irclog2html for #asterisk on 20050819

00:00.07*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
00:00.32*** part/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
00:00.32*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
00:00.49surfduein gstreamer-properties when i change to alsa i hit test, and it says failed to construct pipeline
00:01.28kyooAriel_: What should I check next?
00:01.40kyooWhat does DID mean?
00:02.01DaPrivateerdirect inward dialing
00:02.31*** join/#asterisk anthm[tablet] (i=anthm@208.254.19.131)
00:02.48*** mode/#asterisk [+o anthm] by ChanServ
00:02.48shido6need a DID?
00:02.58shido6number for "ingress" or inbound calls
00:03.05Ariel_kyoo, did you put the number at the end of your broadvoice account sip.broadvoice.com/number
00:03.45kyooAriel_: Yes, but I did not know what to pus, so I used my own extension, 305...
00:03.51kyoos/pus/put/
00:04.01shido6err
00:04.16shido6is 304 in "default" or the context you specified in [general] in sip.conf?
00:04.20Ariel_kyoo, for broadvoice you need your phone number
00:04.34surfduecan asterisk transfer calles to other numbers
00:04.37surfduelike 1500 2020929
00:04.38surfdue?
00:05.19Ariel_surfdue, like a number you dial out too?
00:05.48kyooAriel_:   So it is: <mybvtelno>@sip.broadvoice.com:<mypass>:<mybvtelno>@sip.broadvoice.com/<mybvtelno> ?
00:06.22Ariel_kyoo yes
00:07.12surfdueyes
00:07.17surfduecan u redirect it to another number?
00:07.29Ariel_surfdue, yes with the right dial plan you can
00:07.30kyooAriel_: Still no change...
00:07.38surfduedial plan?
00:07.56Ariel_kyoo, are you using normal asterisk or something like amp added to your setup?
00:08.09kyooAriel_: I am using AMP via A@Home.
00:08.45Ariel_kyoo, did you setup your did route?
00:08.54surfdueAriel_ dial plan?
00:09.01kyooAriel_: No, I did not - I wasn't sure what it was for.
00:09.26Ariel_kyoo, it's for routing your did's. (your phone number given you from bv)
00:09.31nick125why doesnt asterisk start!?!? /me sighs
00:09.49Ariel_nick125, something is missing or not configured correctly
00:11.00kyooAriel_: What should I put there?  It wants DID number and destination... (reception (empty), extension, vm, ring group (empty), queue (empty), custom app, 'use incoming setting')
00:11.18kyooDo I put my BV number in the number field?
00:11.18nick125ah, i think some hold music i put in caused it o.0
00:11.33file[laptop]Side effects may include bleeding from the eyes.
00:11.42Ariel_kyoo, ok put your bv number then send it to were you want it to go too. like an extension.....hint
00:11.49surfduenick
00:11.57surfdueHOLD MUSIC!
00:11.57surfdueLol
00:12.05kyooAriel_: But this is for *outgoing* calls...  Why would I want it to go to an extension?
00:12.13nick125i removed the hold music, and it started
00:12.39Ariel_kyoo, bv number is for incoming calls
00:12.51Ariel_outgoing calls don't take this settings
00:13.03Ariel_you need to create an outbound dialing rule.
00:13.21kyooAriel_: I'm trying to get outgoing calls working.  I've set up an outbound routing...
00:14.02Ariel_kyoo, you picked the broad voice trunk?
00:14.12Ariel_when you dial out what does the CLI say
00:14.49Ariel_nick125, are you using mpg123 or mpg321?
00:15.14kyooAriel_: Yes. the route is named "outgoingbv" no password, dial patterns 1NXXNXXXXXX NXXNXXXXXX NXXXXXX and trunk sequence 0 is SIP/BV1 ...
00:16.42kyooAriel_: I'm sorry - how do I know what the CLI says?
00:16.56Ariel_well the last 2 should have a 1|NXXNXXXXXX and 1areacode|NXXXXXX
00:17.35kyooAriel_: AMP autogenerated those for me.. :)
00:18.10Ariel_kyoo, yes but there dialing to a voip provider are they not?
00:18.23kyooAriel_: They are.
00:19.42kyooThe interface won't accept those numbers - it gives a DB error.
00:22.13Nukemizerany "bug track" admins here that can help me navigating for solution that Digium tech support directed me to look for ?
00:22.18*** mode/#asterisk [+o twisted] by ChanServ
00:23.26NukemizerI am looking for "ill allow you to assign "D" to the Hangup
00:23.26Nukemizerapplication" but am not having success locating in bug tracker
00:23.48Nukemizer"will allow you to assign "D" to the Hangup
00:23.48Nukemizer<PROTECTED>
00:25.23Ariel_kyoo, you added something incorrectly then. or you tried to add bv dial route again. you need to clear that and edit the one you already have.
