IRC log for #asterisk on 20100508

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00:12.48torrancewthanks all the same, jdoe
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01:41.37ruben23hi anyoe used phone booth
01:42.20Kyoshyea back in the 80s
01:44.11ruben23<PROTECTED>
01:45.33*** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk)
01:51.14carrarFREE
01:51.29carrarBe that comes with some excellent SLA's
01:51.31carrarBet
01:59.21*** join/#asterisk lepine (~leprechau@69-196-166-209.dsl.teksavvy.com)
02:00.09lepineI'll be going to the docs shortly, but can someone confirm there's something I'm not grasping yet? http://pastie.org/951037
02:01.02lepineI get as far as the new context, but am told the extension is invalid
02:01.30lepine(and no handler)
02:01.54carrarremove those spaces
02:02.16carrarare you sure they are sending you a +1
02:03.47*** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com)
02:03.51carrardon't need quote marks in the Noop also
02:04.22lepineThanks, that worked (spaces)
02:04.46lepinehowever, it irks me, considering i program most of the time and have spaces between parens
02:04.57lepineand yes, send me the 1
02:05.18carrarBrace yourselffor more IRKS
02:05.23lepinetaking note for NoOp, but I think it makes sense to keep it
02:05.24lepinelol
02:05.29lepineanything in mind?
02:05.52carrarI'm sure you'll find more
02:06.28lepineeh, I've only been fiddling w/ asterisk about 30 hours (very sporadically) ...
02:06.32lepineI'll get back to you :)
02:06.42carrarit's fun
02:07.15lepineyeah, my boss was like "choose, asterisk or ironport (bulk mailing)" ... asterisk, please.
02:07.51lepinelike I care about learning to spam people, i'd rahter play around w/ voip
02:08.28carrarWhat version are you running
02:08.42lepine1.4
02:09.03lepinerunning debian, and that's what they've got. don't love bypassing apt
02:09.31carrarShould also have a underscore before 1888
02:09.36carrarfor pattern matching
02:09.53lepinetried AsteriskNow, trixbox and asterisk/deb + freepbx 3 ...
02:10.13lepine_ is asterisks */% ?
02:10.24lepine(wildcard)
02:10.45lepinenm, I better read about it
02:11.16carrarsearch underscore pattern matching
02:11.18carrarbbl
02:11.25lepinewill do, thanks
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02:43.17*** mode/#asterisk [+o twisted] by ChanServ
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03:31.32*** join/#asterisk dandate2 (~dan@112.206.130.149)
03:31.52dandate2what is thee cheapest way to make outbound calls to the UK?
03:32.33coppicesteal
03:32.50jqlworks for me
03:32.55TJNIIYell out the window.
03:33.10jqlit's a company phone, so it's merely misappropriation of resources
03:33.19TJNII"Hey I can call my maw while I'm up here.  HEY MAW!  GET OFF THE DANGED ROOF!"
03:33.24TJNII~itsp-uk
03:33.29TJNII~itsplist-uk
03:33.30infobotrumour has it, itsplist-uk is UK based ITSps include http://www.voiptalk.org/  http://www.voipon.co.uk/  http://www.gradwell.com/ and a few other tinpot companies you can dig up with google.
03:46.47*** join/#asterisk Netgeeks (~chris@173.11.68.155)
03:50.18carrarGive whoever you want to call in the UK a SIP Phone!
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04:03.52coppiceyou mean a UKIP phone?
04:03.56*** join/#asterisk D-Boy (~Geek@unaffiliated/cain)
04:04.46D-BoyHi, how to get a list of connected sip phones (hard phones and soft phones) ?
04:06.56dandate2sip show peers
04:08.25D-Boydandate2 : thx !
04:10.38*** join/#asterisk ChannelZ (~bobm@burner.com)
04:13.05carrar~book
04:13.05infobot[~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
04:13.17carrardownload that D-Boy
04:16.56*** part/#asterisk ruben23 (~ITadmin@125.212.40.2)
04:16.59D-BoyOh nice ! i guess now i need just a thousand cups of coffee and patience to get the spirit of asterisk :p
04:17.07D-Boycarrar : thank you :)
04:17.46carrarworth it
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05:54.30*** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
06:02.49Vin73hi, i am relatively new to asterisk and would request for some help on an error
06:03.18Vin73hi
06:03.49Vin73is anyone there?
