.39Maliutano, we won't jump start your car for you
16:56.49ntboureyHaha I wish it was that simple
16:57.10piftzafrir_laptop:  Failed to fetch http://updates.xorcom.com/pkg-voip/repo-i386-etch-experimental/dists/unstable/main/binary-amd64/Packages
16:58.02Maliutai386-etch-experimental ... with unstable and amd64?
16:58.14Maliutasounds like a badly named repo ;)
16:58.26ntboureyDoes anyone know how to integrate Perl into asterisk, res_perl doesn't seem to compile with 1.6
16:59.11Maliutaperl for AGI's?
16:59.37tzafrir_laptopMaliuta, yeah. I should have a proper single repo. but I'm lazy to make this a proper setup
16:59.39ntboureyI guess I am still kind of new to this
16:59.55RypPnAt a guess... http://asterisk.gnuinter.net/
16:59.56tzafrir_laptoppif, use the -amd64 repo
17:00.22Maliutantbourey: define "perl in *" if it's not for AGI's
17:00.41*** join/#asterisk stoffell (n=stoffell@d51A4D629.access.telenet.be)
17:01.01ntboureyMaliuta: res_perl I belive, allows you to use regular perl scripts
17:01.02coppiceis it really AMD specific, or is it just general x86_64?
17:01.07ntboureyfrom within a dialplan
17:01.11ntboureycorrect?
17:01.20tzafrir_laptopcoppice, the name just stuck
17:01.25Maliutathat's what and AGI is
17:01.34Maliutait can be anyone of a number of things
17:01.43ntboureyOkay
17:01.45Maliutaperl php ...
17:01.49ntboureyRight
17:01.54coppicetzafrir_laptop: Good. Keep reminding intel that their PR machine lies :-)
17:02.07Maliutantbourey: go google for perl agi
17:02.10piftzafrir_laptop: better
17:02.12ntboureydo I just need to install asterisk-perl
17:02.15ntbourey?
17:02.34*** join/#asterisk axarob (n=ebash@cpc3-barn8-0-0-cust288.brnt.cable.ntl.com)
17:02.35Maliutadon't know your distro or how it's packaged
17:02.45tzafrir_laptopntbourey, res_perl is quite unmaintained
17:03.04ntboureyI installed from source Maliuta
17:03.07tzafrir_laptopasterisk-perl probably refers to the perl asterisk module, which mainly include Asterisk::AGI
17:03.08[TK]D-Fenderntbourey: can you perhaps explain your  expectations for "integration"?
17:03.42ntboureyI just want to be able to use Perl to handle most of my calling features of my PBX
17:03.56Maliuta[TK]D-Fender: I don't think he can ... sounds to me like stuff that could be done with Exec() or an AGI
17:04.22[TK]D-Fenderntbourey: " calling features" is also a very vague statement
17:04.50ntboureyMaliuta: Yes
17:04.59MaliutaI want my phone thingy to use the interthingy to do thingy
17:05.02Maliuta:)
17:05.08ntboureyup until yesterday I had no knoledge of Asterisk other than what it was
17:05.19Maliutantbourey: read the book
17:05.23ntboureyDid
17:05.45[TK]D-Fenderntbourey: then stop trying to hammer oddly shaped blocks into holes you can't even see yet :)
17:06.01Maliuta[TK]D-Fender: I like that one
17:06.18ntboureythis is what I am trying to do
17:06.22ntboureyim looking to derive menus and user number searching from a postgres database using perl
17:06.28Kattyblergh
17:06.37jayteephlegm
17:06.55KattyA... C.... Phlem
17:07.11Maliutantbourey: [TK]D-Fender is right. You should use normal means to achieve your setup, and see if you even need AGI's along the way
17:07.11Maliutantbourey: AGI
17:07.25ntboureyOkay
17:07.47ntboureyI already have my basic system working I can dial in, and get responses back
17:08.05ntboureyI'm trying to use a DB to handle generating menus
17:09.26jayteeI'm trying to use magic creams and lotions to make my hair grow back
17:09.26[TK]D-FenderntIf you want advise you should dummy up a DB table with sample data that you would like to see be used in processing your call and then we can advise on viable ways to do itMaliuta
17:09.39[TK]D-Fenderntbourey: So show us what the table data would look like FIRST.
17:09.48ntboureyCan't
17:09.56[TK]D-Fenderjaytee: I told you already.... ch-ch-ch-chia!
17:10.21jayteeyeah, but I was hoping for a different color than green
17:10.22*** join/#asterisk NOT_guru (n=NOT_guru@24-241-103-142.static.stls.mo.charter.com)
17:10.25Kattyjaytee: you're lovely as you are. you don't need chia hair.
17:10.36Kattyjaytee: it won't make you any more charming.
17:10.41jayteeKatty, awww, gee. thanks!
17:10.45*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
17:11.14MaliutaKatty: how come you never say seductive^Wnice things to me? ;)
17:11.28KattyMaliuta: because you're not as charming as jaytee is.
17:11.42ntboureyhttp://pastebin.com/m345327d1
17:11.43KattyMaliuta: you should work on that.
17:11.47MaliutaKatty: thats only because you haven't met me in person
17:11.51jayteeswells up and swaggers off to slay dragons
17:12.01KattyMaliuta: i've never met jaytee.
17:12.01MaliutaKatty: I'm much more charming in real life than online
17:12.12KattyMaliuta: well isn't that neat.
17:12.17KattyMaliuta: good for you.
17:12.22MaliutaKatty: IRC is for blowing off steam and making lude remarks ;)
17:12.39jayteelude? as in Qualude?
17:12.49Kattylude-a-chris speed!
17:12.50jayteeor lewd as in "hey! nice tits!"
17:13.03KattyMaliuta: I might be a male. ever think of that?!
17:13.17coppicelewd? isn't that a compliment?
17:13.21jayteeludicrous speed, rofl. I watched that again the other nite
17:13.26MaliutaKatty: I might not care ... ever think of that? :)
17:13.31Kattyeww.
17:13.47Maliutajaytee: go comb the desert for me ;)
17:13.56jayteehahaha
17:14.05[TK]D-Fenderntbourey: How is that a "menu"?
17:14.15[TK]D-Fenderntbourey: thats jsut a PIN-list, not a menu structure.
17:14.15jayteePrincess Bride is still better
17:14.22KattyAs you wish!
17:14.26[TK]D-Fenderjaytee: aaassss yoooouuuu wiiiisssshhh!!!!
17:14.28Kattytumbletumbletumble
17:15.03KattyThat makes me think of twisted.
17:15.11Kattythe dread pirate robberson!
17:15.16Maliutaministry of silly walks all over this
17:15.23[TK]D-FenderKatty: "Roberts"
17:15.23ntboureyThey would be able to login and access a list of contacts
17:15.35Katty[TK]D-Fender: yes. i know. it was a running gag with twisted.
17:15.35Maliutadon't make me do the Parrot Sketch
17:15.42[TK]D-FenderKatty: "the Dread Pirate Robert takes no survivors! HA HA HA"
17:15.46jayteeback in a few, gots "stuff" to do
17:15.59[TK]D-Fender</andre>
17:16.25[TK]D-Fenderntbourey: Ok, this is very trivial stuff you can so far do completely in dialplan.
17:17.15[TK]D-Fenderntbourey: Go read the BOOK and pay extra attention to the chapter on func_odbc
17:17.16[TK]D-Fender~book
17:17.17infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
17:17.18[TK]D-Fender^^^^^^^^^^^^^
17:18.00telnettechI hate the company i work for......
17:18.14Kattyjoin the club.
17:18.24telnettechTK all that talking last night about 911 and they shoot me down and tell me to let development work on it
17:18.32Katty:<
17:18.50Kattytelnettech: our company said they weren't going to sell asterisk systems anymore.
17:18.50Nuggettelnet is eeeeeeevil!
17:19.00Kattytelnettech: because there's "no support" for it.
17:19.06Kattytelnettech: how bout that?
17:19.14KattyNugget: shall we hug?
17:19.34telnettechKatty: what do they mean by no support
17:19.47djMaxanybody know how to setup TBCT on *?
17:19.48telnettechfrom the community or within the company
17:19.50[TK]D-Fendertelnettech: If they are afraid of *, and you're "it", who the hell is "development", and can I have some of their crack?
17:19.57*** join/#asterisk grantm (n=grant@68.142.138.4)
17:20.02Kattytelnettech: the real reason they don't want to sell it is because people recognize Samsung and Toshiba.
17:20.05[TK]D-FenderdjMax: "core show application transfer"
17:20.45Kattytelnettech: their excuse is that I'm the only person here who knows asterisk, and there's no "support" i can call if i don't know what to do. regardless of the fact i've offered them at least 10 sources for support.
17:21.10Nuggethuggles Katty
17:21.20coppicethey want more than just the sources :-)
17:21.24[TK]D-FenderKatty: What are we, chopped liver?
17:21.34Kattyhugs Nugget
17:21.48Katty[TK]D-Fender: welll...
17:21.52Katty[TK]D-Fender: heh ;)
17:22.06Katty[TK]D-Fender: it's just an excuse.
17:22.14telnettechI am fighting with the support manager and the development mgr....they have sold a product to a customer and told them that they can provide this but havent and the customer refuses to finish paying the contract cuase of it
17:22.30[TK]D-FenderKatty: Theya re good at that.  Coming up with excuses that is.  Not good ones... just volume
17:22.32coppiceKatty: free stuff pays no kickbacks
17:22.43telnettechthey want the customer to wait about 4 or 5 months once development has figured out how to do it and then test it
17:22.46Kattycoppice: i'm not going to get into it with them.
17:22.52Kattycoppice: it's like talking to a wall.
17:23.37ntboureyThanks for nothing
17:23.39kc8pxywierdo: i got monkeys last night.   i simply had to specify only the sound name,  not it's path. and it worked :)
17:23.46*** part/#asterisk ntbourey (n=ntbourey@c-76-110-3-120.hsd1.fl.comcast.net)
17:23.54[TK]D-FenderKatty: No... walls reflect sound, its jsut gets lost in their vacuum :p
17:24.05*** join/#asterisk Steak_ (n=steak@85.4.120.35)
17:24.09Steak_hello
17:24.27Katty[TK]D-Fender: that too ;)
17:24.40[TK]D-Fender"thanks for nothing".  Brilliant.. totally.
17:24.56MaliutaKatty: how big is the firm you work for?
17:25.14telnettechSo i guess all they want are field techs that know how to push buttons and click on drop down menus and cant think for themselves
17:25.33KattyMaliuta: 30 people, tops.