00:26.37kyooAriel_: This is what shows in the CLI when I attempt to call: http://rafb.net/paste/results/Yknpop52.html
00:27.43*** join/#asterisk spackle (n=spackle@209.234.83.19)
00:27.48kyooI use a full 10-digit number, which should get triggered by the 1NXXNXXXXXX, I think...  But I do not see it getting triggered at all.
00:27.58nick125hmm, when i try to login with kphone, it says forbidden, but, there isnt a place to put a password
00:28.22jerokyoo, did you put a _ at the beginning ?
00:28.36kyoojero: At the beginning of ... ?
00:28.46jeroof the expression
00:29.01jerolike exten => _1NXXNXXXXXX,1,blah
00:29.13Ariel_kyoo, change in your trunk from 0 for max channel to something like 2 or 3
00:29.29Ariel_jero, this is for out bound
00:29.34jeroyes
00:29.38kyoojero: I'm using AMP - I tmay do that part for me, but it's examples do not use that.
00:29.38Ariel_jero, and it's via amp
00:29.45jerooh ok sorry
00:30.05Ariel_kyoo, your report says you dont have enough channels available and it's set to 0
00:30.13jeroamp has 2 places to define that, you may be in the wrong one
00:30.40kyooAriel_: That looks like it ... IT IS!
00:30.55Ariel_Jero correct it does have 2 places
00:31.27jeroand the two places interact quite a bit, I dont remember how, didnt use it for something like 8 months
00:31.46kyooAriel_: The interface says "blank for unlimited" and it defaults to 0.  oops. :)
00:31.52kyooAriel_: Thanks so much.
00:32.19Ariel_kyoo, no problem
00:32.22jeroI'll try to package sflphone for many linux distros and other oses eventually
00:32.31Ariel_jero it's changed in 8 months allot
00:32.49jeroAriel_, theres no doubt :)
00:32.54jeroanyone ever tried sflphone ?
00:33.37nick125ok i just created a extendion on AMP, how do i get x-lite to use it?
00:33.55jerolog in your xlite as this extension
00:34.03tzangerI love the 'babysitting email scanner' someone has on -users
00:34.18tzangermy posts to the list were automaticlaly deleted because of sexual discrimination... ??
00:34.27tzangerI love the us legal system that makes this kind of thing necessary
00:34.39nick125jero: i try to, using the correct pass and everything, and it says forbidden
00:35.09jerohrm
00:35.42*** part/#asterisk spackle (n=spackle@209.234.83.19)
00:35.45*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-71-65-37-102.indy.res.rr.com)
00:37.02Ariel_nick125, if it's saying forbidden then there is something incorreclty set up on xlite
00:38.01Ariel_tzanger, live is so strange.  Sex is all over the place it's just part of life....some do some don't
00:38.09Ariel_live/life
00:39.20tzangeryeah
00:39.20jeronite all
00:40.05*** join/#asterisk otaku_p0pe (i=foobar@blk-222-96-154.eastlink.ca)
00:40.11otaku_p0peoh wow
00:40.27Ariel_jero, goodnight
00:40.41otaku_p0peso i bet this question gets asked all the time.
00:40.45otaku_p0peanyone use asterisk + broadvoice ?
00:41.04*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
00:41.17Ariel_otaku_p0pe, yes
00:41.44otaku_p0pei keep getting 401 unauthorized from their sip gateway. now i know the password and such is correct
00:41.50Ariel_oh it's yes it's asked allot and yes people use it.
00:42.00otaku_p0pethe only things i can think of is the NAT is causing trouble, or it takes a few hours to actually activate the account
00:42.08otaku_p0penote : i've gotten the email saying my account is active
00:42.12otaku_p0peso i dunno.
00:42.31*** join/#asterisk asteriskjohn (n=johnb@ip24-251-151-16.ph.ph.cox.net)
00:42.34Ariel_otaku_p0pe, are you putting the account to an ata and not to an asterisk box?
00:42.53otaku_p0peto an asterisk box.
00:43.09otaku_p0pethe asterisk box works - i have an FXS / softphones that work gloriously. except over gprs.
00:43.25bkw__it won't
00:43.35bkw__grps is too slow... latent and sucks
00:43.38otaku_p0pei know.
00:43.48otaku_p0pewell, it's just latency
00:43.52kyooAriel_: Can you give any suggestions about incoming on the same trunk?  (it goes directly to BV voicemail, and never even shows in the CLI sip debug...)
00:44.02otaku_p0pei got 35k/s on my gprs one time doing apt-get upgrade (don't ask)
00:44.42Ariel_kyoo, remember the did routes
00:45.05kyooAriel_: I have set up a DID route, to the receptionist...
00:45.32otaku_p0pebleh
00:45.34kyooThe DID number is simple my BV number, without the :1:, correct?
00:45.38Ariel_kyoo, you have it registered correct?
00:45.46Ariel_yes
00:45.56kyooAriel_: As far as I know - it reports registered.
00:46.16nick125Ariel_: i cant find anythong incorrectly setup in xlite..
