IRC log for #asterisk on 20090726

00:04.14*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
00:05.26*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
00:06.34*** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net)
00:35.02OrbixxCan anybody recommend a hardphone?
00:46.36AlfioOrbixx you mean ipphone?
00:47.31OrbixxIP phone, yes.
00:51.45Alfiopolycom
00:51.54Alfiothe best of the best sir
00:51.56Alfio:)
00:56.06*** join/#asterisk Iamnach0 (n=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net)
00:59.48*** join/#asterisk pmhaddad (n=pmhaddad@24-247-42-42.dhcp.mrqt.mi.charter.com)
01:00.17OrbixxAlfio: Recommend a particular model?
01:00.45Alfioit depends about your needs or your clients needs
01:01.19Alfioabout how many lines do you want, if you want Poe or not, if you want two ethernet ports or not, etc
01:14.33*** join/#asterisk Iamnacho (n=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net)
01:16.01*** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net)
01:29.06*** join/#asterisk mchou (n=quassel@unaffiliated/mchou)
01:38.34*** join/#asterisk DelphiWorld (n=Miranda@41.201.111.69)
01:38.38DelphiWorldhello all
01:38.44DelphiWorldwhat is a 477 error?
01:43.36*** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net)
01:43.40*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
01:45.48DelphiWorldplease what is a 477 error?
01:46.46jayteeit usually indicates a transmission error when between routers when sending over multiple hops
01:47.35DelphiWorldjaytee: ok thank you
01:48.00DelphiWorldjaytee: i'm getting it from a kamailio server (v1.5.x notls) in voipuser.org while calling troug it
01:50.40DelphiWorldBYE / thanks to all.
02:05.03*** join/#asterisk PhunTelTek (n=PhunTelT@cpe-76-188-233-188.neo.res.rr.com)
02:15.01*** join/#asterisk Alfio (n=Amunoz@adsl-54-26.tricom.net)
02:22.22*** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe)
02:38.57*** join/#asterisk eric_hill (i=eric_hil@204.94.175.11)
02:39.55eric_hillWould someone be able to help me figure out why Asterisk seems to be ignoring the SIP Remote-Party-ID message?
02:42.47*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
02:47.51[TK]D-Fendereric_hill: put "trustrpid=yes" in your peer entry
02:48.11leifmadsenseriously, what is with all the people lately messaging me to look at their stuff?
02:48.35[TK]D-Fenderleifmadsen: What kind of "Stuff" ;)
02:48.47*** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #aster
02:49.04leifmadsen[TK]D-Fender: I presume their asterisk "stuff" :)
02:49.14leifmadsenprefers not to look at male junk :)
02:49.24[TK]D-Fenderleifmadsen: They don't even warrant you're acknowledging their peddling :p
02:49.26[TK]D-Fender\o/
02:49.33[TK]D-FenderSUCKcess
02:50.47leifmadsenhmmm..... topic seems to be cut off now...
02:51.16*** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #aster
02:51.20eric_hill[TK]D-Fender: tx.
02:51.21leifmadsenweird
02:51.30leifmadsen[TK]D-Fender:  please fix if you get a chance.... maybe it's my client
02:52.19jayteewhen I type /topic it shows the whole thing. when you do it it cuts off after #asterisk-dev with just #aster
02:52.57jayteeand it did that yesterday or thursday I think
02:54.05*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
02:54.15eric_hill[TK]D-Fender: Still not seeing the name passed in the Remote-Party-ID.  Drats.
02:54.35[TK]D-FenderThere
02:55.39eric_hill[TK]D-Fender: Let me rephrase that, asterisk is receiving the Remote-Party-ID (sip debug says so)
02:55.55eric_hill[TK]D-Fender: The phone just isn't showing it, and asterisk isn't sending it to the phone...
02:56.44[TK]D-Fendereric_hill: And I see nothing at all
02:57.08*** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender
02:59.11eric_hill@[TK]D-Fender: http://pastebin.com/m55a0d35d
03:00.06eric_hillLine 135 -> received Remote-Party-ID "Hill Eric"
03:01.57eric_hill@[TK]D-Fender: sip.conf sections for the two endpoints: http://pastebin.com/m56e2d166
03:03.53[TK]D-Fendereric_hill: And you (should have) noticed that your incoming call FAILED to match any peer <-
03:04.17[TK]D-FenderNo user 'cellnumber' in SIP users list
03:04.18[TK]D-FenderNo matching peer for 'cellnumber' from '10.0.0.10:51229'
03:09.21eric_hill@[TK]D-Fender: I put a matching peer on the Cisco side and the error is gone, still not seeing cid name sent to phone: http://pastebin.com/m7fe794b2
03:10.18[TK]D-Fendereric_hill: And I see you trying to show me HALF of the story this time
03:10.26[TK]D-Fendereric_hill: Not the originating call coming in.
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03:10.42[TK]D-Fendereric_hill: Don't give me a runaround on this.
03:11.15eric_hill@[TK]D-Fender: I SWEAR I HAD DEBUGGING ON THAT IP!!?!?  <grumble>
03:11.51[TK]D-FenderAnd restricting to IP only debugs one leg...
03:11.56[TK]D-Fenderanother waste of time.
03:12.03*** join/#asterisk sirus (i=scott@bitch.stfu.org)
03:13.40sirusHello folks, I wanted to know if a telephone is actually needed to use asterisk? I'm wondering if a computer + some sort of app and a USB headset could work?
03:13.46eric_hill@[TK]D-Fender: Here, full sip debugging: http://pastebin.com/m1801835a
03:14.30eric_hillsirus: You can use a SIP softphone with asterisk.  Google for 100 different types.
03:15.03[TK]D-Fender~softphone
03:15.04infobot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
03:15.05sirusOk great I will need a functional sound card on the asterisk server?
03:15.43sirusThanks
03:15.50[TK]D-Fendereric_hill:  http://pastebin.com/m1801835a : 49-50 <- STILL no match
03:15.53sirus~zoiper
03:15.54infobot[~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com
03:16.53eric_hill@[TK]D-Fender: Ok, I'm missing something then.  I *have* a matching peer on the Cisco, but not asterisk.
03:17.15eric_hill@[TK]D-Fender: Do I need a peer on asterisk that matches my cell number?
03:17.47sirusIs it possiable to use a blackberry for voip?
03:18.03carrar<PROTECTED>
03:18.11sirusLike the 8330?
03:18.21*** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net)
03:20.10[TK]D-Fendereric_hill: Your inbound call is not matching a peer and not getting the trustrpid from it.  This ain't Raw-Cat Sigh Hence.
03:20.17[TK]D-Fendereric_hill: And my times up.
03:20.21[TK]D-Fenderis off for a while
03:20.33eric_hill@[TK]D-Fender: K.  Thanks for you help.  I'll keep plugging on it.
03:20.35[TK]D-FenderSirIf you find a CLIENT for it, sure
03:21.29sirusahh ok thanks
03:21.44sirusnow I just need to know if I need to install all the options for asterisk
03:25.06carraryes install all help files
03:25.17*** join/#asterisk BadHAL (n=nn@70-5-91-189.pools.spcsdns.net)
03:25.44siruscarrar: i'm talking able zaptel support and odbc support etc
03:25.55carrarodbc
03:25.56carrarheh
03:25.59carraruse postgres!
03:26.26sirushrmm ok
03:26.39sirusodbc sounds like microsoft stuff
03:26.54carrarand to top it off upgrade your kernel to 2.6.30.3
03:26.55carrarLinux kernel 2.6.30.3 #1 SMP Sat Jul 25 12:32:30 PDT 2009 x86_64 x86_64 x86_64 GNU/Linux
03:27.02carrarok maybe don't
03:27.08sirusI use FreeBSD
03:27.19carrarwhy
03:27.32siruscause i like the OS platform
03:27.41carrarI mean I like it too, just not for *
03:27.59sirusi'm new to it so
03:28.10carrarI'm getting that impression
03:29.00sirusasterisk will use up bandwidth i take it? better the bandwidth better the call?
03:29.23carrardepends on the codec and number of calls
03:29.59sirushrmm
03:30.05sirusits just a home project
03:30.23sirusfigurer i'd put asterisk on my dedi
03:30.30sirusand give the family voip
03:30.36sirusand softphones
03:30.47*** join/#asterisk ingenius (n=alektro@186.136.6.218)
03:31.46sirusCan I be my own SIP provider?
03:33.00*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
03:33.02carraryes
03:33.26siruscarrar: what would I need?
03:33.43carraryou need a way to terminate calls
03:33.55carrarpots lines, T1, another SIP provider
03:34.31sirusHrmm so a dedi serv wont be enough?
03:34.41carrardedi serv?
03:34.48eric_hillsirus: Who are you going to call?
03:35.03eric_hillsirus: With 1 phone, you could call and talk to the nice voicemail lady... :)
03:35.06siruseric_hill: everyone friends, family, buisness contacts etc
03:35.14carraryeah she's HOT++
03:35.23carrarhas a smokin asterisk dress too
03:35.23siruslol
03:35.50eric_hillsirus: Follow along here:  [your phone]---[asterisk]---????????---[your friends]
03:35.57eric_hillPlease fill in the blank.
