irclog2html for #asterisk on 20070220

00:00.39JTj0: which state? there's at least 2 or 3 abbotsfords in Australia
00:00.55*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
00:01.11j0JT: NSW
00:01.21teknoprephey how are the cisco 7970 phones and asterisk ?
00:01.21JTright
00:01.29JTso the one in sydney :)
00:01.34j0yep :)
00:03.32jero<PROTECTED>
00:03.51jeroafaik Queue() does not return
00:04.32Sputteringit returns only if the queue is unavailable for some reason
00:04.49tzangerwoo, 2 TDM400P cards, a TE405 and eight FXO modules for the TDM400 cards... I wonder if any of these qualify for HPEC licenses
00:05.20jeroSputtering: yes
00:05.32jeroI'd like to run some action at the end of a Queue() call
00:07.42*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:08.21|ryan|Is there a way to get *67 to work from a SIP phone?
00:08.30*** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com)
00:08.51CunningPike|ryan|: Define 'work'
00:10.07|ryan|I'd like it to suppress sending callerid like it does on a POTS line in the US
00:10.50JT|ryan|: what connection to the pstn?
00:11.17|ryan|I have an IAX connection.
00:11.43*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
00:13.05|ryan|the provider doesn't send caller-id on outgoing calls unless I set it, so I have it set in my outgoing context before dialing out.
00:14.02|ryan|So I guess what I would need to do is set up a *67 extension that somehow bypasses where the CallerID data gets set.
00:14.13*** join/#asterisk hellojoe (n=hellojoe@c-67-160-249-95.hsd1.ca.comcast.net)
00:14.59JTthat's one way i guess
00:15.09JTi'm curious what you mean by "set it"
00:15.16JTby default asterisk sets callerid
00:16.10|ryan|Well, it's set by default as my extension's userid
00:16.20|ryan|which is non-numeric
00:16.28JTah
00:17.02|ryan|I'm trying to figure out how I'd give another dial tone and accept digits after I hit *67
00:18.52*** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net)
00:18.52*** mode/#asterisk [+o anthm] by ChanServ
00:19.00JToh
00:19.02JTthat's easy
00:19.04JTDISA()
00:19.07flenders|ryan| DISA
00:19.39|ryan|Ok.
00:19.43*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)
00:21.38JTmorning flenders
00:28.29ManxPowerIf cell phone companies want to sell ring tones maybe it would help if they made it easy to do so.
00:29.32*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)
00:31.17ManxPowerVerizon wireless required me to select my phone model, let me search and find the perfect ringtone I wanted, I created an account, re-did the search, clicked on purchase, and was told that I can't purchase ringtones for this phone over the web but could but that specific ring tone by using my phone.  However, I don't know which of the 4 or 5 ringtone applications (each costs money) I have to use.
00:31.32JTManxPower: one think i really hate now is people can pay to have music play while their phone is being dialled :(
00:31.44JuggieManxPower, sounds like you have a 1) shitty provider 2) shitty phone
00:31.57ManxPowerJuggie: I have the only provider that works where I live.
00:32.20Juggiea ringtone on my phone consists of 1) connect the phone to pc using either usb cable/bluetooth/IR 2) transfer ringtone 3) enable ringtone in phone ui.
00:32.27ManxPowerJuggie: the web site even has a "content ID" which in theory I can just type into the phone and get the ringtone.  I can't find it.
00:32.50JTi've never downloaded a ringtone
00:33.24ManxPowerJT: I've only ever downloaded 1 ringtone, now I want a different one.
00:33.34JTheh
00:33.51JTi usually look for one in the default set, that just sounds like a phone ringing
00:33.53JuggieManxPower, your phone is totally locked cant transfer anything to it yourself?
00:33.57hellojoeguys, is there a way to just use a simple voicemail to email instead of all of the voicemail stuff in asterisk?
00:34.01*** join/#asterisk litage (n=nick@203.220.55.70)
00:34.04mmlj4hey ManxPower
00:34.15ManxPowerJuggie: I'm sure I could if I spent a couple of hours figuring out how to do so.
00:34.25mmlj4been to the new covington doctor office yet?
00:34.26hellojoei was thinking of recording the message (Record) and then using sendmail to attach the message and ship it out?
00:34.29ManxPowerhello mmlj4
00:34.35JuggieManxPower, CDMA providers are so anal, my phone is wide open.
00:34.42Juggieand its a locked to provider phone too.
00:34.44hellojoeanyone sees any issue with this. No need fo rme to save the vmails on the server
00:34.49ManxPowerhellojoe: sure you can do that.
00:34.54*** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com)
00:34.55Juggiei hook it up over usb, and its a flash drive.
00:35.16Juggietwo flash drives actually, one is the phone memory, and the other is the addon memory card.
00:35.20|ryan|hmm
00:35.20hellojoeAhh... thanks ManxPower. I was mainly worried about the performance/gotchas
00:35.21ManxPowerJuggie: Oh it is so impressive can I hold it?
00:35.32mmlj4ManxPower: all I need to do is screw on the wallplates, and I'm done  # aside from any red phones, etc.
00:35.38|ryan|DISA is just giving me busy tones when I try to dial.
00:35.49ManxPowerI spent $99 on the phone 18 months ago and it is a prepay
00:35.56hellojoenow I need to figure out the command line for sendmail with *.wav as attachements
00:36.14ManxPower|ryan|: then it is not finding a matching exten line in the context specified
00:36.25ManxPowerI'm NEVER EVER doing another cellphone contract again.
00:36.35JuggieManxPower, i told my provider someone else was offering me a better deal and i got this one for 49$ on a 3 year contrac.t
00:36.52Juggieit should/should have been 150$
00:36.58Juggieer, would/should
00:37.00JTmmlj4: red phones?
00:37.14mmlj4"you need a new battery? sure, just sign another contract, and you can have one"
00:37.17|ryan|ManxPower: is there any way to debug what context a line is in?
00:37.19ManxPowerI basically had to lie and back up the lie with paperwork to get out of the 3 x $175 termination fee.
00:37.33mmlj4JT: POTS bat-phone, backing up VoIP
00:37.44ManxPower|ryan|: you have to specfify the context when you run DISA.  "show application dias"
00:37.46mmlj4$client likes them to be red
00:37.51ManxPowerand "show application disa" too.
00:37.57JTmmlj4: ah ok
00:38.24tzangerJuggie: GSM swap-out-the-card-to-a-new-phone is going the way of the dodo too
00:38.29tzangerCDMA was just ahead of the curve there :-(
00:38.39*** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com)
00:38.40ManxPowerI'll replace my phone when it breaks 8-)
00:39.21|ryan|ManxPower: I did specify a context.  Is it going to break it if I set it to use the same context it was accessed from?
00:39.35J4k3sim cards are cute...  but I'd rather have decent RF performance (CDMA)
00:40.19ManxPower|ryan|: paste the DISA line from extensions.conf
00:40.22*** join/#asterisk elriah (n=johnny@adsl-072-149-159-016.sip.bhm.bellsouth.net)
00:40.52|ryan|(this is in the internal context) exten => *67,n,DISA(no-password|internal)
00:41.10*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
00:41.26ManxPower|ryan|: did you forget the leading _ in a pattern match for the exten line you want matched?
00:41.38|ryan|no
00:41.38ManxPowerelriah: you are in Birmingham??
00:41.48|ryan|I get a dialtone when I hit *67
00:42.27ManxPower|ryan|: does the Asterisk console showing DISA being run?  If not, your phone is handleing the *67 and giving you dialtone, not Asterisk
00:42.54*** join/#asterisk docelmo (n=vircuser@c-68-82-181-225.hsd1.de.comcast.net)
00:43.04|ryan|hmm
00:43.13|ryan|let me try this from my softphone
00:43.17JTanyway, DISA is probably not what you want
00:43.22JTnot sure if you want a second tialtone
00:43.24JTdialtone
00:43.43|ryan|I don't really need a second dialtone
00:44.35|ryan|I wat to run some commands (setting caller id presentation to prohibited) then allow any other number accessable from the context to work.
00:44.52*** join/#asterisk PaulTech85 (i=PaulTech@72.29.76.254)
00:45.10PaulTech85So, If I want to get billsec for the number of seconds a user TALKED to a agent, How would one go about that?
00:45.14PaulTech85Insteed of counting total time in queue
00:45.16JTyeah i was trying to work out how to do this a while back
00:45.29JTbut i then realised i didn't really need it and just blocked callerid
00:45.57|ryan|well
00:46.10|ryan|I was going to do a couple other simmilar things
00:46.20PaulTech85Is there no direct method?
00:46.29|ryan|like try and set it up to allow a code to be entered before dialing that will record the call
00:46.44[TK]D-FenderPaulTech85 : One piece of info you have in the queu log is the hold time.  talk time = total time - hold time
00:46.45PaulTech85ah
00:46.48PaulTech85queue_log
00:46.55PaulTech85Just say that Fender :-)
00:47.01PaulTech85saw*
00:47.10PaulTech85'Asterisk does not currently support dumping queue_log data straight to a MySQL table. '
00:47.22*** part/#asterisk jero (n=jerome@modemcable046.23-81-70.mc.videotron.ca)
00:47.52[TK]D-FenderPaulTech85 : Yes, * can store queue data directly into MySQL, or another ODBC database
00:48.04[TK]D-Fender*sigh*
00:48.11[TK]D-Fenderlooks like a bad reference...
00:48.12PaulTech85Maybe I'm reading old pages on wiki ;)
00:48.33[TK]D-FenderPaulTech85 : They look like current pages... only wrong ;)
00:48.42PaulTech85Hehe
00:48.49PaulTech85let me fire up the actual docs
00:49.55PaulTech85Hmm being honest I dont see it anywhere..
00:50.15*** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net)
00:50.21[TK]D-Fenderkeep looking..... (hint : its in asterisk-addons)
00:50.41[TK]D-Fenderanyways, must be off, back in 2 hrs or so...
00:51.11PaulTech85I'd guess cdr_addon_mysql but I know thats not it, *looks at app_addon_sql_mysql*
00:51.23hellojoeany idea how to attach .wav file using (system /usr/bin/sendmail) command
00:52.36*** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net)
01:02.49PaulTech85Hmm not seeing much
01:03.29*** join/#asterisk joaovianna (n=joaovian@ool-4354d1a8.dyn.optonline.net)
01:04.27joaoviannaCan someone help me ? I'm testing video in asterisk but I'm receiving a error message saying "Unknown RTP codec 126 received from ..." Anyone ?
01:07.16joaoviannaAnyone using asterisk+video ?
01:08.02*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
01:11.17NivexI'm building asterisk 1.2.15 on debian sarge and would like to force it to use gcc-3.4 instead of gcc.  How do I go about this?
01:13.25elriahWhy would you want to? (just curious)
01:13.50Nivex3.4 > 3.3, isn't it?
01:14.09Nivexbut sarge insists on keeping gcc pointed at gcc-3.3
01:14.22JTdoes it even matter, if it works after compiling with 3.3?
01:14.48elriahUse the OS default compiler, save yourself some headache.
01:15.02carraryeah, save yourself serious headache
01:15.18carrarunless you are a C coder
01:15.28carrarand know all the ins and out
01:15.38carrarbut even then
01:15.41carrarI wouldn't
01:15.54Nivexok
01:16.15elriahThat's the "ok, but I'm going to try it anyway" ...
01:16.16elriahlol
01:16.20carrarhahah
01:16.32carrarand then it crashing on a dynamic module
01:16.35carrarcrashes
01:16.36carrarheh
01:17.36PaulTech85Isnt 1.4 the stanard now?
01:17.44PaulTech85If not I'm pissed about porting 1400 line extension.conf
01:17.45JTno
01:17.47PaulTech85;p
01:17.57JTmost people still use 1.2.x in production
01:17.59JTmore stable
01:18.04PaulTech85ahh
01:18.56elriahI hear that 1.4 has a few issues, everbody has recommended to me to wait to 1.4.1 for production upgrades.  I'll probably wait until 1.4.4 or so...
01:19.07elriah1.2.x is very stable, imho
01:21.47*** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net)
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01:28.31Juggieelriah, not 1.4 but 1.4.0
01:28.35*** join/#asterisk Ryushin (i=chris@71.33.251.74)
01:28.38Juggieso you can run 1.4svn or wait for 1.4.1
01:29.15JTor just keep waiting :)
01:31.33*** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
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01:34.00mmlj4grr... i'll bet ManxPower didn't recognize my nick earlier
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01:35.58*** join/#asterisk fastfeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com)
01:36.21*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
01:36.52FuriousGeorgehey all
01:36.56*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
01:36.59FuriousGeorgewhat's shakin'?
01:37.23PaulTech85Hehe
01:37.28PaulTech85My SQL got interesting
01:37.38PaulTech85Just for getting the total talk time of one dst number
01:37.40PaulTech85after queue
01:37.42PaulTech85'select sum(asterisk_queue_log.arg2) from asterisk_queue_log  left join cdr ON asterisk_queue_log.callid = cdr.uniqueid  where asterisk_queue_log.event='COMPLETECALLER' and cdr.dst='$dst' and unix_timestamp('time) between unix_timestamp('2007-02-00 00:00:00') AND unix_timestamp('2007-02-31 00:00:00'); '
01:38.14FuriousGeorgeso i installed asterisk at a bar today.  everything is working fine, they need a way to transfer calls between stations.  im teaching them to transfer calls, we must have made 200 calls, everyone gets it, owner is like a kid with a new toy
01:38.48PaulTech85Haha
01:39.02PaulTech85Bar cant exactly be high volume
01:39.02PaulTech85;)
01:39.05FuriousGeorgeso his buddys there and hes showing off his system, he's got a business too, he's interested asking about the price.  goes to make a call to try it out.  gets dropped
01:39.09FuriousGeorgetries to call again
01:39.11FuriousGeorgenot found
01:39.30FuriousGeorgegoes through, the third try, dropped again "well, i cant use this phone"
01:40.02[hC]whoops!
01:40.15[hC]whats the setup there?
01:40.19JTwhy was it dropped
01:41.38FuriousGeorgehe mentioned it "normally takes a week to work these things out" owner says, "well, we just installed it today", i say "well, i dunno, ill look into that, gotta work it out".  i tell owner later, "im concerned about that, and i want you guys to make a lot of calls, write down time and number if it happens again", but he tells me he thinks it was a cell phone he called or something, wasnt the system...  here's the thing though, i know
01:42.10FuriousGeorgei check my messages, but my event log is blank!
01:42.15FuriousGeorgeso no good info thus far
01:42.16fastfeetI am trying to use MySQL for CDR in version 1.4. I set up a database, set a set a user for the DB, but I am not sure what tables I need to create?
01:42.22fastfeetAny ideas? Hints?
01:42.27FuriousGeorgesnom 360's on asterisk 1.4
01:42.29PaulTech85its on the wiki
01:42.44FuriousGeorgefastfeet: ~docs
01:42.48FuriousGeorgeerr
01:42.49FuriousGeorge~docs
01:42.51jbotfrom memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
01:42.52fastfeetI;ve been through the wiki, I must be missing it
01:42.59[hC]FuriousGeorge: sip/iax/zap for outgoing lines?
01:43.11FuriousGeorge[hC]: iax every time
01:43.23FuriousGeorgei meant to say asterisk 1.2.14
01:43.27FuriousGeorgenot asterisk 1.4
01:43.43FuriousGeorgefastfeet: ive never tried "asterisk realtime" or i'd help you
01:43.48FuriousGeorgeis that what you are searching for?
01:43.49*** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com)
01:44.40FuriousGeorge[hC]: ive used the same provider other places, not had this problem.  also ive reboot his phone since then...  im /thinking/ it may have something to do with this AP
01:44.52fastfeetthanks yall
01:45.23FuriousGeorgenetgear cr@p between phone and asterisk/switch
01:46.11FuriousGeorgegood news is guy now wants me to come back and install more phones
01:52.01*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
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02:12.51foobar778yea all is working !!!!
02:14.39FuriousGeorgei mean, now that i think about it out loud to myself...  is it bad practice to use a $30 netgear ap/switch, certainly not "business class" as the interface between my sip phone and my asterisk/switch?
02:14.46*** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com)
02:17.27foobar778ap/switch or router?
02:18.14FuriousGeorgethe netgear ap/switch has firmware
02:18.21foobar778and do u use an ata or ur switch/ap has phone ports?
02:18.22FuriousGeorgeim not using the wan interface of course
02:18.37foobar778??
02:18.53FuriousGeorgeim complicating things by saying ap/switch, i just mean to imply that im using a device that is a router as a switch
02:18.55foobar778whats ur setup modem then then??
02:19.03FuriousGeorge?
02:19.30foobar778my question which is first to the moden ata or router
02:20.38FuriousGeorgelinux running iptables --> switch --> (wall/jack) netgear wireless router device -->  snom 360
02:20.55foobar778what device connects to the ethernet port of your modem????
02:21.18foobar778snom 360 is ur modem????
02:21.21FuriousGeorgelinux running iptables obviously, or why would i specify that
02:21.24FuriousGeorgeother way
02:21.40foobar778what connects to ur ethernet port of ur modem
02:21.51foobar778what device?
02:21.54FuriousGeorgelinux
02:21.57FuriousGeorgecomputer
02:22.03FuriousGeorgeits a "manged switch"
02:22.08foobar778ahh
02:22.11FuriousGeorgeits not asterisk
02:22.24foobar778so thats ur firewall
02:22.30FuriousGeorgeyeah
02:22.33foobar778ok
02:22.50joaoviannaCan someone help me ? I'm testing video in asterisk but I'm receiving a error message saying "Unknown RTP codec 126 received from ..." Anyone ?
02:22.50foobar778and what does ur phone attavh to
02:23.09FuriousGeorgenetgear wireless router
02:23.31foobar778and that router is a voip router?
02:24.01FuriousGeorgeno, routers almost always have switches built into them.  just dont use the wan.  im thinking maybe, for some reason, the switch that is built into that (ports 1-4) is causing certain calls to drop
02:24.44foobar778ur swich has voip phone ports??
02:24.53foobar778or an ata between th4em??
02:24.54FuriousGeorge:)
02:25.06FuriousGeorgelemme link you to what im talking about
02:25.41foobar778want to join my pbx u can then talk
02:26.06FuriousGeorgehttp://www.continent.com.au/images/products/WPN824AU.gif
02:26.12FuriousGeorgethats all it is
02:26.20JTFuriousGeorge: netgear is trash
02:26.20JTavoid
02:26.23*** join/#asterisk n|cotine (i=nicotine@147.202.49.52)
02:26.28FuriousGeorgeJT:  thanks
02:26.45JTmost consumer brands are fairly trashy :P
02:26.48foobar778thats ur ata that ur analog phone goes into??
02:26.57FuriousGeorgefoobar778: its a snom360
02:26.59JT?!???!??!????
02:27.22FuriousGeorgeJT:  me and him are having "ill communications" as the beastie boys would say
02:27.25foobar778Im asking not familair with a snom
02:27.36FuriousGeorgejust a sip phone that happens to be asterisk friendly
02:27.48foobar778do u have the right rtp ports and sip ports fowared to that
02:27.57FuriousGeorgethere is no nat there
02:28.00FuriousGeorgeim using it as a switch
02:28.09foobar778iptables -F
02:28.17foobar778tried that as a test?
02:28.40FuriousGeorgeto find out why the call dropped i would be doing a sip dump, dont you think
02:28.55foobar778well if its a poprt issue
02:29.03FuriousGeorgeits internal only no nat
02:29.15foobar778<PROTECTED>
02:29.25FuriousGeorge:(
02:29.29foobar778iptables -F does that
02:30.28foobar778what linux distro is up front acting as the firewall
02:30.39FuriousGeorgei was just saying that i think the netgear thing is the culprit (even though im only using it as a switch, and no nat is involved) because only this phone does it and only this phone is behind that thing
02:31.03FuriousGeorgeand besides, if the call it is getting dripped it would be the signalling that caused it, not the rtp
02:31.10*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
02:31.19foobar778not my experience
02:31.36FuriousGeorgeyou can make a call with no rtp, why not...  just wont hear anything
02:31.40FuriousGeorgeit wont drop though
02:31.49foobar778I had rtp ports not opened and my call would dreop in 30 seconds consistently
02:32.20foobar778u have nat=no??
02:32.27FuriousGeorgeodd b/c iirc sip uses like two per call, though i could be wrong
02:32.29foobar778localnet=????
02:32.56foobar778in your sip.conf globals
02:33.05FuriousGeorgeits set
02:33.15foobar778as what yes or no
02:33.21FuriousGeorgeyes
02:33.40foobar778and local net sety up
02:33.48foobar778externalip=
02:33.57FuriousGeorgeusing externhost, but same idea, yeah
02:34.59foobar778how many pcs just curious not related behind the switch
02:35.18FuriousGeorgemax two wireless and the snom
02:35.34foobar778does ur isp give u free wan ips?
02:35.36puzzledevening all
02:35.54FuriousGeorgefoobar778: not feasable
02:36.31foobar778if ur on a switch then there is no nat really and for the pcs well u said wirless
02:36.53foobar778so  the negear is a wirelss ap
02:37.23FuriousGeorgeyeah
02:37.24foobar778and the ap does network translation
02:37.35FuriousGeorgeNO
02:37.37FuriousGeorge:)
02:37.39foobar778so u only need one wan ip
02:37.41puzzledanyone know if spandsp-0.0.3pre27 works with asterisk-1.2?
02:38.46FuriousGeorgeright, i set the nat part up correctly, clients can go rempote, i dont need to pay for more ips, which come at a heavy premium around here
02:39.01foobar778wait if it doesnt do nat and its a switch every pc behind a switch must have a uniqque wan ip unless network translation is being done
02:39.46foobar778so therte must be network translation going on
02:39.47FuriousGeorgeno, linux does nat, the netgear is only being used as a switch... i feel like ive said that
02:40.14FuriousGeorgeright, i set the nat part up correctly, as i just stated
02:40.32foobar778sorry its not working for u
02:40.43FuriousGeorgewell its only once in a blue moon...
02:40.54FuriousGeorgebut you know its gonna happen again at the worst time
02:41.07foobar778u dont like routers??
02:41.42foobar778never understood why people by switches when a trouter costs the same and can be both
02:42.37puzzledprolly performance
02:42.40foobar778and better yet a voip router
02:42.56n|cotinefoobar778:  Because you can do much more interesting things if you have a linux router and a normal switch?
02:43.25foobar778love my router wrt54G with custom linux firmaware dd-wrt
02:43.28FuriousGeorgeits a restaurant that has a subnet for wireless with content filtering, subnet for wireless for back office, subnet for DMZ and several devices including security system, and by the way a voip telephone system, which may want some QoS for the first time one of them discover bit torrent in back office
02:44.02foobar778u should see the qos on the dd-wrt
02:44.03FuriousGeorgeoh AND their credit cards go out over a separate proprietary system, under mine, double natted
02:44.19n|cotinefoobar778:  dd-wrt has iproute2?
02:44.51foobar778go to the ddwrt site see all the packes that can be installed\
02:45.04foobar778<PROTECTED>
02:45.09foobar778<PROTECTED>
02:45.29foobar778paackages
02:45.41FuriousGeorgeas the military would say, any attempt to put a asterisk-router into production would be fubar
02:45.59foobar778not so it runs great
02:46.07foobar778read the success stories
02:46.23FuriousGeorgeim sure its scales quite well
02:46.56*** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com)
02:46.56foobar778this guy had so many channels and had it up like 5 years
02:47.16*** join/#asterisk Stp1800 (n=Stp1800@67-22-111-205.atlsfl.adelphia.net)
02:47.28FuriousGeorgeok, well, im not gonna argue this point with you
02:48.27foobar778anyway turning a 40 dollar router into a 600 dollar router just by flashing the firmware is very cool
02:49.41FuriousGeorgeits definitely cool, i just would use it for something "real" sparingly
02:49.52FuriousGeorgeif at all
02:49.59*** join/#asterisk X-Rob (n=Rob@CPE-58-169-100-13.vic.bigpond.net.au)
02:50.07FuriousGeorgeits hard enough to get things just right without complicating matters for yourself
02:50.28foobar778http://clipmarks.com/clipmark/0AC14714-0A93-4D50-82E7-8F6F6729DC36/
02:50.35foobar778read enjoy
02:52.10FuriousGeorgewheres the part on asterisk dimensioning
02:52.22foobar778goto openwrt
02:53.02foobar778http://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html
02:55.43FuriousGeorgewhy dont you just tell me how many simultaneous channels i can run on my linksys router, and we will make an educated guess about its suitability for business use.