17:25.33wierdokc8pxy, oh, cool :) the obvious
17:25.39KattyMaliuta: over half of them are sales reps.
17:25.52MaliutaKatty: I have found there is a size where it will always be crap, and another size where it will always include a brick wall
17:25.53KattyMaliuta: the company's 'core' business is copy machines.
17:26.08*** join/#asterisk bbryant (n=olpc@68.208.65.50)
17:26.08Steak_I have a strange issue (or at least it is strange to me..), I have an asterisk installation with a SPA3102 as PSTN gateway. whenever I receive a phone call from the PSTN, all the internal SIP phone ring correctly (and work perfectly if picked up), but the CLID of the PSTN call is not shown, instead the SPA assigned number is displayed... anybody has an idea? thanks in advance
17:26.13KattyMaliuta: phone systems is just one of their 'side' projects.
17:26.25MaliutaKatty: yeah, it's going to be uber crappy then
17:26.30KattyMaliuta: yep.
17:26.53Maliutathe size of the sales force is directly proportionate to the size of the crap
17:26.59cutlassdoes anyone know how to disable call files?  Is it just a matter of making /var/spool/asterisk unwritable, or is there a more elegant way to do this?
17:27.14KattyMaliuta: usually i get along with people. all sorts of people. i can walk into a party and immediately make friends.
17:27.24KattyMaliuta: i cannot STAND half the sales reps here. literally, can't even tolerate them.
17:27.48KattyMaliuta: they open their mouth, and i want to stab them.
17:29.10eppigy:<
17:29.16eppigyI am dave in sales
17:29.17KattyMaliuta: and they wonder why i'm so pissy in the morning! *hee*
17:30.02Kattyeppigy: i think it just has something to do with them being southern missouri hick morons.
17:30.18jayteeor just because they're in sales
17:30.30Kattyjaytee: doubtful.
17:30.39Kattyjaytee: there are a couple sales reps here i get along with.
17:31.29Kattyi like dave.
17:31.33eppigy:D
17:31.48telnettechsorry didnt mean to open a can of worms here
17:32.00eppigytelnettech: that ship has saled, thanks
17:32.03eppigysailed
17:32.07eppigylol
17:32.09Kattywe're on to can of pringles now.
17:32.43[TK]D-Fendercutlass: rm /usr/lib/asterisk/modules/pbx_spool.so
17:34.35cutlassok...I wasn't sure if that module was responsible for anything else in addition to call files...is it the same thing to put noload => pbx_spool.so in modules.conf?
17:34.38Maliutacutlass: you can unload pbx_spool
17:35.23cutlassok
17:35.31[TK]D-FenderMaliuta : Odds are if they can drop call-files to the FS they can write to *'s configs... best to remove all possibility :p
17:35.51[TK]D-Fenderburns it... WITH FIRE!!!!!
17:36.13cutlassmakes sense...
17:36.36cutlassthanks  to both of you
17:37.00cutlass:q
17:37.05eppigyTHE CRUCIBLE
17:37.34cutlass...sorry...that was intented for vi
17:37.53eppigyyes i see that
17:38.45[TK]D-FenderOMMFG GGGGGGGGGGOOOOOOOOOOLLLLLLLLLLLLDDDDDDDDD!!!!!!!!! http://www.collegehumor.com/video:1907543
17:38.56PazzoCan anyone give me a short hint on how to terminate a Call in early media state by Asterisk with some "soft" response (e.g. 480)?
17:39.39*** join/#asterisk colinm_ (n=colinm@VDSL-130-13-115-7.PHNX.QWEST.NET)
17:43.34*** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net)
17:43.37PazzoHmmm... I discovered that Hangup(18) == 408
17:47.01Maliutahads cutlass a ! for his/her/its :q
17:49.03PazzoHangup(16) doesn't seem to be mapped, I get 602. Hangup(17) gives me 486, as expected
17:49.11cutlassI like to be alerted of unsaved changes :)
17:49.18PazzoNo one here able/willing to help me?
17:50.00KattyI'm having urges to add an S to the beginning of your /nick.
17:50.26*** join/#asterisk Nasra (n=maxshipp@190.166.70.98)
17:51.23eppigyhaha
17:51.35*** join/#asterisk jesselang (n=jesse@h69-21-229-150.mntimn.dedicated.static.tds.net)
17:53.47*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca)
17:54.54jesselangIs there a way to use EAGI and channel state (knowing when the call has been answered) simultaneously?  I want to have early media, but also know at what point the call is answered.
17:55.20[TK]D-FenderKatty: Good laugh, watch link!
17:57.13*** join/#asterisk rob327 (i=1001@209.80.7.124)
17:57.40MaliutaI am so kicking my ISP in the balls in the morning
17:57.44Maliutawell later in the morning
17:57.46Steak_... can't really understand why the phone is showing the trunk number instead of the incoming caller CID
17:58.41[TK]D-FenderSteak_: if you set "callerid=" for your sip peer for the PSTN port then that would override anything it passes.  Next you ahve to configure the SPA to even pass on the CID.
17:59.06[TK]D-FenderSteak_: voxilla.com 's forums have plenty of posts on how to configure it accordingly.
17:59.12Steak_the PSTN port entry in sip.conf hasn't got the callerid entry, and the SPA is configured to pass the CID
17:59.41[TK]D-FenderSteak_: You also have to tell it to wait 2 rings or so.
17:59.48Steak_the funny thing is that if I look inside of the asterisk call logs, the CID number is displayed perfectly, and asterisk replaces it with the PSTN port name
17:59.50[TK]D-FenderSteak_: Another gotcha
17:59.53Steak_(waiting 3 rings)
18:00.29Katty[TK]D-Fender: k
18:00.31[TK]D-FenderSteak_: pastebin a call at verbose 10, SIP debug enabled and NoOp the ${CALLERID(all)}
18:00.31Steak_so I am assuming that it is asterisk remapping the incoming call CID with the PSTN port name
18:00.32Kattywatches.
18:00.46Steak_ok a sec
18:00.47*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
18:01.20Steak_(I have no ${CALLERID(all)} entry in extensions.conf, nor in sip.conf)
18:01.43[TK]D-FenderSteak_: .. theFUNCTION.  noOp it it in your DIALPLAN
18:02.07[TK]D-Fender~pb
18:02.08infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
18:02.11[TK]D-Fender^^^^^
18:02.15[TK]D-FenderSteak_: include your configs
18:02.18Steak_k
18:03.29Katty[TK]D-Fender: cute!
18:03.51[TK]D-FenderKatty: very well produced and so very real...
18:05.02Katty[TK]D-Fender: yep.
18:06.33Steak_[TK]D-Fender: here the extensions.conf and sip.conf --> http://pastebin.com/d3a60b8df
18:06.47Steak_now I'll make the log of a phone call with the output
18:09.26MaliutaSteak_: ummm you know you left the password in?
18:09.34MaliutaSteak_: you might want to change that
18:10.55kc8pxySteak_:  it rings for 60?
18:11.20kc8pxyMaliuta: i noticed that too.
18:11.50Steak_it's all in a private network
18:12.23*** join/#asterisk lanning (n=lanning@nat/yahoo/x-64d93ef7e0a84ced)
18:12.33Steak_sorry, I got a call from a bakery in the meanwhile that was inquiring about an order of 300 breads :P not made by me :P
18:12.42Steak_ok, back again
18:13.58MaliutaAll your breads are belong to us!
18:14.21Kattynot the breads:<
18:14.21*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
18:14.42UQlevSteak_: that must have been Jesus
18:15.20Maliutaforces Katty to listen to Bread for 3 hours
18:15.20Maliuta:)
18:15.36Steak_http://pastebin.com/d64f4f42a <-- here the output of a test call from my mobile to the system
18:15.39eppigyNEIN
18:15.44*** join/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova)
18:15.51Steak_problem is that the baker was speaking in german only :O
18:16.02Steak_and ich spreche nuer ein bisschen deutsch :P
18:16.15Joe_CoThey guys, got a big problem. I updated my kernel and asterisk yesterday, everything seemed fine. Now for some reason, Asterisk won't load sip. Is there anything I should look for?
18:16.20MaliutaKatty: next on the list ... Chicago, then Air Supply
18:16.39Steak_[TK]D-Fender: http://pastebin.com/d64f4f42a <--the call log
18:17.01Ritzeriskasterisk is like SLOOW cracked out the voice back to me even locally
18:17.34UQlevJoe_CoT: what OS?
18:17.38Kattycries over lost bread.
18:17.44Joe_CoTUQlev, linux. ubuntu
18:17.44Steak_sorry katty
18:17.47Ritzeriskvmware but i had another VMware thats good so im not sure why really
18:17.53*** join/#asterisk The_Lightside (n=Lightsid@41.145.103.193)
18:18.05eppigyLets no cry over spilled guineas
18:18.08eppigynot
18:19.22Joe_CoTUQlev, wait, wtf, it just loaded
18:19.28[TK]D-FenderSteak_: add "trustrpid=yes" to your peer entry.
18:19.47MaliutaJoe_CoT: so sorry you're using a hosed debian system
18:19.47Steak_to the pstn trunk ?
18:19.52The_Lightsidehi all, just upgaded to centos 4.7
18:19.53[TK]D-Fendersteyes
18:19.59[TK]D-FenderSteak_: Yes
18:20.01The_Lightsidedahdi seems to compile, but wont load
18:20.19eppigydid you download your new kernel source?
18:20.23kc8pxyRitzerisk: vmware  for server or client?
18:20.33MaliutaCentOS 4.7 is an upgrade? I thought it was past 5.0
18:20.46The_Lightsideafik, the source is there
18:21.08The_LightsideMaliuta, it is, but some of the legacy stuff is still on 4
18:21.27Steak_[TK]D-Fender: that did not solve the problem
18:21.34Kattyi want to plan a vacation to flordia.
18:21.38Kattyand go to sea world.
18:21.40Kattyit just hit me.
18:22.08MaliutaKatty: we have a sea world here on the glod coast
18:22.12[TK]D-FenderSteak_: Reloaded your config?
18:22.13MaliutaGold Coast even
18:22.22[TK]D-FenderSteak_: Also, what # should we be seeing?
18:22.33Steak_the number of the incoming call
18:22.47Maliutawhich is?
18:22.51Steak_in this case, the 079 etcetcetc
18:22.57Steak_041 is the spa
18:23.00Steak_(pstn number)
18:23.03Steak_079 is my mobile
18:23.08[TK]D-FenderSteak_: Do it again, and do not mask #'s
18:23.24Steak_the log you mean?