03:35.58sirusI dont have any phones yet but softphones may work as I have plenty of usb headsets
03:36.05carrareric is that CLOUD?
03:36.31siruseric_hill: phone number?
03:36.32eric_hillcarrar, no that's (oOooOOOooO)
03:36.35carrarheh
03:36.46carrarYou can make up numbers sirus
03:36.56eric_hillsirus: Are you going to give everyone a softphone?
03:37.01carrarmake yourself areacode 1469
03:37.12sirusYes all my family, buisness contacts
03:37.34eric_hillSo you need [your phone]---[asterisk]---[theirsoftphone]
03:37.35sirusI also want to receive calls to
03:37.52carrarcould use 2 digit phone numbers
03:37.59eric_hillsirus: You'll need to set up each softphone with an extension in sip.conf
03:38.07carrarcall me at phone number 13
03:39.41eric_hillsirus: Put each phone in a context
03:39.56eric_hillsirus: Set up a dialplan for that context that calls the "other" sip phone.
03:40.54sirushrmm ok
03:41.02sirusI need to start looking for a softphone client
03:41.14sirusAsterisk is installing atm
03:41.43siruseric_hill: could I do all this on one box? with out using any other provider?
03:42.03eric_hillsirus: Um, yes?
03:42.29carrarxlite makes nice free softphone clients that work well with asterisk
03:42.39carrarworks in windows/linux/osx
03:42.58sirusPeople using providers paying 1cent a minute i was wondering if I needed to get something like that
03:43.09siruscarrar: cool thanks
03:43.51eric_hillsirus: Providers provide an on-ramp to the PSTN so you can call cell phones and land-lines.
03:44.14eric_hillIf you need to call a cell phone or land line, then you need a peer of some sort (ip, local line, etc)
03:44.30eric_hillIf you're just calling IP to IP phone, you do not need a PSTN on-ramp.
03:45.01sirusI'd like to call landlines,long distance, cell phones, and ip phones
03:45.22eric_hillThen you'll need some way to get those calls from the IP world to the PSTN world.
03:45.24siruscan i do the ip to ip then add on the pstn later?
03:45.30eric_hillOf course.
03:45.49sirusI have a box with ips that i can use
03:46.00carrarIP's are cool
03:46.00sirus~xlite
03:46.00infobot[~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/
03:46.33eric_hillIP's *were* cool.  ATM is where it's at.
03:46.40carrarhahah
03:46.50carrarATM is so last century ago
03:46.54eric_hillUses a PIN number to set up a phone call.
03:47.09eric_hillAnd sometimes craps a quarter out of the balen.
03:48.22eric_hillPlus, they use 0, 1, AND 2.  It's way more efficient.
03:48.33sirusdoes xlite work with avaya call center / ip phones?
03:49.01eric_hillsirus: Got me.  I'm too cheap to pay for avaya.
03:49.44carrarsend me a avaya and I'll test for ya
03:50.01siruswell work uses avaya we are looking for a softphone solution the avaya softphone keeps droping me off so perhaps this might work
03:50.25carrarwhy not connect asterisk to you avaya?
03:50.26eric_hillsirus: If the avaya softphone keeps dropping, you have a *connection* problem, not a softphone problem.
03:50.34carrarthen connect shit to your * box
03:51.02siruseric_hill: thats what im thinking cause when i was on the works network it worked flawlessly
03:51.16sirusso i belive its something with how the vpn connects us to the network
03:51.17carrarthen you can phase out the avaya
03:51.36sirusyep thx
03:51.42carrardump it in the trash one nice on the way out
03:51.46carrarnice=night
03:51.48eric_hillsirus: I deployed a few Cisco 800 series routers connected to a Cisco ASA.
03:52.00eric_hillsirus: Had one-way audio on the first call.  Found a bug in the Cisco IOS.
03:52.14eric_hillsirus: I'm sure Cisco hates me.
03:52.48sirus;/
03:53.30sirusSo for PSTN who would you recommend?
03:53.43eric_hillWhere are you on the planet?
03:54.20carrarwhich planet
03:54.32sirusUSA
03:54.44eric_hillDo you have a land-line?
03:55.15eric_hillDo you want to use it with asterisk?
03:55.27eric_hillDo you want a *second* line?
03:55.37eric_hillDo you want a SIP peer instead of a land-line?
03:55.44eric_hillDo you like green eggs and ham?
03:56.17siruslol
03:56.22sirus<PROTECTED>
03:56.31sirusno I have land line myself
03:56.45eric_hillHigh-speed 'net connection though?
03:56.54sirusoh yes
03:57.47eric_hillSo you want a good sip gateway.
03:58.25eric_hillhttp://www.voip-info.org/wiki/view/Sip+Phone+Service+Providers
03:58.43sirusThanks
03:59.05sirusif it works out perhaps I can ditch the cellphones
03:59.52eric_hillKind of hard to strap an asterisk box and cable-modem to your belt, but hey, whatever works for you.
04:00.03eric_hill(the batteries are a bitch)
04:00.41siruswell i'm always near a connection
04:00.49sirusand I always have my pc
04:00.54siruserr laptop
04:01.55sirusasterisk is almost done installing on the dedi box
04:10.23eric_hillBack to my SIP issue: http://pastebin.com/m45e0adbd
04:10.34eric_hillRemote-Party-ID received from the inbound SIP peer, not transmitted to the phone.
04:11.17sirushrmm
04:11.34carrartrustrpid = yes?
04:12.42carrarsendrpid = yes
04:12.46WindowsUseris anything much known about UMA?
04:13.01eric_hillcarrar: yes, and sip show peer cisco_3825 shows Trust RPID   : Yes
04:13.20eric_hillWindowsUser: UMA on T-Mobile works well for me.
04:13.50siruswishes he went with t-mobile
04:14.05sirusthey suck around here tho :<
04:14.06eric_hillwishes I could get a new Palm Pre or iPhone on t-mobile.
04:14.32sirusi like my 8330
04:14.37eric_hillI've got their tmo@home service.  $10/month, unlimited local and LD from home.
04:14.48sirusnice
04:15.01eric_hillI've got an 8820 with UMA.  I get free calls while I'm in our Mexico office on Wifi
04:15.12eric_hillRather than paying the WAY overpriced roaming.
04:15.31sirusnice
04:15.35eric_hillcarrar: sendrpid is not specified.  Will add that.  one sec.
04:16.35WindowsUserI'm in canada, and none of the GSM carriers up here have anything like the tmo@home adaptor
04:16.37eric_hillcarrar: Damn.  No change.
04:17.03eric_hillWindowsUser: http://fonosip.com
04:17.24WindowsUseralthough i guess I could go handset -> asterisk -> bluetooth -> cellphone -> etc
04:18.20sirusasterisk needs alot of packages to install
04:19.37WindowsUserpackages? usually packages is a kitchensink build, but i guess even without everything it is a bunch of deps
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04:21.27sirusHopefully this all works out
04:26.52PhunTelTekis there an ekiga client for windows?
04:28.02OrbixxYes.
04:28.52OrbixxCan somebody tell me how it's possible to programmatically determine in the dialplan whether any SIP clients are online?
04:29.46OrbixxOr rather, available.
04:30.42siruscool asterisk is done postgresql didn't want to install but I dont think I need it
04:33.54eric_hillOrbixx: core show function SIPPEER
04:34.13WindowsUserOrbixx: can you programmatically check if something exists in the dialplan? I know you can have the chan_sip stuff add a dialplan entry for peers that register, but i haven't seen anything using it
04:34.40siruseric_hill: can you help me set this up?
04:34.49WindowsUserregcontext=itsowncontext
04:34.58eric_hillsirus: I'll be around for another 20-30 minutes or so.
04:35.20eric_hillsirus: Then I'm going to get wickedly drunk and you won't want my help.
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04:35.35Orbixxeric_hill: I need to figure out how to translate that into a boolean value for whether or not ANY Sip clients are available.
04:35.37sirushehe i got the same plan
04:36.03siruseric_hill: i got asterisk installed what do I need to do next?
04:36.05eric_hillOrbixx, what are you trying to accomplish?  I mean, what's the purpose of seeing if there are any SIP clients???
04:36.23eric_hillsirus: Have you looked at the sample configs?  (did you even install the samples?)
04:36.35sirusyes I installed them
04:36.37Orbixxeric_hill: To determine whether to further the execution of the dialplan or not.
04:36.39siruslots of options
04:37.02eric_hillOrbixx, Typically you try the extension then follow the attempt with further processing if the Dial fails.
04:37.10WindowsUserOrbixx: noop(SIPPEER(line1,x)) then noop(SIPPEER(somethingthatdoesn'texist,x)) and compare the two lines :)
04:37.47eric_hillsirus: Do like I said earlier: Put entries in sip.conf for your softphones
04:37.56OrbixxWindowsUser: Not quite.
04:38.15OrbixxI'm trying to determine if one or more SIP client is available.
04:38.17eric_hillsirus: Set a context for the peer, and create a simple dialplan in extensions.conf with that context.
04:38.22OrbixxWhich means I need to check all SIP clients.