00:46.23otaku_p0pei wish mine reported registered :(
00:46.42*** join/#asterisk brimston1 (n=brimston@pcp01534724pcs.huntsv01.al.comcast.net)
00:46.50Ariel_nick125, are you on the same network?
00:46.57nick125yeah
00:47.32nick125my ip is 1.9, server is 1.15
00:47.41nick125(all in private block)
00:47.44Ariel_nick125, what does the cli say wen your trying to register
00:47.57kyooAriel_: It actually shows 2 SIP peers for bv -  bv1/<my bv number> and <my bv number>/6034, both connected to the same BV host. it that normal?
00:48.13nick125the asterisk cli?
00:48.36Ariel_kyoo, no
00:48.55Ariel_nick125, yes
00:49.04nick125if i try to make a call out to something like *411, it says 'service unavailable' in the xlite logs
00:49.09Ariel_kyoo, you might have added two trunks
00:49.15nick125nope, dont see anything
00:49.33otaku_p0peman oh man
00:49.38Ariel_f9 on xlite gives you a debug settings
00:49.57kyooAriel_: I don't think so ( no second one is showing) but it may be because it have the entire set of BV connection information in each of the textareas on the trunk setup - Outgoing and Incoming...
00:50.01nick125thats what ive been looking at to find the service unavailable
00:50.39nick125hmm, in amp, what type should i set?
00:50.51nick125oh nvm
00:51.09otaku_p0peAriel_: do you use AMP ?
00:51.23nick125wait, in the diagnostics, aint that the raw stuff of something?
00:51.28nick125i dont see it sending the password
00:52.03Ariel_otaku_p0pe, yes and no. I use it for some of my customers
00:52.11otaku_p0pewith broadvoice ?
00:52.12Ariel_I don't use it for my self
00:52.16Ariel_yes
00:52.33otaku_p0pehmm. i'm betting it's my register line that's messed
00:52.40Ariel_nick125, hummm could it be you did not set it up correctly.
00:52.42kyooAriel_: What information goes in the "Incoming" textarea?
00:52.53otaku_p0peit's number@sip.broadvoice:password:number@sip.broadvoice/number ?
00:53.05Ariel_otaku_p0pe, yes
00:53.36nick125Ariel_: i set it up as this: username/authorization: 200
00:53.40otaku_p0peugh. i really really really do not want to strace asterisk
00:53.43Ariel_incoming text area????
00:53.44nick125Password: <my password>
00:54.08nick125Domain/Realm and SIP Proxy: <my asterisk box>
00:54.20kyooAriel_: On the trunk setup, there is a textarea called "incoming" wher eyou put all of the SIP settings for incoming calls.  Should that be a copy of the outgoing box?
00:54.48otaku_p0pekyoo: are you also using AMP / broadvoice ?
00:54.55otaku_p0peand no, it's not an exact copy.
00:54.56kyoootaku_p0pe: yes.
00:54.56*** join/#asterisk zox (n=zox@ip70-176-64-134.ph.ph.cox.net)
00:55.04Ariel_kyoo, oh I seee you did not set the trunk up correctly.
00:55.31otaku_p0pekyoo: http://geekgazette.com/index.php?option=com_content&task=view&id=20&Itemid=26
00:55.34Ariel_ok everyone with amp and needs help lets go to the #amportal where there are others there that use it as well.
00:55.34otaku_p0pethat's what i used
00:55.35kyooAriel_: It would not surprise me.  I don't know what to put in the trunk boxes, so I copied the setup info from bv.com's asterisk page...
00:55.51otaku_p0pebut my trunk isn't registering correctly with BV's sip proxy :/
00:55.55Ariel_kyoo, no don't one is peer the other is user
00:55.58Ariel_for type
00:56.17Ariel_argh I will be back in a few minutes.... I need coffeee........
00:56.20otaku_p0pehehe.
00:56.35nick125btw, how do i tell if amp correctly added the sip user?
00:56.36Ariel_lets all take this to the #amportal
00:57.09Ariel_nick125, you can view the sip_additional.conf file
00:57.48nick125there isnt a sip_additional.con file :/
00:58.11nick125conf
00:59.28nick125let me guess, there should be one of these files, correct?
00:59.48*** join/#asterisk mago3-cn (i=maxgluck@200.109.166.172)
01:00.16Ariel_nick125, please go do a little reading here is a page that has amp and xlite on it with pictures....http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm
01:00.20kyoootaku_p0pe: Ariel_: Using the page at geekgazette, it is not working.  Thank you both very much.
01:00.26kyoos/not/now/
01:01.00surfduetrust me nick is trying his hardest
01:01.01surfdueon his own
01:01.03surfduei can hear him
01:01.42otaku_p0pekyoo: what isn't working ?
01:01.44otaku_p0pewait. it is working ?
01:01.45otaku_p0pedammit
01:01.49otaku_p0peheh. it's not for me :/
01:01.53Ariel_nick125, if the file is not there and not being writen when you create the sip extension and pressed the red bar then you have rights problems
01:02.04kyoootaku_p0pe: it is working!