06:04.02Vin73quit
06:04.05*** part/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
06:04.08*** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
06:04.15Vin73hi there
06:04.30Vin73I am a new user seeking some help
06:06.52Vin73#asterisk- [freenode-info] channel
06:07.05*** part/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
06:07.13carrarread this
06:07.16carrar~book
06:07.16infobot[~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
06:07.36*** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
06:07.39carrar~book
06:07.39infobot[~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
06:07.42carrarread that
06:08.09Vin73hi I have read that book, but this is a specific error
06:08.18Vin73In fact I have that book
06:08.54Vin73The error is X-Asterisk-HangupCauseCode: 20
06:09.20Vin73has anyone seen this error before?
06:10.38*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)
06:11.10Vin73hello?
06:11.37carrarhttps://issues.asterisk.org/view.php?id=14684
06:11.42*** join/#asterisk voxter (~voxter@c-66-176-249-104.hsd1.fl.comcast.net)
06:12.57Vin73ok thanks
06:13.02Vin73I will have a look
06:13.10Vin73bye
06:13.53sawgoodIf I wanted my Asterisk box to 'drop' all current registrations with my ITSP carriers, and try to re-authenticate ... wouldn't reload now from the CLI do that?
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06:16.19carrarrestart would
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06:20.15*** join/#asterisk biscuit314 (~michael@s72-38-112-54.static.comm.cgocable.net)
06:20.16sawgoodrestart now
06:24.29*** join/#asterisk biscuit314 (~michael@s72-38-112-54.static.comm.cgocable.net)
06:24.49biscuit314Hello!  I'm trying to get into AGI using pyst.  Got a PERL script working fine, but my attempts to get a python script end in "Exec format error"  Can anyone help?
06:25.30p3nguinsawgood: "reload" should never be used.
06:27.17carrarbiscuit314, sounds like you have a python issue
06:27.50carrardebug your script
06:29.13biscuit314I lifted the following from a sample described in the pyst project:
06:29.18biscuit314from asterisk.agi import *
06:29.19biscuit314def main():
06:29.19biscuit314<PROTECTED>
06:29.19biscuit314<PROTECTED>
06:29.48biscuit314running directly with python interpreter (for whatever that's worth) reveals no issues
06:31.50carrarHow about you write a simple script to tell you what variables asterisk is sending your script
06:32.01carrarstart with that
06:32.16carrarthen try setting some variables
06:32.43carraror join #python and ask there
06:32.59biscuit314Perhaps the problem is in the pyst libraries.  I'll take your advice and start simple, build up using stdout/stderr, take it from there.
06:33.18biscuit314But not tonight.... time for bed.  Nice bumping into you! g'nite!
06:40.26*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)
06:49.40sawgoodI have two ITSP carriers ... when I do a sip show peers from the CLI one ITSP shows up with two listings and the other ITSP shows up with only one listing
06:50.08sawgoodfor the ITSP with two listings ... one is for outgoing settings and the other is for incoming settings
06:50.31sawgoodI was wondering 'why' the 2nd ITSP only shows up once ... with sip show peers, but the other ITSP shows up twice
06:50.38sawgoodstrange to try and explain ...
06:51.40sawgoodnever mind I see why now
07:01.16*** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk)
07:03.56ChannelZExactly.
07:10.00*** join/#asterisk VanDyke (~scotch@kernel-panic/vandyke)
07:10.18VanDykegood evening
07:14.58ChannelZright
07:15.00VanDykehas anyone here set up before an asterisk box with a SIP trunk to CommPartners ?
07:17.39carrarSounds impossible!
07:18.59VanDykecarrar, almost
07:19.00tzafrir_laptoplooks like biscuit314 was missing the #! line on the top of the script
07:27.40ChannelZshe bangs! she bangs!
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07:46.12sawgoodI have an ITSP carrier who sends all incoming DID traffic to me on a dedicated IP address (which is my Asterisk box) ...
07:46.26sawgoodCan anyone help me with what my 'incoming settings' for this trunk should be like?
07:46.38sawgoodtheir tech support is closed after 6 PM and on the weekends
07:46.47sawgoodI really don't feel like waiting until Monday to fix this
07:46.57sawgoodI can make calls just fine, but I cannot receive calls
07:47.28sawgoodThe 'only' real instructions I got from them is to 'allow' all traffic from IP xxx.xxx.xxx.xxx
07:47.45*** join/#asterisk Tim_Toady (~moi@188.4.66.69.dsl.dyn.forthnet.gr)
08:25.27*** join/#asterisk halkun (~halkun@officegateway.ethostream.com)
08:26.37halkunI'm done planning out my new home network. Does this look like a good setup with an SPA 3102? ----> http://img85.imageshack.us/img85/5852/neta.png
08:38.19ChannelZnot sure about the phone splitter, you could probably put the FXO into the DSL modem which should automatically filter that port
08:44.29carrarWhy is the internet plugged into a phone jack?