04:38.35WindowsUserwell im suggesting a method to figure out what to do
04:38.39WindowsUserrather than what to do
04:38.54sirushrmm
04:39.13WindowsUsercos it might return nothing if the peer isn't connected
04:39.16eric_hillOrbixx, you may be better off by putting the SIP clients into a queue and ringing the queue.
04:39.50eric_hillOrbixx, and a phone can be registered and still have no network connectivity.  Registration simply renews every so often.
04:41.50siruseric_hill: for the sip.conf thats where I need the provider?
04:42.16eric_hillsirus: start small.  Get two softphones and get them to connect to each other.
04:42.31eric_hillsirus: and yes, you'll put your sip provider in there *when you're ready*
04:42.36sirusI have xlite
04:42.53sirusI dont have another laptop softphone to test with
04:43.31eric_hillThen get the one softphone connected and try calling into voicemail.
04:43.40eric_hillWalk before you run grasshopper.
04:43.54sirussoftphone connected to the asterisk box?
04:44.42WindowsUseryes
04:45.35eric_hillsirus: You know, honestly, you might be better off with a softphone and a sip provider/registrar, and no asterisk box.
04:45.54sirushows that?
04:45.56eric_hillsirus: If you're going to be the only softphone connected to asterisk, there isn't really a need for it...
04:46.12eric_hillsirus: Unless you just want to play.
04:46.29sirusI plan to have multiple users connected to asterisk + i want to play
04:46.46sirusthis could be a new communications solution for me and my group
04:47.15WindowsUseryea I've got asterisk at my place, but I set up an SPA3102 that registers with my itsp for at my fathers house
04:47.24eric_hillsirus: Totally understand.
04:48.32sirusif the softphones work I'll get ip phones unless I can find a client for the 8330
04:49.13WindowsUserblackberry?
04:51.03sirusyep
04:54.59*** join/#asterisk McLazarus (n=McLazaru@dogpile.mcallister.ws)
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04:59.16*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
05:00.36sirusI'm gonna need a walkthrough on this anyone have any suggestions?
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05:06.49Snoogan[TK]D-Fender: thank you for helping me yesterday.  I have a quick question to pick your brain if i may
05:07.49eric_hillsirus: What are you having trouble with?
05:09.02sirusthe extensions.conf file
05:09.21eric_hillhttp://www.voip-info.org/wiki/view/Asterisk+dial+plan+-+working+example
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05:10.02Snooganmaybe someone else can shed some light.  my voip trunk won't register when I have nat=yes externip & localnet set in sip_general_custom.conf, but without it remote extensions suffer from oneway voice
05:10.21siruseric_hill:  maybe someone else can shed some light.  my voip trunk won't register when I
05:10.31sirusbah
05:10.32sirusdamn mouse
05:10.43sirus<PROTECTED>
05:10.43sirus<PROTECTED>
05:10.48siruswould that work?
05:11.09sirusmy extension being 1234 call comes in asterisk ox should answer
05:11.16sirusif not hangup?
05:11.26eric_hillNo, extension priorities need to be 1-2-3-4, not 1-4.  Typically use 1-n-n-n-n-n-n for next-next-etc
05:12.32sgiffordHello, I'm having some trouble with MeetMe and CDRs.
05:12.49sirushrmm
05:13.09sgiffordWhen I bridge a few callers together in a MeetMe room, the time doesn't seem to show up in any of the CDR entries.
05:13.10sirus<PROTECTED>
05:13.11sirus<PROTECTED>
05:13.14siruslike that?
05:13.25eric_hillsirus: yes.
05:13.32sgiffordsirius, the second number indicates an order of steps, so those instructions say "when a user dials 1234, first answer, then hang up"
05:13.39eric_hillsirus: Write these down:
05:13.44eric_hillsirus: core set verbose 5
05:13.53eric_hillsirus: sip set debug on
05:14.00eric_hillsirus: sip show peers
05:14.11eric_hillsirus: core show hints
05:14.38sgiffordsirius, from what you described, I think you want something like:
05:14.44sgiffordexten => 1234,1,Answer
05:15.17sgiffordexten => i,1,Hangup
05:15.37sgiffordThat is, if the user enters extension 1234, answer; if they enter an invalid extension, hang up.
05:15.37sirussgifford: i'm new to this i'm just trying to get it setup so I can make/recv calls
05:16.09siruseric_hill: in the conf?
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05:17.09eric_hillsirus: No, on the asterisk console.
05:17.11sgiffordsirius, you probably want to start with that working example URL eric_hill provided earlier.
05:17.19eric_hillsirus: Helps with debugging.
05:18.41sirushrmm ok
05:18.57sirustries to build an extensions.conf from scratch
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05:22.37sirusDo I have to set a default point of contact?
05:22.48eric_hillis leaving to go enjoy a black russian with grey goose and a splash of baileys coffee
05:25.32siruspeace eric_hill
05:25.36sirusthx for the info
05:32.47sirushttp://pastebin.com/m775681e2 <-- Is this good for my extensions.conf
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05:51.19WindowsUserwow
05:53.32WindowsUsersirus: what version of asterisk are you using?
05:53.41sirushrmm lets see
05:54.04sirus1.4.25
05:54.43WindowsUserah
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06:11.34[TK]D-Fendersirus: Doesn't do too much...
06:12.23[TK]D-Fendersirus: And globals  aren't generally needed for almost anything
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08:59.53datacompboyHi all:) Have anybody got TxFAX/RxFAX working after talk? I have stock Asterisk 1.4.21.2~dfsg-3 from debian, and build spandsp0.0.6~pre7 without core patches. Direct call to TxFAX(..) over Dial(SIP/<number>@<gw>) working fine (<gw> support g711 and t38), but if i do talk before, and then transfer destination to TxFAX(...) its heared nothing on receive side.
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09:07.02GnewtIs there a way to set line 2 of my softphone as in a different context?
09:08.34datacompboyGnewt: easyest way is use prefixes, like 1*NUMBER, and match in dialplan like exten => _1*...
09:09.38eansSIP Provider in Malaysia, any idea?
09:10.51Gnewtdatacompboy: Yeah, got that, was just wondering about lines
09:10.53Gnewtanyways
09:10.55Gnewtthank you! :)
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09:34.58datacompboyHi! Have anybody setup TxFAX/RxFAX in middle of talk?
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11:15.00viraptordoes anyone know how to get the jitter in some normal units (ms?) from the RTCP IA jitter value?
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11:40.45Gnutoohi,I'd like to use the IAX protocol....I've some windows clients too....is there a VOIP client software that does IAX and video and windows?
11:41.52Gnutooor should make them install a local asterisk server so that it does: ekiga->asterisk---IAX--->[Asterisk-server]<---asterisk-----<ekiga ?
11:42.59viraptorGnutoo: why do you need IAX at all? do you have any weird nat problems that prevent you from running sip, or something else?
11:43.29GnutooI want encryption + more efficient protocol
11:44.09Gnutoomore efficient because with ekiga I can't have video+sound because sound isn't good when I use video
11:44.43Gnutoothere is coccinella but it seems not to have video
11:48.20viraptorthat's ekiga's (or asterisk's), not sip's problem... try another client, or if you really want to play with other protocols, then h323 may be more interesting
11:50.14Gnutoook I'll look at h323
11:50.17Gnutoothanks
11:50.47viraptorGnutoo: new ekiga should support it, but I don't know if they handle h323 video too...
11:51.18Gnutoook thanks,by the way does h323 works like a normal protocol on one port only?
11:52.10Gnutoommm that still uses RTP
11:52.19Gnutoobetween the 2 clients
11:52.20viraptormmm... h323 works like a normal protocol - port per channel, IAX is the 'not normal' one
11:52.53Gnutooyes and IAX has DOS problems
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13:52.40intel352hey all, what is the best way to debug an Asterisk configuration that allows a SIP phone to register, but won't ring it?
13:52.40intel352it determines that calls are bound for extension 200, but drops to VM
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14:03.55Alfio~ itsp
14:03.55infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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
14:04.14Alfio~itsplist-us
14:04.15infobotextra, extra, read all about it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
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14:40.08datacompboyHi everybody :) Any way to make t38 work thru chan_local?
14:40.54datacompboyi.e. t38device<->sip<->chan_local<->sip<->t.38 provider<->fax machine.
14:45.38datacompboyor any way to temporary closure, i.e. do forward/redirect/smth other, for only fax conversation, with restore state after?
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14:54.02coppicewhat do you need to do that simple T.38 passthrough won't do?
14:55.02datacompboycoppice: mine system uses chan_local to drive call legs -- all calls are one leg to Dial() and second leg are chan_local, drived by AGI.
14:55.28datacompboySo when i try to pass-trhu, i have device<->sip<->chanlocal<->chanlocal<->sip<->pstn<->fax
14:56.13datacompboyon G711u everything OK, but provider support T38 (tested with direct TxFAX<->sip<->pstn), so want to make T38 works :)
15:00.02datacompboyin theory, i can temporary do that with async_goto, but how to transfer leg back to chan_local?
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15:09.49Gnutoohi I'd like to do the following : sip-client->asterisk-with-iax2->|firewall|----internet---->asterisk(directly connected to the internet)-with-iax2
15:09.59Gnutooso here are my question:
15:10.38Gnutoo*is it possible that one of the asterisk server is behind a nat ?