02:57.14foobar778well a lot
02:57.21FuriousGeorgemore or less?
02:57.27FuriousGeorgewhat would you say?
02:57.30foobar778<PROTECTED>
02:57.36JTblah blah
02:57.41FuriousGeorgeif you dont want to guess i will
02:57.43JTit's clear foobar778 hasn't actually done it, FuriousGeorge
02:57.50JThe's just saying how amazing it looks
02:58.04foobar778I gave u direct links but I cant answer all I have read the sucees stories
02:58.26foobar778<PROTECTED>
02:58.31FuriousGeorgei'd say with no compression, transcoding, pure sip, i MIGHT get 5 channels
02:58.38FuriousGeorgeon a sunny day
02:58.41FuriousGeorgejust a guess
02:58.59JTyeah from memory 7 g.711 sessions max
02:59.03foobar778well if u try u will know but from what I read this guy had a huger pbx
02:59.09JTfrom reading the asterisk dimensioning page
02:59.23FuriousGeorgeJT:  its one of my many character flaws,  i said i wasnt gonna argue it, and yet here i am doing it
02:59.25JTs/u/you/
02:59.31JTheh
03:00.16foobar778I havent put asterisk on yet because currently Im using ddwrt
03:00.33foobar778But Im toyong with flashing openwrt
03:00.40foobar778toying
03:01.05JTfor me it's pointless
03:01.10JTif i can't terminate on it
03:01.12JTno use
03:01.45foobar778after what I have read But sinced I resolved all my nat issues not so enthusiastic about anothert flash and a possible brick
03:02.16FuriousGeorgethe warranty is long, if its covered just tellem you hosed it installing a fw update
03:02.47foobar778But the features I mean the ddwrt really makes u have a 600 dollar type router its truly awesome asterisk or not
03:02.48FuriousGeorgeits not unethical b/c you can blame the device for not appreciating the linux you tried to liberate it wityh
03:02.56foobar778That I can tell u first hand
03:03.18FuriousGeorgeits definitely cool, like i said.  im all for it.  been meaning to rma my POS router
03:03.32JT$600 router, i dunno about tha
03:03.32FuriousGeorgeacts flakey even with openwrt
03:03.46foobar778oh yes definiyley
03:03.56foobar778unlimited port fowardings
03:04.01foobar778<PROTECTED>
03:04.02JTmaybe at cisco prices
03:04.04JTwow
03:04.08JTso does my linux gateway
03:04.10foobar778<PROTECTED>
03:04.11JTit cost me nothing
03:04.12JT$0 router
03:04.13foobar778on and on
03:04.31*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
03:04.38foobar778go look at the ddwrt page
03:04.43JTfucks sake
03:04.44JTNO
03:04.56JTi've looked at it previously, stop advertising it every 2 seconds
03:04.58JTwho cares
03:05.05JTnothing that a normal linux box cannot do
03:05.20JTno idea why i'd want samba on my gateway anyway
03:05.27JTas my gateway is not a fileserver
03:05.34foobar778u need how many network adapyers???
03:05.39JT?!?!?!
03:05.41FuriousGeorgefoobar778: you should check out ipcop
03:05.48foobar778I know it
03:05.50JTfoobar778: what?
03:05.57JTwhat about network adapters?
03:06.00foobar778ipcop firewall
03:06.21*** join/#asterisk thekidrio (n=thekidri@24-205-76-13.dhcp.psdn.ca.charter.com)
03:06.25foobar778it does wirelss???????
03:06.31JTso what's this about network adapters
03:06.32foobar778ipcop think not
03:06.41JTand if you have a question, one question mark is sufficient
03:06.48JTipcop can probably do wireless
03:06.55foobar778how
03:06.59FuriousGeorgefoobar778: its doable, just not supported
03:07.00foobar778need hardware
03:07.03JTyes
03:07.04*** join/#asterisk abv (n=adam@74.72.190.181)
03:07.05FuriousGeorgewireless nic
03:07.08JTomg bbq
03:07.13foobar778well now not so free huh
03:07.22JTalternatively, you could just get an access point
03:07.26JTwhich is pretty cheap
03:07.28JTand solid
03:07.29foobar778$$$$
03:07.32FuriousGeorgeit rained here the other day and a wireless nic fell out of the sky
03:07.32JTcheap
03:07.35JTthey are not expensive
03:07.45JTif you want wireless, it costs money, like anything
03:07.48foobar778well wrt54G ebay 10 dollars
03:07.55JTi use cables for everything at home
03:08.16JTsure, i've used wrt54g, but as aps only, pity you can't really switch off their router function
03:08.20abvhey. i'm an asterisk newbie.
03:08.42FuriousGeorgehey asterisk newbie
03:08.46foobar778u can
03:08.57foobar778turn off router function in ddwrt
03:09.00JTstill not sure what your point is foobar778
03:09.05JTyes if you use unofficial flash
03:09.11foobar778yes
03:09.23JTi couldn't really be bothered
03:09.28abvhi. quick question. i setup asterisk on a vps today (with lylix.net). what's the easiest way to test if it working (inbound)?
03:09.30foobar778thats my point what I have been talking about guess u havent followed
03:09.40*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
03:09.56JTs/u/you/
03:10.14*** join/#asterisk k-man (n=jason@unaffiliated/k-man)
03:10.26JTfoobar778: no, you've been crapping on about how ddwrt is the best thing since sliced bread
03:10.32JTusing wild handwaving
03:10.39foobar778it is
03:10.45FuriousGeorgecrapping is a good word to use there i think :) not to pile on
03:10.49JTit's not bad, but it's nothing that amazing
03:10.54*** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net)
03:10.55k-manso.... after doing a make install, should I do a make samples? or should i just populate /etc/asterisk with the files as i need them?
03:10.57FuriousGeorgegood verb choice
03:11.14FuriousGeorgek-man: make samples is nice when you upgrade especially
03:11.21FuriousGeorgeas configs change
03:11.22k-manFuriousGeorge, why?
03:11.32foobar778well thats ur opinion most think its great Do u create firmware??
03:11.33FuriousGeorgei mean between major releases
03:11.37k-manit won't overwrite my existing configs?
03:12.01FuriousGeorgek-man: oh, ive always backed mine up, i assume they were overwritten
03:12.06JTfoobar778: no, i generally prefer to use linux boxes to do NAT/routing
03:12.08FuriousGeorgebut i think it actually makes .sample
03:12.17FuriousGeorgenot responsible if im wrong about that :)
03:12.36*** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk)
03:13.01foobar778well good for u that make u geekier thew ddwrt in Linux my friend ssh into the os and u will see a full linux diostro at ur disposale
03:13.16foobar778ddwrt is a Linux os
03:13.55JTok i hardly understood that but i think i get the gist
03:14.01JTyes i know it's linux
03:14.10JTjust does not make it worth $600
03:14.13foobar778example the v5 was vxworks and u strip that and put on a Linux ddwrt or opewrt or hyperwrt OS
03:14.17FuriousGeorgefoobar778: let me put it this way, openwrt is cool, but if you want to do some serious stuff, like content filtering/web proxy etc. you just arent gonna do that with openwrt alone
03:14.30foobar778It has it
03:14.45FuriousGeorgehow am i gonna cache a million .jpgs on my linux router
03:14.46foobar778I should let u into my setup page
03:14.49JTalso, the cpus aren't very powerful, so if you have high load they can't handle it
03:14.54FuriousGeorges/l;inux/linksys
03:15.25foobar778U have to dedicate a whole nachine for ur firewall
03:15.38foobar778<PROTECTED>
03:15.40FuriousGeorgeyou have a half a machine for a quarter of a firewall
03:16.01foobar778wrong
03:16.13JT10dollar, that assumes the second hand unit you buy fof ebay works
03:16.17foobar778when u get it u will sing a differnt tune
03:16.18JTwhich may not be the case
03:16.54foobar778how much does ur PC that runs ipcop cost?????
03:16.58JTumm, he's right, a linksys router is shit for a http proxy with anything but light traffic load
03:16.59FuriousGeorgefoobar778: look, you obviously have done your homework, and know a lot about what openwrt can do.  now you have to realize that just because it can do something, doesnt mean it can do it well, and just because it can do some THINGS doesnt mean it can do everything
03:17.13JTit cost me $0, but the market value was probably AUD$50 at the time, or USD$30
03:17.29foobar778well doesnt that apply to ipcop??
03:17.41JTi don't use ipcop either
03:17.42*** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net)
03:17.52*** join/#asterisk orbikitti (n=orbitn@68-119-118-25.dhcp.jcsn.tn.charter.com)
03:17.53foobar778so a low powerd PC is firewalling ur network
03:18.02JTno shit
03:18.06JTat home
03:18.15JTi have a much more powerful one at the office
03:18.17foobar778well I have 6 on my lan
03:18.40JTmy home low powered pc will outperform the linksys
03:18.49JTembedded machines have very limited cpu flash and hdd
03:19.00JTcpu flash and ram i mean
03:19.28flendersahahha, I can't believe you guys are arguing about it
03:19.42JT:)
03:19.42flendersJT: just agree mate...
03:19.46FuriousGeorgefoobar778: openwrt is just not viable for every application.  i would not use it with or without asterisk installed because it is not appropriate for the application that started this whole conversation.  im no expert but i know anyone who claims to be and says it is is wrong
03:19.48JTflenders: haha
03:20.17foobar778i use ddwrt in frony of my asterisk machine
03:20.23JTcool
03:20.27JTi'm sure it works fine
03:20.38flenders:)
03:20.38foobar778this started when u were talking about ur shitty netgear switch
03:21.04foobar778my ddwrt is in front of the asterisk
03:21.04JTumm, that was FuriousGeorge
03:21.09JTi hate netgear
03:21.20JTand most consumer brands of cheap stuff
03:21.20foobar778well that how this talk started
03:21.22FuriousGeorgefoobar778: lets say you have 15 users on a moderately slow internet connection who visit basically the same 7 web pages all day.  lets say those pages have content with a lot of images.  guess what, openwrt is MUCH worse than a pc running linux as a firewall/proxy in that situation
03:21.26FuriousGeorgethats all there is to it
03:21.28flendersI hate the WRT54GP2
03:21.36flendersworst router I've ever seen
03:21.50JTflenders: netcomm NB5
03:21.52JTworst ever
03:22.03JTcannot hold PPPoE longer than 18hours
03:22.08JTthen needs power cycle
03:22.09foobar778hey there are no probs here conet filtering active x all from setup page
03:22.20foobar778contentfiltering
03:22.23FuriousGeorgefoobar778: besides flenders here we all like openwrt.  its ok to like openwrt, its even ok to love openwrt, but stop *loving* openwrt
03:22.34*** join/#asterisk LoneShadow (n=duh@c-24-6-162-76.hsd1.ca.comcast.net)
03:22.34flendersJT: mine needs a power cycle every 3 or 4 days
03:22.45foobar778i love ddwrt acyually
03:22.45flenderswireless starts to get slower and slower
03:22.46JTFuriousGeorge: to be fair, flenders is probably talking about stock firmware not openwrt
03:22.50JTopenwrt is aftermarket
03:22.51FuriousGeorgeand by loving, of course, i mean felating at a rest stop.  you barely KNOW HIM
03:23.06foobar778openwrt is opensource
03:23.12JTFuriousGeorge: ahaha, audiable
03:23.16JTfoobar778: correct
03:23.52JTflenders: netcomm has the worst tech support ever
03:24.03flendersthose firmwares don't work with wrt54gp2
03:24.11flendersI would even give it a go if it worked
03:24.49foobar778maybe not not all wrt65g can be flashed example the v7
03:24.52flendersI almost bought those ethernet over power things from netcomm once
03:25.00flendersjust to get rid of my router
03:25.06FuriousGeorgeJT:  yeah ive installed openwrt to try to get rid of flakiness.  the device was defective so all i got was a flaky device running linux which was much cooler, yet harder to work with :)
03:25.18JTFuriousGeorge: heh :)
03:25.19foobar778the gs with 64mb ram would be best
03:25.21*** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net)
03:25.34foobar778i use a 2.2 with 16mb
03:26.03FuriousGeorgego price that one on ebay, then price a pentium II 500 with twice the ram and a harddrive that isnt solid state memory
03:26.23foobar778kinda of cool to install lynkx on ur router and surf from it
03:27.16JTflenders: hey
03:27.21foobar778I use wget from the router and drop huge files to a samb mount from the router
03:27.40JTsure, standard linux stuff
03:27.55flendersJT: yeah?
03:27.57foobar778yes thats whats so cool the router is a Linux os
03:28.21JTflenders: when you get the PRI, you've got to tell me if optus allows setting callerid to something you don't own
03:28.41flendersJT: yeah, I asked the sales girl, but she didn't know
03:28.45FuriousGeorgelinux on soekris (?) is cooler
03:28.49flendersJT: still waiting for their tech to ring me
03:29.21foobar778solaris u mean??
03:29.21FuriousGeorgeno
03:29.21JTflenders: i think that's something you more try than ask
03:29.26FuriousGeorgeturns out i spelled it right
03:29.34InHisNameanyone install app_backtricks.c in their asterisk ?  I have a need to read in file contents into the library
03:29.35FuriousGeorgei actually mean soekris
03:29.38foobar778well never heared of it
03:29.41flendersJT: I'll tell you that in 6 weeks then
03:29.45foobar778tell us
03:29.45flenders:)
03:29.53FuriousGeorgehttp://www.soekris.com/
03:30.09foobar778tell me about it one sentence
03:30.14JTno
03:30.22InHisNameit is a single board computer
03:30.24JTyou told us to go look everything up
03:30.29JTnow you can do it :P
03:30.45FuriousGeorgefoobar778: its cool
03:30.46*** part/#asterisk orbikitti (n=orbitn@68-119-118-25.dhcp.jcsn.tn.charter.com)
03:30.47FuriousGeorge:)
03:31.02foobar778yes I see
03:31.07FuriousGeorgethat was my sentence
03:31.33FuriousGeorgeyou guys wanna know whats cooler than all this stuff though
03:31.59JTFuriousGeorge: gumstix.
03:32.09JTcan't terminate PRI to them
03:32.12JTbut you can do a lot of stuff witht hem
03:32.22JTand they are way smaller than soekris, linksys, etc
03:32.37JTwww.gumstix.com
03:32.39FuriousGeorgeand since they are EVEN cooler, they cost EVEN more
03:32.39foobar778hey furious IIm using a dwg-1120S ur opinion has an fx0 port
03:32.44FuriousGeorgeseems to stay with the theme :)
03:32.46coppicewhat makes the soekris more interesting than half a dozen other similar product lines?
03:32.58JTFuriousGeorge: they're actually fairly price competitive, all things considered
03:33.53JTanyone have any suggestions for a small form factor computer/embedded unit that has a 3.3v pci slot, x86 arch, and a small size?
03:33.58JTthat's what i'm after
03:34.03JTfound nothing suitable so far
03:34.10flendersahhahahahahaha
03:34.19flendersthe smallest flying webserver in the world!
03:34.27JTeh? :)
03:34.32ping2921how can prevent a sip register timeout?
03:35.02foobar778nice size gumstrix
03:35.12foobar778pricy
03:35.36JTfoobar778: actually competitively priced compared to other embedded linux solutions
03:35.52foobar778real;tive I suppose
03:36.39JTfor the size
03:36.48JTit's not for your standard router
03:36.55JTyou can do much cooler things with it :)
03:37.32foobar778still 9 times what I paid
03:38.23foobar778imagine what u can do with a mainframe
03:38.24flendersI wanna do a router inside a robot!
03:38.51foobar778Hey have u clusterd ur linux boxes yet??
03:39.51JTno, but i'm planning a cluster over 2 machines, with about 16 virtual machines ;)
03:40.08NuggetI run a 100,000 computer cluster.
03:40.21foobar778ok
03:40.32foobar778Im President of the USA
03:40.38NuggetI do!
03:40.42Nuggetdistributed.net
03:40.48JTyou run it?
03:40.54Nuggetyes
03:41.09[TK]D-FenderJT : Get a Shuttle system, or something like an acer M-ATD desktop type case system
03:41.31*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
03:41.40[TK]D-FenderJT : Wide variety of MB's available some w/ dual NIC's etc
03:41.43JT[TK]D-Fender: i'm thinking more the motherboard than case
03:41.51JTi won't be using any standard cases
03:42.02JTmost of the stuff available is VIA
03:42.05flendersJT: what do you want to cluster?
03:42.06JTvia sucks
03:42.18[TK]D-FenderJT : I'm not talking VIA... VIA blows
03:42.27JT[TK]D-Fender: hrm, details? :)
03:42.37JTflenders: some voip stuff
03:43.02[TK]D-Fenderhttp://eu.shuttle.com/en/DesktopDefault.aspx/tabid-2/
03:44.06flendersand why does it need to be small?
03:44.46[TK]D-Fenderhttp://usa.aopen.com/
03:45.00JTwhat the hell, looks like you can't get a shuttle without a case
03:45.14JTflenders: the main cluster will be server class hardware
03:45.51JTflenders: but i want small stuff because rack space is expensive and it's a waste of space and energy using a full size pc to terminate PRIs
03:46.14Hmmhesayssweet geebus the model they used for enterprise D sold for $500,000.00 USD
03:46.16[TK]D-FenderJT : Wait... you're talking racks?  So why not 1U's?
03:46.16JTi am actually thinking it's cheaper to go Redfone bridge than pri cards
03:46.57JT[TK]D-Fender: yes, the idea is for me to build a custom 1RU server which has a xeon mobo inside and 1 or 2, space depending, tiny pcs like mini ITX or similar
03:47.12*** join/#asterisk Johnnie (n=jdlewis@jdlewis.org)
03:47.37flendersJT: I got 2 dells poweredge 850 for 1K AUD each here
03:47.41[TK]D-FenderJT : You can slap 2 x 8-port cards in an appropriately chosen 1U....
03:47.43flendersthey have 2 PCI slots
03:47.48JohnnieAnyone here overly familiar with func_odbc?
03:47.54Hmmhesaysyou can get them with pci slots
03:47.57JT[TK]D-Fender: i don't want to terminate on the same hardware
03:48.05Hmmhesaysi use func_odbc
03:48.12JTi'd never go 8 port anyway
03:48.17JTtoo big a point of failure
03:48.19Hmmhesaysget external pri gateways JT
03:48.24JTyes,
03:48.32JTi'm thinking redfone is a much better solution
03:48.35JohnnieI'm rather confused by the documentation on the VoIP Info Wiki.
03:48.36HmmhesaysI use them with no problems
03:48.38JTtrying to see if it has hardware EC
03:48.41JohnnieIt seems rather ambiguous to me.
03:48.44JTwonder if it has any other drawbacks
03:48.45JohnnieWhat am I missing?
03:48.55JTJohnnie: try the book
03:48.58JT~thebook
03:48.59jbotextra, extra, read all about it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
03:49.05HmmhesaysI don't know Johnnie: what are you missing?
03:49.45JTHmmhesays: does it have hardware EC? notice anything you can't do compared to native pci card?
03:49.46flendersJT: how much for the redfone?
03:50.00JTflenders: about USD$1800 for a 4 PRI model
03:50.02HmmhesaysJT yes
03:50.06JTvery competitive
03:50.09HmmhesaysI use quintum pri gateways though
03:50.10JTHmmhesays: awesome
03:50.18Hmmhesaysi have no idea about redfone
03:50.23JohnnieJT: Thanks, I'll check that out.
03:50.24JTHmmhesays: ah, was the yes for quintum or redfone?
03:50.25JTok
03:50.36JTHmmhesays: how much are the quintums?
03:50.40*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)
03:50.51JTdo quintums give you sip or tdmoe?
03:50.56Hmmhesayssip
03:51.03HmmhesaysI don't know I don't buy them
03:51.05JThmm ok
03:51.17HmmhesaysI just make them work
03:51.23JTalso, what's the RU vs. port density?
03:51.29JTsorry for all the questions :P
03:52.09Hmmhesaysup to for pri's in a 1U unit
03:52.26JT4, ok i guess
03:52.42Hmmhesaysotherwise it goes 2u for 8 and 4 u for 28
03:53.10JTyeah so maybe i'll go for 2RU of servers + 1RU of gateway
03:53.25JTand chuck a gumstix in each server for OOB/LOM
03:54.12JTseeing as for 4 pris, a pci card is almost the same cost as a whole gateway
03:54.18JTprofiteering anyone? ;)
03:55.18Hmmhesayspci card doesn't have all the hardware you need to go from pri to sip
03:55.24JTi know
03:55.30Hmmhesayspeople seem to forget that
03:55.34JThence why i was saying they were so much better value
03:55.39Hmmhesaysahh
03:55.44JTpci cards are overpriced
03:56.29JTmost pci cards of any type that cost more than a thousand dollars each are major profit gouging
03:56.36[TK]D-FenderJT : Really?  How much does a 1U 4-port PRI gateway go for?
03:56.45JTpeople can say "r&d", but it's really supply&demand
03:56.50JT[TK]D-Fender: USD$1800
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03:57.41[TK]D-Fenderhttp://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43638158336.htm
03:57.44[TK]D-Fender^^^^^^^^^^^^
03:58.48[TK]D-FenderJT : Redfone does NOT count.  It is a super cheap PC running Digium's lowest end card without EC, and talk TDMoE which NOBODY cares about
03:59.25bkruse_home[TK]D-Fender: ha, i like the rundown ;]
03:59.33JT[TK]D-Fender: haha they're dreaming
03:59.37JTaww
03:59.39JTno EC?
04:00.04JTdash my hopes why don't you :P
04:00.41[TK]D-Fenderbut.... will it blend? ;)
04:01.08JTi thought it'd have to run asterisk
04:01.17JTdidn't know it ran a low end digium card too
04:01.27[TK]D-Fender<- Ahm da pahty poopah
04:03.08JTUSD$16k for a quintum 4 pri, wtf
04:03.19JTok time to look at making my own again :P
04:03.36[TK]D-FenderEngineered around our unique high-speed SoC (System-on-Chip) TDMoE engine, foneBRIDGE2 provides low-latency delivery of your critical voice traffic. <- Not SIP
04:04.12JTi knew that, it didn't worry me either
04:04.25JTif tdmoe works, fine
04:04.32coppiceits pure E1/T1 to TDMoE. nothing more
04:04.43[TK]D-FenderJT : PRI gatways are pricy, but have a very different implementation design.  they can be highly intelligent, link it with billing systems, have multiply layers of redundancy, etc.  All probably overkill for your needs.
04:04.45JTcoppice: is it a digium card?
04:04.58[TK]D-FenderJT : http://www.voipsupply.com/product_info.php?products_id=2026
04:05.02coppiceno. its a box from redfone
04:05.07JT[TK]D-Fender: billing isn't really what my gateway needs to do
04:05.10JTcoppice: ok
04:05.12bkruse_homeJT: nope
04:05.22JT[TK]D-Fender and coppice, battle! :P
04:05.26puzzledJT: for %16k you can get a MaxTNT on eBay which has a ton more functionality and redundancy than that quintum
04:05.30puzzled$16k even
04:05.37JTyeah
04:05.51JTi have a couple of 3Com totalcontrols, wonder if they do voice
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04:06.07[TK]D-FenderJT : read the specs.  effectively is.
04:06.16JT[TK]D-Fender: yeah i was already on that page
04:06.29[TK]D-Fendercoppice : I remember the 1st gen casing were it looked pretty damn obvious who's card was in there...
04:06.30JT[TK]D-Fender: the specs aren't low level
04:07.02JT[TK]D-Fender: i don't know how they could build it so cheap with digium card unless they got a massive discount
04:07.09[TK]D-FenderJT : Its a funnel between PRI & TDMoE.  Nothing more.  Now HWEC, no transcoding, no nothing.
04:07.24JTright
04:07.43JTsay i could get some echo cancellers, i don't see a problem
04:07.44[TK]D-FenderJT : You DO know that the tormenta2 design wass "Open" right?  there are plenty of compatable 4-port cards for about $1000 USD
04:07.52JTi don't want my gateway to transcode
04:08.12JT[TK]D-Fender: on pci, from who?
04:08.22[TK]D-FenderJT : Lemme dredge up a name...
04:08.27[TK]D-FenderJT : beena  while....