01:02.08otaku_p0pegood !
01:02.21kyoootaku_p0pe: What part is not nowring for you?  In or Out?
01:02.25otaku_p0peeither
01:02.28otaku_p0peit can't register with BV
01:02.53*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone)
01:02.55kyoootaku_p0pe: You have had BV provision you a generic device?
01:03.53otaku_p0pethat's what they said.
01:04.17otaku_p0peoh hell
01:04.28file[laptop]brimstone: Matt!
01:04.37nick125ah
01:04.40nick125found the problem
01:04.43nick125permissions
01:04.52brimstonewerd up file[laptop]
01:04.59kyooEssentially, registering was the easiest part for me: Provision with BV, get then to mail me the password, and set up the trunk so it regestered with mynumber@sip.broadvoice.com:secret:mynumber@sip.broadvoice.com, and then set up sip.broadvoice.com in my hosts file...
01:05.17otaku_p0pehaha hell.
01:05.23*** join/#asterisk reagent (i=mathias@2002:d4fe:b1a1:0:20d:88ff:fef4:66bb)
01:05.24otaku_p0peso i use this password they generate for me; not my login
01:05.44kyoootaku_p0pe:  Correct!
01:06.37reagenthi. I set up askterisk as a SIP client (connecting to a SIP server and waiting for calls from there). When I run asterisk, it tells me that this channel is unreachable. How can I find out why?
01:06.49reagentit says: Peer '41325110504' is now UNREACHABLE!
01:07.03nick125whats the number to test asterisk IAX?
01:07.10bkw__reagent, any number of things can cause that
01:07.17bkw__but don't take that as the TRUTH
01:07.33reagentbkw__: Hm
01:07.52bkw__thats just a sip packet that didn't make it back or the end point didn't respond.. or went braindead
01:07.54bkw__its not a ping
01:08.00bkw__people seem to think that qualify is a ping
01:08.26Ariel_nick125, you mean 500 in the sample setup files
01:08.35reagentbkw__: Hm, maybe because it couldn't properly register?
01:08.36brimstonefile[laptop]: what are you upto?
01:08.39nick125ah yes
01:08.42file[laptop]brimstone: download tunes!
01:08.49bkw__reagent, could be
01:08.50brimstonedownload?
01:09.04nick125Ariel_: what file do i enable it in again?
01:09.10otaku_p0pehmm. so close i can taste it.
01:09.18file[laptop]downloading
01:09.24brimstoneooooohhhhh
01:09.32reagentbkw__: I just saw with SIP SHOW REGISTRY that in waits a long time while 'Packet Sent' and then the registration timeouts
01:11.17reagentbkw__: I can connect to the same proxy with xlite... something must be wrong :)
01:11.33adelashey how can i tell if my cisco 7960 phone is connecting to my pbx server?
01:11.39kyooIAX is more robust and efficient than SIP?
01:11.54bkw__depends
01:12.00bkw__that question would start a ware
01:12.05bkw__SIP scales better than IAX
01:12.11bkw__but that all depends on usage
01:12.17bkw__in some cases iax will win
01:12.34kyooFor small call volume. (2 calls simultaneous max)
01:12.56bkw__use sip
01:13.02bkw__unless you have nat
01:13.04bkw__then use iax2
01:13.11bkw__but in small volume its really up to you
01:13.23DarthClueiax/iax2 has issues that sip doesn't, sip has issues that iax/iax2 doesn't.  use what works best.
01:14.02Ariel_wow DarthClue well said
01:14.03kyoobkw_: NAT at both ends.  Essentially, wondering if SIP to VOIP provider to PSTN   or IAX2 direct to IAX2 is going to be better use of limited bandwidth...
01:14.07Ariel_ok got my coffee.
01:14.23Ariel_kyoo, nat both ends iax2 is better
01:14.30DarthClueAriel_: i'm barely awake, surprised i can even comprehend anything at this point.
01:14.32Ariel_but if done correctly sip will work.
01:14.50bkw__kyoo, use IAx2
01:14.57kyooAriel_: Which one will cope better with limited bandwidth?  (NO QOS routers in place yet.)
01:15.10Qwelland with iax2, you can trunk
01:15.10Ariel_kyoo, none
01:15.16Qwellnice little benefit
01:15.20Ariel_but iax2 will work better via dual nat
01:15.25kyooQwell: What does that mean?
01:15.31DarthClueif you have the exact same version of iax2 on each box.
01:15.40QwellDarthClue: oh?
01:15.50kyooDarthClue: They will be the same version of A@H. :)
01:15.56Ariel_lets not mix things up please
01:16.29Ariel_kyoo, if your talking between two asterisk boxes use iax2
01:16.51otaku_p0pethis is kinda cute.
01:17.24*** join/#asterisk L|NUX (n=linux@202.5.145.14)
01:17.27otaku_p0peit just rings forever.
01:17.31otaku_p0peon outgoing.