08:44.35carrarmakes no sense
08:44.52ChannelZheh yeah arguably that line should come from the DSL modem
08:46.20carrarWhere is your firewall?
08:46.38carrarAlso why do you have two devices handing out DHCP?
08:47.25ChannelZWhat, you don't wear two condoms?
08:47.33carrarTurn off DHCP on your AP and let your DSL modem hand out leases
08:47.44carrarused to
08:47.45carrartoo
08:50.32carrarShould add that to www.ratemynetworkdiagram.com
08:51.47carrarhttp://www.ratemynetworkdiagram.com/?i=10993
08:51.51carrarsee they have a firewall there
08:51.54carraryou are missing tghat
08:52.45ChannelZhe really has 2..
08:53.12ChannelZpossibly
08:53.40carrarhalkun, I would make your asterisk box your firewall so that it has a external outside IP address and put your DSL modem in bridge mode
08:54.06carrarthen the second NIC on your Asterisk box would be your internal network
08:54.24carrarrun a real DHCP server (ISC) on yoru asterisk box
08:54.48halkunOh, I didn't see the replies. :)
08:54.57carrarkeep up
08:55.05halkunI was busy working :P
08:55.11carrarSo you don't FAIL IRC
08:55.19carrarno one works on weekends
08:55.38halkun3rd shift W-Sun
08:55.43halkunit's a suck schedule
08:55.48carrarW-Sun?
08:55.53halkunwed-sun
08:55.53carrarSun MicroSystems?
08:55.56carraroh
08:56.31*** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru)
08:57.23halkunOk, only the D-Link gives out DHCP
08:57.39carrarwell you ahve it specified on the drawing that the AP is also
08:57.49halkunthe AP will not accept a static setting.
08:58.10halkunI think that's my ineptitude though :P
08:58.11carrarAP can get it's IP's from DHCP
08:58.14carrarthen
08:58.18carrarbut thats crazy
08:58.38carrarNever seen a AP that doesn't have a static IP
08:58.49carrarI think it's user error
08:59.01halkunThe AP has a static/ 192.168.0.4
08:59.33carrarSo move your DSL modem to Bridged mode, put your external IP address on the Asterisk box and make it a firewall/NAT server also
08:59.38carrarand a DHCP server
08:59.44halkunif you set you wireless up with a static however, the AP won't accept the connection. (It's more for my own note)
09:00.01carrarAP shouldn't be handing out IP's
09:00.16halkunI have no control over the DSL modem. That's from my ISP. I have a dynamic address from my ISP
09:00.26carrarIs the DSL modem in your house?
09:00.31carrarthen you have control over it
09:00.35halkunyes. This is a home setup
09:01.01carrarYou can do dynamic IP on the asterisk box from your provider too
09:01.12carrara litle more nutty however
09:01.35carrareven more reason not to trust your DSL modem to protect your home network
09:01.53halkunNo, I can't log into the DSL modem, I plug a phone line into it and the other side has PPPoE
09:02.14carrarWhat happens when you open a browser to 192.168.0.1?
09:02.21halkunYou get my D-Link
09:02.50carrarso the Dlink has the external IP address?
09:03.21halkunyes, it's from my ISP.