04:08.31JTheh
04:09.43[TK]D-FenderJT : For reference : http://www.zapatatelephony.org/
04:09.57JTyeah i'm aware of the site
04:10.05JTand the original cards
04:10.12JTi thought they were relics of history
04:10.23JTnot purchasable items
04:10.37coppiceseveral people make and sell those cards
04:11.15coppicetheir main drawback is the lack of bus mastering
04:11.19[TK]D-FenderDamn... Can't recall the name.... one company made a fuss about 2 years ago....
04:11.22JThmm
04:11.25JTrhino?
04:11.27[TK]D-Fendernope
04:11.40[TK]D-Fenderthey were their own company before they started making shit cards ;)
04:11.49coppicethey do report errors, though, which the current Digium cards hide. that makes them much easier to support remotely
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04:11.58JTin any case, i'm still in the market for a small x86 mobo
04:12.30bkruse_homecoppice: things that are not zaptel debug related??
04:12.43bkruse_home(refering to the card of course, pri debug, etc etc)
04:12.44JTvia's mini itx stuff is 5v pci i believe and... VIA
04:12.58coppiceE1/T1 link errors are hidden by the drivers for the current cards
04:13.37bkruse_homegotcha.
04:13.45coppicewhy do people always have a downer on VIA. Intel has produced most of the really troublesome chipsets
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04:14.13PMantisIs there a difference in dialing SIP/hostname/exten  vs  SIP/exten@hostname  ?
04:14.15JTvia has a phenominal habbit of sucking
04:14.20[TK]D-Fendercoppice : just the C3 based uber-low-end junk.
04:14.26JTtheir itx stuff looks interesting
04:14.30JTjust 5v pci :(
04:14.49JT[TK]D-Fender: the C3 processors are overkill for what i need anyway
04:14.52coppicecan you name any 32bit PCI slot which is not 5V?
04:15.04coppice(clue: there is one)
04:15.08JTthey're 3.3v these days
04:15.10JTpci2.2
04:15.19coppicenope
04:15.26JTpci2.2 is 3.3v :)
04:15.52coppiceits 3.3V signalling. the power on *every* board is 5V, except for one Dell design
04:15.52JTunless you're going for something obscure here
04:16.03coppicethis is the same for VIA
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04:16.18JTright, so when someone says "3.3v pci, do they mean power or signalling"?
04:16.43coppicethey are normally referring to power, because evry slot for 10 years has accepted 3.3V or 5V signalling
04:17.02JTsince pci2.2?
04:17.12coppicelong before that
04:17.19JTheh
04:17.32JTi know a lot of wireless cards didn't work on pre pci2.2 boards
04:17.52coppicethat has nothing to do with voltages
04:18.07JTuhuh
04:18.27JTso you can buy digium pri cards in 5v or 3.3v, expand :)
04:18.59[TK]D-FenderJT : found one http://www.mapleleaf-technologies.com/webstore/openvox_voicedatacards.php
04:19.05coppicebecause those cards are so lousy they can't adapt to a 3.3V or 5V supply like 99% of all other PCI cards can
04:19.31[TK]D-Fenderhttp://www.mapleleaf-technologies.com/webstore/varion_cards.php
04:19.39[TK]D-FenderVarion was the name I was looking for...
04:19.51coppicevarin is tormenta 2. openvox is not
04:19.58coppicevarion
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04:20.23JTcoppice: heh
04:21.00[TK]D-Fendercoppice : crapTASTIC aren't they? :)
04:21.19JTopenvox, is there anything actually wrong with them?
04:21.23coppiceif you have a TE410P card, which requires a 3.3V supply there appears to have only ever been one 32bit slot made that you can plug it into. It was a Dell. everyone else uses those cards in 64 bit slots, which are almost always 3.3V powered
04:21.45coppicepeople seem to say nice things about openvox, but I've never used one.
04:21.56[TK]D-FenderJT : Ancient PCI design, probably next to NIL on support.
04:22.08JTancient pci design?
04:22.34[TK]D-FenderJT : Even Digiums lowest end has gone through a number of revisions...
04:22.51JTare you sure the openvox units are not revised?
04:23.01[TK]D-FenderJT : Would you risk it?
04:23.06coppiceopenvox appear to support their stuff OK
04:23.18JTwell, i'd check if it was a legitimate concern first
04:23.27[TK]D-FenderJT : vs buying from a known company that is still producing cards and improving on things?
04:23.42JTheh
04:24.01coppiceopenvox introduces new cards pretty regularly. you are just spreading FUD
04:24.04JTlooks like openvox are still making cards too
04:24.06JTso meh
04:24.24[TK]D-Fendercoppice : No... I'm selling it as certainty... thats MARKETING ;)
04:25.12coppiceno. marketing spread FUD. spreading false certainty gets you sued
04:27.40[TK]D-Fendercoppice : If I sell MY uncertainty as "the word", I'm entitled to be wrong, its still not a lie.  Also I have no economic gain, and the fact that these cards are virtually unheard of does give cause for concern.
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04:28.18[TK]D-Fendercoppice : But I'll take the edge off my comedy.... it is just a little to brisk on this topic.
04:30.51n|cotineWhat driver do Digium TE2xxP cards use?
04:31.58[TK]D-Fendern|cotine : Zaptel.  Or are you referring to the module name more specifically?
04:32.27[TK]D-Fendern|cotine : which would be : wct2xxp
04:32.28n|cotineThe latter.
04:32.52n|cotine[root@voip extra]# ls -la wct2*
04:32.52n|cotinels: wct2*: No such file or directory
04:33.05n|cotineFrom zaptel-1.2.13
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04:33.42n|cotineI am slightly confused.
04:34.07Juggiehave you considered the documentation
04:34.11n|cotineAha.
04:34.14n|cotineAliased.
04:34.43n|cotinealias wct2xxp wct4xxp in modules.alias
04:40.39Snake-Eyesany one know of write up of how to configure a box with two tdm400P cards ? (yes i have tried googleing)
04:41.02QwellSnake-Eyes: it's the same as one, you just configure more ports
04:43.14Snake-EyesQwell, i tried that, but I got chan_zap errors so I figured I might be going down the wrong path as it were
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05:13.36zeeeshhi
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05:15.13PMantisIs there a difference in dialing SIP/hostname/exten  vs  SIP/exten@hostname ? If the first dead-wrong?
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05:23.22JTwell i think you mean SIP/sip.confentry/exten to be technically correct
05:23.28JTit's not necessarily hostname
05:24.47sbingnerJT, I don't suppose you actually had an answer to his question in addition to the inane comment?
05:25.43JTsbingner: inane comment, i'm sorry?
05:26.13JTi am attempting to be helpful
05:26.27JTit's a common misconception that it's SIP/hostname/exten
05:26.37JTso i was partially answering his question
05:26.56JTusing a sip.conf entry, so the former method, is preferable
05:27.20JTas you can set all the options up in sip.conf on a per host basis
05:27.39JTsbingner: so exactly what was inane?
05:32.03[TK]D-FenderPMantis : For a hostname, use the latter, for a peer-name, the former, and it CAN matter and its cause DNS resolution errors when formatted improperly.
05:33.40PMantis[TK]D-Fender, Thanks... so for a peer, the "IAX format" will work. :)
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05:35.21JTPMantis: it will, iax/sip format, whatever you want to call it :)
05:35.28JTit's nicer to use sip.conf though
05:36.34PMantisOh of course... I just recently noticed freepbx uses the SIP/peer/exten format... I always used exten@peer for my own scripts.
05:36.50PMantiswell, scripts, dialplan... whatever. :)
05:37.40JTyeah i never use the exten@peer format
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06:09.53FuriousGeorgehmmm, is there a way to match any digit/character or none?
06:10.20FuriousGeorgespecifically actually i want * or no *
06:10.37FuriousGeorge[ *]wouldnt work would it
06:10.50JTit should
06:10.57JT* should match *
06:11.02FuriousGeorgecool
06:11.08FuriousGeorgethanks
06:11.20JTfor ANY digit/non digit it's .
06:11.40JT. uses timeout dialling when dialling digit by digit
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06:11.59JTas it means 1 or more chars, iirc
06:14.39[TK]D-FenderJT :no "." if for any NUMBER of any digits, not a SINGLE.
06:14.48JTi know
06:14.51JTthat's what i said
06:14.54JT1 or more chars
06:15.12JTand it matches non-dtmf too
06:15.23[TK]D-FenderJT : jsut took you 3 sentence fragments to peice it together ;)
06:15.32[TK]D-FenderJT : yup
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06:41.33Fr0zen_anyone here use Cisco ip phones?
06:53.56FuriousGeorge; Set iaxcompat to yes if you plan to use layered
06:53.56FuriousGeorge; switches.  It incurs a small performance hit to enable it
06:53.56FuriousGeorge;
06:53.56FuriousGeorge;iaxcompat=yes
06:54.14FuriousGeorgelayered switch just means two "daisy chained" switches between you and wan, right?
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07:13.50FuriousGeorgedarndest thing:  i set up 3 asterisk boxes as friends of each other the exact same way.  for some reason, one of them rejects the call as its trying to send it to the 'default' context even though the other box is set up as a friend in the 'inside' context
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07:16.30FuriousGeorgerequest '100@default' does not exist
07:16.31FuriousGeorgewhere is it getting 100@default, from?!
07:19.57Juggiecare to provide more information
07:20.47Juggieobviously, some phone is getting into the default context
07:20.52FuriousGeorgeJuggie: sure, ill pb something
07:21.27Juggieyour sip/iax/extensions would be handy
07:21.34Juggieand the console output
07:22.41Juggiehurry too i'm sleepy
07:22.44Juggieits almost 2:30am :P
07:22.59FuriousGeorgehttp://pastebin.ca/364575
07:23.41FuriousGeorgeJuggie: let me know if you still need extensions, but from the snippets of the other confs and the cli output, i think we can agree that shouldnt happen, no
07:25.02Juggiemore confs would be helpful.
07:25.20Juggieyour extensions.conf plz.
07:25.30FuriousGeorgecoming right up
07:25.53Juggieyour dialstring from the other box would be handy doo.
07:25.55Juggie*too
07:26.47Juggiehomer just finished making up with marge so hurry :)
07:30.14FuriousGeorgehttp://pastebin.ca/364583
07:30.18FuriousGeorge:)
07:30.31FuriousGeorgethat homer
07:31.11JuggieFuriousGeorge, iax is differnt then sip.
07:31.29FuriousGeorgeyou want to see my sip.conf?
07:31.38Juggieno, i already know the problem
07:31.39FuriousGeorgei should mention that this works in the other direction
07:31.54Juggieexten => _61XX,n,Dial(IAX2/Juanita/${EXTEN:1}@inside,60,,t,T)
07:31.56Juggieshould do the trick
07:32.39FuriousGeorgei believe you, but i shouldnt need to do that...  i dont need it on the other guys for some reason
07:32.43FuriousGeorgethere are three boxes set up this way, this is the only one that doesnt accept calls
07:32.59Juggiei dont have an answer to that.
07:33.04Juggieiax is different then sip
07:33.11Juggiesip.conf is this peer has this context
07:33.16Juggiebut iax can have many contexts.
07:33.23Juggieyou can even do context=*
07:34.04FuriousGeorgei guess ill have to do that if i cant figure out why this one isnt acting right
07:34.10FuriousGeorgebut its annoying me atm
07:35.00Juggiei am too sleepy to dive into all the configs
07:35.12Juggieyou would think you would not need to specify it when there was only one
07:35.41kaldemarFuriousGeorge: on the Claudia box, do you have 'username=Claudia' in context Juanita?
07:36.16FuriousGeorgekaldemar: username= in the context?  you mean the peer entry?
07:36.18JuggieFuriousGeorge, and from what i can see that is how it is designed to work
07:36.20FuriousGeorgenow that i think about it
07:36.23kaldemarit may not get the right username when dialing and falls back to default context because of that.
07:36.26Juggieso, something is mucked up somewhere
07:36.32FuriousGeorgethis box used to be called juanita, but now the hostname is edith
07:36.34Juggiebut its most likely not a bug.
07:36.35kaldemarFuriousGeorge: yes, the peer entry.
07:37.05FuriousGeorgeis it possible that changing the hostname of the box would cause this somehow?
07:37.07kaldemarFuriousGeorge: then it uses that username and the defined secret to dial the peer if the Dial application is called that way.
07:38.40JuggieFuriousGeorge, for fun add @inside to the dial
07:38.44Juggiei bet it fails some other way
07:38.49Juggiewhich may be more helpful
07:39.22FuriousGeorgeRejected connect attempt from blah.84.214.199, who was trying to reach '100@'
07:39.31FuriousGeorgethats what happens when i set username so were on to something
07:39.39Juggiebingo, auth problems.
07:39.43Juggiehave fun, i'm going to sleep :)
07:39.53FuriousGeorgeJuggie: thanks for the time
07:39.55FuriousGeorgenight
07:39.57Juggienp.
07:40.01Juggiemust work @ 9am :)
07:40.03Juggieand its 2:30
07:40.10FuriousGeorgegodspeed to you :)
07:40.45kaldemarFuriousGeorge: do you have friend entries in iax.conf or peer and user separately?
07:40.51Juggieyah, its some stupid auth issue.... you'll get it.
07:41.23kaldemarFuriousGeorge: if friends, are the secrets the same?
07:42.21FuriousGeorgeRejected connect attempt from 67.84.214.199, who was trying to reach '100@inside'
07:42.32Juggiealso, set verbose 10
07:42.33FuriousGeorgethats what happens when i tag @inside at the end of the dial
07:42.35Juggieto get more info
07:42.52Juggieyes, the auth is failing
07:42.56FuriousGeorgeno new info there
07:43.15Juggiedo you have iax setup with two peer=friend 's?
07:43.20Juggiesame secret=... ?
07:43.32FuriousGeorgeyeah
07:43.44Juggielook again
07:44.42Juggiepastebin me the iax.conf entries from each box.
07:44.58FuriousGeorgeJuggie: one sec
07:45.07FuriousGeorgejust the peer section or the whole thing?
07:45.19Juggiethe section for each peer
07:47.59FuriousGeorgehttp://pastebin.ca/364591
07:48.11FuriousGeorgei double and triple checked the actual values for secret
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07:48.47Juggieand both dialstrings
07:48.48Juggieplease
07:49.12FuriousGeorgeyou got it
07:50.06Juggiei have it? or you are getting it?
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07:51.21FuriousGeorgehttp://pastebin.ca/364592
07:51.32FuriousGeorgeyou got it, as in "your wish is my command"
07:51.56FuriousGeorgeis it so straight forward and obvious that it is blinding us
07:52.01FuriousGeorgei mean, WTF :)
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07:53.00FuriousGeorgeJuggie: hope you see something there im missing
07:53.10Juggiein shell, if you do a 'ping Juanita'
07:53.15Juggiedoes it resolve to something?
07:53.20Juggieon the box that cant make calls.
07:54.51FuriousGeorgek if i ping juanita (or her gotdns address in this case) from the box that can make calls, im pinging myself
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07:54.54FuriousGeorgeand it resolves fine
07:54.57FuriousGeorgesame other way
07:55.17FuriousGeorgethe one thing that has changes is that claudia's hostname is claudia whereas juanitas used to be juanita and is now edith
07:55.38FuriousGeorgefor some reason im scared to remotely change the hostname, though i dont see what would cause that
07:56.05kaldemarwhat's the deal with all the commas in the dial lines?
07:56.15FuriousGeorgerather:  i dont see how changing the hostname would affect it
07:57.06FuriousGeorgekaldemar: looks like at first i didnt have seconds specified, then i added seconds but didnt remove the comma
07:57.25FuriousGeorgeor is it that i need two commas anyway if i dont use (insert option here) before t,T option
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07:58.15kaldemarDial(...,60,tT) <-- i'd put it that way.
07:59.33kaldemaraccording to wiki there's an URL parameter after the third comma.
07:59.33FuriousGeorgeill change it but i dont think its causing my problem
07:59.44FuriousGeorgehmmmm
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08:00.36kaldemari don't think it directly causing the problem, but you never know if there's a bug or something.. less errors, less confusion.
08:02.44Juggiei dont see anything blatentally wrong, but i am fairly sleepy, ask me tomorow when i'm @ work and mostly awake if your still stuck.
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08:04.01FuriousGeorgethanks Juggie
08:04.23FuriousGeorgekaldemar: took out all the seconds and t and comma since the remote context handles all that anywya
08:04.25FuriousGeorgeno luck
08:04.42FuriousGeorgeback to seeing '100@' in remote cli
08:05.20Juggiehmmmmmmm
08:05.30Juggiethe only way * has to match the incomming peer is the host
08:05.51Juggiecheck that
08:06.04FuriousGeorgenot sure what you mean
08:06.06FuriousGeorgehost= setting
08:06.09FuriousGeorgethat's fine
08:06.27FuriousGeorgeotherwise iax2 show peers/users would reflect
08:07.22FuriousGeorgethere are some dull moments, but never a dull entire day :)
08:08.32FuriousGeorgegasp
08:08.41FuriousGeorgeisnt there a dnsmanager file or something
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08:09.01carrartry adding deny=0.0.0.0/0.0.0.0 & permit=1.2.3.4/255.255.255.255 on each
08:09.18carrarspecify the IP /32 for each
08:09.27Juggiethats not it
08:09.35carrarthen you have it working
08:09.39carrarno problem then
08:10.28Juggiehttp://pastebin.ca/364606
08:10.29Juggietry that
08:10.51Juggiei would have my working configs if i was at the office, but i'm not and havnt worked on * in like a year so i'm hazey :)
08:11.23Juggiezzzz.
08:12.13carrarFor every IAX peer I havem I defined the following:
08:12.14carrartype= user= host= secret= context= peercontext= accountcode= auth= deny= permit= disallow= allow=
08:14.18carrarexten => s,1,Dial(IAX2/IAX-namehere/${ARG1},100)
08:14.22carrarput that in a macro
08:15.02FuriousGeorgeJuggie: looks like setting the usrname on both sides worked, homeslice
08:15.07FuriousGeorgeyouve earned your sleep
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08:15.26JuggieFuriousGeorge, i bet the box that coudnt be called INTO had more then one iax peer defined
08:15.28Juggiecorrect?
08:15.30carrarbesure to set a default
08:15.38FuriousGeorgeJuggie: sure did
08:15.52FuriousGeorgei actually commented out a peer that came in the default config while i was in there
08:16.01Juggieyah, and more then one with the same host=
08:16.11FuriousGeorgei dont think so
08:16.16FuriousGeorgelest i messed up and didnt notice
08:16.22FuriousGeorgei looked for Dups though
08:16.45Juggiesomething is causing the auth to fail, it should go oh, host=blah thats that peer, but its not.
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08:16.52Juggieanyways, you can figure that out later
08:16.57Juggiei allways use username
08:17.00Juggiemakes things easier
08:17.08kaldemarmy guess it that the call doesn't have a proper username.
08:17.10Juggieit will never fail and doesnt depend on any guesswork
08:17.10FuriousGeorgeif you want, after you sleep, ill take out username and see if taking out the demo peer did it
08:17.38Juggieyou can try it, just say what happened, i'll look over last nights irc log tomorow.
08:17.39FuriousGeorgeeverything is identical on both ends and it only worked both ways.  right down to the motherboard and the sipphone
08:17.45FuriousGeorgethe isp is the only var
08:18.03Juggiebut allways user username= its more foolproof.
08:18.07FuriousGeorgeJuggie: its not that i want you to go to bed, im in the same timezone and dont have the energy to mess with this anymore :)
08:18.08Juggieand easier to see whats going on
08:18.21FuriousGeorgeJuggie: will from now on
08:18.45Juggieinstead of * guessing which peer your looking for per hostname
08:18.49Juggieit knows exactally.
08:19.00Juggiezzz
08:19.05FuriousGeorgemakes sense
08:19.14FuriousGeorgelater dude, see you tomorrow
08:19.35Fr0zen_anyone here use cisco ip phones?
08:22.09carrarI do
08:22.20carrar7940/7960
08:23.47carraras well has LinkSys 94X and Polycom 6XX with sidecars
08:24.11carrarand softphones
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08:35.07FuriousGeorgei have the first sipura/linksys model
08:35.10FuriousGeorgethink its an 841
08:35.38FuriousGeorgea monkey on qualudes could not have designed a less intuitive telephone
08:38.11FuriousGeorgespeaking of monkies:  this iaxcompat=yes option that's commented by default in iax.conf...  when it says it can be disabled if i have "layered switches", what does that mean exactly
08:39.11FuriousGeorgeaccording to wikipedia a multilayer switch just means a device switching using a cpu rather than an asic, is that what that refers to?
08:39.22FuriousGeorgeor does that just mean daisy chaining regular old switches
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08:41.01FuriousGeorgeits not that i dont know the difference, i just dont know what that comment in iax.conf is talking about
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08:56.18codeyhi there
08:56.36codeycan somebody help me on a queue? I'm new to asterisk and I just have to add something to my queue ... but I don't get it.
08:56.50codeyIf someone calls our number with an ending 0 he gets into the default queue
08:57.09FuriousGeorgeive never implemented a queue, but maybe if you link me to what you're trying to do i might could help
08:57.18codeydefault queue calls SIP/1 and SIP/2 - and now I just want to get SIP/3 and SIP/4 called if nobody gets on the phone
08:57.37codeyafter 10 secs
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08:58.30codeyor just another question.. what's the second parameter of Dial()? ... maybe this is just what I need to know
08:59.02codeyso i can just do exten => bleh,Dial(SIP/1&SIP/2,10) exten => bleh,2,Dial(SIP/3&SIP/4)
08:59.30FuriousGeorgei know that one off hand dial(tech/chan/exten,options)
08:59.49FuriousGeorgeyou can do that
09:00.00FuriousGeorgebut you want to wait 10
09:00.17FuriousGeorgethere is a queue app you dont seem to be using
09:00.25FuriousGeorgeor am i misunderstanding
09:01.37codeyone sec, i'll paste the part of my extensions
09:02.21kaldemarDial(<channel>,<timeout>,<options>) timeout is how many seconds you let it ring before moving on in the dialplan unless the remote end answers.
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09:04.32codeyhttp://rafb.net/p/X4o6iA84.html
09:05.27codeyif i'm understanding this right - it defines the 4 sip extensions and then goes on to dial them
09:05.48codeyand if noone gets on the phone within 12 seconds it goes to "queuezentrale"?
09:06.09FuriousGeorgehttp://www.voip-info.org/wiki-Asterisk+call+queues
09:06.53FuriousGeorgeif you use goto you have "to go somewhere".  why didnt you show me context [queuezentrale]
09:07.02FuriousGeorgezentrale is italian?
09:09.27codeygerman
09:09.39FuriousGeorgenot even close :)
09:10.36codeyhttp://rafb.net/p/fbXJW579.html
09:11.14FuriousGeorgeyou are using app_queue, and you want more phones than your tutorial teaches?  you probably want round robin, two extensions at a time?
09:11.50codeyi didnt do the configuration, i jsut have to add something :/
09:12.24FuriousGeorgecodey: are you just trying to make a queue using dialplan logic?
09:12.34FuriousGeorgedo you know what i mean by that
09:12.59codeyi think so, yes ;)
09:13.08FuriousGeorgeok, wait one sec
09:13.58FuriousGeorgehow about this, ill design a simple queue so you will learn how it works.  you want to dial sip/1&2 then 3&4 with a 10 second pause?
09:14.04FuriousGeorgethen restart?
09:14.18FuriousGeorgenot necessarily in that order
09:15.05codeysounds good, why not
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09:16.39FuriousGeorgeso, let's say you put someone in the queue by dialing 99
09:20.55codeyokay
09:24.45FuriousGeorgeok almost done sorta
09:28.05codeyokay .. i've got to KILL someone now
09:28.11codeythey've sent me the wrong configfiles
09:28.13codey...
09:28.20FuriousGeorgehttp://pastebin.ca/364667
09:28.48FuriousGeorgei started to do the complicated version where you make a terminating loop happen, but i figured i'd leave that to you
09:29.06FuriousGeorgei /think/ that will work
09:29.17codeyokay, thanks :) i'll take a look at that and try it with that
09:29.28FuriousGeorgelike i said though, ive never used app_queue, or tried to do it myself
09:29.52FuriousGeorgealso, they wont hear music, but ringing
09:29.58codeyoh my f*cking god
09:30.03codeyI'm an asterisk noob - okay
09:30.08codeybut their queue is ... kinda lame
09:30.14codeyeven *I* understand it oO
09:30.24codeyits funny...