18:23.29[TK]D-FenderSteak_: yes, new acll
18:23.31[TK]D-Fendercall
18:23.36Steak_ok redoing
18:23.51The_Lightsidehow much text can i paste in #?
18:23.59[TK]D-FenderSteak_: exten => 0417602xxx,1,NoOP() <- and I told you to Noop ther CALLERID
18:24.25MaliutaThe_Lightside: not much
18:24.31Maliuta~pb
18:24.32infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
18:24.40The_Lightsidei hope this isnt too much
18:24.44The_Lightsideits like 2 lines
18:25.06The_LightsideFATAL: Error inserting wct4xxp (/lib/modules/2.6.9-78.0.13.ELsmp/extra/wct4xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg)
18:25.12The_Lightsidelol, not that much
18:25.36Steak_[ http://pastebin.com/d37677c86
18:25.40Steak_[TK]D-Fender http://pastebin.com/d37677c86
18:25.54Steak_but I saw something that IMHO needs to be changed on the SPA
18:26.15*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
18:26.15The_Lightsidei get that on dahdi start
18:26.16Steak_just look at the first SIP entry in the log
18:26.19Steak_From: 0417602486 <sip:0797063898@192.168.25.30>;tag=eb7dce12c13407c4o1
18:26.29Steak_looks like the SPA is masking the real CLID
18:26.46[TK]D-FenderSteak_: same here
18:26.51Joe_CoTMaliuta, thanks for actually trying to help with my issue, instead of just making some comment about my Distro
18:26.58[TK]D-FenderSteak_: From: 0417602486 <sip:0797063898@192.168.25.30>;tag=eb7dce12c13407c4o1 To: <sip:0417602486@192.168.25.30> <- same from/to
18:27.04[TK]D-FenderSteak_: SPA = misconfigured
18:27.21*** join/#asterisk bpgoldsb (n=bpgoldsb@209.208.6.182)
18:27.25Steak_the funny thing is that incoming and outgoing calls work perfectly
18:27.37[TK]D-FenderSteak_: Sure, CID does not change that
18:27.48bpgoldsbIs there a replacement for valetparking for Asterisk 1.6?
18:28.03Maliutaand dmesg says?
18:28.03Maliutaroll over, roll over.
18:28.07Maliutaso they all rolled over and <insert_dmesg> fell out.
18:28.17MaliutaJoe_CoT: that's fine
18:28.23MaliutaJoe_CoT: anytime
18:28.43*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
18:28.52Steak_well nevermind, I'll disconnect everything and start from scratch again, as soon as my right hand will be healed (got injured today while mounting a 4u server on a defective rack)
18:29.08Steak_so this is the reason why I am so slow, I can only use my left hand :P
18:29.22MaliutaThe_Lightside: what is in your dmesg?
18:29.22MaliutaThe_Lightside: and did you rebuild the dahdi mods for that particualr kernel?
18:29.27Steak_[TK]D-Fender thanks for your help, I'll tackle this thing in the next days
18:29.29Steak_bye all
18:29.36[TK]D-FenderSteak_: You're welcome
18:30.01DavidBerGood afternoon - I am trying to solve a problem that all my extensions are busy.  I checked to see if they had DND set on and do not.  They can call out, but any incoming call shows them as busy.
18:31.08[TK]D-FenderDavidBer: You can't
18:31.12bpgoldsbI'm trying to upgrade from Ast 1.2 to Ast 1.6.  We need to be able to put a call on certain designated spots.  In 1.2, we used the ValetParking add/module.  Is there a similiar way to do this in 1.6?  Via a new module, built-in or not?
18:31.24[TK]D-FenderDavidBer: "Busy" from the phone is a response, not a "state"
18:31.42The_Lightsidehttp://pastebin.ca/1391944
18:31.51DavidBerOk - then they are responding busy
18:32.46DavidBerI had a 1.4.21.2 setup that was acting a little strange with results coming back that the line was busy - so I upgraded to 1.4.24.1
18:33.25[TK]D-FenderDavidBer: * doesn't make the other side say "busy"...
18:34.14DavidBergot sip respons 486 :)
18:37.53*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
18:38.30MaliutaThe_Lightside: are these modules you built yourself?
18:38.30MaliutaThe_Lightside: because it looks like something wasn't linked properly
18:38.30MaliutaThe_Lightside: or there is a lib missing
18:38.42Maliutathat's it ... I give up. My net connection is being sooooo shitty I can't class it as usable
18:38.44MaliutaI'm going to try and get more sleep, and then kick the ISP in the balls
18:39.56Kattywtb summer.
18:40.37Kattyand some palm trees.
18:40.42Kattytwo bahama mammas
18:40.43Kattyand a towel.
18:41.40The_LightsideMaliuta: i did compile them oon the machine, yes
18:43.35*** join/#asterisk oej (n=olle@ns.webway.se)
18:44.01Ritzeriskyou know that lady on asterisk ??/// she sounds like shes on cRACK and its playing really SLOW for some reason but the webmin is not slow at all so im not sure what it is
18:44.20*** join/#asterisk jeffgus (n=jeffgus@green.zimage.com)
18:44.42*** join/#asterisk WarptwistDK (n=chatzill@0x4dd49295.adsl.cybercity.dk)
18:44.53kc8pxyRitzerisk: asterisk on vmware guest?
18:44.57Ritzeriskis there like a CPU stat command or something i have about 2 gbs of memory to it
18:45.36Ritzeriskyea on vmware but i had it on another vmware same setup ANd same thing Really odd
18:46.04Ritzeriski meant the other Vmware was no problem ....
18:47.01kc8pxyRitzerisk:  mebbe I've just got too many spare boxen NOT to,  but i 'd say toss it on a spare REAL box(like i did), and you should have no proble.   in the meantime,  try `core show sysinfo` iirc
18:48.03*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
18:48.57Ritzeriskhmm my other vmware was fine either bad install or what but 2gbs well over ....
18:51.11The_Lightsideor use xen and pass real hardware to the gues
18:51.18The_Lightsideguest
18:51.40Ritzeriskxen never really used it do i have to have a linux box to do that
18:52.10Ritzeriskohhh shoot the only thing i can think of is 64bit windows where the vmware is installed
18:52.25Ritzeriski might have to download a 64bit version
18:55.07*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
18:55.48The_Lightsidebleh, winblows....
18:58.12lesouvageI'm testing a routine using the cityhall number during out of ffice times. For sme strane reason the channel variables dialtime and answered time don't have a value in the h extension while they have a value when I test the routine on y home number or mobile number. I'm out of options and hoping that one of you can give a hint to get this problem (an h extension that doesn't wrk all the...
18:58.14lesouvage...time) solved. Thanks in advange.
18:58.51lesouvageand excuse for all the typos.
18:58.52eppigyALLO
19:06.30*** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer)
19:07.12*** join/#asterisk ilowe (n=ilowe@modemcable230.43-82-70.mc.videotron.ca)
19:07.29JerJerhas anyone figured out a dialplan way to know if progress has been made on a call ?    Dropping a call file into dialplan - wana kinda wait around until we know progress is being made
19:08.03JerJerhmm - then again we might not make it to the dialplan until its connected -  grrrrrr
19:09.00lesouvageJerJer: have you checked the status variables?
19:10.11*** join/#asterisk a-s (n=user@92.81.117.113)
19:10.42a-sI wish to use the echo - canceller from *. Does someone use it?
19:11.06a-sI did not succeed to start it.
19:12.56JerJerlesouvage:  this box is running biz edition - i don't see anything status
19:13.00JerJeroutside of like AGI
19:13.02JerJeror AMI
19:14.32a-sI need to write an echo canceller for a soft written by me. I wish to see one working before...
19:15.08JerJerdahdi has various echo cans
19:15.11JerJernot asterisk
19:15.56a-sJerJer: I looked over dahdi. I cannot start it over a SIP call.
19:16.09UQleva-s: prohibit using loudspeakers and that's it
19:16.27a-sUQlev: I am not writing it for me :)
19:16.32a-sit's a MUST
19:16.48UQlevahh, ok
19:16.57UQlevkeep doing it
19:17.14a-sUQlev: yes, but where should I start from ?
19:18.03UQleva-s: from a client's sound card, it is useless to try to filter it out
19:18.06JerJerdahdi has nothing to do with sip
19:18.14JerJer~book
19:18.15infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
19:18.21JerJer^^^ read it
19:18.36a-sJerJer: thanks a lot. let's see...
19:19.07*** join/#asterisk ck_28 (i=CK@93.185.235.237)
19:19.13ck_28hi ppl
19:19.35ck_28How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway
19:21.07*** part/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova)
19:21.46ck_28Corydon76-dig denon file jtodd leifmadsen putnopvut Qwell russellb Kindly guys i am working on this case for more than 2 weeks can you help me
19:22.07russellbO.O
19:22.15*** mode/#asterisk [+b %ck_28!*@*] by russellb
19:22.21russellbPlease do not ever do that again.
19:23.04denonman I hate it when people do that
19:23.05russellbThis is a volunteer help channel.  People that are willing to help and have time to help are already here watching.  Please do not grab the attention of every op in the channel as your problem is not everyone else's emergency.
19:23.14a-sJerJer: waw. thanks. seems that the book explains very clearly.
19:23.25*** mode/#asterisk [-b %ck_28!*@*] by russellb
19:23.39Corydon76-digIf you want paid support, there is a tollfree number available for that.
19:23.41ck_28thanks
19:23.48leifmadsen~consulting
19:23.49infobotit has been said that consulting is Having a problem with Asterisk that you need resolved quickly? Hire an Asterisk consultant! You can find several 'for-hire' Asterisk consultants in #asterisk-consultants.
19:23.55leifmadsenand there :)
19:24.05russellbDigium also offers support subscriptions.
19:24.10leifmadsenso many options!
19:24.18ck_28its for my univ project
19:24.35russellbaw, you shouldn't have told me that.  That means you should be doing it yourself for sure, or else it's probably cheating!
19:24.36russellb:-)
19:24.47leifmadsen;)
19:24.56Corydon76-digYep, plagiarism
19:24.57ck_28ok ser concider it done :)
19:25.08lesouvageck_28: Hope you learn from the suggestions. What is your system, plain asterisk?
19:25.45ck_28sure asterisk V 1.4.24
19:26.39lesouvageGoogle for dial() cmd asterisk and do some reading. I think you will find your solution.