15:11.33pifhi, what is chan_vpb ?
15:11.48Gnutoo*does asterisk 1.2.26.2 have iax2?
15:13.18sirusHrmm
15:13.37sirusI have my extension.conf / sip.conf setup just can't seem to make any calls ;/
15:14.21datacompboyGnutoo: yes, you can connect * behind nat
15:15.01datacompboyif any one of it have full connection. otherwise, try to use external proxies / tunnels (very difficult solution, and may be unreliable for quality)
15:15.40sirusdatacompboy: I have my asterisk box setup along with the extensions.conf / sip.conf i'm using xlite for my softphone I've plugged in the information but I'm unable to make calls can you help me debug?
15:16.13Gnutoodatacompboy, ok thanks, so is there a comment in iax.conf telling how to connect to another asterisk server running iax?
15:16.54datacompboysirus: what exactly wrong with dial out? have you checked context, to where placed calls?
15:17.20sirusHrmm I dial an extension and it acts like it doesn't see it
15:17.35datacompboysirus: check context
15:18.02Gnutooah found it
15:18.06Gnutooit's named register
15:18.20datacompboyGnutoo: yes, you need to register, then dial @iaxpeer
15:18.46sirushrmm ok
15:19.44datacompboysirus: check context= in [yourpeer] in sip.conf, it must be same, as [somecontext] in extensions.conf
15:20.12Gnutoook thanks
15:23.50sirushrmm
15:25.28sirusdatacompboy: perhaps you can give me an example?
15:28.37datacompboysirus: http://pastebin.com/m736d0e28
15:29.16sirusthx
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15:32.03sirusjust doesn't get it
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15:46.59pifafter upgrading to 1.4.26 I get "ast_carefulwrite: write() returned error"
15:47.51datacompboypif: when?
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15:48.07pifwhen munin logs in to the manager interface
15:49.00datacompboypif: dunno. seems, googling doesn't help?
15:49.11pifso far no
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16:01.04Gnutoommm  my setting for the iax seems correct but ekiga can't register to the IAX client asterisk server
16:01.51datacompboyGnutoo: its in same net than asterisk server, where it tries to register?
16:02.05Gnutooyes 127.0.0.1:5160
16:02.25GnutooI've tried several things for a while and none worked
16:02.30Gnutoothe account seem correct
16:02.40Gnutooit's binded on udp port 5160
16:02.49datacompboy:)) well, on what ip asterisk listens, and on what port ekiga listens?
16:02.53GnutooI binded sip on that port because ekiga uses 5060
16:03.11Gnutooboth on 0.0.0.0 I think
16:03.16pifhow does a polycom phone 'knows' it has messages?
16:03.24pifis it mailbox= in sip.conf ?
16:03.28Gnutoobut I changed asterisk's port
16:03.33Gnutooto 5160
16:03.36Gnutoobecuase of that
16:03.41datacompboyGnutoo: check with "netstat -nap" as root
16:04.13Gnutoonetstat -anup gives: udp        0      0 0.0.0.0:5160            0.0.0.0:*                           14869/asterisk
16:04.14datacompboypif: http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
16:04.30pifkay
16:04.48AlfioGnutoo with that address it wont register
16:04.50datacompboyGnutoo: well. you register with sip or iax? since 5060 is sip
16:05.03Alfio127.0.0.1:5160
16:05.04Gnutoobasically here's my setip
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16:05.29Gnutooekiga->asterisk-iax-client--internet----asterisk-iax-server
16:05.41Gnutooekiga and asterisk iax are on the same machine
16:05.58GnutooAlfio, what address should I use?
16:06.17datacompboyGnutoo: why not get just IAX client instead of ekiga ? :)
16:06.38Gnutoobecause the person that will use it is used to ekiga and because of video
16:06.43Alfioother diferent than 127.0.0.0 network
16:06.53Gnutoook
16:07.08Alfiothat address is for loopback
16:07.26GnutooAlfio, yes indeed
16:07.42GnutooAlfio, ekiga and asterisk-iax-client are on the same machine
16:08.06Gnutoobasically I want to use asterisk as a bridge to iax
16:08.37GnutooAlfio, 192.168.1.106 doesn't work either
16:09.00Alfiodid you change in the ekiga client the port
16:09.03Alfio?
16:09.34datacompboyGnutoo: why not use SIP on remote asterisk?
16:09.39Gnutoodatacompboy, I want better bandwith usage,video and encryption
16:10.02Gnutoodatacompboy, I tried other client than ekiga...such as sip communicator,the audio was laggy as hell
16:10.21Gnutoomaybe 30s or 1min later I had the audio
16:10.38datacompboyGnutoo: better bandwith just means you should use other codec. iax/sip have no difference there
16:10.52AlfioGnutoo did you change the sip port listen in ekiga?
16:10.54pifdatacompboy: I removed "mailbox=" in my phone's sip.conf section, yet MWI keeps blinking
16:11.06pifI want to disable MWI for that line
16:11.09GnutooAlfio, no I didn't change ekiga's port but I changed asterisk's port
16:11.23Gnutoodatacompboy, ok thanks I use speex
16:11.31Alfioand how ekiga knows that?
16:11.32Gnutoodatacompboy, how about encryption...?
16:11.43datacompboyGnutoo: brr... poorest codec that I have heared.
16:12.02GnutooAlfio, because I typed 127.0.0.1:5160 in the registar box
16:12.04datacompboyGnutoo: G723/G729 better for low bitrate quality
16:12.13Gnutoook I'll try that at once
16:12.25datacompboypif: Asterisk sip checkmwi = Number : Global interval (in seconds) between mailbox checks. Default 10 seconds. (New in v1.2.x)
16:12.50datacompboypif: also, may be not remove, but put line "mailbox=" ?
16:13.41pifI tried that
16:13.55datacompboypif: agh, i see your post on google.
16:14.04pifheh
16:14.29AlfioGnutoo ok, continue with that address
16:15.22*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
16:15.58datacompboypif: http://www.voip-info.org/wiki/view/Asterisk+Realtime+MWI+Hacks, may be allowsubscribe=no in phone section, or subscribecontext = <context_name> with non-exists context_name ?
16:17.58*** join/#asterisk eric256 (n=Administ@c-67-165-208-191.hsd1.co.comcast.net)
16:18.50datacompboypif plus, may be, also null vmexten
16:19.12eric256hey...i've got a phone i formated and i'm trying to get it to reload the app files, but the tftp server keeps erroring with "Requested file to big, increase BLKSIZE" but in.tftp doesn't seem to have a way to increase the block size...any ideas?
16:19.19pifthx, looking
16:19.52datacompboyeric256: what os?
16:20.11eric256CentOS release 5.2
16:20.38eric256i'm thinking i might have to setup ftp and use that to provision from..only solution i can come up with
16:21.08Gnutoodatacompboy, I tried G722 and It's a lot better thanks a lot
16:21.34datacompboyeric256: agh... BLKSIZE seems to set on TFTP client side...
16:23.10datacompboyeric256: what cmdline do you use to send?
16:24.36Gnutoowhat should I test now?
16:25.09datacompboyGnutoo: can you pastebin ekiga debug log and asterisk console output while your phone tries to register?
16:25.11eric256its running through xinetd so i don't run anything to send it... "server_args             = --logfile /var/log/atftpd.log  --no-blksize"
16:25.41eric256i just added the --no-blksize to try and get around the issue. same error with and without it though
16:25.48Gnutoodatacompboy, the test client has asterisk 1.4...so how can I have debug on asterisk 1.4?
16:25.59Gnutoothe real client will run asterisk 1.2
16:26.00datacompboyGnutoo: sip set debug
16:26.04Gnutoook thanks
16:26.22Gnutoobut sip set debug require someone registered
16:26.57datacompboyGnutoo: it shows all SIP messages :) even tries
16:27.17datacompboyeric256: add "--option blksize 1428"
16:27.17Gnutooah ok without the on
16:27.19Gnutoothanks
16:28.09eric256i've tried somethign similar, but happy to try that too. lol
16:28.18eric256same error.
16:28.37drmessanoAnyone know a good serial <> ethernet server.. ?
16:28.54datacompboyeric256: also, maybe use other server? like tftp-hpa
16:29.11Gnutoodatacompboy, http://pastebin.com/d677add31
16:29.29eric256okay, yea i'll see if i can find another tftp, if not i guess i'll just setup an ftp for this upgrade
16:29.30eric256thanks
16:30.30datacompboyGnutoo: show output of "sip show peers"
16:31.17Gnutoo102/102                    192.168.1.106    D          5060     Unmonitored
16:31.26Gnutoo+ others things with others clients...strange
16:31.36Gnutooah yes
16:31.43Gnutoothe others client are not connected
16:31.44datacompboyo - enable qualify=yes :) in sip.conf, its cool to see is it OK :)
16:31.45Gnutooso no problem
16:32.00Gnutoothere is also
16:32.01Gnutooekiga/GNUtoo               86.64.162.35                5060     OK (66 ms)
16:32.40AlfioGnutoo asterisk is listen in 5060
16:32.48Gnutooouch...