09:30.56codeyhttp://rafb.net/p/4eqQMc14.html
09:31.08codeyam I the only thinking that *this* is totally useless? :>
09:31.42FuriousGeorgearent those parts of different but related files you are pasting
09:31.50codeythe first on is extensions.conf
09:31.50FuriousGeorgeone is extensions.conf and the other is queue.conf
09:31.52codeythe second part is queue.conf
09:33.04FuriousGeorgethe only problem i see is that youll never get into the queue b/c there is no priority one
09:33.19FuriousGeorgei dont think thats useless, its the right way to do what its intended to do
09:34.34FuriousGeorgelike i said i dont use queue's so im not sure as to the exact behavior, but the general idea is clear
09:34.56FuriousGeorgebut you want round robin and different agents
09:35.37FuriousGeorgeor send them to another queue when/if they return
09:35.37FuriousGeorgehope that helps, got to run
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09:44.46BobHenkDoes anyone have a working DUNDi configuration for Trixbox 2.0? I can't get it to work, so every help is very welcome.
09:44.50*** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com)
09:45.13phearlesshello !
09:45.57phearlesswhen I press 00 I got call ended on my Sipura 942 phone, very strange
09:46.09phearlessLinksys/Sipura SPA942
09:46.21phearlesswith (0[1-9]xxxxxxxxx|4xx) as a dialplan
09:46.47phearlessexten => _0XXXXXXX.,1,Dial(Zap/g1/${EXTEN}) and this for outbound calls in extensions.conf
09:46.52phearlesshow can I debug this?
09:49.38*** join/#asterisk friedrich| (n=friedric@e177248017.adsl.alicedsl.de)
10:00.26phearlessI tried with : (0[1-9]xxxxxxxxx|4xx|00xxxxxxxxxxx)
10:00.30phearlesson the phone dialplan
10:00.33phearlessand same problem
10:05.13*** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com)
10:05.15Stephniehi
10:05.39Stephnieis there any way to secure AGI PER SCRIPTS? like If I can compile or something like that?
10:06.11StephniePERL*
10:08.07simplexiohow i get channel status from sip user, like availabe or busy(if there is call going)
10:09.17simplexioor more like sip extension. ChanIsAvail(SIP/2009|sj) returrs allways 0? and goes to next priority
10:25.02kippihi, I have added pickup to 3 of my extenstions, two of them will pickup, the other one says nothing to pickup, any ideas?
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10:31.41phearlessthis "00" is crazy
10:31.53Stephnieis there any way to secure AGI PERL SCRIPTS? like If I can compile or something like that?
10:31.56phearlessI can not dial any 00something number
10:32.09phearlessStephnie: I do not know!
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10:34.09phearlessit does not seems to be a vertical activation code problem
10:37.18*** join/#asterisk ipguy (n=ipguy@124-168-21-206.dyn.iinet.net.au)
10:37.39ipguyhi all, i just setup my first asterisk box :-)
10:37.54ipguyall working
10:37.54phearlesscool !
10:38.01ipguyexcept
10:38.20ipguyi can't sip call other sip users
10:38.36ipguyas in from one asterisk box to another
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10:38.54ipguyall local accounts work fine, just no remote access
10:39.02phearlessnot an NAT problem ?
10:39.11ipguyno NAT or firewall
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10:39.36kippii am getting this problem with codec, how can I sort this out? http://www.pastebin.ca/364714
10:40.00XIN01OZAnybody know by chance why after a call is placed through a2billing it calls the number and hangs up on pickup?
10:41.15ipguyphearless: does one need to setup something in extensions.conf to allow sip call from two remote asterisk boxes ?
10:41.29kippialso getting this: http://www.pastebin.ca/364717
10:41.29phearlessipguy: I do not know
10:41.58ipguyphearless: ok thanks
10:44.47*** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au)
10:44.57kezza491Hi
10:47.27kezza491Would some one be able to explain to me if there is like a free service so you can like ring your computer to check if your astrisk set up is ok?
10:49.50Poincarekezza491: I think fwd had such a service
10:50.02kezza491hmm k
10:50.09Poincarefreeworld dialup
10:50.23kezza491Just i want to fiddle around with asterisks before i get commited to anything
10:50.26Poincarebut that's only for the sip/iax accounts
10:50.53kezza491?!
10:51.05Poincarehow do you want to ring your computer?
10:51.07kezza491Have'nt actual started to work on asterisks as yet
10:51.11Poincarevia the PSTN line?
10:51.27kezza491Ehh i dont mind it wont matter in the end will it that much?
10:51.28PoincareOr do you want to test an 'incomming' SIP/IAX line?
10:51.52kezza491ok what is a SIP/IAX line?
10:52.05PoincareSIP and IAX are voice over IP protocols
10:52.19kezza491my idea is that i connect the phone line to the comp and also have the comp dial out using the net but for learning purposes just the net for now
10:52.55Poincarein that case: register with FWD
10:52.59kezza491Ok
10:53.23kezza491is it possible to have asterisks work with the net and a phone line?
10:53.31Poincareif your asterisk is working, it won't matter that much if its a 'real' line or not
10:53.33Poincaresure
10:53.42kezza491Ok
10:54.39kezza491So no hidden costs or anything with this freeworld?
10:56.09Poincareno, as long as you don't want to call PSTN lines it's completely free
10:56.31Poincareand they have several test numbers and a 'call back test service'
10:56.51kezza491Ohh k
10:57.14kezza491so say if i have an asterisks box how do i interface that with say another computer like a windows machine?
10:57.35Poincareand if you ask arroun here, maybe someone will call your fwd number to test
10:57.53kezza491ahh k
10:58.19kezza491Hate to be a pain but what is some really bare basic easy to understand asterisk doco?
10:58.20Poincareasterisk <=> windows boxes: SIP clients or IAX clients
10:58.39kezza491k
10:58.41kezza491thanks
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11:26.19shadebobhi,
11:28.34shadebobI need some help with multiple registration on the same SIP provider. Problem is asterisk take the context of the last registered peer for incoming calls for this provider. I don't want to create a single context with multiple exten=>number,... because I have dev many GUI. Can asterisk can manage the good context in this case?
11:28.57shadebobI see an old project in the svn tree (oej/register) but it seem abandonned
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11:52.32kezza491What is some good documentation for astrisks for people just starting off that is simple and easy to understand
11:53.27E-bolathe book was a good starter for me
11:53.38E-bolato get a basic understanding of the different terms and concepts
11:53.44E-bolaits however not enough to actualy setup a system
11:55.08kezza491hmm
11:55.22kezza491mabe a tutorial is my best bet then fiddle around until i get my desired effect
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12:09.19XIN01OZwhoa im having it rough this morning for some reason when a call from SIP is placed the call makes it through but once picked up by the called number asterisk hangsup after 2 secs and I get a declined
12:09.27XIN01OZcause code 16
12:09.42XIN01OZanybody familiar with what is going on
12:11.17*** join/#asterisk RoyK (n=roy@80.239.107.70)
12:13.20XIN01OZRoyK: Good morning we talked awhile back .. might you know where i could get some more information about why a call is being dropped 2 seconds after it is picked up by the called number
12:13.27XIN01OZall im seeing is cause code 16
12:13.57*** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au)
12:14.21XIN01OZdidnt expect this .. planned something for someone this morning thought all was well
12:14.42JT16 is normal clearing
12:15.00RoyKXIN01OZ: no idea
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12:15.21JT16 means successful call, hanging up, from memory
12:15.53XIN01OZyeah that was all i could find as well but my sip client gets Declined
12:16.08mickey9Hello All, I've decided to buy ABE. I've been using Sangoma cards until now. Are they supported under the ABE support package?
12:16.22XIN01OZas well through IAX
12:16.38JTmickey9: doubt it
12:16.42JTat least not officially
12:17.00InHisNameYou get cause code 16 with more than one call destination ?
12:17.14InHisNameThat is only 2 sec long ?
12:18.52XIN01OZactually i now see it maybe a spawn extension error : http://pastebin.com/885158
12:18.54mickey9JT: What would you suggest for a digium channelized E1 with Hardware echo canc.?
12:19.10JTmickey9: have you had problems with sangoma cards?
12:20.43XIN01OZJT: Could you please look at that pastebin and see if it says anything otherwise to u
12:21.04JTit says you're using freepbx :/
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12:22.23XIN01OZalright most likely a freepbx issue then- im looking for more debug info
12:22.42JTi can't determine what it's done
12:23.01XIN01OZverbose at 69 did not reveal anything more
12:23.11XIN01OZappreciate it
12:23.19JTlol i think the max usefulness is around verbose 6
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12:23.22JTi usually use 10
12:24.37InHisNameanyone install app_backtricks.c in their asterisk ?  I have a need to read in file contents into global var
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12:43.29tzafrirInHisName, there's the function SHELL in recent versions
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13:15.32kippihow comes my phone keeps on ringing and not going to voicemail? http://www.pastebin.ca/364858
13:16.30headstonehello all. Does anyone have a B410p up & running? I'm getting insane with this card
13:18.59headstonecan anyone help me ? I'v the lights blinking but ztcfg shows no channels
13:23.53Nobbieheadstone: what card ? what lights/colours
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13:25.07Nobbiekippi: can you paste your log file for the relevant call ?
13:25.23puzzledhi
13:29.01kippiNobbie: http://www.pastebin.ca/364879
13:31.42Nobbiekippi: does agent 1153 ring for 12 seconds ?
13:31.56kippijust keeps ringing
13:31.59Nobbiesince that's your timeout value
13:32.09Nobbieit should say something like: nobody picked up in 12000ms
13:32.39kippiwhat would be stoping it from doing that
13:32.58Nobbieare you sure you're waiting for 12 seconds ?
13:33.22Nobbie_at least_ 12 seconds.
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13:34.45kippiNobbie: waited until the phone stoped rining but it did not go to voicemail, just had music on hold
13:35.01headstoneNobbie: Digium B410P 4BRI
13:35.08Nobbieand how many seconds was that ?
13:35.31headstoneNobbie: it has 4 lights on the back (one 4 each BRI channel)
13:35.42Nobbieheadstone: do you have bristuff installed ?
13:36.58Dr-Linuxany voicemail expert active
13:37.11kippi44
13:37.12headstoneNobbie: no. On all the forums I'v been (including Digium's intall intructuions for thar card, it was never mentioned
13:37.58headstoneNobbie: for all I'v read it seams to use di mISDN drivers
13:38.13Dr-Linuxwhen i get voicemail, in subject voicemail id and in body id always differet
13:38.21headstoneNobbie: gonna try it anyway
13:38.54Dr-Linuxlike if message in subject is 0005 it will be voicemail 6
13:39.03Dr-Linuxalways one difference
13:39.10Dr-Linuxany body understand my problem?
13:39.39headstoneNobbie: question: arn't does drivers for the junghanns interfaces?
13:39.58headstoneor they work as well with Digium's?
13:45.46kippiNobbie: anyideas?
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13:48.37joelsolankiHello All.
13:48.46joelsolankiHave question regarding cdr.
13:49.08joelsolankiI have allowed my client to make calls on basis of IP. host = IP
13:49.17*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
13:50.28Nobbieheadstone: i think they're for most BRI cards. worked with my Duxbury too
13:51.02Nobbieheadstone: mISDN may be the way to go *shrug*
13:51.15Nobbiekippi: try removing the 'n' option
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14:00.34kippiok
14:03.14tzangermorning
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14:08.47kippiNobbie: done that, still nothing differnet
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14:20.10Ahrimanessomewhere on the great internet, i found someone who made a mysql connection pooler for asterisk, anyone here seen that?
14:20.58e-ddiethere is a great internet?
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14:21.12ManxPowere-ddie: Not that I'm aware of.
14:21.22Ahrimanesyeah, as opposed to the sucky one
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14:22.06e-ddiedoh
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14:22.29Ahrimanesdamnit.. need that pooler, but cant remember which software package had it
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14:37.05MarkWDMorning, we are moving our server from our RD area to our test area and we will be behind a real fire wall what are the range of ports that need to be opened in order to use sip? or is there another way...
14:37.39MarkWDshould we be look at iax
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14:39.30[TK]D-FenderMarkWD: 5060-5070,10000-20000 all UDP
14:39.43MarkWDThanks
14:39.58[TK]D-FenderMarkWD: Unless you have another server on the outside or an ITSP that uses IAX2, I wouldn't bother
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14:54.38iqHi
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15:19.18atnhi everyone, im trying to see how many users a daemon can handle and how many CPU/MEM a daemon need, any help on that? maybe a redirection to a document?
15:19.56ManxPoweratn: there is no way to tell because there are too many variables involved.
15:20.19ManxPowerAsterisk does not use a large amount of memory, so your main concern would be CPU usage.
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15:20.47atneven
15:21.12ManxPowerFor example, SIP to SIP on the same LAN with reinvites uses virtually no CPU.  If 1 leg of a call is G729 and 1 leg of the call is iLIBC, then the call will use significant amount of CPU.
15:21.41ManxPowerIf you are going SIP with ulaw codec to/from PSTN line connected to the Asteirsk server in the USA then CPU usage will be very low.
15:22.24atnwhat about iax?
15:22.37ManxPowerThe protocol has almost no impact on CPU usage.
15:22.55atnso a celeron 800Mhz can handle 100 users ? :P
15:22.58ManxPowerCompression (i.e. codec conversion) is what takes the CPU.
15:23.07ManxPoweratn: is that 100 users or 100 calls at the same time?
15:23.22atn1 call per user
15:23.24atn100 calls
15:23.54ManxPowerA celeron 800Mhz MIGHT be able to handle 100 calls if they are all using ulaw, reinvites are enabled, and there is no communication to the outside world, including PSTN
15:24.03wunderkinWow.  I had a package sent FedEx ground from NY to PHX.  Expected delivery 5 days.  Ends up being 3.  Wow... They aren't screwing around anymore (hence their new ground commercials, I guess)
15:24.16ManxPoweroh and if there are not many call setup/teardowns per min
15:25.00ManxPowerIf you need 100 calls, stop being a cheap ass and buy a decent server.
15:25.28wunderkinheh
15:25.41*** join/#asterisk infernix (i=nix@spirit.infernix.net)
15:26.36ManxPowerYou are basically asking "Can I use my Ford Escort in a race and expect to win?"  The answer is "Yes, if all the other cars in the race are broken down."
15:27.08ManxPoweri.e. The only situation where the answer to your question is "yes" is so unlikely to happen the answer might as well be "no"
15:27.26mercesteswhat if the escort has NOs?
15:27.33*** join/#asterisk shinux_ (n=shinux@196.220.30.98)
15:27.56ManxPowermercestes: There's yet another variable 8-)
15:27.58filetranslation: "what if my system is overclocked?"
15:27.59mercestesand rocket engines
15:28.03[TK]D-Fendermercestes: The when it EXPLODES you have a chance of winning ig the judge qualifies little BITS of you arriving at the finish line first...
15:28.30fileI remember overclocking a Celeron 300MHz CPU... I jumpered the wrong pins and melted the jumper
15:28.39mercesteslol@file
15:28.43[TK]D-Fenderfile: I remember OC'ing my 300 to 450 :)
15:29.03[TK]D-Fenderfile: Best bank/buck OC ever
15:29.03mercestesI remember overclocking a 1.3ghz.  I had a process "freeze" and I set it to realtime and melted my proc.
15:29.19file[TK]D-Fender: quite
15:29.46[TK]D-Fenderfile: Celery's were stable then....
15:30.02[TK]D-Fenderfile: Now I seriously can't be bothered with OCing anything...
15:30.54file'tis already fast enough
15:30.57atnok thanks
15:34.09*** join/#asterisk Juppers (i=imagine@phantom.blueone.net)
15:34.43Juppersanyone familiar with using FXS ports on a cisco 2610 with asterisk? How do you register them? The 2610 doesn't support sip authentication that I can find.
15:40.22*** join/#asterisk Sputtering (n=Keelan@192.197.213.245)
15:41.03*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
15:46.39*** join/#asterisk ocnarfid (n=ocnarfid@nylug/member/ocnarfid)
15:47.56*** join/#asterisk marv[work] (n=timr@24.214.206.254)
15:51.04kippican anyone see why my calls are not going to voicemail? http://www.pastebin.ca/364999
15:52.09danpSet(CALLERID(name))=Hall Bookings)
15:52.18danpi think you have an extra ) there
15:52.54danpbut that might not be what's keeping it from going to voicemail
15:53.46kippiwhat could be doing this?
15:53.59Sputteringtry fixing the bracket and testing again
15:54.06kippihave
15:54.20kippiso its like Set(CALLERID(name)=Hall Bookings)
15:54.26mercesteskippi:  Do you have a timeout= in queues.conf??
15:54.41*** join/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de)
15:54.43unicehi
15:54.56mercesteskippi:  Dump a timeout=6 in queues.conf under [hall] and do an reload app_queue.so
15:55.07mercesteshello, unice.
15:55.10Sputteringkippi: with no-o ne signed into the queue, what happens?
15:55.24n|cotineI setup DUNDi according to the documentation, and everything works fine - except if I call an extensions that does not exist, but another PBX is advertising.  Then the channel stays in State Busy with application data Congestion(), and the channel never drops.  Any idea what's wrong here?
15:56.13*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
15:56.51*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)
15:57.07unicei've upgrade from 1.2.10 yesterday.. asterisk 1.2.14 seems to ignore the astrundir statement in asterisk.conf - i'm unable to connect the socket using asterisk -r  (Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?))
15:57.24kippiit works with the timeout in the queue
15:57.45uniceastrundir => /var/run/asterisk - file is there and has the permissions set accordingly
15:58.04unicedid i miss a chage?
15:58.09unicechange, even
15:58.24ManxPowerunice: only if you didn't read the Changelog
15:58.44unicewell i grep-ed it for astrundir, socket and .ctl :)
15:59.23mercestesunice:  That error message is very uninformative.  It doesn't really mean that asterisk.ctl is there, or not there, or that asterisk is or is not running.
15:59.33mercestesunice:  what that error message really means is "oh crap, something went wrong.  sorry."
15:59.41uniceyeah i know
15:59.47ManxPowerI assume you can start asterisk as "asterisk -cvvv"
16:00.06uniceasterisk.conf gets parsed fine, as the output of asterisk -cvvv shows
16:00.17mercestesunice:  Try doing a hard shutdown on asterisk (killall -9 asterisk anyone?) and removing the .ctl file from /var/run/asterisk and do the asterisk -cvvv and check for real error messages.
16:00.36*** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose)
16:00.44mercestesunice:  Are you doing this via ssh?
16:01.12unicei've checked for errors, none obvious
16:01.17uniceyes, ssh
16:01.38mercestesunice:  do your asterisk -cvvvvv in one window, and with asterisk still up, try your asterisk -r
16:02.10uniceok haven't tried that
16:02.50unicebut i have to wait... people are using the server now :)
16:03.50unicei forgot to mention: asterisk tries to write the pidfile to /var/run instead of /var/run/asterisk as configured by astrundir statement
16:03.56*** join/#asterisk Johnnie (n=jdlewis@jdlewis.org)
16:04.18unicewell i'll check the -cvvvv output.. thanks so far!
16:05.08mercestesunice:  Check your config file permissions and make sure * can read the config file
16:05.22mercestesunice:  And make sure /var/run/asterisk/ exists and has the proper permissions.
16:05.55unicethe config files are owned by the asterisk user... as well as /var/run/asterisk
16:06.12uniceaccording to -cvvvv output the files are read fine
16:07.02ManxPowerunice: time to report it on bugs.digium.com
16:07.39uniceyeah i'll do it after i've checked the output again tonight
16:07.50*** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
16:07.50*** mode/#asterisk [+o russellb] by ChanServ
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16:08.48*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)
16:09.44*** join/#asterisk anthonyl (n=fbfff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
16:12.19ManxPowerunice: I left the paths the default and have no problems
16:14.45atnwhere can i find asterisk specifications?
16:14.51Qwell[]atn: for what?
16:15.12atninstallation specifications
16:15.29mmlj4please explain
16:15.35atnwhat it need to be installed
16:15.41atnwhat software i need
16:15.49mmlj4you mean system requirements?
16:15.55mercestesatn:  Depends on what your trying to do.
16:16.26atnno i mean software requirements
16:16.42mercestesatn:  an operating system.
16:17.01Qwell[]a unix/unix-like operating system
16:17.04mercestesatn:  and a text editor.
16:17.17mercestesor a windows one if you *really* must.
16:17.18atnyes on linux
16:17.20atnwhat else
16:17.23mmlj4you need gcc and make
16:17.39Qwell[]and a shell
16:17.53mercestesatn:  And hot babe to monitor your CPU usage
16:17.58fileand power
16:19.26mmlj4hey ManxPower: you know I'm Joey, right?
16:19.52[TK]D-Fenderatn : www.asterisk.org
16:19.54[TK]D-Fender~docs
16:19.55jbotmethinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
16:19.57[TK]D-Fender~book
16:19.58jbotbook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
16:21.05*** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)
16:22.09DrukenLPYmorning everyone
16:22.29*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
16:22.35mercestesmorning
16:23.13*** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it)
16:23.35*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
16:29.13*** join/#asterisk variable_office (n=variable@cerberus.iswan.net)
16:33.38[TK]D-FenderWow... I scared him right off :)
16:34.31mercestesGood job.
16:35.20mercestesbkw_:  Aww....just find out she's a fraud?
16:37.42[TK]D-Fendermercestes: We can't all be psychic like you! ;)
16:38.02[TK]D-Fender.... er .... make that PSYCHOTIC ;)
16:38.07mercesteseither or.
16:38.23mercestesBut I do offer dirt cheap communications through Telepathy telco.
16:38.57*** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com)
16:41.27[TK]D-Fendermercestes: I'll also take for granted that the echo is entirely natural :O
16:43.58DrukenLPYthe echo makes it "magical"....
16:44.58*** part/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de)
16:45.17mercesteswhen is the next astricon thingy?
16:45.39mercestesYea, that's just my multiple personalities trying to speak over each other.
16:45.49filemercestes: September I do believe
16:46.32*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
16:49.39mercestesSweet.
16:49.58*** join/#asterisk VampBoi (n=VampBoi@68-187-206-043.dhcp.ahvl.nc.charter.com)
16:50.30*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
16:50.41mercestes:D
16:50.44VampBoi:P
16:50.56kippiwhy would asterisk be striping off the 0 when showing the callerid when coming in?
16:51.30mercesteskippi:  I don't think * is doing that, but a 0 at the beginning of a valid caller ID makes no sense.  Exactly where/how are you seeing a striped 0?
16:51.57mercesteskippi:  And what's wrong with stripping?
16:51.58mercesteskippi:  and look for "strip most significant digit" rules.
16:52.00*** join/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de)
16:52.20kippiwill that be on the zaptel.conf? or extenstion?
16:52.43mercestesVampBoi:  That's scary.
16:52.57*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
16:53.10mercesteskippi:  If your worried about zaptel.conf then ask you rtelco why they are stripping off the 0 in your spoofed and invalid callier id.
16:53.51mercestesVampBoi:  Aside from loosing your CDR's and your voicemails everytime you reboot your system.....it *should* be ok.  You'd have to somehow update your changes onto a slax cd or you would loose your * install and config changes every reboot oo
16:54.01mercestesVampBoi:  Unless you have that flash memory crap
16:54.26VampBoiwell I'm going to install Slax onto the box directally I'm going to do the server version
16:54.35mercestesVampBoi:  Weird memory rules there too to avoid CDR thrashing.  I wouldn't plan on running anything real intensive because your RAM is going to be dedicated to yoru kernel.
16:54.57mercestesVampBoi:  oh.  in that case, should be standard linux stuff
16:55.09VampBoilol thought so just wanted to make sure :P
16:55.13mercestesVampBoi:  I suggest gentoo tho
16:55.34[TK]D-FenderVampBoi: Just install Slackware and be done with it.
16:55.40VampBoimy uncle has * running on one of his servers at his place and I got addicted to the music on hold and stuff :P
16:56.25cpmmultigenerational asstriks hacking, this can't really be a good sign
16:56.36*** join/#asterisk bmg505 (n=leon@c1-181-4.rndf.isadsl.co.za)
16:56.48kippior is there away I can add them?