09:03.24carraragain you should make yoru asterisk box your outside firewall/NAT/DHCP server
09:03.38carrarswap the dlink with your asterisk box
09:04.12carrarand put a mini switch on the backside of your asterisk box for interal connectivity
09:04.35carrarBut I suspect thats too difficult for ya :)
09:04.58halkunI kind of see where you are going, but yes
09:05.51halkunI'll be doing some port forwarding so that the important bits of the * server can be seen by the outside world
09:06.07carrarsee you wouldn't haev to do that my way
09:06.15halkunIf course, I also have to deal with the fact my ISP also blocks all ports under 1024
09:06.23halkun(except 22)
09:06.30carrarwhats wrong with that
09:06.47halkunmakes running servers on it a pain
09:07.08carrarnot a aserisk server
09:07.22halkunI'm still fleshing out my understanding of Asterisk. I know the ports are pretty high there
09:07.31ChannelZthey is
09:08.03*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)
09:08.08halkunThis is my "Lab" set up so I can break things generally screw things up so I can learn
09:08.25carrarYou also have to realize running a "server" with a dynamic IP makes no sense also
09:08.43ChannelZdyndns
09:08.47halkunWell, I'llprobably get dynedns
09:08.49carrarSo this is probably why your ISP offers a buisnes class dns service
09:08.50halkunyea
09:08.59carrardsl
09:09.06ChannelZNo, that's to extract more money from you
09:09.17halkunI work from my ISP. I get basic home service for free
09:09.23halkun6M DSL
09:09.26carrarwell Freeinternet.com doesn't exist anymore
09:09.33carrarheh
09:09.45halkunI'm doing this on the insanely cheap
09:10.05halkunThe idea is to get my droid to work on wifi so I don't have to by cell service
09:10.26*** join/#asterisk HorizonXP (~xitij@69-165-140-215.dsl.teksavvy.com)
09:10.36ChannelZwell you may or may not have luck with that on free wifi hotspots
09:10.47carraron your wireless network or other peoples wireles network?
09:10.52halkunYes
09:11.02carrarYes A or Yes B
09:11.06halkunI pretty much have Wifi wherever I go
09:11.24carrarSo you're gonna wish your asterisk box wasn't behidn nat
09:11.58halkunMy friends places, my work, and also I work for a comapny that does Wifi for hotels. I have the keys
09:12.48halkunI'm donna try and have as much of the * server available. I'll do everything short of moving into the DMZ
09:12.51carraranyways, you got my opinion
09:13.05carrardon't come back if it doesn't work and you didn't take my suggestions :)
09:13.32halkunOh, no, I have a second network card for the * server just in case
09:14.00carrarand use ChannelZ suggestion for dynamic DNS for your droid * server hostname
09:14.02halkunI'll have the server make the PPPoE connection and hand off to the dlink if worst comes to worst
09:14.42carrarwell you might be able to use the switch portion of the dlink just for the switch part
09:14.42halkunOrigally, I had the * server hooked up to 3102
09:14.49carrarbut I wouldn't do that
09:14.55carrarI'd dump it and get a real switch
09:15.45carrar5 or 8 port switches are cheap
09:15.50halkunI found a 5 port in my car. I'll have to find the power plug.
09:15.58carrarsee
09:16.00carrarfree
09:16.08halkunIt's why the laptop is wireless. I ran out of ports on the d-ling
09:16.36carraruse DHCPD 4.1.1 (http://www.isc.org/software/dhcp)
09:16.40halkun4 port + an WAN port
09:17.12carrarswitches don't have WAN ports
09:17.13halkunnever ran a DHCP server before... scary!
09:17.18halkunno, my D-link
09:17.30carrarwell ISC is the best
09:17.32halkunI've always had my routers handle it
09:17.48carrarplenty of example configs on google also when you need help
09:17.51carraror jsut ask me
09:18.23carrarRouters suck for DHCP servers
09:19.05carrarYou have the perfect box for a DHCP server
09:20.27halkunso, bet case is set DHCP on the * server, give it PPPoE so it can use the DSL modem and ditch the D-link for a nice 5 port switch
09:20.58carraryup
09:21.12halkunthat way I also have a direct link to the 3102 from *
09:21.15halkunIt makes sense.
09:21.38halkun"I'll have to redraw the map again!"
09:21.48carrarwell the 3102 plus into your internal switch
09:21.52carrarlike everythign elswe does
09:22.13carrarwhich means everything internal has a "direct link" to it
09:22.29halkunRight, I'll give that a go then. Thanks for the advice :)
09:22.30carrarand by direct link I mean  on the same logical network
09:26.45carrarThen once it's setup, get a IPv6 Tunnel form your Asterisk box to a IPv6 tunnel provider!
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12:21.22beefpastryCan anybody point me in the direction of how I might figure out how to get a polycom to display the callerid instead of the dialed parking extension (is this even possible)?