01:17.35kyooAriel_: I'm debating ig it is worth installing a second asterisk box. (They already have a PBX on location, but if it will help with bandwidth issues, I will attach a * box to the pbx for the time being...)
01:17.52kyoootaku_p0pe: You have registration now?
01:17.56DarthClueQwell: if you don't have the same version of iax2 on each box you will see issues.  sometimes consistent, sometimes random.
01:17.56otaku_p0pekyoo: yup
01:18.01otaku_p0pei was using the wrong pw ^_^
01:18.09kyoootaku_p0pe: Was it the "special" password?
01:18.13kyoootaku_p0pe: yup!
01:18.13otaku_p0peyeah :(
01:18.37kyoootaku_p0pe: From there, your "geek" instructions worked perfectly.
01:18.47Ariel_kyoo, well the bandwidth issue depends more on the codec your using. not if it's iax2 or sip.
01:18.48otaku_p0peheh.
01:18.57nick125in my firewall, what ports should i open up so other people can login to the system?
01:19.00Ariel_But if your trunking more then 3 calls on one connection iax2 is better.
01:19.19Ariel_5060/61 and 10,000 to 20,000 for sip
01:19.25kyooAriel_: Oh, of course.  Can I choose the codec, or does my VOIP provider or FXS do that for me?
01:20.02Ariel_kyoo, depends on provider bv only supports ulaw. But between your own box you can do what codec you have.
01:20.08otaku_p0peoh. do i need a DID route for the outgoing ? or incoming ? or am i drunk again ?
01:20.36kyooAriel_: Do other providers provide other codecs, or is ulaw a standard?
01:21.08Ariel_kyoo, others do provide other codec. But ulaw is the best due to sound. But it takes the most bandwidth.
01:21.57kyooAriel_: I heard somthing about a g-something-7-something codec in here earlier.  It is claimed to sound as good, but much better on bandwidth?
01:22.13Qwellg729
01:22.14nick1255060/61?
01:22.14adelashey dose anyone know a way to clear out like the asterisk in system, so that i can reinstall it and start over?
01:22.16nick125oh
01:22.16nick125nvm
01:22.22otaku_p0peoh this is odd
01:22.32otaku_p0peso it will ring, but when the other party answers the connection craps out
01:22.38kyooQwell: Are there voip providers I can use g729 with?
01:22.43Qwellsure
01:22.55otaku_p0peisn't g729 a pain with asterisk ?
01:22.55kyooQwell: Are thery considerably more expensive? ;)
01:22.59Ariel_rm -rf /var/lib/asterisk/modules
01:22.59Qwelldunno
01:23.12otaku_p0peAriel_: hee.
01:23.17nick125Ariel_: udp or tcp?
01:23.21Ariel_g729 is great codec. it's extra cost with asterisk it's not freeeee
01:23.26kyooAriel_: That sounds like a dangerous command to just throw out here. :)
01:23.29Ariel_udp
01:23.33nick125ah ok
01:23.44Ariel_they asked
01:23.51kyooAriel_: Oh, so they did. :)
01:24.11*** join/#asterisk grimse_ (n=grimse@p5481EF73.dip.t-dialin.net)
01:24.15Ariel_now it's up to people to use it correctly just a gun. The gun does not kill it's the person firing it.
01:25.01otaku_p0pehahaha.
01:25.35Drukenit's actually the bullet that kills
01:25.50Ariel_Druken, Symantec
01:25.54Druken:)
01:25.58otaku_p0pewait. broadvoice claims i don't need to open anything in my firewall
01:26.01otaku_p0peare they lying to me ?
01:26.07Qwellotaku_p0pe: usually
01:26.07Drukenunless you beat them with the gun
01:26.09Ariel_lier lier
01:26.14kyoootaku_p0pe: I have nothing open.
01:26.16otaku_p0pecause that would explain the issue with it crapping out when they other party answers
01:26.19otaku_p0pekyoo: does outgoign work fine ?
01:26.19leoncamelgood morning . :)
01:26.22kyoootaku_p0pe: (related to this anyway. :) )
01:26.40kyoootaku_p0pe: yes, perfectly.  (A little quiet...)
01:26.43otaku_p0peheh.
01:26.45Qwellotaku_p0pe: If your router doesn't pass related packets, it'll break entirely
01:26.54otaku_p0peQwell: like it won't ring even.
01:26.55otaku_p0pe?
01:27.04Qwellnothing will happen
01:27.08Ariel_if something does not see it how can it ring
01:27.14otaku_p0peyeah exactly.
01:27.17otaku_p0peok. so it's something else.
01:27.53otaku_p0pewierd.
01:28.58otaku_p0pei know. i bet i need to have some sort of human sacrifice going on.
01:29.15QwellWhat?  You didn't sacrifice a sheep?
01:29.24Qwellits in the install instructions...
01:29.26otaku_p0peno. just a lowly virgin.