19:26.52*** join/#asterisk grantm (n=grant@68.142.138.4)
19:27.19ck_28lesouvage did you get my quest
19:27.43ck_28lesouvage you cant find it at google -2 weeks of searching
19:28.24[TK]D-Fenderck_28: * does not take audio from the gateway unless the gateway answers *'s call and is passing their progress inband.  In short, NO.
19:29.06The_Lightside[TK]D-Fender, what about early media?
19:29.23[TK]D-Fenderthe_Not sure on the fine points for yours...
19:30.00lesouvageck_28: Yes, it is about moh when phone is ringing. It seems that you have the m parameter in your dial string.
19:30.02ck_28[TK]D-Fender i cant edit chan_sip.c to tellhim that when * recieves 183 or 180 stop early media
19:31.03ck_28lesouvage yes _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m)
19:32.16[TK]D-Fenderck_28: Stop using "m", thats why you have MoH
19:32.44*** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db87385.pool.einsundeins.de)
19:32.44lesouvageck_28: just remove the m if you like you can replace it for something else. Just google for dial() cmd asterisk and read the info on different paramters. Some say you shouldn't use the r paramter but if you do you will hear a ring.
19:33.06ck_28[TK]D-Fender  i want to  let the caller hear MOH untill it rings
19:33.21[TK]D-Fenderck_28: what do you mean UNTIL it rings... its already RINGING
19:33.51*** part/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer)
19:35.32ck_28[TK]D-Fender some times i have a high pdd so i want to cover the dead air by moh
19:35.46lesouvageck_28: more than you already have is not available as an answer to your question.
19:37.56*** join/#asterisk smbrienz (n=guest@host202-204-dynamic.1-87-r.retail.telecomitalia.it)
19:39.39Kattyhas...
19:39.43Kattystrawberry
19:39.46Katty...limeaide
19:39.57[TK]D-Fenderck_28: You Early media should be "answered", not "ringing".
19:40.16*** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43)
19:41.03pmhaddad-workok, so i have asterisk 1.6.0.9 installed from a tarball, and i just compiled and installed dahdi and dahdi tools from svn, that all works fine, i added the rcbfx module to /etc/dahdi/modules, and restarted dahdi and asterisk
19:41.09ck_28[TK]D-Fender i should answer the call and put him on hold and then dial
19:41.17pmhaddad-workbut how can i do like "dahdi show channels" from the asterisk cli?
19:41.41pmhaddad-workdo i have to run dahdi_cfg first?
19:42.16[TK]D-Fenderck_28: Answer what call?  You're the one CALLING
19:42.53[TK]D-Fenderpmhaddad-work: * has to be compiled AFTER DAHDI
19:43.21ck_28[TK]D-Fender scenario is like that  UserA--->*--->Termination GW ---->User B
19:43.52*** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net)
19:44.06pmhaddad-work[TK]D-Fender, yep, just noticed that
19:44.09pmhaddad-workrecompiling now
19:44.25[TK]D-Fender[15:37]<ck_28>what i need is How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway
19:44.44[TK]D-Fenderck_28: As this sys, you DON'T.  * can play * INSTEAD of ringing.  not BEFore, INSTEAD <-
19:46.05ttl-does anybody has experience in configuring the Linksys SPA3102 ATA please?
19:46.38[TK]D-Fenderttl-: www.voxilla.com <- plenty of guides there
19:47.10ttl-i'm trying to get it to work for 5 days, followed all kinds of guides
19:47.38ttl-[TK]D-Fender: ok i'll try www.voxilla.com again
19:48.11ttl-this thing drives me crazy
19:48.38djMaxThis line: exten=>9565,1,Transfer(ZAP/1115551212) (number was changed) does that look proper?
19:48.47ck_28when I used directrtpsetup=yes I heard the MOH and the ring back tone together when I used directrtpsetup=yes I heard the MOH and the ring back tone together
19:48.49djMaxit just seems to run right over it.
19:49.16[TK]D-FenderdjMax: No, that is certainly not correct.
19:49.25djMaxok, what'd I blow up?
19:49.32djMaxthe Zap invocation?
19:49.43[TK]D-FenderdjMax: does taht look like the proper way to reference a CHANNEL or GROUP of channels to you?
19:50.03djMaxI've been through too many asterisk, freepbx, and sundry configs to remember anymore.
19:50.10djMaxI thought it was.  But I go dig now.
19:50.56djMaxused to be easy to see examples in the configs, but with FreePBX is variabled-to-death
19:52.30Kattydeposits strawberry limeaide on jaytee's desk.
19:53.24lesouvageCan there be any reason that channelvariables doesn't get a value if a channel is set up to certain numbers while with other numbers they got here value as expected. ( ${DIALEDTIME} and  ${ANSWEREDTIME} )
19:53.34*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:53.49ck_28[TK]D-Fender by the way why * makes two invites when he want to make a call
19:54.14Kattywhat does lesouvage mean?
19:54.16[TK]D-Fenderck_28: Could be a re-invite through the gateway
19:54.36ck_281st invite to the GW and when the GW sends ACK * resend an invite to the caller
19:55.44ck_28lesouvage even on bye message same behavior ,that because its a b2bua?should be acting like that
19:56.23lesouvageKatty: I'm testing a routine that uses the ${DIALEDTIME} and ${ANSWEREDTIME} variables in the h extension to get some sort of logging straight. I thought it works like a charm but this evening I found out that the variables doesn't always got values.
19:56.47[TK]D-Fenderlesouvage: "h" often loses access to channel variables
19:56.50Kattylesouvage: what does that have to do with my question? ^_-
19:56.57*** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net)
19:57.00Kattylesouvage: oh, i see what you did.
19:57.06Kattylesouvage: you parsed my question imporperly.
19:58.06Kattys/imporperly/improperly/
19:58.06Kattylesouvage: define lesouvage.
19:58.06ck_28[TK]D-Fender you said chan_sip can you pleasw specify in which part
19:58.07ck_28or just chan_sip
19:58.07Kattythere are parts ot chan_sip now?!
19:59.04*** join/#asterisk jtexter3 (n=jtexter3@72.242.229.213)
19:59.46*** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net)
19:59.47*** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net)
20:00.16djMaxOk, so Transfer(DAHDI/g0/12125551212)?
20:00.51[TK]D-FenderdjMax: Looks a lot more valid... assuming you're running DAHDI now and not Zaptel
20:00.59djMaxyes, * 1.6
20:01.10lesouvage[TK]D-Fender: I know, but it is kind of strange that must of the times the routine works and sometimes it doesn't because the variables don't have a value. Is there any change that it is because a read() cmd timed out.
20:01.17[TK]D-FenderdjMax: You're samples should be labeled "mixed nuts" :p
20:01.18*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
20:01.20[TK]D-Fenderyour*
20:01.38djMaxsamples?
20:01.50*** join/#asterisk seb- (n=seb@li30-51.members.linode.com)
20:01.58seb-[TK]D-Fender: are you around?
20:02.07djMax(i'm sure that was a joke that flew by me)
20:02.55*** part/#asterisk smbrienz (n=guest@host202-204-dynamic.1-87-r.retail.telecomitalia.it)
20:03.04djMaxHmmm... Received an unknown call with DID set to <x>.  I'm assuming that means the transfer failed.
20:03.06*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
20:04.35pmhaddad-workok now when i start dahdi up it loads everything but "rcbfx: error"... "no hardware timing source in /proc/dahdi" i looked in syslog, and it shows dahdi loading, but nothing on rcbfx
20:05.05Kattythis makes me want to do shots of vodka.
20:06.09*** part/#asterisk cr4z3d (n=cr4z3d@unaffiliated/cr4z3d)
20:06.22telnettechKattty: I am ready too....miserable day
20:06.33Kattytelnettech: let's go have drinks at applebees!
20:06.47Kattytelnettech: we can drown the day with long iceland ice teas.
20:06.53Kattyiceland?!
20:06.57Kattysighs
20:07.06Kattyi give up.
20:07.10telnettechim a rim and coke guy
20:07.22pmhaddad-workis a martini guy
20:07.23telnettechsee i did the same thing.....rum and coke
20:08.25Kattycan't do rum and coke anymore.
20:08.32Kattybad memories.
20:08.41Kattygot real sick.
20:08.50Kattybut good memories before that!
20:09.27ck_28why asterisk sends to invite to complete a method
20:09.36ck_28invite or cancle for example,if the gw sends him 486 --* will reply to the Gw with ACK and then send and invite to the sip client
20:09.42djMaxIf at priority 1 I have Transfer(|j), and it fails, it should go to priority 102 right?  It seems to just quit (I also have n after step 1 to play some monkeys)
20:10.03telnettechwell you drink the long island and i will drink rum and coke
20:10.03Kattyi've never used transfer() before for anything.
20:10.20Kattyoh wait
20:10.22Kattyi have
20:10.27djMaxI assume it's the only way I can get blind transfer
20:10.39Kattyit was a call file, that played a prerecorded message to the number dialed. and then transfered them into a 911 conference call.
20:11.04bpgoldsbI'm trying to upgrade from Ast 1.2 to Ast 1.6.  We need to be able to put a call on certain designated spots.  In 1.2, we used the ValetParking add/module.  Is there a similiar way to do this in 1.6?  Via a new module, built-in or not?
20:12.08KattydjMax: can you do a transfer to an internal extension?
20:12.10pmhaddad-workanyone got an idea on my dahdi error? i've googled to no avail
20:12.24KattydjMax: does this just happen when transfering to dahdi channel?
20:12.30eppigy8[]
20:12.35Kattyhi dave.
20:12.39Kattyyou should join us at applebees.
20:12.40eppigyHI
20:12.43djMaxtrying.  The overall goal is to transfer an inbound DAHDI call to an outbound one.
20:12.46eppigyi would like to
20:12.47Kattywe are drowning the day away.
20:12.51djMaxi.e. to log the call and shove it off.
20:12.52eppigyNICE
20:12.58eppigyi could go for some whiskey
20:13.03KattydjMax: oh.
20:13.14KattydjMax: that should be accomplished by a simple Dial
20:13.33Kattypastebins some stuff
20:13.35djMaxw/o taking 2 chans?
20:13.40Kattyoh.
20:13.44Kattyno. it will take two channels.
20:14.02djMaxyeah, would rather not that.
20:14.10Kattysounds like..flashhook.
20:14.14Kattyor whatever it's called, i forget.
20:14.45Kattyit's something the samsung 7100 boxes are supposed to be able to do, but the telco has to let you do it
20:14.53*** join/#asterisk mweichert (n=mweicher@216.13.154.21)
20:15.01djMaxXO says they have, but of course I don't believe it
20:15.25Kattyidk how to do it with asterisk
20:15.43Kattynever needed to
20:15.47mweicherthello. I'm trying to write an extension which dials a number a plays back a sound...