16:58.41mercesteskippi:  this is so ppl can just hit 'dial' on "missed calls" because you require a 0 for outbound calls, huh?
17:00.25VampBoimercestes may I pvt u?
17:00.53mercestesVampBoi:  Sounds lurid.  Sure.
17:01.00VampBoilol
17:04.32*** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
17:08.37*** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)
17:17.15*** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no)
17:17.36ryantanyone know of any nice high res wallpapers/images of asterisk logos, etc?
17:17.57mercestesdo a google image search on asterisk   with safe search off
17:18.01mercesteswhat's the worst you could get?
17:18.08ryantthat didn't being back much of value
17:18.13wunderkin[TK]D-Fender: woot i finally got my ip501! hopefully the customer gets theirs today too
17:18.29ryanthow about someone internally at Digium? You guys have to have high res logos somewhere!!! :)
17:18.33mercestestry digium
17:18.39[TK]D-Fenderwunderkin: Good to hear.  Great phone....
17:18.44Juppersanyone familiar with using FXS ports on a cisco 2610 with asterisk? How do you register them? The 2610 doesn't support sip authentication that I can find.
17:18.48mercestesor gratis6
17:18.55mercestesoh, oops.  That has nothing to do with asterisk.
17:19.33kippiwhat is the best way to be able to set night serice/
17:19.46kippiwant them to be able to dial a * code
17:19.59mercesteskippi:  google asterisk hire a consultant
17:20.15*** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk)
17:20.30kippiis it that hard/
17:20.30mercesteskippi:  or read the book.  dial a * code is just an exten => *code,1,Do(something)
17:20.58kippiI should be able to use a string?
17:21.03mercesteskippi:  night service is just a GotoIfTime,*********,1,Hangup()
17:21.15mercesteskippi:  Sure!  If your device can dial *code then yes.
17:22.07*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
17:23.50mercesteskippi:  Otherwise I suggest something more numeric like *1234 or *812  or *6969696969 or something.
17:25.17*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
17:25.29kippijust need to work out how to get the things in a string
17:25.56mercesteskippi:  I use a tin can and a button
17:29.50*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
17:38.39*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)
17:47.15headstoneanyone with Digium B410P?
17:47.32*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
17:48.30puzzledheadstone: no but what's the problem
17:48.36florzWhat mechanisms are there to let a SIP -> PSTN gate know that one wishes to block caller ID presentation?
17:49.27puzzledflorz: think there is a command built in. check show applications
17:49.40bkruseflorz: you can match that in the dialplan of asterisk
17:49.44bkruseheadstone: i have a few :]
17:50.11bkruseheadstone: if it has to do with installation and configuring, i can help.....anything beyond that (errors) i have no idea
17:50.30florzpuzzled: If you mean SetCallerPres (== Setting From: to "Unknown"), that doesn't work ...
17:50.43bkruseflorz: what do you want to do now?
17:50.46florzbkruse: Hmm? What do you mean?
17:50.48bkruseblock people with certain caller ID's?
17:51.03puzzledflorz: yeah there was something with that. have you tried the 0x... values?
17:51.30florzbkruse: I want to tell a SIP -> PSTN gate that it should block CID presentation to the callee
17:51.34*** join/#asterisk So3kris (n=jan-will@217.170.33.70)
17:52.12florzpuzzled: Erm, no!? It does set the From header, though, the PSTN gate just doesn't care ...
17:53.09bkruseflorz: easy gotoif(${CALLERID(num)} == "2564233142"? badpersoncontext : goodpersoncontext
17:53.19bkrusewhoops missed my ) at the end, my apologies
17:53.32puzzledbkruse: he means outgoing, not incoming
17:54.04bkruseoh, i think the same principal can be applied
17:54.06*** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no)
17:55.30bkrusejust look at the dialplan variables, and you can use gotoif's
17:55.48puzzledflorz: do you have "secallingpres=yes" in zapata.conf (found that on voip-info.org)
17:56.14florzpuzzled: ?! - how is that supposed to help with SIP?!
17:56.28puzzledflorz: don't ask me about the logic the developers used :)
17:57.15florzpuzzled: You got some URL?
17:57.30puzzledhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetCallerPres
17:57.33puzzledunder CLIR
17:57.48*** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it)
17:58.20florz; for zap channels I needed to put "usecallingpres=yes" in the zapata.conf to get <- !?!
17:58.33*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
17:58.40puzzledah sorry, missed the "for zap channels"
18:02.04*** join/#asterisk paoleela (n=paolope@217.7.206.10)
18:02.25*** join/#asterisk notoriousrab (n=robert_m@207.47.34.74.static.nextweb.net)
18:06.04saschi recompile my asterisk
18:06.16saschand when i lunch asterisk now return Segmentation fault
18:06.24saschwhy ??
18:06.46mafkeesdid you do: make install while asterisk was still running ?
18:07.17saschyes i run make clean && make && make install
18:07.27*** join/#asterisk psyferre_ (n=psyferre@host-prestigemag-105-10.customer.ntelos.net)
18:07.35mafkeesthen asterisk will die
18:07.39mafkeescan you start it now ?
18:07.40*** join/#asterisk NL5124 (n=BlubBlub@port-87-234-153-49.dynamic.qsc.de)
18:07.45bkrusesasch: type this: rm /usr/lib/asterisk/modules/*
18:07.52bkrusei totally tried to tab complete that.
18:07.57saschok
18:07.58saschi try
18:08.00bkrusethen make install
18:08.02saschone moment
18:08.16bkruseyou dont have to make clean again, or make even
18:08.46*** join/#asterisk infernix (i=nix@spirit.infernix.net)
18:08.46saschnow i run root@centres:/usr/src/asterisk# rm /usr/lib/asterisk/modules/*
18:08.47saschrm: cannot remove `/usr/lib/asterisk/modules/*': No such file or directory
18:08.50saschops
18:09.02saschi post a lines ....... ops :-P
18:09.18bkruseinteresting.....
18:09.21psyferre_hey folks, i hope everyone is well today :)  Would anyone have a moment to help an asterisk n00b?  I registered some new codecs and all seemed to go well, but now asterisk will not start - "Illegal Instruction" right after finding the codec...
18:09.24bkruseok, make install and try to run asterisk
18:09.37*** join/#asterisk bmd (n=bmd@72.54.252.34)
18:09.50bkrusepsyferre_: is it for the right architecture? and  is it for the right ast version?
18:10.03bkrusesasch: try to make install
18:10.08bkruseand tell me the output, if it errors
18:10.10psyferre_bkruse: i think so, definitely the right asterisk version
18:10.14bkrusethen ls /usr/lib/asterisk/modules
18:10.20bkrusepsyferre_: tell me more about the codec
18:10.22bkruseg729?
18:10.25psyferre_yup
18:10.36bkrusewhat machine, and what g729 codec did you get
18:10.40bkruseand what ast version are you running
18:10.53psyferre_one moment while I get all that
18:10.59bkrusei just downloaded the g729 codec 2 days ago, and used it
18:10.59bkrusek
18:11.01bkrusethanks
18:12.21saschok now run all
18:12.24saschthanks
18:12.35*** join/#asterisk infernix (i=nix@spirit.infernix.net)
18:13.06psyferre_running on an older dell poweredge 500sc with a celeron processor, codec g729a (32-bit), 32 bit register utility, and asterisk 1.2.7.1
18:14.07paoleelaHello. Dial tones from mobile phone sometimes aren't recognized. message log looks like this:
18:14.09paoleelaHuh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)?
18:14.13docelmoQuick question for someone when asterisk says it wants -lssl it wants the libs for openssl right?
18:14.56tzafrirright
18:15.00*** join/#asterisk EmleyMoor (n=phil@topdeck.tinsleyviaduct.com)
18:15.02tzafrir-lssl -lcrypto
18:15.02paoleelaIt's because I set up a callback and have to dial the number after being called by asterisk.
18:15.17psyferre_when I try to run asterisk -rvvv I get an error that it can't connect and I should check to make sure /var/run/asterisk/asterisk.ctl exists.  It does.  when I run asterisk -vvv it gives illegal instruction right after "Found total of 5 g.729 licenses"
18:15.39EmleyMoorI seem to have a high echo on calls made on IP phones via my FXO port - what should I check to fix this?
18:15.41psyferre_sorry, all that to bkruse
18:15.45docelmoWhat lib is crypto
18:15.51EmleyMoorIs there an echo test on a UK PSTN number?
18:16.04tzafrirpsyferre_, what is your cpu?
18:16.41psyferre_i know it's a celeron, let me see if i can find out what speed
18:16.52tzafrircat /proc/cpuinfo
18:17.33psyferre_tzafrir: thanks, that's a fantastic command to know - 1097.113
18:17.38mafkeescat: /proc/cpuinfo: No such file or directory
18:18.22*** join/#asterisk infernix (i=nix@spirit.infernix.net)
18:18.33[TK]D-FenderEmleyMoor: What card?
18:19.07EmleyMoorTDM400P
18:19.13ChicagoBudgood morning.  can I have an extension like 7750 and 77XX in my dial plan, where a number like 7724 would match 77XX?  Dialing 7750 is fine but as soon as I hit 772 I get an invalid extension.
18:19.31ChicagoBudwhat am I missing
18:19.35[TK]D-Fenderpaoleela: You didn't set the appropriate DTMFMODE for your channel.
18:19.56[TK]D-FenderEmleyMoor: Got "echocancel=yes"?  how about "echotraining=800"?
18:20.09*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
18:20.32[TK]D-FenderChicagoBud: Yes you can, and you have probably made a minor & silly goof
18:20.54ChicagoBud[TK]D-Fender, yeah I knew that
18:20.55paoleela[TK]D-Fender: What should be the right mode for dial tones? I mean, sometimes it works, sometimes not.
18:21.09EmleyMoorGot echotraining=400 - you recommend I try 800?
18:21.18docelmothanks tzafrir I got it.  damn ldconfig
18:21.19*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)
18:22.01ChicagoBudI see 77XX added as an exten to my context when I reload
18:22.22*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
18:23.07psyferre_bkruse, tzafrir: any idea?
18:23.13paoleela[TK]D-Fender: It's set to dtmfmode=rfc2833
18:23.18tzafrirEchotraining is not much of a difference
18:23.44*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
18:23.45tzafrirpsyferre_, file /usr/lib/asterisk/modules/*
18:23.54paoleela[TK]D-Fender: But I also tried the other modes.
18:24.03tzafrirhmm... ignore that
18:24.22psyferre_okay
18:24.26tzafrirpsyferre_, chances are you have a faulty module there
18:24.38tzafrirunless you get segfaults with other operations
18:25.07psyferre_i haven't seen any other problems, though that's the only thing we're using this box for...
18:26.11*** join/#asterisk gatuno (n=gatuno@145.red-82-158-215.user.auna.net)
18:26.42psyferre_how can i check to see if a faulty module is the problem?
18:26.50EmleyMoorCan time ranges on GotoIfTime span overnight?
18:26.52psyferre_redownload the g279a.so file?
18:27.56wunderkinis echotraining used on a pri?
18:28.26[TK]D-FenderEmleyMoor: I believe so, but if won't cross into the next DAY.
18:28.44psyferre_tzafrir:  I'm guessing rm /.../asterisk/modules/codecg729a.so and then do the register process over again
18:28.53[TK]D-FenderEmleyMoor: what version of Zaptel are you using?
18:29.07[TK]D-FenderEmleyMoor: You may want to consider upgrading.
18:29.18EmleyMoor[TK]D-Fender: Your answer appears to be "yes and no" - so I do need to stick with overnight in two parts
18:29.25[TK]D-FenderChicagoBud: paste your dialplan line that isn't working.
18:29.57EmleyMoor[TK]D-Fender: 1.2.11
18:30.09[TK]D-FenderEmleyMoor: Try an upgrade
18:30.25ChicagoBud[TK]D-Fender, just got it.  _77XX not 77XX
18:30.38EmleyMoor[TK]D-Fender: That isn't easy
18:31.03bkruse[TK]D-Fender > all
18:31.05bkrusejbot: [TK]D-Fender++
18:31.52ChicagoBudextension.conf syntax is awful
18:32.08EmleyMoorChicagoBud: It's not bad when you get used to it
18:32.57ChicagoBudEmleyMoor, just seems it could have been better - more like a language we've seen before
18:33.37EmleyMoorI've managed to do away with numbered priorities, except of course 1.
18:33.45psyferre_just to be sure i'm on the right track... celeron 1Ghz would be i386 architecture for the G729 codec download?
18:34.43ChicagoBudoh well, it is what is it.  I think the openpbx guys are doing something different but it remains to be seen if they become mainstream
18:36.02EmleyMoorI'm just trying to see if I can implement anything in macros - but my peculiar incoming number requirements make it difficult
18:37.27ChicagoBudEmleyMoor, I try to use macros too
18:38.32EmleyMoorI managed to get my voicemail checking line down to just 10 priorities last night
18:42.03*** join/#asterisk CJLinst (n=CJLinst@209-221-212-010.qnet.com)
18:42.32*** join/#asterisk funxion (n=nunya@63.214.236.169)
18:42.52CJLinstIs there a specific trick to getting multiple sip registrations working from an snom to a single asterisk?
18:43.30EmleyMoorCJLinst: Each registration probably needs to be a different peer
18:43.38funxionCan anyone tell me whether its possible to get wmi working using realtime I have read articles stating yes and no,
18:43.55funxionI have rtc enabled in sip.conf
18:43.59funxionit still doesnt werk
18:43.59CJLinstThat's how I have it.  But only one of the sip peers can make calls, authentication fails on the other one.
18:44.11psyferre_tzafrir, bkruse: it was a corrupted or incorrect module, after deleting it and redownloading the i686 version instead of using the generic "32-bit" from the readme file it all worked like a charm.  Thanks for your help!
18:44.20CJLinstchan_sip.c:8065 check_auth: username mismatch, have <246>, digest has <226>
18:44.23*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
18:44.58bkrusepsyferre_: sorry, i shoulda caught you and said use i686
18:45.07bkruseits pretty generic in terms of which one to use :]
18:45.09bkrusegl!
18:45.22CJLinstCalls from other extensions go to the correct Identity button
18:45.33psyferre_bkruse: no biggie :)  I needed to look it up anyways to get the full deal :)  Thanks again!
18:45.38psyferre_Cheers!
18:45.59saschto activate DND in my grandtsream ??
18:46.19[TK]D-FenderEmleyMoor: How is it not easy?
18:46.24[TK]D-Fenderbkruse: y0
18:46.37*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
18:46.40bkruse[TK]D-Fender: wut up :]
18:47.05EmleyMoor[TK]D-Fender: I lose package management if I go to a later version
18:47.22[TK]D-FenderEmleyMoor: You say that.... as though it were a BAD thing ;)
18:47.29*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net)
18:48.08[TK]D-Fenderbkruse: Work, the usual.... Brought all my blades in to work as I'm going stright to martial arts after.  Had some fun showing my katana to our engineers :)
18:49.34*** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-145-140.ny325.east.verizon.net)
18:50.16funxioncan someone help me with WMI and realtime?
18:51.21*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)
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18:52.58[Airwolf]funxion, what seems to be the problem ?
18:53.20funxionI setup realtime sip exten and voicemail all seems to be working fine with teh exception of wmi
18:53.44funxionI have the rtccacchefriends or whatever set to yes in sip.conf
18:54.03[Airwolf]Can you pastebin your sip config and an sql query of one of your users ?
18:54.05funxionit will do everything but sen notification
18:54.32funxionone sec
18:54.43psyferre_just another quick question folks :)  After installing the G729 codec is there anything else to do?  If I purchased a 5 channel license, will it just support that codec on 5 simultaneous connection automatically, or do i ahve to pick which 5 extensions and tell them to use that codec manually?
18:55.02Qwell[]psyferre_: it "just works"
18:55.08[Airwolf]psyferre_, it wil do that automaticly
18:55.08mafkeesQwell[]: !
18:55.13Qwell[]mafkees: I didn't do it
18:55.14mafkeescongrats with commit 55555
18:55.16psyferre_lol
18:55.17Qwell[]thanks :D
18:55.25funxion[Airwolf] is there a database field in realtime that turns mwi on and off?
18:55.36Qwell[]mafkees: I was TRYING to find a typo to fix, but I found a real bug instead. :P
18:55.42psyferre_thanks, i really appreciate your help and patience.  have a good one all :)
18:55.47mafkeeslol Qwell[]
18:55.53mafkeessometimes luck is just with you
18:55.55psyferre_*buys you all a beer*
18:56.00[Airwolf]funxion, not really. But you have to make sure everything is filled in correctly
18:56.16*** part/#asterisk psyferre_ (n=psyferre@host-prestigemag-105-10.customer.ntelos.net)
18:56.16[Airwolf]Such as your mailbox number and voicemail context.
18:56.23funxionthose are
18:56.33funxionI can call and leave a message
18:56.36[Airwolf]euh one moment
18:56.39funxionand can check messages
18:56.50funxionjust no message waiting indicator
18:57.53[Airwolf]the mailbox field should be  <number>@<context>  (the last one if it's not default)
18:58.24funxionyes
18:58.27funxionit is
18:59.10*** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net)
18:59.13*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
18:59.28funxionhmm
18:59.32funxion@context?
18:59.37funxioni have @voicemail
18:59.44[Airwolf]that is incorrect.
18:59.55funxionnow it werx
18:59.57funxionthnx
19:00.00[Airwolf]np :)
19:00.16funxionbtw do you have any experience with voicemail odbc storage?
19:03.42*** join/#asterisk champster (n=asterisk@AH.tescogroup.com)
19:04.07*** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-145-140.ny325.east.verizon.net)
19:06.46EmleyMoorWhat happens when a macro exits?
19:07.38mmlj4isn't a macro like a function?
19:08.05[TK]D-FenderEmleyMoor: It returns to the point from which it is called and resumes.
19:08.15froguzit's like a C function
19:08.21froguza sub-rutine
19:09.38*** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net)
19:10.32*** join/#asterisk mega (n=mega@217.201.163.156)
19:11.44CJLinstCan anyone explain exactly what's happening when I get this: chan_sip.c:8065 check_auth: username mismatch, have <246>, digest has <226>
19:13.01[TK]D-FenderCJLinst: Check your phone.
19:13.25CJLinstThanks, but what is happening?
19:14.04mafkeesdigest and username does not match
19:14.21CJLinstAnd the digest refers to?
19:16.33[TK]D-FenderCJLinst: Your settings are hard to swallow ;)
19:17.13[TK]D-FenderCJLinst: Just go look at your phones setup in extreme details and you're likely to see something thats "off"
19:17.21*** join/#asterisk giasai68 (n=administ@ip-3-145.sn2.eutelia.it)
19:17.29*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)
19:17.54CJLinstWhat I'm trying to do is get two Line buttons on an Snom 300 working as two different SIP peers.  It's working great for calls to the phone, but outbound calls into the context are only working for onw of them.  The other gets the authentication failure.
19:21.53*** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net)
19:22.43*** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net)
19:23.22GaVakI'm having problems with a PRI turn up... the provider says that they see the d-channel come up... then drop after I reload *.
19:23.27GaVakI'm getting this error: 3
19:23.27GaVakWrite to 44 failed: Unknown error 500
19:23.27GaVakShort write: 0/5 (Unknown error 500)
19:23.55GaVakis there a command that will show the status of the PRI link?
19:24.06GaVakzap show status only says 'OK'
19:24.27wunderkinpri show span x
19:26.19EmleyMoorI've now got a macro to do my colourlist checking
19:27.17*** join/#asterisk jm|laptop (n=jm@zen.jamiem.com)
19:27.33GaVakMeh, it looks like their DSL dropped. Well, thanks for the command wunderkin.
19:28.30[TK]D-FenderDSL?!
19:29.26wunderkinapparantly short loops are frequently done over hdsl
19:29.30tzangeruh
19:29.37tzangerall T1s are HDSL2 loops these days
19:29.47tzanger5 years ago they were HDSL loops
19:30.13tzangerthe real physical T1 as a long-haul transport doesn't exist anymroe
19:31.16tzangerit doesn't terminate to a DSLAM like consumer-grade DSL but the technology (coding) is all the same
19:31.51tzangerthey can easily tweak the tx/rx buckets for the DMT transport in DSL to get you more than 800kbps uplink but they won't do it, even as VDSL is rolling out
19:32.19cpmtzanger, are you sure?
19:32.28tzangercpm: about what?
19:32.50cpmrealy physical T1 as a long-haul transport doesn't exist anymore
19:33.17tzangercpm: yep, we had a 40km PTP T1 10 years ago that was actually a DS1-over-HDSL
19:33.35tzanger(two powered loops, but not T1 as in electrial+logical spec)
19:33.57cpmerrr, why did I think that ds1 over hdsl , , ,ah, okay.
19:33.57tzangerour PRI upstairs is an HDSL2 (1 loop) DS1-over-HDSL2
19:34.02ManxPowerIt's not suprizing.  Real T-1 needed a repeater every 1,800 feet, modern "T-1s" do not.
19:34.18tzangermodern T1s are every 5000ft IIRC
19:34.28tzangerI think that's the distance, I can't remember now
19:34.30ManxPowercpm: shine a flashlight thru the smokey window on the telco T-1 box at work.
19:34.42tzangerManxPower: don't do that
19:34.46tzangeryou'll startle the hamster
19:35.06cpmI just use my key, all T1 smartcard cages use the same key I think
19:35.13ManxPowerYou'll see Pairgain or Paradine or Westel card with some form of DSL on the label
19:35.17cpmyup
19:35.20tzangeryup
19:35.25tzangerHRU1 in my case :-)
19:35.32ManxPowerWell if you have a key why are you asking these silly questions? 8-)
19:36.02cpmall this time, and I never made the hdsl/dsl connection at all
19:36.17cpmalways considered dsl a dry line alarm circuit or something
19:36.33tzangernope
19:36.42tzangerDSL runs over powered circuits in most cases
19:36.47tzangerSDSL is typically dry copper though
19:36.51tzangerI set up a number of those over the years
19:37.05EmleyMoorCan one macro call another? If so, what happens to the MACRO_ variables?
19:37.19tzafrir_laptophmmm... any idea why suddenly asterisk does not respect "secret" in iax.conf? (asterisk 1.2.13)?  Whatever I put in "secret" I get an "empty secret" in 'iaxs show users" with Authen value of 3
19:37.19cpmlike the onces covering rotary switch and crossbar
19:37.56*** join/#asterisk s1gny|wrk (n=s1gny@p54915087.dip.t-dialin.net)
19:38.07ManxPowerEmleyMoor: 1: Yes.  2: Try it
19:38.28*** part/#asterisk s1gny|wrk (n=s1gny@p54915087.dip.t-dialin.net)
19:38.42tzafrir_laptopoops, was running it from a wrong set of config
19:40.14tzangerhmm I could have sworn that ssh IP packets had low-latency bit set
19:42.45Juppersanyone familiar with using FXS ports on a cisco 2610 with asterisk? How do you register them? The 2610 doesn't support sip authentication that I can find.
19:43.01*** join/#asterisk [Mr_X] (i=1000@88.118.97.205)
19:43.05florztzanger: For interactive sessions and openssh this is usually true, yes.
19:43.08mercesteswhy would you want to register SIP over an FXS port?
19:43.17tzangerflorz: that's what I'm staring at right now in ethereal
19:43.22*** join/#asterisk bmd (n=bmd@72.54.252.34)
19:43.23tzangerflags of 0x10
19:43.27mercesteswhy would you want to register an FXS port for that matter?
19:43.31ManxPowerJuppers: it does not support any kind of sip authentication, you need to use IP based authen when getting calls into asterisk from a cisco like that
19:43.55tzanger0x10 = D bit set, minimize delay... hmm
19:44.04Juppersmy asterisk admin says it has to register so it can be in the same context as every other phone in our system
19:44.15ManxPowertzanger: I think you want 0xb8
19:44.20tzangerflorz: ethereal seems to misrepresent the bits
19:44.33ManxPowerJuppers: your admin is an idiot
19:44.41tzangerit calls the precedence "Differentiated Services"
19:44.56mercestesJuppers:  seconded.  Fire his stupid ass and hire a *real* asterisk admin
19:44.59ManxPowerJuppers: you can out it in whatever context you want by using permit/deny
19:45.01JuppersManxPower - he is new to asterisk. we are trying to move away from CCM.
19:45.07florztzanger: Well, that depends on which of the interpretations you choose ...