01:29.29otaku_p0pe:(
01:29.42Qwellthose aren't mutually exclusive
01:29.47surfduehey i cant talk in xlite
01:29.49surfdueanyone know why
01:29.50Ariel_virgin's you mean there are still some available
01:29.51otaku_p0pehmmmm.
01:30.01otaku_p0pesurfdue: yes. something is not configured correctly.
01:30.06Ariel_xlite not working correctly.
01:30.11surfduecan anyone help?
01:30.13otaku_p0pexlite was a pain for me to get working.
01:30.20Ariel_xlite is easy
01:30.30otaku_p0peAriel_: yes but you appear to be a ninja.
01:30.35surfdueum
01:30.37surfduei can hear
01:30.39surfdueand everything
01:30.44surfduebut he cant hear me
01:30.50surfdueand i can see my bar goes up as i talk 2
01:31.09Drukenvirgins? ya 10 and under,... and sometimes not even then...
01:31.14otaku_p0peAriel_: hee
01:31.34twistedwtf
01:31.38*** join/#asterisk Barmal (n=info@c-24-92-153-118.hsd1.ga.comcast.net)
01:31.48otaku_p0peAriel_: any ideas on the issue i'm having though ? or places i could check for problems ?
01:31.52Ariel_nat
01:31.58otaku_p0peyeah it is nat'd
01:31.59otaku_p0pehmm.
01:32.08otaku_p0pei added an exter_ip thing in sip.conf
01:32.20otaku_p0pemaybe port-forward back in ?
01:32.24Ariel_externip=
01:32.25Drukennat can be such a pain
01:32.26otaku_p0peahh tcpdump. why didn't i think of you sooner ?
01:33.25Ariel_otaku_p0pe, it uses udp
01:34.04Ariel_otaku_p0pe, most of the sound ports are 10,000 to 20,000 rtp for sip
01:34.38nick125wee :)
01:34.41otaku_p0peoi vei
01:34.44nick125my asterisk works :)
01:34.50otaku_p0peso i have to like forward all of those ? that sounds odd.
01:34.52otaku_p0pebut hey
01:35.29Ariel_well sip uses them for sound iax2 only uses 4569 for sound and registration.
01:36.01*** join/#asterisk tengulre (n=tengulre@61.185.238.166)
01:36.08otaku_p0peyeah.
01:36.11otaku_p0pe<3 iax2
01:36.43otaku_p0peyeah that's it
01:36.49otaku_p0pei have to forward 10k-20k to my pbx
01:36.49otaku_p0peugh
01:37.04otaku_p0pethat is =ridiculous=
01:37.05Ariel_you want to hear them don't you
01:37.09otaku_p0peyeah.
01:37.12otaku_p0pebut guh.
01:37.37otaku_p0pei think that once i move i'll setup the pbx on a public ip, only with a bridged firewall between it and the internets
01:37.39Ariel_if you don't have too many sip users you can cut that down and edit the rtp.conf file
01:37.46*** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net)
01:37.46*** mode/#asterisk [+o bkw__] by ChanServ
01:37.58otaku_p0peyeah.
01:38.36Ariel_ok so now we have kyoo, nick125 and otaku_p0pe working finally?????
01:38.44otaku_p0peyeah !
01:39.22otaku_p0pesweeet.
01:39.23Ariel_NEXT! NEXT!  I take donations at arielb27@gmail.com paypal hehehehehe
01:39.32*** join/#asterisk santiago (n=santiago@63.245.86.202)
01:39.41otaku_p0pehey if i wasn't so poor my only phone line was a BYOD Lite i would totally do that Ariel_
01:39.44otaku_p0pe=totally=
01:40.27otaku_p0peoh i like how you =cau= use multiple SIP connections with broadvoice. just they charge you for it
01:40.56Drukennetwork engineer, that's as bad as a starving artist...
01:40.59Ariel_they allow you to have 2 channels with them.
01:41.23Ariel_Druken, yep but I have an house full of girls to feed.
01:41.44Drukenso you really are starving... :)
01:42.14otaku_p0peok i have some perfomance tuning
01:42.36*** join/#asterisk pbd (n=pbdavids@12.144.118.37)
01:42.40otaku_p0pebut it's mostly due to the fact that osX is a memory hog and i only have 512m of ram :/
01:42.52pbdEven', all.
01:43.06Ariel_otaku_p0pe, you can do noload on what you don't need in asterisk.
01:43.18Ariel_look at modules.conf
01:43.28Ariel_pbd, hello
01:43.50otaku_p0peyeah that might help too
01:43.51pbdAnyone out there seen a case of one-way DTMF tones on an h.323 (nufone) channel? (Head, of earlier this week)
01:44.27pbdRTP is fine, signalling is fine- asterisk RECEIVEs the oob dtmf, but doesn't SEND it.
01:44.32surfduewhats the extention to dial to get the main menu u setup
01:44.51Ariel_surfdue, amp?
01:44.52nick125if i want to get to the main menu (digital receptionist), how do i access that from the internal network (off my sip phone)?