16:32.54Gnutoook I'll look why
16:33.57datacompboyAlfio: nope, he changed asterisk to 5160
16:34.04GnutooAlfio, udp        0      0 192.168.1.106:5060      0.0.0.0:*                           17270/ekiga
16:34.10Gnutoofrom netstat -anup
16:34.50Alfiodatacompboy nope
16:34.55Alfioits listen in 5060
16:35.08datacompboyAlfio: [23:04:14] <Gnutoo> netstat -anup gives: udp        0      0 0.0.0.0:5160            0.0.0.0:*                           14869/asterisk
16:35.37Alfiodatacompboy did you see the sip debug
16:35.41Alfio?
16:35.43*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
16:36.04datacompboy<--- SIP read from 192.168.1.106:5060 ---> REGISTER sip:192.168.1.106:5160 SIP/2.0
16:36.18Alfiohi [TK]D-Fender
16:36.24datacompboythat means that it reads from ekiga on :5060, that tries to register @:5160 :)
16:36.29datacompboyso, asterisk on 5160
16:36.35datacompboyHi [TK]D-Fender!
16:37.03[TK]D-Fendersenses not-so-quiet desperation...
16:37.11Alfiohehehehehhehhe
16:37.22Alfiocan you check please this sip debug
16:37.25Alfiohttp://pastebin.com/d677add31
16:37.25datacompboyGnutoo: its seems to be OK in log -- your ekiga registered:)
16:37.33Gnutoook
16:39.13[TK]D-Fenderyup
16:39.29[TK]D-FenderAlfio: it OK after the re-auth
16:40.13Alfiowhat port are using asterisk to listen?
16:40.45Alfioto sip listen*
16:41.19[TK]D-FenderAlfio: What does your sip.conf say?
16:41.27[TK]D-Fenderwhat does netstat say?
16:42.47Alfiowe were trying to help Gnutoo with one configuration
16:42.56datacompboy[TK]D-Fender: its on 5160 :) sip.conf and netstat says 5160
16:43.02[TK]D-FenderAlfio: And as it does appear that you're running Ekiga on your * box they should be on different ports
16:43.27Gnutoomaybe the problem is asterisk-iax-client->asterisk-iax-server then
16:43.46*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)
16:44.21datacompboyGnutoo: so, what -exactly- problem you have now?
16:45.26GnutooI understand...
16:45.47Gnutooif I call @192.168.1.1 it pass trough the iax and seem to work...
16:46.10Gnutoothanks a lot!!!
16:46.30Gnutooi was calling @127.0.0.1
16:46.34Gnutoowith the IAX dial plan
16:46.48datacompboyGnutoo: well, glad to see it working :)
16:46.51Gnutoo(IAX/101) instead of (SIP/101)
16:47.21*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
16:47.30*** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
16:47.31mbrevdahelp sip show
16:47.32mbrevdaNo such command 'sip show'.
16:47.38mbrevdawhat does that mean?
16:48.23mbrevdanm
16:48.30Gnutoommm 102/102                    192.168.1.106    D          5060     Unmonitored
16:48.34Gnutooon my main ekiga server
16:48.37GnutooI'll try something
16:50.59datacompboynow, again: does anybody knows how to ask call leg to execute some extension _and return back_ after? Here we have call <chan_local>-<sip>-<pstn>; I'm drive over <chan_local> with AGI script. If i do Exec RxFAX /to/somewhere, everything ok, but fax received over G711. If I do ChannelRedirect, it received over T38 ok, but it doesn't return back to local chan :(
16:55.09eric256datacompboy: thanks for the help, got vsftp up and working and seems to be transfering the app...just gotta wait till it finishes loading now
17:01.15*** join/#asterisk eans (n=eans@60.53.236.108)
17:02.51*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
17:05.13*** join/#asterisk Da-Geek (n=Da-Geek@nsabfw1.nsab.se)
17:05.59Da-GeekHi all.....
17:06.08datacompboyhmmm... does Parking can be done for chan_local channel?
17:07.01Da-GeekAnyone got any advice about getting a CP7941 working if it need to connect to a non local * box ?
17:07.22jqlpray?
17:07.25Da-Geekhas been banging his head against a wall for 2 days now...
17:07.42Da-Geekjql: thats my next optin after IRC :-)
17:08.01jqlcisco sip via NAT only works through ridiculously awesome sip-aware firewalls
17:08.03jqlor a PIX
17:08.10Da-Geekis 99.9% sure it has something to do with NAT but just can not get it going... :-(
17:08.55Da-Geekjql: i do have a PIX-501E sitting under my desk, but am not using that....
17:09.16jqlif you were using it, it'd work
17:10.31Da-Geekjql: if is is that strait forward i might just have to dust it off and plug the ol' girl in
17:10.38*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:10.46eppigyDa-Geek: what are you using now?
17:11.10*** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com)
17:11.12jqlyeah. as long as the PIX has a public IP, and you turn on sip fixup, you're at least going to have the possibility of communicating with asterisk
17:11.25jqlon the peer config of asterisk, turn off nat
17:11.31Da-Geekeppigy: dont ask :-(, Speedtouch 716
17:11.41jqlas far as asterisk is concerned, the phone will be directly connected to the internets
17:11.43eppigylol never heard of it
17:12.11Da-Geekjql: everything i have read says i should have nat=yes
17:12.46Da-Geekjql: i get it, nat=no 'IF' the PIX is connected..
17:12.47jqlwith a cisco phone, the situation is different. cisco is asymmetric
17:12.51jqlyes
17:12.52jqlexactly
17:12.56Da-Geekeppigy: ISP supplied...
17:13.08eppigyyeah dude you cant use that
17:13.12jqlif you use the PIX, you can pretend the phone isn't behind NAT at all
17:13.28eansWhat is digium card?
17:13.32eans?? digium card
17:13.49eppigydigium.com
17:13.50[TK]D-Fendereans: www.digium.com
17:14.00eppigyDOT COM
17:14.10[TK]D-FenderTRABAJO
17:14.18eppigyDONDE
17:14.24[TK]D-FenderESTA!
17:14.29eppigyMEUI BIEN
17:14.31eansok
17:14.45eppigyeans: FRIJOLES
17:15.09eansFRIJOLES?!
17:15.10eansWhat is that lol
17:15.33eppigyME AMIGO EL USTED DAVE CASEY
17:15.37Da-Geekthinks he now has to change his crappy ISP supplied router with the PIX 501E he has...
17:15.42eansI cant understand at all
17:15.48*** join/#asterisk Dovid (n=annon@213.8.121.90)
17:16.01Dovidanyone know if I can download the code to the Asterisk ADA ?
17:17.47eppigyare you flying that plane?
17:22.43*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
17:24.39[TK]D-FenderDovid: ADA?
17:26.47*** join/#asterisk xpot-mobile (n=james@174-23-132-93.slkc.qwest.net)
17:27.56*** join/#asterisk errotan (n=errotan@5403E450.catv.pool.telekom.hu)
17:28.16drmessanoDovid: Not that I am aware of
17:28.52drmessanoeppigy: MI ESPOSA ES BEN TRAMPLADO POR TOROS
17:29.02eppigyAHHH SI
17:29.04eppigySI ME GUSTA
17:29.58drmessanoeppigy: EL [TK]D-Fendero ES LOCO
17:31.44eppigySI MUCHO LOCO
17:31.47eppigyEL BANO DIABLO
17:32.29Gnutooah...I have put the client's asterisk server on 5060 it worked...on 5160 it didn't anymore
17:32.34GnutooI'll try to investigate
17:32.45Gnutoobut is the autentification separate?
17:39.54eppigyhttp://vimeo.com/1743331
17:40.01eppigydude the video of the now has me lollin
17:42.33DovidTK: The windows plugin to outlook that connects to Asterisk
17:44.54mbrevdaits dosent do jabber...
17:46.50[TK]D-Fenderok/fine/sure
17:47.48*** join/#asterisk voxter (n=voxter@190.241.15.56)
17:48.20*** join/#asterisk stope (n=ejc@i209-195-113-143.cia.com)
17:48.31stopeanyone how to get rid of this error message:   WARNING[3396]: loader.c:400 load_dynamic_module: Error loading module 'app_fax.so': /usr/lib/asterisk/modules/app_fax.so: undefined symbol: t30_set_header_info
17:49.06stopeI have spandsp 0.0.6 pre 8 installed
17:49.22stopeand asterisk 1.4.25.1
17:49.30stopethe fax modules just wont load!
17:49.42*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
17:50.55Corydon76-digstope: clearly, you don't have EXACTLY the version that was specified in the README
17:51.33stopegoes to read the README file
17:51.41Corydon76-digspandsp isn't a library that you can use "X or above".  The API changes so frequently that you need EXACTLY the version recommended
17:51.52[TK]D-FenderCorydon76-dig: Have you considered getting that giant flashing neon sign for it? ;)
17:52.02stopeah, I take it the readme in *
17:52.10stopereads
17:52.21Corydon76-dig[TK]D-Fender: do you EVER take a day off?  ;-)
17:52.34[TK]D-FenderCorydon76-dig: It happens...
17:52.58[TK]D-FenderCorydon76-dig: Though there is almost always some time of day I'm at home
17:53.12[TK]D-FenderCorydon76-dig: And I'm on at work, so thats mon-fri
17:54.16*** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net)
17:54.42Corydon76-dig[TK]D-Fender: what is it that you do, that you can help out on here?