19:45.21ManxPowertzanger: diffserv is a more modern QoS method that builds on ToS
19:45.33tzangerManxPower: ahh
19:45.35mercestesThen he's not an asterisk admin.  he's an asterisk acolyte.  not even an acolyte really...more an apprentice to an acolyte.
19:45.42tzangerheh
19:45.56florztzanger: DS is probably state of the art. But in the end, what counts is how the routers that are to use the info so interpret it ...
19:46.03ManxPower0xB8 SEEMS to be DiffServ "EF" and ToS Low Latency/High Priority/whatever
19:46.06mercestesjuppers:  you jsut set the context in zapata.conf for the fxs interface.
19:46.13florztzanger: s/so/do/
19:46.14tzangerI'm the receptionist's associate for the apprentice for the acolyte, studying under the great guru of all things Asterisk, ManxPower
19:46.19ManxPowermercestes: the port is on a cisco
19:46.27JuppersI'm running a 2610 with fxo and fxs so I can bring my home lines and work lines together and have my own dialplan that doesn't include 9
19:46.42ManxPowerJuppers: you need to be on a static IP
19:46.51JuppersManxPower - I am on static
19:47.27ManxPowerJuppers: other than that on the asterisk side [yourhomecisco] permit=yourip/255.255.255.255 deny=0.0.0.0/255.255.255.255 host=yourip the rest of the settings
19:47.43ManxPowerJuppers: this is off the top of my head, check the sample configs that come with Asterisk
19:48.15ManxPowerJuppers: that will make calls from your cisco match [yourcomecisco] and the context= line you have in that section
19:48.24Juppersso it should be set up in zapata.conf and not sip.conf
19:48.33ManxPowerJuppers: no, in SIP since it is SIP
19:48.56ManxPowermercestes was not paying attention to your odd and terribly painful to thinkabout setup.
19:49.06Juppersso in sip and type peer with specific IP rules
19:49.07ManxPowerI'd just use a SIPura box to do the same thing
19:49.11*** join/#asterisk hellojoe (n=hellojoe@natint3.juniper.net)
19:49.20ManxPowerJuppers: I would make it a type=friend in this case.
19:49.41mercestesManxPower:  probably.  In every scenario I've seen fxs != sip.  =/
19:50.46ManxPowerWhen I first started with Asterisk oh so many years ago I started out with FXO/FXS on a Cisco 1750 as my PSTN gateway device.
19:50.56Jupperswhat he tells my is in the default context he has to append my ip to get calls routed to me. and somehow that messes up things when trying to log me into the call queue.
19:51.09tzangerhmm, this filter should be catching it then
19:51.13ManxPowerJuppers: he is confuzed.
19:51.21tzangerethereal's seeing flags 0x10 and my tc filter is ip tos 0x10 0xff
19:51.32hellojoeHi folks, is it true that Jajah is using Asterisk?
19:51.46ManxPowerJuppers: "context" is the wrong word for sections in sip.conf unless he is talking about context]
19:51.53JuppersManxPower - thanks a bunch. I'm trying to get our current config files so I can run a test box here and see if I run into the same as he is saying
19:52.22JuppersManxPower - he is talking about the context sections in extensions.ael
19:53.13Juppersthat instead of adressing me as Dial(SIP/224) he has to do Dial(SIP/224@xxx.xx.xxx.xxx)
19:53.55ManxPowerJuppers: then he is REALLY confused.  If your sip.conf is set up correctly then the call will go into the correct extensions.conf/extension.ael context with the correct destiantion number just like a real SIP phone.
19:54.07ManxPowerJuppers: your set up is not very common, but should work.
19:54.31ManxPowerJuppers: Nope!  That is why you have a [244] section in sip.conf for your router with host=yourip
19:54.57Juppersok.. so the whole thing is he has the sip.conf wrong for me. Thanks a bunch again
19:55.41ManxPowerJuppers: He's going to have massive problems if he keeps thinking that an extension is a device.  We actually use the MAC of the device as the SIP account ID
19:55.57[TK]D-FenderJuppers: if he's doing "Dial(SIP/224@xxx.xx.xxx.xxx)" then you are not authing at all.
19:56.06EmleyMoorDamn - 1 macro calling another sets the extension to s
19:56.10ManxPowerSince it is easy for 4 different devices to have the same extension in a real corporate enviroment with whiney users that don't want to hear "no"
19:56.16*** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no)
19:56.41ManxPowerEmleyMoor: Set(SAVED_MACRO_EXTEN=${MACRO_EXTEN} before calling the 2nd macro
19:56.42[TK]D-FenderEmleyMoor: this is where you get to take a minute to think about the magic of PARAMETERS.
19:57.03[TK]D-FenderMACRO_EXTEN = waste.  Just pass the value as a parameter
19:57.18ManxPower[TK]D-Fender: you mean the magic of if you use ARG1, ARG2, and ARG3 when calling the first macro and then when you call the end macro with ARG1 and ARG2, then ARG3 is STILL set?
19:57.26funxionanyone know why if I connect to voicemail then hangup asterisk restarts?
19:57.32ManxPowerend == second
19:57.51ManxPowerfunxion: you are running 1.4
19:57.54funxionno
19:57.56funxion1-2
19:58.07[TK]D-FenderManxPower: Should "stack", shouldn't it?
19:58.09ManxPowerfunxion: never seen that problem ever
19:58.15ManxPower[TK]D-Fender: it doesn't.
19:58.23funxionlast message in console is "Extension 9999, priority 1 returned normally even though call was hung up
19:58.23funxiondebast-vm*CLI>
19:58.23funxionDisconnected from Asterisk server
19:58.34[TK]D-FenderManxPower: Silly.  MACRO_EXTEN should be just as vulnerable then
19:58.55ManxPowerall args set when calling a macro stay set when you call any other macro from within the first macro unless you overrite them
19:59.22ManxPowerfunxion: hanging up a call IS normal
19:59.34funxionyes but then it restart asterisk
19:59.35funxionhmm
19:59.51EmleyMoorI knew ARG1 could be used to fix it - just wasn't sure until just now how to pass it with ExecIf
19:59.52[TK]D-FenderManxPower: Ok, and does the nested on get ITS ARG1 where appropriate?  And upon return, is the original valure reinstated?
20:00.15ManxPower[TK]D-Fender: I don't know.
20:00.18[TK]D-FenderManxPower: (I will of course test this when I get home)
20:00.48EmleyMoorAll my permission checking is now done by macros :-)
20:00.53[TK]D-FenderManxPower: Now it #1 used ARG3, and #2 doesn't, SURE, #2 could access ARG3, but it WOULDN'T (by any sane coder)
20:01.11[TK]D-Fenders/it/if
20:01.17EmleyMoor(except checking of whitelist... that is done to bypass their calling)
20:01.32ManxPowerMacro(happy-1,FRED, BARNY, WILMA) calls Macro(happy-2,FRED) then in [macro-happy-2] ARG2 is BARNY and ARG3 is WILMA, whereas I think they should not exist or be empty
20:01.37[TK]D-FenderEmleyMoor: All of your dialout's should be macro's as well
20:02.09hellojoeGuys, can anyone point me to AGI script that works like Jajah? I need this for a group of folks I am working with. Any pointers wouldl be appreciated.
20:02.22EmleyMoor[TK]D-Fender: I think some of them are a little too complicated to be macros
20:02.41ManxPower[TK]D-Fender: I got stung by this when I had a macro call itself with a different number of params and the macro did different things depending on the number of params
20:02.43hellojoeBy the way, it doesn't have to be AGI. I have tried this with .call files too (not very reliable though)
20:03.03hellojoeany suggestions (AGI Vs .call?)
20:03.07ManxPowerhellojoe: Does anyone here even know what the heck JaJah is?
20:03.08[TK]D-FenderManxPower: as long as happ-2's ARG1 takes precedence and gets set back upon completion, who cares?  Why would you reference a parameter not provided from within the macro?  That'd be sloppy coding
20:03.15hellojoehuh!
20:03.16*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
20:03.17hellojoejajah.com
20:03.39ManxPower[TK]D-Fender: because I don't know how many options a macro was called with.
20:04.02hellojoebasically, enter two telephone number on an online portal and then connect the two parties
20:04.05hellojoethat's all I want
20:04.10[TK]D-FenderManxPower: I guess if you're making variable parameter macro's,but thats a strange animal....
20:04.14*** join/#asterisk x86_ (n=x86@p3m/member/x86)
20:04.22Juggiethen you shuold include a flag to tell the macro what it should expect
20:04.25hellojoeso a script to which I can feed 2 telephone numbers and expect it to connect
20:04.35ManxPower[TK]D-Fender: it was elegant and simple -- or would have been
20:04.35hellojoethe parties using .call or AGI
20:04.43Juggieso Macro(type of action,var,var,var)
20:04.53ManxPowerhellojoe: did you check the mailing lists or the wiki?
20:05.03hellojoeyes.. no luck :-(
20:05.06[TK]D-FenderManxPower: http://www.sofaswitch.org/d/
20:05.08ManxPowerJuggie: I coded around the issue
20:05.21*** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net)
20:05.28*** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar)
20:05.30Juggieits not so much an issue as it is a pitfall :)
20:05.37[TK]D-FenderManxPower: Increasingly imperfect solutions for an imperfect world ;)
20:05.58[TK]D-FenderJuggie: Pitfalls wouldn't be so bad... if not for the giant spikes at the bottom ;)
20:06.14BrianR___When I call Dial() from a zap channel, is the PRI hangup cause passed back to the calling channel?
20:06.28Juggieyes.
20:06.35Juggiewell, the hangupcause
20:06.37L|NUXis there any one who do have global crossing dids
20:07.14BrianR___Ie, if I dial a number that would result in a hangup with a "This number is not inservice..." recording, does the magic #1 code pass through automagically, or do I need to do some magic like Hangup(PRI_CAUSE) or something?
20:07.36Juggieyou mean to pass it back to the dialing channel?
20:07.51Juggieso the phone will play the proper error?
20:08.35BrianR___Juggie: Yeah. Let's take the following hypothetical: I have an asterisk box with two T1 linecards, one connected to the PSTN and the other to a second asterisk PBX. I want the second asterisk PBX to be able to see the hangup cause, but all of its calls are going via the other asterisk PBX (over T1)
20:09.04ManxPowerBrian What signaling between the two asterisk's?
20:09.16BrianR___is it enough to have in the [fromotherasterisk] context a rule like _X.,1,Dial(Zap/2/${EXTEN})
20:09.20BrianR___ManxPower: PRI signalling over T1
20:09.23Juggieso far as my expierence you need to forcefully pass it
20:09.26*** join/#asterisk thekidrio (n=thekidri@66.107.42.13)
20:09.29ManxPowerbrian: it should pass it back
20:09.57ManxPowerthen you need to decide what to do based on the value of hangupcause on the originating box
20:10.14BrianR___Juggie: Do the HANGUP_REASON, PRI_HANGUP, and Hangup(ARG1) use the same set of magic numbers?
20:10.17Juggiei have something like this, http://pastebin.ca/365231
20:11.03ManxPowerBrianR___: Hangup(number) only works in 1.2 AFIK
20:11.11ManxPower..er... ONLY in 1.4
20:11.20*** join/#asterisk apardo (n=apardo@87.217.144.34)
20:11.29BrianR___The local CO plays recordings for some of the messages - I'm not sure how to detect if the CO has a recording or  I need to play my own also...
20:11.36ManxPowerBrianR___: all of asterisk's hangupcauses are based on Q.931 hangup causes
20:11.37Juggiethat macro should probally be expanded to include other dialstatus's but its what i use for sip->pstn
20:11.53ManxPowerBrianR___: always assume you have to play your own on PRI
20:12.24BrianR___The current system is a rather ancient 1.0.6... Without any special handling, it plays a reorder tone for all of the error conditions.
20:13.03BrianR___I'm testing with a 1.2.12 system, trying to figure out if it needs any hacking or not.
20:13.12ManxPowerBrianR___: yes, it would as you are supposed to handle your own audio messages when dealing with OOB signaling like SIP or PRI
20:13.44*** join/#asterisk Vec (n=Vector@dsl-243-93-201.telkomadsl.co.za)
20:13.46*** join/#asterisk orcimrepus (n=orcimrep@74-130-224-149.dhcp.insightbb.com)
20:13.54BrianR___ManxPower: How does that apply to things like SIP phones, which could potentially receive a hangupcause?
20:14.23ManxPowerBrianR___: the phone won't play "The number you have dialed, 555-1212 has been disconnected".
20:14.58*** join/#asterisk swamig007 (i=boom@59.92.194.37)
20:15.04ManxPowerIf you don't do anything the translated cause code from Q.931 should be translated into whatever the equiv SIP code is and the phone would prolly give a reorder/congestion tone
20:15.14BrianR___Looks like the polycom phones I have just play the reorder tone for all non-zero hangup causes.
20:15.29ManxPowerBrianR___: I'll bet most phones would do that
20:15.38*** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93)
20:15.41*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
20:16.12[[blah]asfdhas anyone experienced static on the call when no one is speaking... and then when you speak, it goes away, then comes back when you stop speaking? is there a way to correct this?
20:16.24ManxPower[[blah]asfd: no.
20:16.28[[blah]asfdphones are connected to another server via SIP
20:16.40[[blah]asfdthe server that it is connected to is connected to pri t1s.
20:16.42PaulTech85Is there anyway to get a trigger when a call goes connected in the queue to call a command
20:16.46ManxPower[[blah]asfd: the phones might have VAD/CNG enabled that might cause such a problem I guess.
20:16.49[[blah]asfdbut the phones on that system do not have this.
20:16.49PaulTech85I want to send info to the agent that answered the phone
20:17.02BrianR___ManxPower: I suppose it may be enough to just make a dial macro that jumps to Busy 41 and lets the phone play reorder for everything else...
20:17.08[[blah]asfdthey are the polycom spountpoint ip 301
20:17.13[[blah]asfdwhere would i find that setting?
20:17.14ManxPowerBrianR___: standby
20:17.16PaulTech85I think mysql is the only way, constantly monitoring the queue_log
20:17.26ManxPower[[blah]asfd: it is a setting ON THE PHONE
20:18.11[[blah]asfdlooking... you mean in the menu and screen on the phone as opposed to the sip.cfg or mac.cfg right?
20:18.30ManxPowerBrianR___: some of this is 1.2 specific http://www.fnords.org/~eric/macros.inc
20:18.44ManxPower[[blah]asfd: what brand of phone?
20:18.56ManxPower[[blah]asfd: ah.  VAD/CND is NOT on by default
20:19.17ManxPower[[blah]asfd: no, I mean it is set in the phone configuration, not the asterisk configuration
20:19.32BrianR___Ok.. My 1.0.6 system is also patched with bristuff, which might explain why it allows hangupcause to be passed as an argument to hangup()
20:20.10BrianR___Here, SetVar(PRI_CAUSE=1) and Hangup(1) produce the same result.
20:20.56*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.15 (Feb. 9, 2007) Zaptel 1.2.14 (Feb. 19, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=-
20:21.03*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.15 (Feb. 9, 2007), Zaptel 1.2.14 (Feb. 19, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=-
20:21.36*** join/#asterisk ez` (n=ez@c207.134.229-230.clta.globetrotter.net)
20:21.36[TK]D-Fenderyay, new zaptel.....
20:21.52BrianR___Did the zaptel echo canceller ever get fixed?
20:22.02[TK]D-FenderBrianR___: Which? :)
20:22.02ManxPowerBTW, for that pastebin the macro-dial-result is what you want to look at
20:23.00BrianR___[TK]D-Fender: Uh...
20:23.19BrianR___[TK]D-Fender: I finally broke down and bought a Sangoma A104D card with the Octasic echo canceller thing...
20:23.38[TK]D-FenderBrianR___: You say that... as though it were a BAD thing ;)
20:23.56BrianR___[TK]D-Fender: In testing so far, it stops echo dead silent...
20:24.15[TK]D-FenderBrianR___: Ain't it grand?
20:24.39BrianR___[TK]D-Fender: Also does the HDLC in hardware, which is nice. This Dell 1850 has only two PCI slots and the A IRQ wire in both of them is shared with other devices...
20:25.14[TK]D-FenderBrianR___: I've well versed with that card....
20:25.18[TK]D-FenderI'm*
20:25.40[TK]D-Fenderrussellb: Which of thhose 3 entries in the changelog was key to such an early release?
20:25.56*** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net)
20:26.05*** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no)
20:26.16russellb[TK]D-Fender: the driver fix for the TDM800P
20:26.19BrianR___[TK]D-Fender: Right now I have it mounted in a seperate PC and I'm transferring calls through it to test the echo canceller. Going to move it to a passthrough role tomorrow to stave off echo complaints until I have a chance to build up and test a new PBX around the A104D...
20:26.35BrianR___[TK]D-Fender: Thus all of the questions about hangupcause :)
20:26.47[TK]D-Fenderrussellb: I sort of figured that...
20:27.26russellbBrianR___: interrupt sharing should no longer be a problem.  If you have any problems related to that after zaptel 1.2.13, let us know ...
20:27.42wunderkino rly?
20:27.47russellbyes
20:27.51wunderkinzomg
20:27.57wunderkin:d
20:28.19russellband also, the digium quadspan and dualspan cards do hdlc in hardware, as well.
20:28.54mercestesnummeh
20:29.28BrianR___russellb: Hmm.. Is that dependant on the zaptel driver version?
20:29.42russellbas long as it is anything not ancient ...
20:29.58*** join/#asterisk tdi (n=tdi@seth.kill-9.pl)
20:30.13tdihi, is this a channel to ask for t38 software?
20:30.20russellbo.O
20:30.32[TK]D-Fenderrussellb:  Would that be defined as pre /topic? ;)
20:30.40russellbno
20:30.41[TK]D-Fendertdi: No.
20:30.46russellblike, over a year ago or something
20:30.49BrianR___russellb: Ok. This machine Asterisk 1.0.6 and Zaptel 1.2.0 on it, which would probably make it sufficiently ancient to not work well
20:30.52[TK]D-Fenderrussellb: :O
20:30.55tdi[TK]D-Fender: aha
20:31.06russellbBrianR___: yes, that is pretty old.
20:31.14*** part/#asterisk tdi (n=tdi@seth.kill-9.pl)
20:37.30ChicagoBudI'm about to dive into fax to mail via a sangoma a200d -- any advice -- looks like there are a bunch of ways to go...
20:39.49n|cotineQuestion on app_queue - why does the queue ring a dyanmic agent who is already on a queue call?
20:40.00n|cotineshow queues even shows that the queue manager knows that the agent is (In use)
20:40.26mercestesn|cotine:  I like that question.
20:40.50mercestesn|cotine:  I would add to yoru question with, "why does it ring a static agent when *they* are on the phone?"
20:40.58PaulTech85n|cotine Yes it still rings a In Use agent
20:41.04PaulTech85Same for static
20:41.05n|cotinePaulTech85:  This seems undesired.
20:41.12n|cotineIs there an easy fix in the source?
20:41.12mercestesPaulTech85:  We agree.   how to fix?
20:41.30*** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer)
20:41.32PaulTech85I wrote for a patch for mine but many people like the use of it
20:41.40n|cotinePaulTech85:  Available anywhere?
20:42.15PaulTech85It was done privately but I dont see why I cant release, What branchs are you guys patching against?
20:42.20n|cotine1.2.15
20:42.23mercestes1.2.13 here
20:42.33n|cotineI can hand-apply if it's not too bad, though.
20:42.40n|cotineI can read C, just not write it. :)
20:42.47*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)
20:42.59PaulTech85let me look over it, Mine was patched against 1.4 I dont think much changed
20:43.31n|cotinePaulTech85:  wouldn't it be in ring_entry - just check call status when you're checking wrapup time?
20:43.57PaulTech85Yep, same place
20:44.43PaulTech85When it checks 'ringinuse' against parent
20:45.06`SauronDoes anyone know offhand where the sound files are stored in openpbx? (I know this is #asterisk, bite me)
20:45.37mercestes`Sauron:  no, you *BITE ME* and go crawling back to those asterisk hating source code stealing CLOWNS that can't fix their own shit.
20:45.41mercestesWTF is wrong with you?
20:45.47`SauronYawn.
20:45.58mercestesgah, your the type to walk into a vegan joint and go "man, where can I get some bacon?"
20:46.00JerJer<PROTECTED>
20:46.27froguzwhat is the "plancomment = " in the extensions.conf for?
20:46.36`SauronJerJer: Thanks. I found that on my asterisk install at home.
20:46.44JerJermercestes:  lets feed the trolls with relevant info for Asterisk   :)
20:46.47`SauronApparently they brainf*ed that part too.
20:47.37`Sauronmercestes: bitter?
20:47.59fetcherIs there a way to get SIP phone presence / BLF working when one of the phones is behind a 2nd Asterisk server?  (linked to 1st via IAX)?
20:48.03Corydon-w`Sauron: off topic, please take it elsewhere
20:48.33hellojoeGuys, anyone has an opinion on .call Vs AGI scripts for connecting parties given their telephone numbers? I have found .call less reliable in the past
20:48.41*** join/#asterisk dasenjo (n=dasenjo@190.24.24.34)
20:48.56hellojoeI used to generated .call files using a custom perl script and dump them on the same extension
20:49.02ChicagoBudShould I be looking at HylaFax or AsterFax or something else?
20:49.21`Sauronhellojoe: make sure you move .call files into the directory, so * detects them properly
20:49.31`Sauroni.e. create them elsewhere, then drop them in
20:49.46froguzAsteriskNOW has put a  "plancomment = DialPlan1" on my extensions.conf and it's not documented anywhere
20:49.52hellojoeYup.. did that. Becomes an issue when there are too many calls
20:49.53fetcherChicagoBud: I use Hylafax, behind an AS5300 w/ t38modem and H.323.  Works OK in the end, but it's a real mess to set up.
20:49.57JerJer`Sauron: we should illistrate your point
20:49.58froguzwhat is that for?
20:50.20hellojoeSauron: Have you tried it in higher call volume scenarios?
20:50.30`SauronJerJer: I use * at home. Have done so for the last 4 years. Where do I have to do the * cheerleader dance for y'all to stop being so bigoted?
20:50.54`SauronJerJer: the day * can originate and terminate faxes, I'll switch the work stuff back to *
20:50.59JerJerwhen dealing with call files one should always MOVE files into the spool directory.  Copying the file ~can~ lead to asterisk grabbing partically done copies
20:51.11ChicagoBudfetcher, Interesting.  All I really need to do is terminte a fax call on a sangoma a200d an convert it to an email
20:51.21mercestes`Sauron:  My * handles faxes just fine.
20:51.28JerJerwhile it is rare, it has happened to me
20:51.31swamig007hi guys i am trying to get a SIP account configured as a trunk (this is an outgoing only account from primus india) i keep getting all circuits are busy ...
20:51.33`Sauronmercestes: origination including termination?
20:51.38froguz`Sauron, then move to * 1.4
20:51.39hellojoeJerJer: yes, it happened to me
20:51.40mercestesyes.
20:51.44`SauronLast I looked at 1.4, they said "use t.38"
20:51.46froguzwith T.38 support
20:51.46fetcherI've used the app_rxtx FAX extension for a single line
20:51.46`SauronI'm at 1.4
20:51.49Mavvie!_(#_!(@_#
20:51.53Mavviefscking IPS.
20:51.56`SauronMavvie
20:51.57mercestesin fact.  I fail to see how switchign from * to * based derivatives "solved your fax problem."
20:52.02hellojoeJerJer: how to move files to the spool directory (Sorry, basic question)
20:52.12MavvieThe Tipping Point is blocking the Asterisk Keep Alive checks for SIP hosts.
20:52.21JerJer`Sauron:  bloody tampons man - you made a good point and I was making the point you made
20:52.23Mavvieseventeen alerts on nagios right now :-)
20:52.32ChicagoBudmercestes, waht are you using for fax?
20:52.32`Sauronmercestes: openpbx put work into both analog fax origination/termination and t.38 origination/termination
20:52.43mercestesopenpbx is a joke.
20:52.46`Sauron* 1.4 can't originate/terminate t.38, nor can it originate analog fax calls
20:53.16JerJeri fax almost every day via asterisk 1.2, TDM Card and a PRI
20:53.16`SauronMavvie: sucks, can you beat them with a black stick?
20:53.31Mavvie`Sauron: nah, will just make an exceptoin for our PABXs.