01:44.55Drukendoes anyone know the guidelines for any other numbering plans than NANPA ?
01:44.57pbdAriel- good choice.  Unfortunately, I'm stuck with it- integrating with Cisco Callmanager.
01:45.15Ariel_argh dtmfmode=inband
01:45.15DrukenH323 is EVIL!
01:45.23Ariel_h323 is not evil
01:45.32pbd(and yes, I know I can use SIP- I do, and it works fine- but CCM only does compressed codecs with h.323, not SIP)
01:45.35Drukenis in my mind...
01:45.42Ariel_digital recp 777 in amp
01:45.45pbdCan't use inband- I'll be using 729 in the final mix.
01:45.46nick125is there an extension to access it?
01:45.59Ariel_rfc2833
01:46.20pbdI've got that set- and it works, INBOUND... but doesn't send it.  Codecs appear negotiated correctly.
01:46.31pbdI'm halfway to filing a bug on mantis.
01:46.37Drukennick125: try dialing the main incoming number?
01:46.41otaku_p0pepdb : i read that as martinis
01:46.45Ariel_pbd, there is a channel called #nufone
01:47.11Ariel_no one seen what I typed   777
01:47.17nick125_lappyDruken, what if its a pstn number?
01:47.24otaku_p0peso incoming doesn't work.
01:47.30otaku_p0pe<3 rfc's
01:47.31nick125_lappybut, the pstn number isnt setup yet
01:47.50pbdYeah, but that's for nufone's business services.. this is an asterisk channel driver issue, and, like it or not, nufone's channel driver is what's packaged in CVS- and the one I've found actually works best.
01:47.54Drukennick125: well if your dialplan is done correctly, it'll route it to where it needs to go
01:48.19nick125_lappyooo, would DID routes work?
01:48.31nick125_lappyget the DID number to something like *10, and forward it to the menu?
01:48.48nick125_lappywill that work from the internal phones?
01:48.53*** part/#asterisk Uther_P (n=uther_p@66.180.120.82)
01:49.15Ariel_nick125_lappy, in what program?
01:49.38nick125DID in AMP, using xlite as a sip client to my asterisk
01:50.38Ariel_nick125, if you dial exten 777 it takes you to the digital recp.
01:51.13*** part/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
01:51.22Ariel_nick125, did route work if you have a trunk for inbound setup
01:51.28*** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
01:51.31nick125so, i would type '777' into xlite, and it would treat it as a incoming pstn call?
01:51.36*** part/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
01:51.44*** join/#asterisk TeknoKosh (n=evangeli@ip157.net65.ipnetworks.net.au)
01:51.47Ariel_nick125, yes
01:51.52*** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net)
01:51.56nick125_lappyneato
01:52.36nick125does asterisk have like a guide for all these codes and everything lol
01:52.43otaku_p0pewhat does BV need udp:69 for ?
01:52.47Ariel_rofl
01:52.51Ariel_~doc
01:52.51jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
01:52.51Qwelltftp?
01:53.10*** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
01:54.49Ariel_otaku_p0pe, that is for if you use an ata with them.
01:55.01otaku_p0peoh ok.
01:55.05otaku_p0pehmmm.
01:55.14Ariel_you don't need it with an asterisk box
01:55.49otaku_p0peso who wants to call my number
01:56.27*** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net)
01:57.51*** join/#asterisk BrainSurg (n=paul@69.158.227.13)
01:58.03BrainSurgGreetings.
01:58.08otaku_p0pehi
01:58.41BrainSurgotaku_p0pe: How goes it?
01:58.47otaku_p0penot too bad.
01:58.56otaku_p0pejust fighting with incoming on my shiny new sip trunk
01:59.25*** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net)
02:00.17BrainSurgnifty.
02:00.25*** join/#asterisk Tili (i=Tili@219.137.38.125)
02:00.31BrainSurgGot any pointers to setting up a basic asterisk config?
02:00.39BrainSurgI'm just starting to play with it.
02:01.42otaku_p0peyes. use the asterisk management portal
02:01.46otaku_p0peor just go for asterisk@home
02:01.52otaku_p0pethat's what iw oudl have done, but i didn't have a spare pc
02:01.55pbdBrain: Spend a few quality hours with www.voip-info.org
02:02.02otaku_p0pewhich is somewhat annoying
02:02.25krisguyI'm glad govt auctions happen quite a bit around here
02:02.39krisguygreat way to get spare boxen cheap
02:02.44otaku_p0peyeah
02:02.47otaku_p0pei actually have a spare box now
02:02.49Ariel_ebay
02:02.52otaku_p0pebut no time to swap the system :D
02:02.57otaku_p0peAriel_: see previous statement about being broke
02:03.10krisguyI started with Asterisk@Home, good way to get wet
02:03.11Ariel_otaku_p0pe, so am i
02:03.16otaku_p0pehehe
02:03.21Ariel_so what did you use vmware?