17:55.25[TK]D-FenderCorydon76-dig: I am the IT dept for my company, and I multi-task all day
17:55.33*** join/#asterisk superc (n=superc@port-87-234-216-82.static.qsc.de)
17:56.59Gnutooit was a problem in ekiga
17:57.09Gnutooekiga didn't listen on 127.0.0.1
17:57.16*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:58.09stopeok, which readme? the asterisk or the spandsp?
17:59.28AlfioGnutoo i told you at the first moment
17:59.37Alfiodo you remember that?
18:03.24[TK]D-Fenderstope: Think for a second as to which depends on which
18:05.04stopeasterisk depends on spandsp in that its the correct version and if not, then asterisk wont load it, so the * docs have to be read
18:05.20*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
18:06.48supercbtw. any hints on my bugpost https://issues.asterisk.org/view.php?id=15578 regarding spandsp/fax?
18:11.27*** join/#asterisk many (n=many@dslb-088-067-174-219.pools.arcor-ip.net)
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18:32.18*** join/#asterisk tamiel (n=tamiel@ip-230.net-82-216-103.rev.numericable.fr)
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18:34.54*** part/#asterisk sjobeck (n=Adium@75-164-154-24.ptld.qwest.net)
18:35.25*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
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18:45.35Alfio~ itsplist-usa
18:45.54eans?
18:46.08Alfio~itsp
18:46.09infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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
18:46.17Alfio~itsplist-us
18:46.18infobotwell, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
18:46.35eansAny of them offer free trial?
18:46.41[TK]D-Fenderlol
18:46.43Alfio:{
18:46.47[TK]D-Fendereans: And why would they do that?
18:46.53Alfiohahahahahahhaha
18:47.00*** join/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com)
18:47.01[TK]D-Fendereans: Think you're getting a free lunch?  Don't bet on it
18:47.04eansI just wanna confirm if my system is will work.
18:47.16[TK]D-Fendereans: test with ekiga.net
18:47.29Alfiowww.gafachi.com they will give you one dollar i think
18:47.52Alfionot for your pocket, just for test
18:47.55Alfio:)
18:48.02eansof course lol.
18:48.13eansIm not going to make any outgoing calls by the way.
18:48.24eansI just want to see if I can receives call by DID in U.S
18:48.51[TK]D-Fendereans: then : ipkall.com
18:57.12mchou[TK]D-Fender: so a friend has an interesting scenario.  He can connect to his ITSP with twinke, but not asterisk (and quite a few othe soft phones)
18:58.16mchou[TK]D-Fender: the sip trace is a bit hard to make out, but it seems the itsp requires an additional proxy auth that somehow ends up getting "lost" in *
18:58.17[TK]D-Fendermchou: You know whats remarkably common about this scenario?  I ahve jsut as much real debug and information to go on as just about every other schmuck brings in here :)
18:58.32[TK]D-Fendermchou: "show me the money"
18:59.09mchou[TK]D-Fender: I can show you twinkle sip log but I dont have * log
18:59.26mchouand since it's a friends acct I can't log in atm
18:59.56[TK]D-Fendermchou: Believ me, I don't really care about twinkle.... * is failing, so debug that
19:00.00mchouI just want to know how asterisk handles additional proxy auth challenges
19:00.17[TK]D-Fendermchou: When you've got something to show, then I'll have something to say
19:02.54*** join/#asterisk SlicerDicer (n=SlicerDi@24.138.244.145)
19:08.40mchou[TK]D-Fender: http://pastebin.com/d7bb7c2c7
19:10.22mchou[TK]D-Fender: here is twinkle log for comparison: http://pastebin.com/d72ffde2
19:15.46mchou[TK]D-Fender: you there?
19:18.50eppigyare you flying that plane?
19:20.21mchoueppigy: lol
19:20.24*** join/#asterisk BenCecka (n=bcecka@ip72-208-162-32.ph.ph.cox.net)
19:20.37mchoueppigy: it's on autopilot now :)
19:20.40[TK]D-Fendermchou: - Executing [15053010785@MarkDialGroup:1] Dial("SIP/6001-08408f78", "SIP/phonemaxpr,15053010785") in new stack
19:20.45[TK]D-Fendermchou: not a valid dial <---
19:20.56mchouhmm???
19:20.57[TK]D-Fendermchou: you are not timing out for MILLIONS of seconds <-
19:21.08[TK]D-Fendermchou: s/,//
19:21.23[TK]D-Fendermchou: You aren't passing the number because of the formatting error
19:21.36[TK]D-Fendermchou: Silly if you ran around all this time for such a clear mistake
19:23.12[TK]D-Fendermchou: To: <sip:cwferrel25@phonemaxpr.com>        <- should have notived the # missing from the "To:" as well
19:23.18[TK]D-Fendernoticed*
19:23.31eppigySILLY BILLY
19:24.00mchou[TK]D-Fender: http://pastebin.com/m34fb676b
19:24.07mchoucorrected, same dealio
19:25.14[TK]D-Fendermchou: I'd have to see the peer
19:25.36mchouwhat, in sip.conf?
19:25.43[TK]D-Fendermchou: Where else.....
19:25.55eppigyin the lost gnostic gospels
19:25.59eppigynaturaly
19:26.21mchouyo, there's no need to be snide
19:26.32eppigywhoa calm down
19:26.35eppigyim just doin me
19:26.46eppigyyoure among friends here
19:27.44[TK]D-FenderImagine if you had enemies ;)
19:27.53[TK]D-Fender:F
19:28.31eppigyyesh
19:29.28mchou[TK]D-Fender: http://pastebin.com/d59bb93c4
19:29.38eppigyit just astounds me that a man as great as beethoven is the leader of los angeles
19:31.05BenCeckaare either of the asterisk gui's (freepbx or asteriskgui) dependable enough to use or am I better off sticking with the core asterisk build?
19:31.44eppigyyou are better off sticking with the core build
19:31.48eppigycompilign from source
19:31.53eppigycompiling
19:31.59eppigyand learning from the ground up
19:32.14eppigyonly when you have mastered something should you use shortcuts and GUI's to save time
19:32.32BenCeckathat's what I thought. I've tried every gui and iso that I could find and have recently settled on building it from source on a slackware box.
19:33.08BenCeckajust trying to put together some budgets for how I'll get our infrastructure up and running
19:33.56[TK]D-Fendermchou: fromdomain=phonemaxpr.com , type=peer
19:34.21BenCeckais there a current book anybody would recommend? voip-info.org seems a little outdated in parts
19:35.32[TK]D-Fender~book
19:35.33infobot[~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
19:35.40eppigyeverything that I know of is outdated
19:35.59mchou[TK]D-Fender: ok, thx.  that works.  care to explain?
19:36.19eppigythe book only covers 1.4
19:36.25eppigybut is enough to get you going for sure
19:36.33mchou[TK]D-Fender: what does fromdomain actually do?
19:36.35[TK]D-Fendermchou: fromdomain ID's you as being part of their proxy network as opposed to just some call randomly directed at them
19:36.50mchouok. cool
19:37.00BenCeckayeah, i've downloaded that and have been going through some of the basic setup. at least I can stop trying to find something newer then
19:37.11BenCeckathanks eppigy and [TK]D-Fender
19:37.14eppigyyes
19:37.30eppigythere was supposed to be an asterisk cook book released like early this year
19:37.35eppigybut that never happened
19:37.39eppigyor maybe even last year
19:37.43eppigyI forget
19:38.09BenCeckaoh well. I suppose the changelog between 1.4 and 1.6 should tell me about any major differences right?
19:39.05eppigyhttp://www.oreillynet.com/etel/blog/2007/01/help_create_the_asterisk_cookb.html
19:39.10eppigyhaha just reading the first comment
19:39.22eppigyis probably a pretty good indication of why it never made it
19:39.40eppigyBenCecka: yes
19:39.51eppigybravo for knowing what a changelog is
19:39.59eppigyI mean that sincerely
19:40.15BenCecka:) thanks. I try
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20:41.16DigitalDaz21Hi, anyone know anything about using outbound sip proxies?
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20:57.02ManxPowerDigitalDaz21: you almost never need to specifically configure a SIP proxy.  Just set it up at a peer/friend.
20:57.38ManxPowerRemember SIP proxies just proxy the call control (SIP), not the audio (RTP)
20:59.12DigitalDaz21The host in this case is sitting behind a BT openser proxy and is not accessible from the net
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20:59.44ManxPowerso set up that as a peer and send the calls to the proxy.  It will handle the rest
21:00.11DigitalDaz21I've managed to get it working but only through a mistake :)
21:02.20DigitalDaz21I'm using freepbx and it appears that if I put everything in correctly, no sip traffic is generated, if I then put the peer as a resoleable dns name then I get sip traffice but failure as expected. Then if I put the correct info in again and reload, everything works perfectle
21:03.17ManxPowerYou should ask on the FreePBX channel.
21:03.23ManxPowerwhat is the proxy you need to connect to?