12:42.33carrarbeefpastry, polycom is making a call
12:42.44carrarcallerid would be the polycom
12:43.09beefpastryI am aware...was wondering if it could be overridden
12:43.12carrarthere is no magical transfer of the parked call's caller id
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12:44.37carrardon't even think it's possible since the polycom initiated the call
12:45.27carrarunless you can send some sort of message to the phone outside of the parked call in some goofie mannor
12:45.47carrarwhich there isn't any support that I know of for that
12:46.34carrarcould however write a app you dial
12:46.45carrarwhch then sends that parked call to you as a new call
12:46.53carraron  your second line
12:47.10carrarthen you would get the caller id of whatever your app sent
12:47.27carrarhaving the app extract that caller id out of the park call info
12:48.08carrarlet me know when you have that written :)
12:48.58carraror have it send it to you have you hang up that app you dialed
12:49.06carrarhave=after
12:49.28carrarSo I think it's doable
12:49.35carrarit a different way
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13:05.47beefpastrycrap...ran a yum update to pull 1.6.0.27 and I got 1.6.2
13:05.55beefpastrynow seg fault
13:06.04beefpastrygood times
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13:15.05smooth_penguinalways hated rpms
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13:38.46beefpastryphew
13:44.41riddleboxman I was just reading on scenarios with findme/followme, who wants to sit on the phone waiting while the system rings 5 phones on after another!
13:44.49riddleboxI would hang up after 4-5 rings
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15:50.10[sr]howdy
15:50.26[sr]to install asterisknow in raid on IDE disks, there's a boot command, i forgot :(
15:50.33[sr]can you remind me?
15:56.00[sr]ok forget, wasnt needed :)
16:00.57TJNII#asterisknow
16:01.00TJNII~asterisknow
16:01.01infobotmethinks asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons.  Please seek support in #asterisknow instead.
16:04.32[sr]im there
16:04.33[sr]:P
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16:38.14omesdoes anyone know about a howto/guide on how to set up an asterisk server that shares a sip phone number amongst multiple phones/user accounts?
16:38.33omesim completely new and have no idea where to start with asterisk
16:38.47Kyoshasteriskdocs.org
16:38.52Kyoshgo read the asterisk book
16:38.58Kyoshgreat start
16:39.27[TK]D-Fenderomes: Define "share"
16:40.35omes[TK]D-Fender: I have one sip account and multiple phones around my house, it would be nice if they could all use that line
16:41.37omesor i can set up multiple lines, but it would be nice if asterisk could do the routing from the phones to the sip accounts
16:42.09omesas the information there change quite frequently and i don't want to change the info on my phones that often, it's easier to do it all in one place
16:42.18Kyoshand they can, but you need to set up the asterisk box
16:42.18Kyoshor any ip pbx, but asterisk should be easiest
16:42.18KyoshasteriskNow would be easy too
16:42.22Kyoshomes, you want something that is as easy as an out of the box solution, right?
16:42.35omesno
16:42.43Kyoshno?
16:42.51omesi want to know how to edit the config files to do that
16:42.53Kyoshso you want the challenge of the learning curve?
16:42.58omesindeed
16:43.03Kyosheither case you can
16:43.13Kyoshits just how fast to you want to be up and running
16:43.24omesdoesnt matter if it takes a lot of time
16:43.36omesi have time, but i want to learn it the "proper" way ;)
16:43.40[TK]D-Fenderomes: this is all basic dialplan.  Time to hit the books
16:44.05omesah i see, so the book Kyosh gave is good for that then? :)
16:44.08[TK]D-Fenderomes: Here's some "inspiration" material for you to see what a simple setup might look like
16:44.11[TK]D-Fender~jerjerguide
16:44.11infobot[~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/
16:44.15[TK]D-Fender^^^
16:44.41[TK]D-Fenderomes: Do not follow the installation procedure, but use this as a sample to help you understand the book as implemented in a small setup
16:44.57omes[TK]D-Fender: that link was perfect!
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17:06.10omesexten => _1NXXNXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
17:06.11*** join/#asterisk neurosys (~neurosys@adsl-8-187-44.mia.bellsouth.net)
17:06.19omeswhat does the underscore and X's mean here?
17:06.31p3nguin_ indicates a pattern match
17:06.41omesi mean N's
17:06.42p3nguinX represents numbers 0-9
17:06.44omesthe X'es i mean
17:06.51omeshaha, the X'es i understand
17:06.57[TK]D-Fenderomes: "_" means that what follows is a pattern and not just literal chars.  X= any digit.  N = 2-9
17:07.07p3nguinN represents numbers 2-9
17:07.12[TK]D-Fenderomes: Time to go read chapter 5 <--------
17:07.18omesah, thanks :)
17:07.38omes[TK]D-Fender: i will definitely read the whole book!
17:07.42p3nguinand Z represents numbers 1-9, if I remember right.