20:54.05*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)
20:54.08[TK]D-Fender<`Sauron> Does anyone know offhand where the sound files are stored in openpbx? (I know this is #asterisk, bite me) <- as opening lines go, this is NOT a winner in the "how to win friends and influence people" book.  YOU started the trash talk.  Please sopt.  just... STOP.
20:54.08`SauronJerJer: then go flesh out the voip-info.org documentation on it. Last I looked (3-4 weeks ago) everybody was saying "Can't be done"
20:54.15JerJerThen  Hylafax+Asterisk makes a functional solution - I don't like it but it does work - using PRI
20:54.38JerJeryou need the right magic sause that nobody has totally figured out yet
20:54.38`SauronFender, I'm not in it for winning friends. I'm in it for getting shit to work.
20:54.39mercestesJerJer:  Works with iaxmodem as well.
20:54.44[TK]D-FenderSHUP.  Both of you.
20:55.05JerJerthose that do know aren't talking
20:55.10[TK]D-Fender`Sauron: Good.  Now you need to work on getting it to work in a CIVIL manner.
20:55.16mercesteswtf.  I stopped talking five minutes ago...why am I getting lumped in with troll-boy?
20:55.30JerJerbut faxing via g.711 over the net will simply never be reliable
20:55.33funxionanyone know of a reason that asterisk would restart when issueing a odbc show ?
20:55.36`SauronYawn.
20:55.55J4k3or learning how to use the one they have.
20:56.15`SauronJerJer: Seriously. If you can get some sort of docs together on how to both originate and terminate fax calls (to a PRI, I don't care) - I might even buy you a sixpack.
20:56.47JerJeri'll add it to my list of blog/article/whitepaper topics
20:57.03BrianR___russellb: How is the firmware for the digium quad t1 cards handled anyway?
20:57.03[TK]D-FenderStrangly ironic how drinking 6-packs lead to a beer gut.....
20:57.26`SauronAnd how well does MeetMe scale? :)
20:57.33J4k3[TK]D-Fender: sheeit, I got a 30 pack then.
20:57.34J4k3;)
20:57.39JerJer`Sauron: via PRI, very nicely
20:57.42file[TK]D-Fender: crazy circles?
20:57.46*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
20:57.49`SauronJerJer: And over sip trunks?
20:57.51[TK]D-Fender`Sauron:  Not too bad, but it cooks more evenly if you toss it in the pan WHOLE ;)
20:57.52JerJerSIP maybe not so nicely - depends
20:58.20J4k3why would you want to send 'fax tones' over voip
20:58.33`SauronI dunno
20:58.36`SauronI don't. :)
20:58.37J4k3when you can just send 1's and 0's instead, and let the far end deal with the modem tones?
20:58.49Corydon-wJ4k3: because people are cheap bastards
20:59.08JerJeri've had 8 full E-1 spans into one meetme
20:59.23`SauronJerJer: the scaling, is due to the timing stuff?
20:59.34`Sauronscaling issue?
20:59.38BrianR___for fax I've been using hylafax and of those T1 fax cards plugged into my asterisk box's quad pri card with a crossover cable...
20:59.45JerJerone issue in VoIP land is transcoding
21:00.03J4k3Corydon-w: I'd say "shoot 'em all in the face then take their wallets" but I think the wallet thing might be mostly pointless :)
21:00.05JerJerif everyone runs ulaw then its not an issue
21:00.12`SauronBlech, people actually still use hylafax?
21:00.15JerJerbut if everyone wants to run G.729, it can become an issue
21:00.25JerJer`Sauron: many of my customers do
21:00.27BrianR___I also have a few fax machines plugged into ATA's...
21:00.45`SauronLast I looked at it.. it was poo.
21:00.53`Sauronthen again, that might be 4-5 years ago
21:01.04J4k3hylafax has been poo for like >10 years
21:01.04JerJerits not overly friendly
21:01.08BrianR___`Sauron: hylafax is dirt simple faxing that "just works"
21:01.26`Sauronmaybe I'll look at it when I get an extra couple minutes
21:01.30JerJerbut once its setup it  simply works
21:01.39`Sauronlast time, it was unable to do what I needed it to do.. route faxes based on LDAP lookups
21:01.45BrianR___We have DID's mapped to email addresses for incoming fax... And there's a email-to-fax gateway for sending faxes.
21:01.54Mavvie`Sauron: we have hylafax here too. And mgetty+sendfax.
21:02.02BrianR___And printer driver for the windoze PC's that pops up a box asking for phone number and cover page info.
21:02.10`SauronMavvie: I ended up writing some perl super-glue to do the magic for me.
21:02.23Mavvie`Sauron: happily connetced to a Patton Dual E1 card, hooked up to a QUad PRI on the * server.
21:02.30J4k3BrianR___: what you need is a terminal server that can handle fax traffic and NOT VoIP service.
21:02.35J4k3:P
21:02.54J4k3I mean shit, a $20 Livingston/Lucent Portmaster 3 can handle 48 (or 60) fax calls without a problem.
21:03.00BrianR___The hylafax faxdispatch thing is just a shell script anyway....
21:03.01JerJerBrianR___:  we do very a similar operation as well
21:03.06J4k3and yes, you can get them loaded with DSPs for about $20 off ebay.
21:03.30JerJerBrianR___:  mine is perl
21:03.42JerJerlike maybe 2k lines
21:03.58J4k3now if you've got PRIs terminated to your box, thats different... although I doubt peecees handle the dsp tasks very well.
21:04.02BrianR___I bought on3e of those divaserver T1 fax cards... Basicly a bunch of faxmodems on a PCI card...
21:04.15J4k3brian: yuck... and you paid more than $10 for it?
21:04.16JerJeryup
21:04.43BrianR___They're way more than $10...
21:04.54JerJeryeah, but its paid for itself time and time again
21:05.26[TK]D-FenderJerJer: Then it wasn't "too much", just "more than you'd have liked"
21:05.38BrianR___But a lot less agrivating than futzing with half-baked soft dsp stuff or obsolete modem-pool-in-a-box appiances...
21:05.52tzangerhmm
21:06.02*** join/#asterisk x86 (n=x86@p3m/member/x86)
21:06.45BrianR___Thing just plugs in an shows up as 24 pci faxmodems...
21:06.56tzanger[incoming] has an extension that Dial(SIP/500,gt).  SIP/500 picks up.  Now I'm trying to use # to transfer.  I hit #, I hear "Transfer" but when I go to type the extension it bombs out, saying that the extension doesn't exist in [sipphones]
21:07.17JerJerBrianR___:  sorta -  I got anxcious and bought one from a supplier...then a month later a customer wanted their own solution but wanted it cheap, so I found the same card on EBay for over 1/2 what I paid   :(
21:07.20J4k3BrianR___: maybe for you.
21:07.21tzangerthis is true, the extension does not exist in [sipphones] but why's it looking at SIP/500's sip.conf entry?
21:07.28tzangeractually that makes some sense
21:07.44Mavviehttp://lists.digium.com/pipermail/asterisk-users/2007-February/180535.html
21:07.48Mavvieso, I can go back to bed!
21:07.55[TK]D-Fendertzanger: "transfercontext" in sip.conf?
21:07.59tzangermakes it hell for me to Dial(SIP/500&Zap/g1/${CELLPHONE},,gt) and have my cell phone able to park the call so I can pick it up at home
21:08.23tzanger[TK]D-Fender: does a similar thing exist in zapata.conf?
21:08.35GaVakStatus: Provisioned, Down, Active
21:08.53GaVakDoes that mean that the card is good to go, but the other end isn't up?
21:08.54JerJertzanger:  there is a global value for transfer context
21:09.00GaVak(For PRI)
21:09.11tzangerJerJer: is there?
21:09.16[TK]D-FenderGaVak: You mentioned your D-chan bouncing on you.  pastebin - "cat /proc/interrupts"
21:09.32JerJeron PRIs i use a global channel variable
21:09.42JerJernot sure about sip though
21:10.43mercestesok, asterisk 1.2.13, when you get a voicemail, and then forward it with a prepended message, the "notification email" app_voicemail sends out says I have a new message of "0.00" duration.  Voicemail itself is fine.  Is this a known "feature?"
21:10.46tzanger[TK]D-Fender: transfercontext is not listedn in sip.conf.sample
21:10.52[TK]D-FenderGaVak: Means the card is ready, but the link is dead (no claims for fault)
21:11.05[TK]D-Fendertzanger: I ran into is somewhere, just check your file.....
21:11.13*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)
21:11.32GaVakTK, I'm posting it now... The provider said they saw the D-Channel come up, then drop.. I reloaded *, and they said they saw it happen again.
21:11.42elriahHi all.  Is it possible to refer to audio files via full paths in an extensions context?  i.e., background(/some/path/to/audio) instead of putting everything in the asterisk sounds dir?
21:11.43tzangerJerJer: I see ${TRANSFER_CONTEXT}, is that what you're talking about?
21:11.43bkruseGaVak: ztcfg -vv
21:11.48GaVakhttp://pastebin.com/885437
21:11.51JerJerthe word transfer doesn't exist in my sip.conf.sample  v1.2.13
21:11.55mercestesusers are complaining that the duration in the emails are wrong.  ;.;   why?  I don't know.
21:11.55JerJeryes
21:12.01bkruseGaVak: do that.
21:12.07JerJermercestes:  don't send the duration then    :P
21:12.37GaVakBkruse: 25 channels configured 1-12 = pots, 25-37=b, 48=d
21:12.48bkruseGaVak: ztcfg -vv gives no errors?
21:12.54GaVaknone at all.
21:13.01bkruse25 channels configured? what mode and framing are you using?
21:13.04mercestesJerJer:  Nice.   How to fix?
21:13.07mercestes>.>
21:13.15GaVakthere are 2 cards
21:13.21bkrusemercestes: rm -f `which asterisk` ?
21:13.25funxiondoes anyone have odbc voicemail storage setup in here?
21:13.33mercestesGaVak:  Awesome!  Thanks!
21:13.39bkruseGaVak: post your zaptel.conf and zapata.conf, i have time to take quick look while im waiting for this to load
21:13.42bkrusemercestes: that was me :]
21:13.57*** join/#asterisk anthonyl (n=fbfff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
21:14.00mercestesbkruse:  Ah, okies.  gah, does it *have* to patch all these files?     Do I need to recompile/restart?
21:14.01bkrusemercestes: ive seen that also, not remember what i do to change it, or i fixed it
21:14.11bkruse:P
21:14.17funxionwhat modules need to be loaded to get voicemail odbc working?
21:15.29BrianR___Still interested in russellb's mention that my TE410 cards might become more usable with newer zaptel drivers...
21:15.35tzangerJerJer: that works much better :-)
21:15.36elriahAnyone:  Can I use pathnames in the BackGround command for sound files?  If I want to orgranize sound files into directories?
21:15.55JerJerelriah:  yes
21:16.10GaVakpasting the config files now.
21:16.11JerJeri do   Background(prompts/foo)
21:16.11wunderkinelriah, just dont specify the file extension
21:16.21elriahThanks.
21:16.42JerJerthen symlink  /var/lib/asterisk/sounds/prompts/  to a different location (if necessary)
21:17.02bkruseBrianR___: there is a problem with your te410?
21:17.08funxionwhat modules need to be loaded to get voicemail odbc working?
21:17.27bkrusefunxion: ls /usr/lib/asterisk/modules | grep odbc
21:17.29CJLinstRe: multiple extensions from one phone:  Got it working by defining the sip devices as type=friend.  But that broke subscription hints for outgoing calls.  Hints still process for incoming calls (busy-level=1).
21:17.32GaVakbkruse: http://pastebin.com/885452
21:17.47funxionbkruse do i need to preload or just load?
21:18.03bkrusetry to just load, because if they depend on another module, theyll fail
21:18.07bkrusetry to just load
21:18.15bkruseor even better yet, load from an asterisk already running
21:18.15BrianR___bkruse: echo, causes PCI erros from time to time, sometimes has what appear to be timing slips.
21:18.24bkruseinteresting......
21:18.28bkrusesetup?
21:18.44bkruseswitchtype=5ess?
21:18.46bkrusecool
21:19.48bkruselooks fine to me GaVak, check zttool to see if they come out of alarm
21:20.42funxionif I try to do odbc show and it restart asterisk would that mean there is a problem with my dsn?
21:20.59GaVakok
21:22.10GaVakNo alarms, internally clocked.
21:22.11tzangeris there a way to reload features.conf without restarting asterisk?
21:22.20tzangermodule reload res_features.so?
21:22.30tzangeryep
21:22.31tzangerheh
21:22.50bkrusegavak, so it says its configured correct, and in green?
21:22.55bkruse(pri)
21:22.57*** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au)
21:23.24GaVakIn green, I didn't see a green, but the customers DSL just went down again. *sigh*
21:23.27GaVaklost my access
21:23.47GaVakI was getting some strange errors though
21:23.53GaVakWrite to 44 failed: Unknown error 500
21:24.10GaVakShort write: 0/5 (Unknown error 500)
21:24.14GaVaknot sure if it was related
21:24.20bkrusewoah...
21:24.47tzangerparkcall => #72 in features.conf does not seem to work
21:24.53tzanger*3 for blindxfr does, but not that
21:25.04ManxPowertzanger: you confirmed that your phone is not eating #72
21:25.19tzangerit doesn't eat *3 (cell phone)
21:25.58funxionanyone know asterisk would restart after issueing odbc show
21:26.00ManxPowerah, so not a SIP phone.
21:26.13ManxPowerfunxion: you have a serious build or hardware problem
21:26.32ManxPowerand no amount of debugging asterisk is going to fix your problems.  What ELSE makes asterisk crash?
21:26.34funxionits most recent stable build
21:26.35funxion1.2
21:26.40funxionno hardware issue
21:26.43tzangerManxPower: nope not a sip phone
21:26.51ManxPoweryou already have two totally different things voicemail and show odbc which is making asterisk crash.
21:27.07ManxPowerfunxion: start asterisk as "asterisk -cvvv" replicate the problem, look at the error message
21:27.21bkruseturn on debugging in logger.conf
21:27.25ManxPowerfunxion: asterisk does not just magically crash
21:27.25funxionit worked fine until i decided to add odbc voicemail
21:27.33ManxPowerbkruse: I think he's getting a segfault
21:27.37bkruseouch
21:27.38bkrusenvm
21:27.41funxionpart of checking to see if the voicemail is configured is using the odbc show
21:27.55funxionI have debugging in logger.conf
21:28.01funxionManxPower not segfault
21:28.06mercestesfunxion:  Then you did something wrong.  lol
21:28.13funxionobviouly
21:28.18funxiontrying to figure out what
21:28.24bkruseok.
21:28.25bkruseand?
21:28.26funxionI followed the hoto
21:28.27ManxPowerfunxion: what is the actual message on the console when running as "Asterisk -cvvv"
21:28.32funxionit no werkie
21:28.47ManxPowerremember, you won't get all the message when running "asterisk -r"
21:29.10funxionits segfault
21:29.40funxionur right
21:29.40hellojoeHi folks, how do I get one of my macros to return some value?
21:29.40funxionwhat would cause that
21:29.40ManxPowerfunxion: I have a bit of experience with Asterisk
21:29.41funxionI know
21:29.42ManxPowerfunxion: a seg fault generates a core file.  see the README.backtrace
21:30.18[TK]D-Fendertzanger: You sure TRANSFER is not eating your "#"?
21:30.24ManxPowerPersonally I'll bet you have a buggy ODBC library -- a version not tested with Asterisk, but I don't use ODBC
21:31.17funxionIll tell you what Im trying to do, let me know if you can think of a better way
21:31.22hellojoeThis is something I have been looking for. If asterisk macro can return a True/False, it could help in dialplan
21:31.53ManxPowerfunxion: that requires that I think -- I charge for thinking
21:32.20funxionI ahev an asterisk cluster with a central vm server and was hoping to make mwi werk by adding a odbc voicemail storage to all cluster nodes
21:32.34ManxPowerhellojoe: it can't, but you can Set(BOOL=TRUE) before you exit the macro, then check that in your dialplan.  You can do this with any variable
21:33.01ManxPowerpretty much any variable you set is a global channel variable
21:33.10tzanger[TK]D-Fender: nope my featuremap has no # except for #72
21:33.10mmlj4hey ManxPower
21:33.15tzanger[TK]D-Fender: all others are *codes
21:33.15hellojoei see.. thanks a lot...
21:33.24ManxPowerhello mmlj4, yes I know you are joey
21:33.24funxionI know I can probably just write a script to create a msg.txt and msg.wav on each server it would prolly werk but I wanted it be built in
21:33.28mmlj4ManxPower: um, no visits to the new covington UMC yet?
21:34.18mmlj4you are on that job, right?
21:34.30ManxPowermmlj4: I installed the router and got everything going, then they told me about the admin office about 1 week before they moved in so I had to take the covington router for the admin office.  I told JB to order a replacement router
21:34.42ManxPowerI had MY stuff installed before they even had power outlets working
21:34.42mmlj4heh
21:35.10mmlj4well, all my stuff is done, except I haven't screwed the wallplates on yet
21:35.12ManxPowerI had to use an extension cable to get power from an outside outlet for testing, etc
21:35.22mmlj4probably done by next weekend
21:35.42ManxPowerIf they are not numbered I will be referring the users to you if they need to figure out which wall plate goes to which patch panel outlet
21:36.02mmlj4they are numbered... the wires are, and the plates will be
21:36.28mmlj4hey, freezer tape is cheap :-)
21:36.38ManxPoweroh, the switches were also ordered months ago.
21:36.49mmlj4not in yet?
21:36.52ManxPowerthe channel bank was installed months ago as well.
21:37.00mmlj4yeah, that's there
21:37.01ManxPowermmlj4: They arrived months ago too
21:37.10Bobthehunterwhats the order of god phones ? polycoms then waht what astra and cisco ?
21:37.15*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)
21:37.40mmlj4astara or whatever looks interesting, Bobthehunter
21:37.58[TK]D-FenderBobthehunter: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x
21:39.00mmlj4ManxPower: according to the soekris website, they're putting out a faster board shortly, maybe by next month... it's expected to be fast enough to run a T1 card (sangoma or whatever) and a bunch of SIP channels
21:39.03*** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net)
21:39.27mmlj4the existing soekris stuff can either route T1, or handle a bunch of SIP channels, but not both
21:39.36thinwireshi, could someone help me out real quick?
21:39.43[TK]D-Fendermmlj4: And support PCI 2.2?
21:39.47mmlj4anyhow, I want to play with the new board, once it comes out
21:39.53*** join/#asterisk Skarmeth (n=Skarmeth@201009014171.user.veloxzone.com.br)
21:39.55mmlj4[TK]D-Fender: looking now
21:39.58[TK]D-Fenderthinwires: Ask a specific question and you might get a specific answer.
21:40.05thinwires:-)
21:40.42mmlj4[TK]D-Fender: not sure, it doesn't say yet
21:40.43thinwiresI'm having issues when Asterisk Sends out a voicemail, in an email... it comes out all ad and the headers and message is garbleded
21:41.46ManxPower[TK]D-Fender: you need to teach that to jbot
21:41.51*** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl)
21:42.07*** join/#asterisk greendisease (n=jack@fedora/greendisease)
21:42.24ManxPowerthinwires: the message headers are messed up or the audio file headers are messed up?
21:42.25[TK]D-FenderManxPower: I know how, I just hadn't repeated it enough to bother yet :)  I think I may fill up the bot shortly...
21:42.49ManxPowerjbot phones is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x
21:42.51jbot...but phones is already something else...
21:42.55ManxPower~phones
21:42.56jbotrumour has it, phones is at http://bani.anime.net/phones/
21:43.12Qwell[]That's a good link :D
21:43.24Qwell[]and has anybody seen bani lately?
21:43.30Bobthehunterthanks
21:43.31ManxPowerjbot  phones is also  is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x
21:43.33jbotManxPower: okay
21:43.35thinwiresboth, the only thing I can read is "you have a message from userX" the header and sender don't work and the voicemail.wav is just a bunch of text
21:43.36ManxPower~phones
21:43.38jbotwell, phones is at http://bani.anime.net/phones/, or  is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x
21:43.38GaVakBkruse: you still around?
21:43.43greendiseaseQwell[]: ping
21:43.56Qwell[]greendisease: I'm not really here.
21:44.00ManxPowerthinwires: I've never experienced that issue
21:44.19mmlj4anyhow: # 433 to 600 Mhz AMD Geode LX single chip processor with CS5536 companion chip
21:44.19mmlj4<PROTECTED>
21:44.33Qwell[]well, okay, I guess I can be here for you
21:44.36*** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar)
21:44.38bkruseGaVak: hey sup, if its green
21:44.41Qwell[]greendisease: what's up?
21:45.01mmlj4the only gotcha I see is, does * require a sound card to process calls and/or voicemail?
21:45.03tzangerheh
21:45.04Z_Godis anyone here using either the gtalk or jingle channel? I get only audio for 1 sec to Google Talk
21:45.09tzangerI think I found a bug, but maybe not
21:45.30GaVakbkruse: how would I know if its green. (I'm not on site.) The show command shows: Primary D-channel: 48
21:45.30GaVakStatus: Provisioned, Down, Active
21:45.51tzangerI call a DID on my PRI from a SIP phone.  The DID dials my Cell phone.  My cell phone picks up, I dial *3, 700 to park the call (from the cell, so the sip phone is parked)
21:45.59ManxPowerGaVak: pri debug span 1
21:46.02ManxPower..er.. sorry
21:46.05tzangermy cell is now hung up, so I call in to my DISA and hit 701 and pick up the call just fine
21:46.07bkruseGaVak: it would say green in zttool
21:46.18ManxPowerpri debug span 2 since your PRI is on span 2
21:46.36ManxPoweryou should be getting messages from the telco
21:46.42tzangerI have exten => _8XX,1,Goto(parkedcalls,7${EXTEN:1},1) in my DISA context for other reasons
21:46.51GaVakzttool says no alarms
21:46.59ManxPowerif you are not then you call the telco and say "I'm not getting anything on my D-Channel.  Fix it or I will kill you."
21:47.13ManxPowerGaVak: zztool know nothing about PRI, only the physical layer T-1
21:47.15GaVakpri span debug 2, no logging
21:47.26tzangerI call the same DID from the SIP phone, call is picked up by cell as before.  This time I dial *3, 800 to park the call (from the cell, so the sip phone is parked, as before)... now my cell phone is hung up and the SIP PHONE hears "seven oh one" but the SIP phone is parked!
21:47.33tzangerI call in on the cell again and dial 701 and it works
21:47.38GaVakMax: lol, any suggested methods of death to offer them?
21:47.38ManxPowerGaVak: I assume you are in USA or Canada?
21:47.42tzangerit's playing it to the wrong end :-)
21:47.43GaVakUsa
21:48.18ManxPowerGaVak: I always find the threat of being transfered to sales usually works.
21:48.38GaVakLol, ok, I'll take it back to them. This is my first PRI install and I wasn't sure how to do the debugging my end.
21:48.44GaVakBut as you say, I don't see any data.
21:49.04ManxPowerGaVak: make sure you have all debugging enabled for the console in /etc/asterisk/logger.conf first
21:49.04*** join/#asterisk mega (n=mega@217.201.132.7)
21:49.11ManxPowerthen a logger restart or logger reload
21:50.10tzangerwhat's funnier is that defining extension 800 as exten => 800,1,ParkAndAnnounce(PARKED) works as it should (i.e. the cell phone hears the announcement, not the phone that is about to get parked)
21:50.45GaVakuncommnted the section about console=yadda,yadda,yadda,debug
21:50.47GaVakreloaded the logger
21:52.19GaVakwow
21:52.35GaVakFeb 20 16:52:21 NOTICE[18411]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2
21:52.46GaVakspamming like crazy
21:53.44Corydon-wSounds like a timing problem
21:55.10pigpencould someone tell me where to put the TRANSFER_CONTEXT definition for blind transfer...
21:56.18pigpenI see there is a __TRANSFER_CONTEXT...but once again...I am unsure where to stick it....
21:56.18GaVakwhat do you do on timing problems? Is it a setting, or a service call to the provider?