21:03.28DigitalDaz21I have, no reply yet
21:03.52DigitalDaz21www.bbvservice-560129.bt.com:5060
21:04.00ManxPowerpastebin your config for that proxy/peer from sip.conf
21:04.27DigitalDaz21like I say, its working perfectly now but if I restart, I will have to mess withit again before it works
21:06.20ManxPower~pb
21:06.21infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
21:07.04ManxPowerchances are you are behind NAT so the translation times out.  qualify=yes in the peer/friend/user sip.conf entry should help with tht.
21:07.42ManxPower~nat
21:07.42infobotsomebody said nat was Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
21:07.44DigitalDaz21http://pastebin.com/m5291c0f5
21:08.02DigitalDaz21The server is not behind nat and no firewall
21:08.45ManxPowerWhat happens if you replace "outboundproxy=www.bbvservice-560129.bt.com:5060" with "host=www.bbvservice-560129.bt.com:5060"
21:08.50*** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
21:09.39DigitalDaz21that config I have pasted will not work but if I change host to sipgate.co.uk then do a sip reload, then change host to bmnha-01.bt.com after another sip reload everything works perfectly
21:10.02DigitalDaz21tried that and it fails
21:10.06ManxPowerIt sucks to be you.  I wish you the best of luck.
21:10.41DigitalDaz21lol, like I said, its working its just annoying me that I can't figure out why I have to do what I'm having to
21:12.10Gnutoohi,is that correct: exten => 101,1,Dial(SIP/101)
21:12.15Gnutooin extensions.conf
21:12.19Gnutoobecause I have that:
21:12.31Gnutoo[Jul 26 22:59:36] NOTICE[6640]: chan_sip.c:14035 handle_request_invite: Call from '101' to extension '102' rejected because extension not found.
21:12.42Gnutooand I have context=home on both users
21:13.02GnutooAlfio, ok sorry
21:13.58ManxPowerGnutoo: Do you have a [101] section in your sip.conf?
21:14.04Gnutooyes
21:14.06GnutooI have one
21:14.28ManxPowerand you have a [home] section in extensions.conf?
21:14.39[TK]D-FenderGnutoo: DIALPLAn error
21:14.47Gnutoook
21:14.57[TK]D-FenderGnutoo: You do not have a match for 102 in the context its looking it
21:14.58[TK]D-Fenderin*
21:15.00ManxPowerand what is the type= set to in the [101] section?
21:15.15[TK]D-FenderManxPower: Doesn't matter.  It auths
21:15.21[TK]D-FenderManxPower: C'mon you know this...
21:15.26Gnutootype=friend
21:15.28[TK]D-FenderManxPower: straight up 404
21:15.37[TK]D-FenderGnutoo: Your peers are fine, your dialplan is not.
21:15.38ManxPower[TK]D-Fender: they have changes how type= works so often I don't trust it at all.
21:15.44Gnutoois the context home?
21:15.51ManxPower[TK]D-Fender: but you are welcome to help him
21:15.57[TK]D-FenderManxPower: it confirmsgnugo pastebin a failed callw ith SIP debug.
21:16.02[TK]D-Fendersklfjkl;sjfse;rlktj gm[werty
21:16.07[TK]D-Fenderugh
21:16.10ManxPowerI did not see his pastebin
21:16.16[TK]D-FenderGnutoo: PASTEBIN the failed call with SIP debug enabled
21:16.21Gnutoook
21:16.23[TK]D-FenderManxPower: neither do I
21:16.34Gnutoo[Jul 26 23:11:34] NOTICE[7578]: chan_sip.c:14035 handle_request_invite: Call from '101' to extension '102' rejected because extension not found.
21:16.50Alfiohehehehehhehe
21:16.50Gnutoothat's all I have
21:17.08GnutooI'll add debuging
21:17.46Alfio[TK]D-Fender we need to create some entrys in the infobot to deal with those situations
21:17.57[TK]D-FenderAlfio: Which?
21:18.10Alfiolike paste sip debug
21:18.13[TK]D-Fender~wmmfpb
21:18.14infobot[~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!?
21:18.14Gnutoohttp://pastebin.com/m30365fe8
21:18.15[TK]D-Fender^^^^^^^^
21:18.18[TK]D-FenderAlfio: there
21:18.23Alfiohehehehehheheh
21:18.55[TK]D-FenderGnutoo: Looking for 102 in home (domain 192.168.1.106)
21:19.00Gnutooyes
21:19.08[TK]D-FenderGnutoo: no match for "102" in [home] just like it says
21:19.17Gnutooah I'll verify that
21:19.22[TK]D-Fender"SIP/2.0 404 Not Found"
21:19.57DigitalDaz21From a newbie, should that nat be on??
21:20.37Gnutoommm I bet exten => 102,1,Dial(SIP/102) is not sufficent in [home] in extensions.conf
21:20.52GnutooI'll look at the doc
21:21.07DigitalDaz21Sending to 192.168.1.109 : 5060 (NAT)
21:21.19ManxPowerI'll bet he's typoed something.
21:21.27ManxPowerDigitalDaz21: But there's no NAT!
21:21.36ManxPowerlooks over his glasses at DigitalDaz21
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21:21.55ManxPower[TK]D-Fender: My bet is he added extra spaces in his exten line.
21:22.12[TK]D-FenderManxPower: I see 1 exten line, and I don't trust what I DO see..
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21:22.33[TK]D-FenderGnutoo: ........pastebin your f-ing dialplan.
21:22.37[TK]D-Fender*argh*
21:22.37Gnutoook
21:22.39Gnutoosorry
21:22.52[TK]D-FenderGnutoo: Of course you can't see the problem.  If you could, it'd be FIXED already
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21:26.08Gnutoohttp://pastebin.com/m4e7cdd7f
21:26.16Gnutoos/IAX/SIP
21:26.43Gnutoochange exten => 101,1,Dial(IAX/101) for exten => 101,1,Dial(SIP/101)
21:26.51Gnutoobecause it was just changed
21:26.56Gnutooin order to try something
21:28.03[TK]D-FenderGnutoo: go "reload"
21:28.25[TK]D-FenderGnutoo: and retry
21:29.28Gnutoothanks a lot
21:29.29GnutooI'll try
21:31.22Gnutoook it nearly work with IAX
21:31.35Gnutoo[Jul 26 22:28:21] NOTICE[16714]: chan_sip.c:18101 handle_request_invite: Failed to authenticate device "gnu too" <sip:ubu@192.168.1.106>;tag=348c75c4-9878-de11-91f4-00148511ef02 on the main server
21:31.54Gnutooah chan_sip...
21:32.09ManxPowerGnutoo: go read the book
21:32.10ManxPower~book
21:32.11infobot[~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
21:32.14Gnutoook
21:32.31Gnutoois it up to date?
21:32.43Gnutoodid a lot of thing change?
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21:39.16Gnutooah I get it thanks
21:39.20GnutooI'll try something
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21:50.39Dovid<PROTECTED>
21:50.39infobot[~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY
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22:05.14moos3anyone using zultys zip 4x5 phones?
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22:29.22eric256any pointers on what to research if i want to have line 2 on a polycom 330 show the status of a different extension? (and idealy be able to be used to pickup calls)
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22:57.54ManxPowereric256: the terms you should search for are BLF and SLA  There is an SLA document in the Asterisk source code.
23:04.48*** join/#asterisk ScarEye_ (n=scareye@cpe-72-229-213-183.nyc.res.rr.com)
23:06.22eric256thanks ManxPower yea i've been searching and everything i'm finding is years old and sayings SLA doesn't work yet....hmmm
23:06.27eric256slaps google around a bit
23:06.28eric256lol
23:06.45ScarEye_hey guys I am putting together a asterisk box. which version should I work with?  1.4.26 or 1.6.1.1 ?
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23:09.50ManxPowereric256: look in the ASTERISK SOURCE CODE directory in the doc/ directort
23:10.53[TK]D-FenderManxPower: * SLA doesn't do "phones" or pickup
23:11.28[TK]D-Fendereric256: And last I checked IP3XX phones don't support presence indication on a line-key
23:11.29Alfio~sla
23:11.30infobotfrom memory, sla is service level agreement, or shared line appearances
23:11.45[TK]D-Fender* SLA = hack, not SIP-B
23:11.54Alfio~blf
23:11.54infobotfrom memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing.  hint extensions are static mapped to SIP or other channels.
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23:13.00eric256guess i'll do something different ;)
23:13.22ManxPower[TK]D-Fender: learning about both is a good start for anyone wanting those features.
23:15.47eric256yea i just used the shared setup for the line in the polycom and it lets me make calls from that number...but doesn't alert when they are incoming.  no big deal in this case but would be cool if its possible to learn how to do it
23:19.02ScarEye_is that a good phone to get for asterisk?  polycom or can you guys recommend something else?
23:19.09ManxPowereric256: neither of those two features require SLA/BLF
23:19.11ScarEye_that works great with asterisk
23:19.42ManxPowerDial(SIP/phone1&SIP/phone2)
23:20.01ManxPower~phones
23:20.02infobotWhile personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream phones.  Ever.
23:20.44eric256um...no Manx i don't want them both to ring, there are a dozen ways to do that, i want it like on my intertel where i can setup a button to show the status of another extension, and use that button to call the extension or pickup calls holding/ringing on that extension
23:21.14ManxPower"but doesn't alert when they are incoming"  Sounds like you want it to ring.