17:07.43omesasterisk is great!
17:10.27omesexten => _XXXXXXXX,1,Dial(SIP/Phonzo/0047${EXTEN}) <-- so this would make a call to norway?
17:11.00p3nguinIf the number starts with 0047 and needs another 8 numbers after that, probably.
17:11.30omesyeah, numbers in norway are 8 digits long and +47 sends them in the right direction ;)
17:12.19omeslocal, fetching registrations failed: 403 Forbidden
17:12.20[TK]D-Fenderomes: Also depends on the format your provider requires
17:12.25omesseems i made a mistake somewhere
17:12.39omes[TK]D-Fender: i know 0047XXXXXXXX will work
17:12.56omesthat's the error twinkle gave me
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17:14.06omeswould one have to make a users.conf too?
17:14.23p3nguinOnly if you wanted to use users.conf instead of the normal methods.
17:14.43[TK]D-Fenderomes: your auth is bad.don't
17:14.55[TK]D-Fender~users.conf
17:14.55infobot[~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system.
17:16.08omeswell it still hates me :\
17:21.40omesThe canary is no more.  He has ceased to be!  He's expired and gone to meet his maker!  He's a stiff!  Bereft of life, he rests in peace.  His metabolic processes are now history!  He's off the twig!  He's kicked the bucket.  He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!!  THIS is an EX-CANARY.
17:21.45omeshahaha, great quote there
17:26.10omesERROR[9760]: pbx.c:3309 __ast_custom_function_register: Function MAILBOX_EXISTS already registered. <-- apparently that's where it kicks it
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17:34.42[TK]D-Fenderomes: PASTEBIN "ls -la /usr/lib/asterisk/modules"
17:34.44[TK]D-Fender~pb
17:34.44infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
17:35.00omes[TK]D-Fender: I had to use the default modules.conf
17:35.17omesthen it worked, now i just need to get it running on a different ip so i can get my sip client up again
17:35.57omesagain it forbids me access with the 403 error
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17:36.48omeshttp://pastie.org/951603
17:38.25[TK]D-Fenderomes: Twinkle is running on the same box as *?
17:38.30omesyes
17:38.37omesbut i changed the ip for asterisk now
17:38.49omesso they're both up and running, back to the user authenticatoin problem
17:39.01[TK]D-Fenderomes: pastebin your sip.conf entry for that device
17:39.09omess/authenticatoin/authentication/
17:39.24omeshaha, nice bot :D
17:39.55omeshttp://pastie.org/951607
17:43.04[TK]D-Fenderomes: Try another client like Ekiga
17:43.17omes[TK]D-Fender: sure
17:45.09omesno sigar "Could not register"
17:46.00omesim gonna go eat, brb
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17:58.13omesthere
17:58.20omesis there any way to debug the thing?
17:58.36omesso that i can see everything that's going on? like if its a password issue, username issue and so on
18:03.54[TK]D-Fenderomes: use SIP DEBUG from * CLI, not an external dump.  Try again, pastebin the attempt and your sip.conf
18:04.08[TK]D-Fenderomealso your softphone should be running on a different port than 5060
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18:06.47omeshaha sorry guys
18:06.53omesit *was* a twinkle problem after all
18:07.02anny__hi all
18:07.05omesapparently twinkle binds to 0.0.0.0 too...
18:07.17omesRegistered SIP '100' at 10.0.0.100 port 5061
18:07.19omesvoi voi
18:07.24omesim gonna try a call now wish me luck
18:07.52anny__i'm new to astersik, can anyone tell me where can i find some documents to start reading
18:08.12anny__possibly not as detailed as the official documents
18:08.14pabelanger~bppl
18:08.18pabelanger~book
18:08.18infobot[~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
18:08.35anny__thx
18:08.47omesanny__: http://jeremy-mcnamara.com/asterisk/how-to-configure-asterisk/ <-- this really helped me too
18:09.00anny__thx omes
18:09.08omesthe book is not something you can do in an evening me thinks :P
18:09.18anny__i know :)
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18:09.28anny__just looking for a place to start
18:09.41anny__but i've been browsing the official docs on the asterisk site
18:09.59anny__i need something less detailed just to grasp the global picture
18:19.06omeshttp://pastie.org/951644
18:23.28omesapparently that was a problem with my sip provider...
18:23.43omesis there some way to find out that didn't work with one of them and why it worked with another one?