20:15.51mweichertso far I have the following:
20:15.52mweicherthttp://rafb.net/p/wfjL1O14.html
20:16.16mweichertI can get a sound to play - but it doesn't place the call
20:16.33Kattymweichert: err that won't work
20:16.41pmhaddad-workhttps://support.rhinoequipment.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=92&nav=0,34 <-- lets see if this helps
20:16.41eppigySTUNTIN LIKE MY DADDY
20:16.47lesouvagemweichert: just check the M parameter of the dail() cmd.
20:16.48Kattymweichert: because it won't go to the next step after dial, until the conversation is done.
20:17.04Kattymweichert: what exactly are you trying to accomplish?
20:17.26Kattymweichert: play a brief audio snippet at the beginning of a call?
20:17.47mweichertKatty: when I dial an extension, I want a call to be initiated and when they answer, a message played back to them
20:18.24mweichertfor example, when I dial extension 1111, I want the number 555-555-5555 dialed and a sound played back to them when they pickup
20:19.01mweichertis this possible?
20:19.03lesouvage<PROTECTED>
20:19.10Kattymweichert: well.
20:19.19Kattymweichert: you mean like...
20:19.25Kattymweichert: please hold while i connect your call?
20:19.42Kattymweichert: and then it calls 555-555-5555
20:19.54jayteejust got back to my desk.
20:20.04jayteeKatty, thanks for the strawberry limeade
20:20.46mweichertKatty, no... calls 555-555-5555, then when they pick up plays "You have just been pinged"
20:20.51*** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net)
20:20.51*** mode/#asterisk [+o mog] by ChanServ
20:21.02Kattymweichert: that would be a call file then.
20:21.15Kattymweichert: ext 1111 would execute a system() command which copies a call file, and creates a call
20:21.23mweichertwhere can I get some documentation that lists what applications such as Dial() are available and what parameters they accept?
20:21.26Kattymweichert: the call file would dial the number, and then play an audio file to them
20:21.42mweichertKatty, yes, that's what I'd like to do
20:21.43djMaxhmph, so I'm getting transfer failed (better than unsupported I suppose)
20:21.54Kattyinfobot: call file?
20:21.55infobotACTION looks around and then screams out file as loudly as possible
20:22.01Kattyfile: disregard that.
20:22.36Kattymweichert: have a look at this: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
20:22.59mweichertah, I see the Asterisk - The future of Telephony book has "Appendix B. Application Reference"
20:23.15Kattymweichert: example 6 will probably be most useful.
20:23.56_ShrikEmweichert: core show application dial
20:25.07Kattymweichert: per example six... they put in extension 1 (but you could make it 1111), and it copies a call file from a directory into the spool directory which makes asterisk excute it much like an email server would send an email if you dumped it into the spool folder.
20:25.22Kattymweichert: the call file says to go to extension 10 in context [pa-call-file]
20:25.43Kattymweichert: in extension 10, you do some like 10,1,Dial(somephonenumber) 10,n,Playback(someaudiofile)
20:25.57Kattymweichert: then hangup or transfer it or do whatever
20:27.03Kattymweichert: we use it for 911 here. someone dials 911, andasterisk calls the receiptionist and transfers her into a meetme with the person who called 911
20:27.08*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
20:30.24*** join/#asterisk propellerhead (n=yogurt2u@host110.190-136-62.telecom.net.ar)
20:32.32Kattyeppigy: what's for dinner.
20:32.32*** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix)
20:32.38eppigyhrmmmmm
20:32.38Kattybbkt-trix: ohai
20:32.42eppigyi do not know
20:32.45Katty:<
20:32.48eppigyi know
20:32.50Kattywe are going to starve! /tear
20:32.57KattyTHINK OF THE CHILDREN
20:33.01eppigyI am thinking a wendys double cheeseburger
20:33.03pmhaddad-workomg i'm annoyed
20:33.07eppigyI am on a wnedys kick
20:33.09Kattypmhaddad-work: whatsammater
20:33.11bbkt-trixKatty: ohai  :-)
20:33.12eppigythey are so good
20:33.23pmhaddad-workKatty, trying to get dahdi working with a rhino card
20:33.27pmhaddad-workwith rcbfx
20:33.32mweichertKatty, thanks - I'm gonna give a couple of things a try
20:33.35Kattyyou should talke to jameswf
20:33.37*** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv)
20:33.37Kattyhe works for rhino
20:33.39djMaxboy, I have no idea how one goes about debugging the Transfer app.
20:33.41pmhaddad-workKatty, followed: https://support.rhinoequipment.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=92&nav=0,34
20:33.48pmhaddad-workand now i can get the module loaded
20:33.50pmhaddad-workbut
20:34.13pmhaddad-workthe dummy is still the only thing loaded
20:34.19pmhaddad-workwhen i do dahdi show channels
20:34.21aiksa[LV]just a quick question: is bri_cpe signalling available in main dahdi release or is it something I should use bristuff for
20:34.25aiksa[LV]?
20:34.29Kattypmhaddad-work: mmmyeah i'd talk to jameswf
20:34.31pmhaddad-workand syslog has a LOT of rcbfx errors
20:34.45aiksa[LV]if tzafrir_laptop were around he would be able to answer this under a sec :)
20:34.59Kattydon't use bri.
20:35.00pmhaddad-worki don't see him in the channel :|
20:35.21djMaxok, so Dial(Local/501) works, but Transfer(Local/501) does not.  Does that tell anybody anything interesting?
20:35.26Kattypmhaddad-work: hrmm.
20:35.29Kattypmhaddad-work: so it seems.
20:35.32tzafrir_laptopaiksa[LV], bri_cpe is now also part of asterisk >= 1.6.0
20:35.34Kattypmhaddad-work: hes usually here tho
20:35.45aiksa[LV]tzafrir_laptop: but starting with 1.6
20:35.46aiksa[LV]?
20:35.53KattydjMax: can you transfer to a SIP?
20:36.05aiksa[LV]and you are online :)) hi friend
20:36.14pmhaddad-workKatty, could you take a look at the errors in syslog if i pasted them?
20:36.26pmhaddad-worknot sure how hard/easy this is to fix, but its mission critical
20:36.27tzafrir_laptopthis is unrelated to dahdi
20:36.36tzafrir_laptopit is userspace
20:36.37aiksa[LV]Katty: were I live and work in most cases bri is the obly option
20:36.43Kattypmhaddad-work: go visit #asterisk-consultants
20:36.48BlargMaN00mweichert: you'd probably be better off having the extension call a script that produces a call file...  I think that would be easier for you to implement...
20:36.50aiksa[LV]tzafrir_laptop: ok.
20:37.03aiksa[LV]tzafrir_laptop: through libpri?
20:37.14Kattypmhaddad-work: i don't use rhino equipment, so i'd be useless for you.
20:37.19Kattypmhaddad-work: someone in there might know.
20:37.25Kattypmhaddad-work: especially if it's urgent
20:37.28pmhaddad-workok
20:37.33pmhaddad-workthanks Katty
20:37.34djMaxNo, Dial(SIP/2001) works, Transfer(SIP/2001) does not.
20:37.38pmhaddad-worki'll drink a beer for you tonight
20:37.49KattydjMax: what if you make an auto attendant.
20:37.55KattydjMax: and then it answers, and then transfers
20:38.10mweichertBlargMaN00, call a script... what kind of script? AGI?
20:38.15djMaxyou mean just answer the call first?
20:38.28Kattyyes. like exten => 444,1,goto(autoattendant,s,1)
20:38.39Kattyand then [autoattendant] s,1,Answer s,2,Transfer(SIP/4001)
20:38.51Katty<PROTECTED>
20:39.04lesouvagemweichert: if you give me a moment I can pass something that i working.
20:39.21mweichertlesouvage, wow, I'd really appreciate that - thanks!
20:39.26tzafrir_laptopaiksa[LV], this requires libpri >= 1.4.4
20:39.46*** join/#asterisk UQlev (n=yuriy@91.184.221.31)
20:40.02BlargMaN00mwichert: no, just a bash script that will echo everything you need into a tmp call file, and then --> move <-- it to the /var/spool/asterisk/outgoing dir
20:40.29aiksa[LV]tzafrir_laptop: ok, many thanks. Does that mean that with libpri > 1.4.4 I should be able to compile something rather fresh from 1.4 branch and still get bri signalling for chan_dahdi?
20:40.50djMaxnope, answer doesn't seem to have any affect
20:40.58djMaxeffect? can never freakin' get that right
20:40.58tzafrir_laptopaiksa[LV], no. for 1.4 you still need bristuff
20:41.05KattydjMax: pastebin me the cli
20:41.10aiksa[LV]tzafrir_laptop: ok. that explains it.
20:41.23aiksa[LV]many thanks
20:41.33*** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe)
20:43.22djMaxhttp://pastebin.com/m413308c3
20:45.23Kattywhat's 1 + 101
20:45.38djMax102
20:45.46Kattywhere is your 102?
20:45.57djMaxdon't have one, shouldn't it be 2+102?
20:46.00djMaxTransfer is step 2
20:46.07Kattylabel it as 2
20:46.09lesouvagemweichert: see http://www.pastebin.be/17827
20:46.35djMaxsame diff
20:46.35lesouvageMaybe not the niciest way to do it but it is working without any problems.
20:46.38KattydjMax: forget the transfer status, just play weasels
20:46.45Kattyand take out the other ns
20:46.45djMaxit doesn't even go there anyhow, just prints failure.
20:46.46*** join/#asterisk profXavier (n=jezus@unaffiliated/neverblue)
20:46.52profXavierwhere can I download 1.4.4 ?
20:47.02Kattyrecreates aa on her box.
20:47.19djMaxit falls straight through the Transfer line basically.  I assume it'd just play weasels.  I try now.
20:47.49mweichertlesouvage, why is 'moving' the file so important?
20:48.02mweichertwhy not just write to the outgoing/ directory?
20:48.10*** join/#asterisk cesar_CR (n=cesar@200.91.75.67)
20:48.51lesouvagemweichert: the /var/spool/asterisk/outgoing is checked so agrassively that if you copy the file there i a change that the file is processed before completion with unpredictable results.
20:48.51OctothorpeI know this is #asterisk, not #freepbx... but, that said, FreePBX does seem to be a fairly popular frontend to *. Hopefully someone would be able to help. If I wanted to move the FreePBX content from /var/www/html to, say, /var/www/html/freepbx, I know I have to edit some FreePBX config file like amportal.conf (for example) or somesuch, (1) so that it doesn't break completely and (2) so that future updates don't go sideways.