21:56.19bkw_GaVak, chances are its IRQ misses
21:56.26bkw_which in turn cause frame slips
21:56.27bkw_which are the root of the errors you're seeing
21:56.50JuggieOR, it could just be a bad cable
21:56.57bkw_or bad card
21:57.01bkw_or bad computer
21:57.03pigpenor low signal
21:57.04GaVaklol
21:57.07bkw_but i doubt its the cable
21:57.15pigpenor bad smartjack
21:57.24bkw_now I have had that happen
21:57.32GaVakwow, this is a hard error to pin down, no?
21:57.36*** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net)
21:57.53vader--does anyone know how to reload the zapata.conf in the asterisk console?
21:57.59n|cotineI noticed the discussion on asterisk-user regarding Dell servers - does anyone NOT have a problem utilizing a 2850 or a 2950 as a PBX with Asterisk?
21:58.00tzangerhaha
21:58.01tzanger<PROTECTED>
21:58.01tzanger<PROTECTED>
21:58.04bkw_vader--, just restart
21:58.18vader--i don't wanna shut the phone system down
21:58.19Corydon-wvader--: you can do a reload chan_zap.so
21:58.23bkw_n|cotine, buy sangoma cards and it will work
21:58.34Corydon-wvader--: that'll work as long as you aren't changing signalling
21:58.45pigpenn|cotine, hmm...I have several Dell 6850's.....and I had a dell 2850 running great too.
21:58.59n|cotinepigpen:  What type of PSTN hardware?
21:59.00vader--ya i wanted to change signalling
21:59.02mafkeesn|cotine: they work great here
21:59.04CrashHDhints for local parking extensions doesn't seem to be working, any ideas?
21:59.05Corydon-wn|cotine: it can be done, you just need to isolate the IRQ
21:59.06CrashHD1.4?
21:59.15pigpenn|cotine, digium, 2port pri/4 port pri
21:59.26mafkeesCorydon-w: with quadbri I did not have to isolate IRQ at all
21:59.37pigpenn|cotine, it is helpful to have a gentoo kernel dev in house.   :)
21:59.42mafkeesjust popin the card, boot debian, compile bristuff and it's working great
21:59.43Corydon-wmafkees: depends upon your load
22:00.00bkw_pigpen, you mean linux kernel dev.. I wasn't aware gentoo wrote a kernel :P
22:00.03*** join/#asterisk fibs- (n=chrisk@c-24-20-45-4.hsd1.mn.comcast.net)
22:00.05mafkeesand the sangoma A101 works great in it as well
22:00.10Corydon-wHeavy system load needs a separate IRQ
22:00.11*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
22:00.13fibs-Hola
22:00.21bkw_sangoma cards don't have these IRQ issues
22:00.25bkw_they can even share
22:00.32n|cotineIf I have POTS lines, are people seeing these issues with the TDM400P also in Dell hardware?
22:00.41mafkeeswe have 5 2850's with quadbri and 2 2850's with sangoma A101
22:00.45mafkeesall machines run fine
22:00.57mafkeesno need to fidle with bios or anything
22:00.58thekidriocan i get asterisk to dial two numbers via outbound IAX2 and then connect the two calls together?
22:01.17mafkeesthekidrio: yeah, look at callfiles or the manager Originate call
22:01.36pigpenbkw_, bla.
22:01.36thekidrioreally i want asterisk to call one number, and if that number answers to dial out to another number
22:01.38JTn|cotine: digium cards favour dell hardware
22:01.40JTas a general rule
22:01.51fibs-I have a questions regarding incoming SIP and asterisk, I have it setup (i think) and when i call my number I hear a male voice saying "testing" and then some DTMF tones
22:01.54PaulTech85thekidrio Use the manager interface
22:01.56thekidriomafkees: thanks will do
22:01.59pigpenn|cotine, we also have 4 port pots and the tdm2400
22:02.11thekidrioi assume i need to out going lines right?
22:02.17Corydon-wJT: no, Digium cards favor white boxes, really
22:02.22mafkeesthekidrio: yeah
22:02.28thekidriothanks mafkees and PaulTech85
22:02.37JTCorydon-w: hrm, well of brand names anyway
22:02.41Corydon-wJT: Dell boxes have their own set of problems, because they like to hard assign slots to IRQs
22:02.42*** join/#asterisk stack_ (n=stack@63.239.190.203)
22:02.44JTi hear dell gets the best zttest results
22:02.47fibs-Does a male voice saying "testing" and then some tones sound familiar to anyone?
22:02.57thekidrioDell power edges are a pain in the arse
22:03.07JTthekidrio: in what respect?
22:03.09*** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net)
22:03.13thekidrioi have had 3 power edge machines fully see my digium card
22:03.14Corydon-wJT: we get better results with zttest with white boxes with Intel motherboards
22:03.38Vecthekidrio : ahhh, I am getting a poweredge 2900, hope the thing works ?
22:03.40thekidrionot full see i mean
22:03.42fibs-I know asterisk is seeing the inbound sip "Looking for s in sipcall (domain 24.20.45.4)
22:03.48mafkeesmost dell stuff is intel based as well
22:03.50JTit's all fractions of a difference, are we talking 100% across the board?
22:03.50stack_Our Grandstream 386 burned up today... should we go with another one is there something better for hooking a physical fax into our asterisk system?
22:04.06JTmafkees: based... but they do make their own boards
22:04.07mafkeesstack_: cisco ata
22:04.10thekidrioi think its less of an issue in the newer dell power edges
22:04.14mafkeesJT: indeed
22:04.19bkw_haha
22:04.22stack_mafkees: any specific model?
22:04.32mafkeesstack_: nah, they all are great
22:04.39thekidriomy model numbers are 2650 and 4600
22:04.49fibs-Anyone setup a Gizmo Voip with asterisk before?
22:04.55thekidrio2 2650's 1 4600
22:05.01GaVakQuick question: the timing errors I'm getting. Could it be because I'm using a TDM card in the same system?
22:05.10mafkeesboth rollouts with dell were on customer request. I would never buy dell machines myself
22:05.15GaVakcould it be the tdm card eating the interrupt bandwidth.
22:05.49bkw_actually you dont have to interrupt 1000 times a second
22:06.05bkw_you can actually lower that by 95%
22:06.12bkw_and accomplish the same results
22:06.15bkw_and better performance
22:06.18russellband increase latency.
22:06.19mafkeesbkw_: I never understood that 1000 interrupts/second
22:06.22bkw_no
22:06.24russellbyes.
22:06.25bkw_no
22:06.29bkw_youdo not
22:06.36bkw_you're doing 20ms packets on the voip side
22:06.43bkw_so why are you filling up 20ms packets 1ms ata time
22:06.48bkw_slice that bad boy off there in 20ms chunks
22:06.50mafkeesthat 1000 interrupts are for timing right ?
22:06.56bkw_muchmore efficient
22:07.17GaVakshow do you change that?
22:07.18mafkeestiming like meetme and zap trunks
22:07.19GaVak*how
22:07.26bkw_russellb, 20ms doesn't kill anything.. ifit did then faxing from LA to London wouldn't work
22:07.31thinwiresdoes anyone here use AsteriskNOW?
22:07.34*** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net)
22:07.36Corydon-wGaVak: You don't.  He's trying to start a flame war.
22:07.40Corydon-wbkw_: please stop.
22:07.41bkw_no i'm not
22:07.42bkrusethinwires: join #asteriskNOW
22:07.44GaVakRoger that.
22:07.45bkw_I'm stating a fact
22:08.00thinwiresbkruse: it's dead as a doornail over there, no one is responding :-(
22:08.07*** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net)
22:08.21Corydon-wYou're stating something you you believe to be true, contrary to existing advice
22:08.29bkw_no I know its true
22:08.37bkw_I'm doing it right now on 16 T1's
22:08.37Corydon-wIf you want to debate that, this is not the forum
22:08.57bkw_its only the forum for that when you guys feel like it is..
22:09.12Corydon-wPlease take it elsewhere
22:09.22bkw_I'm just stating that 1000 interrupts persecond is NOT a requirement
22:09.42*** mode/#asterisk [+b %bkw_!*@*] by Corydon-w
22:09.50Corydon-wPlease take it elsewhere
22:10.01bkrusethanks.
22:10.30*** mode/#asterisk [-b %bkw_!*@*] by Corydon-w
22:10.31*** join/#asterisk bkw_ (i=brian@adsl-70-143-62-84.dsl.tul2ok.sbcglobal.net)
22:10.46*** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines)
22:13.45n|cotineThis isn't very clear in the cookbooks on voip-info - how exactly does call flow in DUNDi work?
22:14.46*** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com)
22:14.53*** join/#asterisk AJaymn (n=boiwonde@24-159-236-181.dhcp.mdsn.wi.charter.com)
22:15.04*** join/#asterisk Umaro (n=umaro@68.142.142.105)
22:15.53UmaroHey guys.. I'm not looking for something free, or to circumvent the g729 patent, but is there a softphone I can buy that runs on linux and supports g729?
22:16.03*** join/#asterisk flenders (n=fserto@unaffiliated/flenders)
22:16.24thinwirescan anyone tell me which is better? *1.4.0 or *1.2.15?
22:16.47cpatry1.4.0
22:17.15J4k3testing and experimentation is the best method to know whats "best"
22:17.26*** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
22:17.30J4k3technically 1.4.0 is superior but as its fresh and new you may find a bug or an incompatability that you cannot live with
22:17.38*** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no)
22:18.08thinwiresI'm actually using AsteriskNOW atm but I can't find any support for it...
22:18.12mogormanUmaro, digium.com
22:18.26mogormansells hardware and or software for g729 transcoding
22:18.27*** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net)
22:19.34Umaromogorman: yeah, but not a g729-enabled linux softphone
22:19.46mogormanoh i misread
22:19.47mogormansorry
22:19.52mogormanasterisk can be a softphone
22:19.54Corydon-wUmaro: the text console of Asterisk can be used as a phone
22:20.12Corydon-wUsing chan_oss or chan_alsa
22:20.19J4k3or bluetooth
22:20.38Corydon-wThe bluetooth stuff is not in there, yet, though
22:20.49J4k3ah.
22:21.11J4k3I need to work on that, maybe thats what I'll dedicate the rest of today to
22:21.46flendershey, do you guys think the voice quality on a cisco 7905 would be the same as on a 7960?
22:23.12UmaroCorydon-w: Is that the only option?
22:23.21Corydon-wUmaro: I can't answer that, as I don't know
22:23.32Corydon-wIt's certainly one option
22:26.39UmaroCorydon-w: yeah, just not sure how feasible it is to have 50 call center agents running asterisk on their workstations
22:26.56Corydon-wUmaro: A quick google search reveals that it's not the only option
22:27.01Corydon-whttp://voxilla.com/voxilla-stories/voxilla-stories/free-linux-soft-phone-released-397.html
22:27.07CrashHDanyone have a problem with parking hints?
22:27.17CrashHDin 1.4
22:27.19*** join/#asterisk amdtech (i=amdtech@nat/digium/x-1fe7d0b17b740e8b)
22:27.24CrashHDmine seem to have stopped working
22:29.01JTUmaro: you might have to get a commercial softphone
22:29.10*** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net)
22:29.15JTanyway softphones suck, is there any reason they must have a softphone?
22:33.36[TK]D-Fender* != softphone.  Holy Jeebus
22:34.16cpatryJT: cause some compagnies dont want to buy all hardware phones?
22:34.56*** join/#asterisk infernix (i=nix@spirit.infernix.net)
22:35.07JTcpatry: is there a good reason or are they just being stingey?
22:35.29JTare the softphones connecting to asterisk? if so, why on earth do they need g.729?
22:35.31cpatrythey probably too cheap, but i understand that.
22:36.15thinwirescheap only buy's cheap :-)
22:36.31fibs-I'm trying to do a EXEC DIAL SIP/user:secret@siphost/numbertocall, and I'm getting siphost/numbertocall hostname invalid
22:36.40fibs-which means its trying to resolve the hostname and the number im trying to dial out
22:36.44fibs-any ideas folks?
22:37.32cpatrywhen compiling zaptel: i see: checking for ZT_TONE_DTMF_BASE in zaptel.h... yes   but theres no app_meetme.so after the make, bug?
22:37.36fibs-so basically:  No such host: proxy01.sipphone.com/<phonenumber>
22:38.26UmaroJT: I don't mind buying a commercial softphone, I just can't find one that runs on linux and supports g729
22:38.42JTthat's because the codec is commercial
22:39.04Umaroright, but I don't mind paying for the softphone, AND paying for the g729 codec
22:39.11Umaroit just has to run on linux and do g729
22:39.29JTUmaro: so the question is, if you have an asterisk server on your lan, why is it necessary to use g.729 on the softphones?
22:39.33JTnot sure if any are available
22:40.09UmaroJT: I'm using g729 with the voip provider, and don't want asterisk to transcode
22:40.22JTit's the easiest option
22:41.03JTand if you do internal calling too, the staff will appreciate the superior voice quality
22:41.24thekidrioanyone have a good guide on how to get two outbound calls to be patched together?
22:41.39cpatryumaro: xten doesnt support 729?
22:41.56thekidrioi think i need to use the call manager or the fop but i am too noobish to know for certain
22:42.36*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
22:43.09thekidrioany ideas?
22:44.58JTcpatry: do they have a linux release?
22:45.08*** part/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net)
22:45.15cpatryJT: yeah
22:45.22thekidrioeven just the correct terminology to search on how to connect two active outbound calls would help
22:45.37JTthekidrio: either the manager interface or .call files
22:45.51cervi<PROTECTED>
22:45.52JTthekidrio: umm, are the calls pre existing, or being set up when they require patching?
22:46.16cerviHow can I start in the opposite order?
22:46.20cpatryhttp://www.xten.com/index.php?menu=download
22:46.36thekidriojt, i would like the calls set up when they  require patching,  i am hoping to eventually use a click to call with the following method
22:47.15JTok, so when you click, both ends are called?
22:47.21thekidriouser clicks on a link, it takes them to form where put in the number they can be reached at, after that the system calls the number attached to the link they clicked and connects them both
22:47.25thekidriojt, yeah
22:47.41e-miliohello all
22:47.51thekidrioi am fairly certain i can get the php agi working
22:48.03pigpenI may have a DID issue.  I have 400 did's.  I am only getting the CID number passed to the DID's, however I am getting the entire CID (name and number) passed to our primary, published numbers.
22:48.04thekidriojust not sure the syntax to connect them
22:48.16pigpenCould asterisk be stripping this, or more likely the Telco?
22:48.20e-milioIs there any actual difference in the asterisk on AsteriskNow than just asterisk ??
22:48.36*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
22:48.39Corydon-we-milio: no
22:48.54flendershey, if I change settings on zapata.conf, does a reload work? or I need to restart asterisk?
22:49.02JTthekidrio: asterisk manager interface or .callfiles
22:49.03thekidrioi could probably just write the .call files, but i don't know the syntax to get the two outbound routes to talk to each other
22:49.09thekidriook cool
22:49.11cpatryflenders: need to restart it.
22:49.15Corydon-we-milio: they're built out of exactly the same source tree
22:49.19e-milioCorydon-w: For a callcenter enviroment IYHO it is the same ?
22:49.23thekidrioi will just read up on .call files as that seems the most simple solution
22:49.25JTthekidrio: there's a sample call file in the docs
22:49.29JTof the source
22:49.41JTand the book mentions them too
22:49.45thekidriojt, great hopefully i can just template that
22:49.53thekidrioyeah i am reading the future of telephony atm heh
22:49.58*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
22:50.06JTheh, my hardcopy is shipping
22:50.21JTthe second edition will be out in a couple of months
22:50.31thekidrioyeah i can't wait for that
22:50.35thekidriomine is filled with update notes haha
22:50.52JTi use it as a reference at the moment
22:51.52*** join/#asterisk flipdesk (n=flip@ip68-2-210-190.ph.ph.cox.net)
22:54.49e-milioCorydon-w:thanks
22:55.23Umarocpatry: xten doesn't have a linux client
22:55.32cpatryfor x-lite yeah.
22:56.47flipdeskso yeah, here's an interesting one
22:57.00*** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net)
22:57.01flipdeskonce in awhile, a phone will register fine
22:57.13JTUmaro: x-lite doesn't have g.729 anyway
22:57.16flipdeskpasswords are fine, sip show peers shows the phone registered
22:57.29flipdeskbut when the phone tries to make a call I'll get in the logs:
22:57.40cpatrywhen compiling zaptel: i see: checking for ZT_TONE_DTMF_BASE in zaptel.h... yes   but theres no app_meetme.so after the make, bug?
22:57.53flipdeskFeb 20 15:43:34 WARNING[21087] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"That Guy" <sip:5605@67.139.179.230>;tag=as6c3eccf5'
22:58.06flipdeskwith sip:5605 being the extension that is making the call
22:58.07GaVakIf I install a new zaptel version, do I have to remake asterisk?
22:58.30flipdeskphone  is registered though
22:58.32flipdeskweird eh?
22:59.05cpatryGaVak: yes
22:59.09GaVakthnx
23:00.28flipdeskanyone have any ideas?
23:03.20flipdeskis there a way to get the current state of a device?
23:03.24flipdeskfrom the CLI
23:05.06*** join/#asterisk MaartenB (n=Maarten@84-105-196-31.cable.quicknet.nl)
23:05.13MaartenBhello everyone
23:05.49MaartenBI have a question about setting the outgoing callerid correctly with CALLERID(num)
23:06.01MaartenBI want to do this based on the sip account, but I have no idea how to do it right
23:06.26EmleyMoorMaartenB: Hold on, I've done that
23:06.36MaartenBgreat
23:07.24EmleyMoorJust put callerid="CallerID name" <CallerID number> in the relevant entry in sip.conf
23:07.58MaartenBok, the problem is that sometimes, a call is being forwarded
23:08.01flipdeski think he wants to use the callerid command
23:08.07MaartenBand I have to set the callerid in that cases too
23:08.11flipdesknod
23:08.25EmleyMoorAh, that's easy too
23:08.47MaartenBI like easy things :)
23:09.01EmleyMoorSet(CALLERID(num)=number)
23:09.20MaartenByes, I know that
23:09.36EmleyMoorUse GotoIf or ExecIf commands to execute it conditionally
23:09.41MaartenBah, ok
23:10.15MaartenBthen I think my question is, how can I test with GotoIf or ExecIf if it is, or it is not a call initiated from the sip accounts
23:10.18EmleyMoorYou'll probably have some CUTting to do to get useful information about the channel
23:11.09EmleyMoorSet(TECHNOLOGY=${CUT(CHANNEL,/,1)})
23:11.25MaartenBok, thanks
23:11.26MaartenBtrying...
23:11.29EmleyMoorYou can then check if that is SIP
23:11.46EmleyMoorIf you need to check more specifically, you can do similar things
23:12.04JTMaartenB: couldn't you just use different contexts for different technologies, and set a variable?
23:12.25*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
23:13.01flipdeskwrong password on authentication for INVITE
23:13.05flipdeskman that's odd
23:13.18JTseen it before
23:13.22flipdeskhow could the INVITE password be different from the REGISTER password?
23:13.23JTcan't remember what caused it
23:13.35JTit might be an asterisk or provider bug
23:13.41*** join/#asterisk qdk (n=qdk@90.184.3.249)
23:13.52flipdeskyeah, I found a bug report
23:14.02flipdeskbut it couldn't be reproduced
23:14.11JTit occurs randomly
23:14.16flipdeskvery
23:14.54flipdeskI bet if I restarted asterisk, it would go away. At least for that extension
23:15.10JTmaybe
23:15.15JTor just chan_sip
23:15.27flipdeskgood point
23:15.55*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
23:16.08flipdeskgoogling has turned up nothing productive :-/
23:20.37*** join/#asterisk pingboy12 (n=idatoo@CPE0004e24967bd-CM001692fb03c2.cpe.net.cable.rogers.com)
23:21.07*** join/#asterisk X-Rob (n=rob-x@CPE-58-169-113-201.vic.bigpond.net.au)
23:22.00mercestesok, when I forward a voicemail to another user, the emailed voicemail notification says that the duration is 0.00.  Is there a way to fix this behavior??
23:22.32*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
23:23.38pigpenIf I am getting the number of the CID, and knowing that I have no special CID rewrite commands in my dialplan, would you assume that my Telco is not passing the CID Name ?
23:23.38pingboy12having probs compiling asterisk 1.4
23:24.24pingboy12i get makeopts error, when doing make install
23:25.16pingboy12says configure script must be run, before running make
23:25.20pingboy12but i did that
23:26.21ChicagoBudpingboy12, was it successful?
23:26.32ChicagoBudpingboy12, the ./configure
23:28.47ChicagoBudhas anyone built app_txfax.c and app_rxfax.c for * 1.4???  How did you do it since the build process is so different???
23:29.04*** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl)
23:32.52*** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux)
23:33.18*** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com)
23:35.08Dovidmorning all
23:35.41*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
23:38.58ManxPowerpigpen: on PRi or FXO?
23:39.08*** join/#asterisk backblue (n=moo@87-196-109-88.net.novis.pt)
23:39.23pigpenpri.
23:39.28pigpendam goofy.
23:39.35pigpenI have done some more testing:
23:39.39ManxPowerpigpen: PRI name arrives as a message AFTER call setup
23:39.55pigpenerr....continue.
23:39.57ManxPowerso if you put a Wait(.5) as the first priority of whatever extens match the incoming calls, then it should wokr
23:39.59pigpenor explain...
23:40.09ManxPowerWait(.25) might work.  It comes in pretty fast.
23:40.17pigpenk.
23:40.24*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)
23:40.27pigpenWell, just in case...this is what I tested.
23:40.38pigpenI dial the main number, type in the exten...works.
23:41.03pigpenI redirect the main number to "Dial(IAX2/6333,20,twW)...no dice....
23:41.27pigpenbut..typically, yes...the main number goes through...well..alot of stuff.
23:41.33pigpenk..adding it now.
23:42.47pigpenHa!.
23:42.53*** join/#asterisk welby (n=welby@ivonova.whmcr.com)
23:43.06pigpenToo dam easy..thanks for the years you have been helping me.
23:43.20ManxPowerfeel free to send money to eric@fnords.org via PayPal
23:43.22pigpenYou Qwell and many others.
23:43.41pigpen:)
23:44.37*** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140)
23:45.45*** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net)
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23:47.34luke-jr_workfile: did 8821 re-break 7051?
23:47.48*** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com)
23:47.49*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
23:48.19blitzragefile did it
23:48.28fileluke-jr_work: I knew it would creep up when I put 7051 in, I knew it
23:48.55luke-jr_workfile: 8821's configuration is broken
23:49.16ManxPowerI suppose if I filed a remote on bugs.digium.com asking to remove a config option because it is not documented and nobody knows what it does would be closed?
23:49.21luke-jr_workfile: the fix for 7051 is necessary to not break my setup
23:49.39ManxPowerSo, when is kram going to come back to IRC?
23:50.13*** join/#asterisk tuxd00d (n=tuxd00d@128.187.128.38)
23:50.20CrashHDhints are not working in 1.4, any suggestions?
23:51.40fileluke-jr_work: an option, that's the only way I can keep both of you happy
23:52.01luke-jr_workfile: oh, there's an option now? works for me, I guess
23:52.09fileno, I mean I'm going to have to code one now
23:52.15luke-jr_workoh
23:52.23fileyou need externip to be compared against localnet, he needs it not
23:52.29luke-jr_workor he could set localnet to include his local IP
23:52.43*** part/#asterisk tuxd00d (n=tuxd00d@128.187.128.38)
23:53.21ManxPowerCrashHD: remove the call-limit option from your config
23:54.05CrashHDmanx none of my sip entries have a call limit
23:54.20CrashHDand show hints actually displays everything as unavabile
23:54.26CrashHD(unavailable)
23:54.39CrashHDwhere as working configs in 1.2 show those entries as idle
23:54.53flendersis there a huge difference on txgain if I change it from -6 to 4 on zapata.conf?
23:55.18ManxPowerflenders: yes.  it's a logarithmic scale
23:56.45CrashHDany other ideas ManxPower?
23:57.28*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
23:57.58*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:58.39ManxPowerCrashHD: you know my policy on 1.4, right?
23:58.55CrashHDheh, I don't, but I have a feeling your about to enlighten me
23:59.13blitzragehockey time!!! GO LEAFS GO!
23:59.24ManxPowerCrashHD: If digium doesn't run 1.4 on their corporate PBX then I'm not going to consider running it on my client's corporate pbxs
23:59.32ManxPowerand I've seen no indication that digium is doing so.
23:59.43ManxPower1.4 == 1.4 released version

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