23:21.26ScarEye_Do not consider Grandstream phones.  Ever. <----------- really?
23:21.27ManxPoweryou can always set the ringer off for that line appearance
23:21.31eric256alert can be blinking
23:21.44ScarEye_I have a few grandstreams ATA's I was going to use with asterisk.
23:21.48[TK]D-Fendereric256: Pick a higher model
23:22.07ManxPowerScarEye_: just don't expect anyone here to help you with the GS stuff.
23:22.09eric256ScarEye_:  i have several grandstream ATAs and they work very well
23:22.41eric256they are obvious budget items, but they work ;)
23:22.55ScarEye_Okay guys I am putting together a budget and I have asterisk support inside my budget
23:23.07eric256[TK]D-Fender:  yea, one with more features, just had this extra button sitting here hoping i could put it to use
23:23.18ScarEye_So for phones I will start with polycomm
23:24.01eric256i saw how to make that button a message button and wondered what else it could do
23:24.34[TK]D-Fendereric256: * SHOULD support BLF on it, but my last check still says "no"
23:24.44[TK]D-Fendereric256: s*/polycom/
23:25.21[TK]D-Fendereric256: Which is to say I can't see why it isn't supported, but it is not.
23:25.52ScarEye_I have then credit card type terminal that dials up to transmit data so on this device I was thinking about hooking up an ATA to it.
23:25.56eric256hmm okay. so i'll read up on BFL and see if i can make it work
23:26.20ScarEye_I have a * ^
23:26.20ManxPowerJust about every Asterisk install I've done had BLF on Polycoms
23:26.20[TK]D-Fendereric256: As I said, the line keys don't support it on IP 3XX's
23:26.21ManxPowerI've not done SLA, however.
23:26.32[TK]D-FenderManxPower: And * SLA.... forget it.
23:26.34[TK]D-Fendereric256: ditto
23:27.27eric256oh sorry i thought you said the last check...made it seem like you where sure...and then manx said he did BLF on polycoms...so confused
23:27.28eric256lol
23:27.31ManxPowerI don't think I've done it specifically on the 3XX, just the 500, 501, 600, 601, 650
23:27.36eric256ohhh
23:27.36[TK]D-Fender^^
23:27.38eric256gotcha
23:27.41[TK]D-Fender3xx = no-go
23:28.22ManxPowerhttp://www.fnords.org/~eric/polycom-config-examples/
23:28.27ManxPowertaken from production systems
23:28.34eric256ahh well i've got a 650  not setup yet, so i'll try on it instead...still sad about my second line unused ont he polycom, anything cool to do with it?
23:28.46ManxPower[TK]D-Fender is usually right about polycom stuff
23:29.24[TK]D-Fenders/polycom//
23:29.25[TK]D-Fender:D
23:29.55[TK]D-Fendereric256: You can use it as a speed-dial, just no presence support
23:29.55eric256oh i wasn't doubting i was misunderstanding ;)
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23:30.55ManxPower[TK]D-Fender: both job prospects are for jobs that require me to actually work with and configure Asterisk again.  Unlike my last job.
23:31.02eric256what about the soft buttons?  and where the heck do you find this info? i have the polycome admin manual pdf but it i'm not making heads or tails out of it yet...still waiting for that aha moment when i get to start speaking the polycom config file languag hehe
23:31.34ManxPowereric256: I suspect maybe 10 people in the entire Asterisk world understand the softbutton configs.
23:32.39eric256ManxPower: well that isn't exactly encourageing ;) unless the asterisk world is 10 people + myself (:
23:32.45ManxPowerLearn to crawl before you try running a triathlon.
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23:32.57pdanihi
23:34.12ChainsawHello.
23:34.13eric256yea the problem with most these things is that the begginers (me) don't know the different between the tasks that are walking or running, it all looks the same
23:34.29ScarEye_hey guys do most of you use asteriskNOW or compile from scratch?
23:34.46eric256for instance the admin guide says i can program that button as a buddystatus function, but doesn't tell me what that means
23:36.00ManxPowercrawling = making calls, receiving calls, configuring messages button., speed dials  Walking = BLF, custom ringers  Triathlon: SLA, reprogramming buttons.
23:36.31ManxPowerScarEye_: nobody here uses Asterisk GUIs.  Or if there are, they are too embarrassed to admit it.
23:36.51ManxPowerI bet you will get a different answer on #AsteriskNOW
23:37.11[TK]D-FenderScarEye_: Or get flogged for daring speak of it ;)
23:37.18ManxPowereric256: Buddy Watch is part of BLF.  See my config examples RUL I just posted.
23:37.41ManxPowerand the URL too
23:38.03AlfioManxPower can post the url again please
23:38.14ManxPower(6:28:21 PM) ManxPower: http://www.fnords.org/~eric/polycom-config-examples/
23:38.24Alfiothx
23:42.00eric256thanks all...dinner time.
23:42.33eric256i wonder if i can do BLF on a the line key and call pickup on the softkey...hmm lol
23:42.38eric256sprains a leg
23:42.40eric256later
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23:46.40Snooganhello folks. can anyone offer assistance with an error msg i'm getting trying to setup dahdi?
23:46.47SnooganNotice: Configuration file is /etc/dahdi/system.conf
23:46.47Snooganline 0: Unable to open master device '/dev/dahdi/ctl'
23:46.53pdaniexten => _06[237]0XXXXXXX,n,MixMonitor(/home/pdani/test.wav,,) <-- i have this line in my extensions.conf, but when i make a testcall from CLI, i get this message: Executing [06306429159@from-sip1:2] MixMonitor("Console/dsp", "/home/pdani/test.wav||") in new stack, but the file isn't there...
23:47.04Alfio~pastebin
23:47.07infobot[~pastebin] 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
23:47.07pdaniit's 1.4
23:47.15ManxPowerSnoogan: you don't have the kernel drivers loaded.
23:47.27SnooganManxPower: how do i load them?
23:47.41ManxPowerpdani: chances are Asterisk does not have permission to write to that directory.
23:47.54pdanii run it with root
23:48.01pdani(for test purposes)
23:48.22ManxPowerSnoogan: *I* load them by running "chkconfig dahdi on" to load them on boot and "service dahdi start" and "service dahdi stop" to manage it without rebooting.
23:49.05ManxPowerI imagine you would load them using whatever the standard tools are for your OS.
23:49.06pdanimaybe this message is relevant too: [Jul 27 01:41:44] WARNING[15407]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such device
23:49.49SnooganManxPower: can i paste some output in pastebin.ca for you to look at? tried /etc/init.d/dahdi start and it gave an error output
23:49.49ManxPowerSnoogan: "make config" will usually install the required files for your OS from DAHDI
23:49.58ScarEye_Soundpoint IP501 what do guys think about that phone?
23:50.05ManxPowerSnoogan: pastebin the error
23:50.20ManxPowerbut that sounds very debianish -- if that is the case I won't be able to help you much.
23:50.40ManxPoweron Redhatish distros it's in /etc/rc.d/init.d/dahdi
23:50.52Snooganhttp://www.pastebin.ca/1508153
23:51.03Snooganyes, it is debian
23:51.09ManxPowerSorry, an error has occurred. Reason: That is an invalid ID, or the post has expired.
23:51.45pdaniany idea?
23:52.38Snooganhmm, tried again and same error.  is there another pastebin site?
23:52.52Mango_http://www.pasteonce.com/
23:53.01Mango_While you're pasting...
23:53.05Mango_I have a dial plan question.  Is it possible to mix wildcard and non-wildcard in the same plan?  I want to do something like this where the first line(s) of the plan is non-wildcard and the rest is wildcard.  The reason is that I want to use regexten.  Here's an example: http://pasteonce.com/42
23:53.24Mango_(though the example doesn't work)
23:53.34ManxPower~pb
23:53.35infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
23:53.59ManxPowerMango_: It is a very bad idea.
23:54.21Mango_Ah, really?
23:54.35Snooganok... try this: http://pastebin.com/m4cbd9196
23:54.38Mango_Best to handcode each individual extension then?
23:54.51[TK]D-FenderMango_: Generally fine.  Don't use "n" though, use "2" <-
23:54.58ManxPowerSnoogan: DAHDI is not installed
23:55.16[TK]D-FenderMango_: Though there is usually little point to using regexten period
23:55.31Snooganso am i right in thinking i need to go to /usr/src/dahdi-linux-2.2.0.2 and recompile?
23:55.37ManxPowerMango_: it is very very very easy to have calls jump all around the dialplan because you forgot something matches
23:55.46ManxPowerSnoogan: what does the DAHDI readme say?
23:56.52ManxPowerSnoogan: just reinstall, watch for errors
23:57.21*** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv)
23:58.44Snoogandone.  http://pastebin.com/m2ca43046 for output
23:58.53Snoogancan you take a look and see if it looks good?
23:58.56Mango_[TK]D-Fender and ManxPower: Thank you.
23:59.25*** join/#asterisk mace (n=mace@debian/developer/mace)
23:59.53[TK]D-FenderSnoogan: Says everything is fine so far

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