20:49.30djMaxok, so doing that from an outbound line to a sip target just quits after Transfer().  Doesn't go to step 3 or 103
20:49.45djMaxspawn extension exited non-zero
20:50.03KattydjMax: okay
20:50.07KattydjMax: here's a working example
20:50.09Kattypastebins.
20:50.13lesouvagemweichert: one thig that can happen is that the callfile isn't removed after setting up the call and the call is set up time after time.
20:51.34lesouvagemweichert: This once cost me my complete sms credit saldo because the callfile keeps on sending sms messages to my phone until I run out of credits while I couldn't do anything.
20:52.02jayteequittin time, be back later
20:52.55lesouvagemweichert: make sure that you have a proper linefeet/return on the last line of the call file because without strange things might happen.
20:53.02djMax(did I miss the pastebin?)
20:53.39mweichertlesouvage, heh, that's quite the problem to have occur!
20:56.22*** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net)
20:56.44djMaxeither this place just went crazy quiet or I've died.
20:57.31lesouvagemweichert: does the pastebin make sense to you?
20:57.37mweichertlesouvage, I don't think the call file is being loaded.
20:57.44mweichertthere is nothing in the log
20:58.26mweichertwhat's the ${PAD} variable?
20:59.40djMaxKatty are you messing with me or did you actually pastebin?
20:59.43ttl-i think my spa3102 is broken
20:59.46ttl-dunno
20:59.57lesouvagemweichert: You can start without th mv line and see if the call file is generated. The call file points to a context/extension/priority that has to exist. The ${PAD} variable holds the complete file name wit the directory included.
21:00.15mweichertand when I want to dial a number using one of my ITSP trunks, am I right to use the following as channel: SIP/name-of-trunk/15551234567 ?
21:00.36ttl-when pstn line rings it does not send anything to asterisk, even monitoring with wireshark now
21:01.14KattydjMax: sorry got a call.
21:01.17KattydjMax: http://pastebin.com/m1524dc7e
21:01.28djMaxthx, thought it was a late APril fools joke
21:01.29KattydjMax: transfer() doesn't support dahdi/zap transfers
21:01.38ttl-i don't know what to try anymore
21:01.42lesouvagemweichert: it is working for me, but you have to adjust it to your setting with a proper local channel or trunk (sip/iax2/dahdi etc.) and pointing to a context/extension/priority that is available.
21:01.54djMaxBut there was all this discussion of * support TBCT... how else would that work?
21:02.22mweichertlesouvage, yes, I know I have the context/extension/priority correct - I'm just not sure about the channel
21:02.38ttl-thinking of throwing the damn thing into the bin...
21:03.03BlargMaN00mwichert: yes, you would use SIP/name-trunk/phone#
21:03.22ttl-any ATA devices which are good and compatible with asterisk 1.6 ?
21:03.47KattydjMax: probably uses a different command()
21:04.42rob0Throw it here ttl- I've been called a garbage disposal before.
21:04.52*** join/#asterisk botox93 (n=botox93@213.221.82.242)
21:05.24mweichertBlargMaN00, do I have to include the pound symbol # at the end?
21:05.52KattydjMax: maybe you're thinking of ChannelRedirect?
21:05.55BlargMaN00mweichert: no...  unless your dialplan requires it, then yes
21:05.57ttl-rob0: It's just one week old, but it's driving me crazy!
21:06.19djMaxI'll check.  I see this too: http://wiki.sangoma.com/Asterisk-FAQ#TBCT
21:06.23rob0Are you new at this, or have you successfully done it before?
21:06.25ttl-rob0: There is even no decent documentation
21:06.38djMaxzapata.conf is now dahdi.conf?
21:06.42rob0I found a pretty good PDF for my SPA2000.
21:06.44BlargMaN00ttl-: such a young ATA...  maybe it just hasn't learned yet...  8)~
21:07.07rob0and my SPA 2000's work fine
21:07.07KattydjMax: yeah i think that's when you use ##
21:07.13KattydjMax: if it's setup in features.conf
21:07.15KattydjMax: could be wrong tho
21:07.43rob0(under * 1.4 and 1.6, the version is not a factor)
21:07.46KattydjMax: sangoma people are readily accessible for questions like that
21:07.58djMax(I don't even know what Sangoma is)
21:08.03ttl-BlargMaN00: yeah, lol, dunno what i'm doing wrong anymore, maybe it's broken, don't get it anymore...
21:08.09Kattyumm
21:08.11Kattysangoma is a brand
21:08.13Kattyof cards.
21:08.22Kattylike digium is another brand.
21:08.25rob327i just installed the asterisk-gui with svn and i'm getting stuck in some sort of loop while its loading 'creating a config file to store GUI preferences'...right now i am connecting through a firewall however, forwarded a port to the host machine's port 8088
21:08.26Kattyand rhino.
21:08.26djMaxI have a digium card, wonder if perhaps it doesn't support it
21:08.28rob0oh I thought it was a type of cancer.
21:08.35KattydjMax: you could ask Qwell
21:08.41KattydjMax: he would probably know
21:08.43rob327anyone know if that would make a difference?
21:08.51rob327or is this a common issue?
21:08.53rob0Lung sangoma ... brain sangoma ...
21:08.57mweicherthere is what I have so far - I feel I'm close:
21:08.57mweicherthttp://pastie.org/446582
21:08.58Kattyrob0: we don't do gui's here.
21:09.00Kattyoh
21:09.06Kattyrob327: we don't do gui's here. they're just problematic.
21:09.22mweichertlesouvage, can you have a look here please: http://pastie.org/446582
21:09.24Kattyrob327: i would suggest contacting whoever developed it for support.
21:09.31rob327Katty: thanks
21:09.32*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
21:09.40BlargMaN00ttl-: are you using it to connect an analog phone, or analog phone line??
21:09.47Kattyhi anthony
21:10.30*** join/#asterisk nicoAMG (i=asgalt@201.203.96.42)
21:10.47ttl-BlargMaN00: both but i'm now trying to get the pstn work
21:10.53rob327i didn't want to go with a gui myself but i'm getting pressure from my boss :\
21:11.06rob327it does seem to be more problematic than it should be
21:11.34djMaxQwell, out there?
21:11.38ttl-BlargMaN00: but when the analog line rings i can't get the spa3102 to send it to asterisk, it just does nothing, monitored with wireshark
21:11.55rob0Doc, I have a large sangoma growing on my middle digium! What can I do?
21:12.14*** join/#asterisk jblack (n=jblack@pool-71-181-243-204.sctnpa.east.verizon.net)
21:12.15Kattyyour doc is a phoney
21:12.23BlargMaN00ttl-: you might have to register the fxo side as a seperate trunk in sip.conf...  not too sure tho
21:12.28rob0ba-da BOOM ching
21:12.40BlargMaN00ttl-: does it allow for that??
21:13.38ttl-BlargMaN00: don't think so, but the device is not sending anything to anywhere, i looked with wireshark
21:14.11BlargMaN00ttl-: found this article --> http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
21:14.26BlargMaN00ttl-: i have never worked with one personally, so see if that helps you out...
21:14.54ttl-BlargMaN00: followed that one without any success
21:15.34ttl-BlargMaN00: Thanks anyway :)
21:15.56BlargMaN00ttl-: oh...  np...  i'll keep looking...
21:17.06BlargMaN00ttl-: I think this is one of those times where I would have to see it to fix it...  8/
21:17.50rob0Did ttl- not download the PDF docs from linksys/sipura?
21:18.07rob0Is the device getting an IP address?
21:18.16rob0Can you access its web interface?
21:18.28djMaxok, so using "##", does it make sense that only 1 dahdi channel would be consumed by a transfer?
21:18.38djMaxI would've assumed it was 2 or 0
21:18.58ttl-BlargMaN00: Thanks
21:19.11mweicherthow do I use the originate command in CLI to place a call from 1102 to 1101?
21:19.18ttl-rob0: yes it has a static i ip
21:19.34ttl-rob0: otherwise i would not be able to configure it
21:19.51ttl-rob0: it has like hundreds of options
21:19.56ttl-lol
21:20.31rob0And the defaults are fine for most uses, at least for getting it up and running.
21:21.20ttl-rob0: Which PDF docs?
21:21.35ttl-rob0: There are no pdf docs for that device
21:21.42lesouvagemweichert: console dial extension@context
21:22.05rob0I don't have your device, but I did find them for my device, which is probably older than yours.
21:23.20mweichertlesouvage, "No such command console dial 1111@ext-local"
21:24.02BlargMaN00mweichert: the problem is with your dialplan, is that you are trying to go to priority 1 when you only have n priorities...  You must have a Priority 1 before anything else...
21:24.44BlargMaN00mweichert: change exten => 1111,n,Answer() to exten => 1111,1,Answer()
21:25.31mweichertBlargMaN00, makes no difference... that's what I had originally, but then I noticed examples had the .call file take action on priority 1
21:28.16lesouvagemweichert: it is working for me with console dial 505@inbound
21:29.00mweichertlesouvage, you're entering that command in Asterisk CLI ?
21:29.06lesouvageyes
21:30.03*** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net)
21:30.10BlargMaN00mweichert: you are creating the call file in another directory, and then moving it into /var/spool/asterisk/outgoing??
21:30.12*** join/#asterisk SparFux (n=raoul@e182025245.adsl.alicedsl.de)
21:30.55SparFuxHi! I have an ATA and an analogue phone connected to it. Astoundingly the time of the phone is automatically set when a call arrives, but it is one hour early. What could be the cause of that?
21:31.25mweichertBlargMaN00, yes
21:31.47BlargMaN00mweichert: are you running asterisk as root or as another user??
21:32.01eppigystarving
21:32.11mweichertBlargMaN00, asterisk
21:32.12Katty:<
21:32.27eppigyi gotta stay till this att dude gets here
21:32.37eppigycause we had another internetr circuit installed today
21:32.51eppigyand the dude must have knocked power out to one of my smart jacks
21:32.51Katty:<
21:32.52eppigy:[
21:32.55BlargMaN00mweichert: are the files disappearing from /var/spool/asterisk/outgoing??  or are they just sitting there??
21:33.02Kattymy feets hurt.
21:33.03eppigyhis children will pay
21:33.19mweichertBlargMaN00, they are just sitting there. And I wouldn't want them to go away.
21:33.21Kattyshouldn't have worn heels today.