18:24.05omesprobably i should read the book and then ask the questions
18:24.26omeselse #asterisk will be a concat of the pdf :P
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18:27.40[TK]D-Fenderomes: pastebin the CLI SIP DEBUG (you eed to enable this) for your failed call
18:28.56Nixy33Hi all, can anyone help a noob out here, got my asterisk setup working in our small office and trying to configure a remote extension. Using a Snom M3, have set the extention up and it's registering on the PBX. I can get my home number to ring (by dialing the prefix for the local analogue lines)  but when I pickup there's no sound. If I try and use the voip prefix it all works fine. Also no core messages such as voicemail menu are audiable
18:28.56Nixy33<PROTECTED>
18:29.24omes[TK]D-Fender: NICE
18:29.32omes[TK]D-Fender: you're an angel! it all works now anyways :)
18:29.42omesill find out why the other provider borked later on
18:31.19omesjust two last quick questions; how do i enable that, and how do i reload config files without restarting asterisk?
18:32.01p3nguinsip set debug
18:32.03p3nguinsip reload
18:32.06p3nguindialplan reload
18:32.10[TK]D-Fenderomes: Syntax varies by version.  recently "sip set debug"
18:32.18omesNo such command 'sip set debug'
18:32.53omeswhat about the extentions file, how do i reload that without restarting asterisk?
18:33.00p3nguin(1332.06) <p3nguin> dialplan reload
18:33.05omesah, ok thanks :)
18:43.16Nixy33any one got any ideas?
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18:48.38pabelangerNixy33: Firewall or NAT issue
18:49.42Nixy33thanks, so not a codec issue then? I'm baffled as to why the call when routed over VOIP(from the pbx)  works but no local messages like "enter your voicemail password" work
19:01.20[TK]D-Fendernixpastebin a failed call with SIP DEBUG enabled
19:01.22[TK]D-Fender~pb
19:01.22infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
19:01.24[TK]D-Fender^^^^^
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19:23.10seanbrightanyone have any experience with lumenvox or vestec?
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19:23.42seanbrighti have a requirement from a client that a customer be able to supply an arbitrary password to the system by speaking it
19:24.07seanbrightbut from the research i have done it looks like if the word isn't part of the supplied grammar, it's not getting recognized
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20:12.41biscuit314Hi all!  I'm hoping someone can help me with my AGI python script problem...
20:13.07biscuit314I have a simple python script properly chmod'ed in the correct folder: /var/lib/asterisk/agi-bin/
20:13.31biscuit314my dialplan includes: exten => 789,n,AGI(agitest.py, 'test1', 'test2')
20:13.41biscuit314The script itself (agitest.py) is as follows:
20:14.08biscuit314import sys
20:14.08biscuit314f = open('/home/michael/agidump.txt', 'w')
20:14.08biscuit314for arg in sys.argv:
20:14.09biscuit314<PROTECTED>
20:14.09biscuit314f.close()
20:15.03biscuit314I can run the script from terminal - no problems.  But when I dial the extension I get...
20:15.05biscuit314<PROTECTED>
20:15.05biscuit314<PROTECTED>
20:15.37biscuit314I have successfully run PERL scripts, but no luck at all with python.
20:15.43biscuit314What am I doing wrong?
20:16.20seanbrightnot putting a #!/usr/bin/python as the first line?
20:16.48biscuit314slaps forehead...  I'll try that...
20:17.05seanbrightand it should be executable
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20:17.15seanbrightand when you paste stuf like that in here, use a pastebin
20:17.16seanbright~pb
20:17.16infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
20:19.32biscuit314That did the trick....  thank you so much!  (and my apologies for pasting 3+ lines... newbie mistake...)
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20:39.39shaderwhat's the bare minimum of configuration required to get asterisk to start?
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20:41.30xhelioxHey! It's really ambiguous day, everyone join in!
20:42.21seanbrightshader: a cpu
20:42.32xhelioxsome sort of storage device.
20:44.40shaderlol
20:45.27shaderI cleared out the config folder, because I wasn't sure if any of the generated sample config files was interfering with my test extensions.conf and sip.conf, but then of course asterisk wouldn't start
20:45.49shaderso now I'm using the samples again, and I wish I knew which files I didn't actually need
20:46.03shaderand which I have to fill in myself
20:51.36[TK]D-Fendershader: modules.conf.  ANy module you load may require its config file or seize
20:52.17[TK]D-Fendershader: Beginners should install the samples and just trash the 2 files you said you already did yourself.