21:33.37eppigyme2
21:33.41djMaxok, so lemme try a different version of this boondongle: who could/would I pay to solve this TBCT problem?
21:33.45Kattythey're such a pain, aren't they
21:33.57eppigyyesh
21:34.00BlargMaN00mweichert: if they are working correctly, the will disappear as soon as they go in there...  that means that * read and processed them, and is done with them...
21:34.02eppigyits like man
21:34.07eppigybeing sexy comes with such a price
21:34.15mweichertBlargMaN00, hmm...
21:34.18BlargMaN00mweichert: if they are not disappearing, that means * can't read them at all...
21:34.26djMax(separately, if ## works, what command simulates ##?)
21:34.35mweichertBlargMaN00, I'll try restarting Asterisk
21:34.45Kattyeppigy: wonderboy.
21:34.51BlargMaN00mweichert: do a 'ls -l /var/spool/asterisk/outgoing' and pastebin for me...
21:35.18eppigythanks!
21:35.44Kattywhatfordinnerwhatfordinnerhrmmm.
21:35.59eppigyI will eat ATT guys children
21:36.08eppigythat is the only way to solve both problems
21:36.12Kattyeppigy: crazy eddie.
21:36.14*** part/#asterisk SparFux (n=raoul@e182025245.adsl.alicedsl.de)
21:36.19mweicherthttp://pastie.org/446613
21:36.28eppigyTHE SINS OF THE FATHER
21:36.30mweichertBlargMaN00, btw - thanks for your help with this. I appreciate it
21:36.57lesouvageI found out that if othing is touched, no menu option, no hangup on the called side just a time out then the channel variables dialedtime and answered time aren't set and no info is available for counting the length of the call etc.  Does this make any sense?
21:37.23*** join/#asterisk BadHAL (n=nn@66.194.174.11)
21:37.39BlargMaN00mweichert: try 'chmod asterisk:asterisk /var/spool/asterisk/outgoing/1111.call' and see if it disappears...
21:39.57mweichertnope, still there. Do I have to enable the outgoing directory somehow?
21:40.03mweichert(even restarted)
21:40.26eppigyWENDYS
21:40.46Kattyeppigy: http://www.junkfoodnews.net/baconator.JPG
21:41.01ttl-goodnight
21:41.08eppigyway too much bacon :[
21:41.16eppigyI am not a fan of bacon on burgers
21:41.27eppigyI like bacon with eggs 8[]
21:41.56BlargMaN00mweichert: can you pastebin the CLI output of what happens when you dial the extension that spawns all of this??
21:42.06Kattyeppigy: http://d1.biggestmenu.com/00/00/24/5a0321911779c1ee_m.jpg
21:43.54mweichertBlargMaN00, aha, in the log file:  Unable to request channel SIP/link2voip/11234567890
21:43.57eppigyDUDE
21:44.05eppigythat looks really grood
21:44.11djMaxis switchtype=national at all similar to 5ess? i.e. blind transfer only works w/5ess seemingly, not sure how screwed I am
21:44.27eppigywell
21:44.30BlargMaN00mweichert: ok...  so now you just need to put the real info in there, and it should work...
21:44.32mweichertBlargMaN00, the file is gone now
21:44.49eppigyi have found that in most cases 5ess and national are interchangeable
21:44.53Kattyeppigy: http://dixiedining.files.wordpress.com/2008/09/popeye.jpg
21:45.09djMaxthx, exactly what I was wondering.
21:45.10BlargMaN00mweichert: also, make sure you change back to 'exten => 1111,1,Answer()'
21:45.15eppigySNAP
21:45.15mweichertBlargMaN00, yes, I put the real info in -  just didn't post that number.  link2voip is the name of my outgoing route defined in FreePBX
21:45.27eppigyARE YOU TRYING TO DRIVE ME IN TO A BEV-RAGE
21:45.43BlargMaN00mweichert: oh, ok...
21:45.57mweichertBlargMaN00, is that not correct?
21:46.21djMaxIt's strange because the docs seem to say blind transfer is "automatic" with 5ess, which I don't see why that'd be a good thing
21:46.55Kattyeppigy: http://www.mrpizzaiolo.com/wp-content/uploads/2008/05/cc_pizzabeer.jpg
21:47.15BlargMaN00mweichert: you are going to have to show me what your CLI output looks like for me to be able to finish helping you...  with out that, i won't be able to see where the issue is now...
21:47.33mweichertBlargMaN00, okay, thanks. What should the priority be in the .call file?
21:48.29Kattyeppigy: http://www.atoasttothese.com/wp-content/themes/roundbox/roundbox/images/posts/grilled-cheese.jpg
21:48.43BlargMaN00mweichert: the call file can stay like it is, you just need to change the first line in your context to have a priority 1 instead of priority n
21:49.51mweichertwaiting for .call file to disappear
21:51.26Kattyeppigy: http://www.littleshamrocks.com/image-files/baked_pineapple_ham.jpg
21:52.02mweichertthe .call file disappeared... when I dial extension 1111, this is what is displayed in Asterisk CLI:
21:52.02mweicherthttp://pastie.org/446641
21:52.33Kattyeppigy: ;)
21:52.51BlargMaN00mweichert: then according to the CLI, it should be working correctly...
21:53.03eppigy8[]
21:53.06eppigyD:
21:53.28mweichertBlargMaN00, the call is initiated as the .call file specified :(
21:53.35eppigyi do not know how to spell crimanitley
21:53.36mweichertBlargMaN00, no call is initiated as the .call file specified :(
21:53.43eppigycrimanittley
21:54.07*** join/#asterisk ingenius (n=alektro@host216.201-253-178.telecom.net.ar)
21:54.35BlargMaN00mweichert: are you testing this with a cell phone, or a phone that is also attached to the PBX??
21:54.56mweichertBlargMaN00, a phone that is also attached to the PBX
21:55.44*** join/#asterisk juanIMP (n=juan@200.71.41.22)
21:55.50*** join/#asterisk UQlev (n=yuriy@91.184.221.31)
21:55.54BlargMaN00mweichert: then this is more than likely why it is not working...  try using 'Channel: Local/1234' instead...  (replacing 1234 with a valid phone extension)
21:56.42mweichertBlargMaN00, sorry, the number I'm wanting the .call file to dial is a land line using ITSP trunks...
21:57.03mweichertbut to initiate extension 1111, I'm dialing it from a PBX-attached phone
21:57.47BlargMaN00mweichert: oh...  so you aren't using the call file???
21:58.54mweichertBlargMaN00, I *assume* I'm using it. I thought that if I dial extension 1111, that the .call file will be read at the priority specified in the file, and dial the channel specified in the file. Is that incorrect?
21:59.20BlargMaN00sighs...
21:59.30BlargMaN00mweichert: i understand where the disconnect is now...
21:59.58mweichertfeels like an idiot
22:00.19BlargMaN00mweichert: what happens, is that you want to dial extension 1234 so that it calls 1234567890 and plays the message from extension 1111  <-- is this correct??
22:01.03mweichertwhen I dial extension 1111, I want 123456789 to be dialed and a message played to them
22:01.33BlargMaN00mweichert: ok...  now i am understanding completely....  we'll get you fixed up now...
22:01.47mweichertsorry
22:02.14BlargMaN00mweichert: no worries...  i wasn't seeing the big picture, but now i do, and now I can make sure that you get the help you need...  8)~
22:02.22mweichert:)
22:02.54BlargMaN00mweichert: can you pastebin your dialplan, so I can see how you are generating the call file???
22:03.20mweichertBlargMaN00, I'm manually creating the .call file and then manually moving it
22:03.58BlargMaN00mweichert: how are you going to create it when you actually put this into a production environment??  we might as well go ahead and iron that part out too
22:05.35mweichertBlargMaN00, does it have to be created for every time extension 1111 is dialed?
22:05.58mweichertbtw, this is the contents of my dialplan and such (though I changed the priority as you said)
22:05.58mweicherthttp://pastie.org/pastes/446582
22:06.09BlargMaN00mweichert: yes...  each call file represents one phone call attempt...  it can not make multiple calls...
22:06.11mweichertI actually have exten => 1111,1,Answer()
22:06.23mweichertBlargMaN00, ah ok - I didn't realize that.
22:06.47*** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net)
22:07.03BlargMaN00mweichert: depending on how clean you want your dialplan, you can use a script, or create it inside your dialplan... i prefer script...  but that's just me...
22:07.18knarflycan I just copy my *-1.4.24 conf files over and make them work on *-1.6.0.9?
22:07.20mweichertBlargMaN00, script sounds good
22:07.42BlargMaN00mweichert: ok...  gimme a few minutes to whip something up...
22:08.09mweichertBlargMaN00, thanks so much!
22:15.18*** join/#asterisk theodred (n=nohost@205.207.102.154)
22:15.53BlargMaN00mweichert: i am going to leave and then come back as I have to get on the bus to go home, but I will get back on here as soon as i get on the bus downstairs...  shouldn't be more than 10 minutes...
22:16.23BlargMaN00mweichert: i almost have everything finished and ready to pastebin...
22:17.09mweichertBlargMaN00, thanks man
22:17.31BlargMaN00mweichert: no worries...
22:22.34*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
22:25.22*** join/#asterisk BlargMaN00-lap (i=BlargMaN@212.sub-70-218-70.myvzw.com)
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22:25.38*** mode/#asterisk [+o russellb_] by ChanServ
22:25.39BlargMaN00-lapmweichert: alright...  back to work...  8)~
22:26.52knarflyI'm new to * on linux...where does linux store the moh files?
22:27.30mweichertBlargMaN00, that WAS fast
22:28.10BlargMaN00-lapmweichert: good timing...  the bus was waiting on me...
22:32.54eppigyknarfly: /var/lib/asterisk/moh/
22:32.56Chainsawknarfly: In /var/lib/asterisk by default. moh or mohmp3.
22:33.08eppigyyou can search with "locate" of find
22:33.17eppigy*or
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22:35.23mweichertBlargMaN00, btw, this was my attempt when you were on the bus:
22:35.23mweicherthttp://pastie.org/446677
22:37.45BlargMaN00-lapmweichert: http://pastebin.com/m7b8cd460  <--  try this and let me know what happens...
22:37.57mweichertsorry, that was a previous version: http://pastie.org/446679
22:38.03mweichertok -I'll check it out now!
22:38.49seb-anyone mind testing my * server? i need someone to call and see if they hear "hello world" 5x
22:41.21Chainsawseb-: Address?
22:41.57ChainsawHm. How do I dial alphanumerically on a Siemens C485 IP.