20:52.26[TK]D-Fendershader: so start with the samples for the rest
20:56.20shaderok
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21:09.01krenelhello
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21:10.23wcselby~sipnat
21:10.23infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:10.54krenelwould it be possible to use a separate asterisk box for voicemail only and still get working MWI on the first box?
21:13.03xuser~itsp
21:13.03infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
21:13.14xuser~itsplist-us
21:13.14infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
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21:53.09MiserySoftJust bought a TDM400 clone card on eBay with 2 FXO + 2 FXS for $80 .. very impressed.
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22:30.42Bartockbatzanyone familiar with the SIP provider Axialys?
22:32.00Bartockbatzfor some damn reason, I cannot get my Asterisk server to register
22:33.50BartockbatzI can get my Sipgate trunk to work just fine - something I should know about non -NANPA numbers? Does it make a difference?
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22:38.09rare1980_hi all ...... on asterisk CLI i am getting this error msg
22:38.10rare1980_ast_carefulwrite: write() returned error: Broken pipe
22:38.19rare1980_any idea what this msg means?
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22:49.23omeshahaha, asterisk is great fun
22:49.31omesim seding faxes over my VoIP line now...
22:49.52omesif they're longer than just a line of text they tend to fail though ^^
22:50.28omesBartockbatz: try to change the useragent
22:50.41omesmy sip provider blocked asterisk, so i had to change it....
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22:53.10BarthezZyeah omes, FoIP is just a pain in the ass which tends to fail :p
22:53.57omesi suppose FoIP would be good (T.38) :P
22:54.21omesthis is Fax over VoIP :P
22:54.21TJNIIWhy would a provider block asterisk?
22:54.21TJNIII'd tell them to FOAD.
22:54.21omesTJNII: i ask myself the same question :S
22:54.24BarthezZyeah true omes :p, still sucks ^^
22:54.51omesindeed :P
22:55.36BarthezZfaxes should just be banned
22:55.42BarthezZuse freaking email people!
22:55.56omeswell, you could tunnel your emails over fax
22:56.11omesEmail-to-fax and Fax-to-email on the other end
22:56.27omesprays, please dont die this time :(
22:56.39omesmy quality setting is set to 10, that doesnt help much lol
22:56.39BarthezZor just leave the fax part out, email to email :)
22:56.41TJNIIAnalog faxes should be banned.  Put a friggin NIC in them already.  Its 2010!
22:57.01omesyay, it works!
22:57.19omesHAHAHA
22:57.26omesthe quality is so bad i just have to laugh :P
22:57.39BarthezZquality of the fax? :p
22:57.43omesyeah :P
22:58.09BarthezZwhat app do you use to send/receive them?
22:58.25omesill have some show and tell soon
22:58.28omessendfax
22:58.48omesa client for hylafax, with iaxmodem as the modem and asterisk as the phoneline
22:58.59BarthezZhehe oke
22:59.22omesthat is, asterisk as my operator
22:59.31omesmy WLAN is my phone line, at least parts of the way
22:59.38omesi guess that doesnt really improve the quality :P
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23:08.47omeshttp://i40.tinypic.com/11vma6w.jpg
23:08.51omeshere's my show and tell
23:08.56omesFax over VoIP
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23:13.21BarthezZomg there was a glitch in the (power) matrix :p
23:14.23omesindeed :P
23:14.27omesoh noez! :(
23:14.32omes05/09/10  02 07 48 < omes> http://i40.tinypic.com/11vma6w.jpg
23:14.32omes05/09/10  02 07 52 < omes> here's my show and tell
23:14.32omes05/09/10  02 07 57 < omes> Fax over VoIP
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23:14.54BarthezZlol, fail :p
23:14.54omesi find it hillarious i even got to send an iamge over my wlan....
23:15.00omesit shouldnt be possible...
23:15.17BarthezZyeah, totally beats tcp/ip with ftp/http/<about any ip protocol> ;p
23:15.17omesactually its great success!
23:15.39omeswell... considering what could go wrong im amazed it went through...
23:15.57omesSIP/Phonzo-00000001 answered IAX2/fax-3638
23:15.59BarthezZtrue :p
23:16.05omeswaiting for it to finish then im going to bed :P
23:16.25omesHungup 'IAX2/fax-3638' yay
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23:17.54omesi think im gonna write a howto on how i did this
23:18.01omesit might even be phun :D
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23:42.13Slugs_hi
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