22:42.09seb-Chainsaw: just sent you a msg
22:42.50seb-Chainsaw: thanks btw
22:45.06*** join/#asterisk war9407 (i=war@liquidswords.org)
22:45.09war9407<PROTECTED>
22:45.09war9407[Apr 14 18:44:27] WARNING[2643]: file.c:635 ast_openstream_full: File dir-pls-enter.gsm does not exist in any format
22:45.09war9407[Apr 14 18:44:27] WARNING[2643]: file.c:936 ast_streamfile: Unable to open dir-pls-enter.gsm (format 0x4 (ulaw)): No such file or directory
22:45.14war9407how do I tell where it is trying to play the file?
22:45.42mweichertBlargMaN00, shit, I'm getting kicked out of the office now - so I'll have to work on this when I get home. Thanks for your help BlargMaN00
22:45.45seb-war9407: search for hello-world.gsm and see if you can play that
22:45.47BlargMaN00-lapwar9407: it should be lcated in /var/lib/asterisk/sounds/en/
22:46.07war9407I used --prefix when installing asterisk (latest version)
22:46.22BlargMaN00-lapmweichert: no worries...  i'll be on later tonight, so see if you can catch me then...
22:46.36mweichertBlargMaN00, thanks a lot
22:46.40war9407asterisk-1.6.0.9-x86_64/var/lib/asterisk/sounds/en/hello-world.gsm
22:46.40war9407yep its there
22:46.48BlargMaN00-lapmweichert: if not, i'm always in here, and willing to help...
22:46.48war9407how to tell asterisk where the sounds are?
22:47.06seb-war9407: first see if you can play hello-world
22:47.18war9407seb-: ok,
22:47.31BlargMaN00-lapwar9407: just put the file name without the extension (i.e. hello-world) and it automatically looks there
22:48.04mweichertBlargMaN00, how does asterisk know that the .call file is for extension 1111?
22:48.32BlargMaN00-lapmweichert: that's what the Extension: 1111 tells *
22:48.39war9407kick ass!
22:48.41war9407hello world worked!
22:48.42seb-war9407: what BlargMaN00-lap said
22:48.53mweichertBlargMaN00, but you have Extension: play
22:48.54seb-war9407: great! now just put your file in there!
22:49.11*** join/#asterisk _BBV_ (n=buklov@213.138.71.254)
22:49.26BlargMaN00-lapmweichert: oh yeah...  i changed it to that, because they have to be seperate...
22:49.29war9407silly question, how do I dial into the pbx from my analog phone?
22:49.51mweichertBlargMaN00, hmm, but then I don't see how 1111 is linked to the .call file?
22:50.11BlargMaN00-lapmweichert: 1111 is what you are dialing, and 'play' is where the call file sticks the channel when your SIP truck gets answered...
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22:50.19war9407[Apr 14 18:50:13] NOTICE[2778] chan_sip.c: Call from 'line1' to extension '*11' rejected because extension not found.
22:50.19war9407ah
22:50.22BlargMaN00-laps/truck/trunk
22:50.40mweichertBlargMaN00, so what if I had two .call files and I only wanted one to occur when extension 1111 was dialed?
22:51.22BlargMaN00-lapmweichert: you will only have the .call files when you dial 1111...  they will not show up until you dial 1111
22:51.37mweichertah, right.
22:52.14BlargMaN00-lapmweichert: if you wanted another one to do something different, then you would create another script, or modify mine to use arguments...
22:52.28BlargMaN00-lapmweichert: if you need that, let me know, and I will modify it for you...
22:52.30dverzollaHas anyone running Asterisk in Solaris+SUN?
22:53.30war9407the extensions can only be two digits? when I try to dial the extension on an analog phone, it cuts it off after (*)11
22:54.38war9407instead of hello world, how do I get it to get me into a call tree, this is my next step :)
22:54.55BlargMaN00-lapwar9407: what exactly are you trying to accomplish??
22:55.14war9407BlargMaN00: first? make asterisk a nice answering machine
22:55.19war9407BlargMaN00: I have it working w/ hello world atm
22:55.29war9407BlargMaN00: I would like it to say, dial 1234 for bob and 1235 for sam
22:55.34war9407and then leave a voice message
22:56.04BlargMaN00-lapwar9407: ok...  so for now, you just want it to be a voicemail system??
22:56.15war9407yes
22:56.23*** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com)
22:56.26war9407I have 1.6.0.9 installed
22:56.29war9407and asterisk gui 2.0 (svn)
22:57.10BlargMaN00-lapwar9407: i don't use the gui, so i wouldn't be much help there, but i can help you get your dialplan written out to do what you want...
22:57.17war9407that'll work
22:57.26war9407I assume I edit this in extensions.conf ?
22:57.59BlargMaN00-lapwar9407: that or extensions_custom.conf...  depends on how clean you like your files and dialplan...
22:58.09war9407what is best practice?
22:58.10BlargMaN00-lapwar9407: but basically yes...
22:58.50war9407; Example "main menu" context with submenu
22:58.54war9407something to try?
22:58.59BlargMaN00-lapwar9407: there really isn't a "best practice"...  i like seperate files for different things, but that's just because I'm a neat phreak...
22:59.31war9407do you have an example template that will work with the default install?
23:00.11BlargMaN00-lapwar9407: yeah...  gimme a sec, and i'll whip something up for you...  than you can look over it, and see what i did, and hopefully it will point you in the correct direction...
23:00.13*** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net)
23:00.17war9407k
23:01.02knarflyI just switched to Fedora from FreeBSD for my * server...the install + dahdi went well. but I cannot seem to get my sip phone to register
23:02.23knarflycore show dialplan is not working...has the command changed in *-1.6.0.9?
23:04.36*** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au)
23:05.58knarflymust have been the firewall with fedora...disabled it and now I'm working but not sure why core show dialplan doesn't work?
23:07.44knarflyhears only crickets on this channel
23:07.45*** join/#asterisk kerx (n=kerx@adsl-69-104-67-217.dsl.irvnca.pacbell.net)
23:08.37BlargMaN00-lapwar9407: http://pastebin.com/m637318f0  <--  check that out, and see if you understand what i did...
23:08.45war9407k
23:10.35BlargMaN00-lapwar9407: BRB
23:10.40war9407replace john-exten with 4567 or what not I assume
23:10.40war9407k
23:10.51BlargMaN00-lapwar9407: yes
23:11.42war9407[pstn]
23:11.42war9407include => voicemail-menu
23:11.42war9407exten => s,1,goto(voicemail-menu,s,1)
23:11.44war9407trying
23:13.05war9407[Apr 14 19:12:53] WARNING[5226] pbx.c: Channel 'SIP/pstn-00c639d0' sent into invalid extension 's' in context 'voicemail-menu', but no invalid handler
23:13.17war9407s=the extension
23:13.19war9407need to fix,
23:13.20war9407sec
23:14.04war9407[Apr 14 19:13:51] WARNING[5294] pbx.c: Channel 'SIP/pstn-011b1e00' sent into invalid extension 's' in context 'voicemail-menu', but no invalid handler
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23:17.00war9407it does not like the ,s
23:17.09war9407[pstn]
23:17.09war9407include => voicemail-menu
23:17.09war9407exten => 123,1,goto(voicemail-menu,s,1)
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23:20.10war9407I did it verbatim like you have it written
23:20.14war9407it worked ;)
23:20.36war9407I left a voice message but it didnt send it anywhere, I will have to look into it more tomorrow
23:20.39war9407one other question though
23:20.44war9407it didnt ask me where I wanted to go
23:20.50war9407it just said persom at extension 1111 is unavailable
23:21.01war9407rather than say, 1111 or 1112
23:21.18*** join/#asterisk BlargMaN00-lap (i=BlargMaN@94.sub-70-216-251.myvzw.com)
23:21.29BlargMaN00-lapwar9407: back...
23:22.03war9407the choose extension part not working
23:22.08war9407it just says person at extension is unavaialble
23:22.33BlargMaN00-lapwar9407: how many digits is it letting you put in before it says that?
23:22.37war9407also caller id is not getting passed through to the phone
23:22.39war9407sec
23:23.09war9407calling number.... ring..... person at extension 1234 is unavailable, please leave your message after the tone, when done please hang up or press the pound key
23:23.47war9407it does not give me the option tfor one or the other
23:24.10BlargMaN00-lapwar9407: do you have a phone registered to the extension 1234??
23:24.19war9407I am using an FXO (3102)
23:24.22war9407no VoIP phones
23:24.29BlargMaN00-lapwar9407: oh...  ok
23:24.33theodredI am new to asterisk, a quarter of the way through the asterisk manual, and am trying to wrap my head around a few of the concepts there..
23:24.33war9407traditional phone line -----> spa3102 -----> inbound
23:24.52war9407BlargMaN00: it says 'PSTN' on my caller id, how do I allow that to pass-thru?
23:24.58war9407BlargMaN00: and second question, why no voicemail-menu?
23:25.51BlargMaN00-lapwar9407: what context are you sticking the calls from the 3102 into??
23:26.11war9407http://graham.doel.org/knowledge-base/?View=entry&EntryID=9
23:26.40war9407http://blog.pathennessy.org/2009/01/01/configuring-linksys-spa-3102-for-asterisk/ <- this one worked
23:28.22BlargMaN00-lapwar9407: ok...  change [incoming] to [pstn] in what i pastebin'd
23:29.08war9407that is what it is currently
23:29.21war9407[pstn] | #include "voicemail_menu.conf" | include => voicemail-menu | exten => s,1,goto(voicemail-menu,s,1)
23:29.27war9407| = return
23:29.53BlargMaN00-lapwar9407: yes
23:30.23BlargMaN00-lapwar9407: except no quotes around voicemail_menu.conf
23:31.30BlargMaN00-lapwar9407: what version of * are you using?
23:32.19war94071.6.0.9
23:32.39BlargMaN00-lapok
23:33.15war9407more serious issue
23:33.25war9407messages:[Apr 14 19:20:35] WARNING[5502] file.c: Failed to write frame
23:33.31war9407(when I left a voice message)
23:33.33war9407it went to /dev/null
23:33.57knarflyhow can I check the status of an X101P card in my system....excuse me but I'm just converting over from zaptel to dahdi
23:34.07war9407BlargMaN00: thx for getting my started
23:34.11war9407BlargMaN00: will look more into this tomorrow
23:34.13war9407my->me
23:34.18war9407BlargMaN00: will tty tomorrow if you're around ;)
23:34.31BlargMaN00-lapwar9407: no worries...  anything to help
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