02:03.26BrainSurgkrisguy: What is Asterisk@Home?
02:03.36*** join/#asterisk wulfy814 (n=lorentz@c-67-165-37-20.hsd1.pa.comcast.net)
02:04.03Ariel_BrainSurg, it's a ISO that comes with OS , asterisk and a nice gui and many more nice things for asterisk.
02:04.08krisguyISO with CentOS, Asterisk, MySQL, etc.
02:04.13BrainSurghmm.
02:04.24nick125since sometimes a softphone is kinda diffcult
02:04.27BrainSurgGot Ubuntu running with asterisk installed already:)
02:04.35krisguynick125, shouldn't be a problem
02:04.36Ariel_nick125, yes
02:04.47Tiliis the challenge sent by asterisk always the same or something random
02:04.53Ariel_BrainSurg, ok then see the docs and make samples
02:04.53krisguyATA boxen will work with Asterisk
02:04.58Ariel_~doc
02:04.58jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
02:05.25BrainSurgjbot: thanks.
02:05.25jbotmy pleasure, BrainSurg
02:05.50krisguyBrainSurg: voip-info.org is a godsend
02:05.57BrainSurgI'll check it out.
02:06.07wulfy814BrainSurg: how about "make config" it didn't work for me in Ubuntu
02:06.36nick125i might see if i can find someone with a ATA box or buy one on ebay or something
02:06.38Ariel_wulfy814, that is due to make config is more for rh type of system. ubuntu is debian based
02:07.00*** join/#asterisk santiago (n=santiago@63.245.86.202)
02:07.00Ariel_nick125, make sure it's unlocked one
02:07.23BrainSurgwulfy814: I just put asterisk on using apt.
02:07.35Ariel_rumor has it the adp-get has one some place
02:07.51Ariel_apt-get
02:07.58otaku_p0pehmm
02:08.07Ariel_argh I need to go to bed.... coffee did not help.
02:08.15otaku_p0pewould a pIII/500 with 256m be good enough for asterisk ?
02:08.20Ariel_yes
02:08.24otaku_p0pethought so
02:08.35otaku_p0peyeah. so i just have to migrate all the apache crap off here and it will be fine
02:08.41Ariel_otaku_p0pe, I have seen a wrt54g running asterisk
02:08.42otaku_p0peoccasioally you get pop's and stuff
02:08.47otaku_p0peAriel_: hott
02:08.57nick125whats a good, free, low b/w codec for asterisk?
02:09.04wulfy814Ariel_ : ok, hints on how to make asterisk start automagically in ubuntu then?
02:09.04Ariel_gsm
02:09.13Ariel_g726-32
02:09.17nick125that would be able to support least 5 calls at once on a 1mb up/down line
02:09.21wulfy814BrainSurg: you just needed to do an "apt-get install" right?
02:09.27Ariel_ilbc works also but has heavy cpu load
02:09.41BrainSurgwulfy814: yeah
02:09.43adelasAriel_ will you be my best asterisk friend :P
02:09.44Ariel_nick125, 5 calls ulaw would work
02:10.06Ariel_adelas, ????
02:10.09BrainSurgwulfy814: Autostarts, too.
02:10.22adelasneed amp help :P
02:10.57wulfy814BrainSurg: zaptel hw or ztdummy?
02:11.17Ariel_adelas, what is your issue with amp?
02:11.25*** join/#asterisk Mycroft1 (n=reece@whatda.whatthe.net)
02:11.31Mycroft1hi pplz :)
02:12.20adelaseverything ;)
02:12.34adelasthe AMP installation guide is way too outdated :|
02:12.36adelasas it seems heh
02:12.55Ariel_do you have a box just for the asterisk?
02:13.21adelasyes
02:13.44Ariel_then get yourself a major time saver and put Asterisk@home on it.
02:13.47adelasi'v beeen jumping back from fedora with asterisk to asteriskAtHome, then asteriskwin32
02:13.55adelasthen now i'm back to fedora :|
02:14.09adelasi have asterisk running
02:14.17adelasjust now the amp stuff
02:14.23Ariel_fedora is beta from RH CentOS is actual RHEL 3
02:14.37Ariel_adelas, amp needs lots of other things added.
02:14.40*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
02:14.43adelasyea thats the thing :|
02:14.58adelasi did all that "yum install package" stuff
02:15.19adelasfollowed the steps, then bam it skips things and tells to do things out of nowhere
02:15.30Ariel_adelas, did you install asterisk::perl, and all the others as well?
02:15.42adelasyea
02:15.54adelaswait, u mean perl, and perl-CPAN
02:16.08adelasor a acutally "asterisl::perl"
02:16.35adelaso wait, i did do like perl -MCPAN -e "install IPC:Siglal"
02:16.38adelasnand stuff like that
02:16.41adelasand*
02:17.04pbdOk, I'm a cvs dummy.. can anyone tell me how to check out the Makefile for -HEAD from Digium?
02:17.52Ariel_pbd,  it comes with asterisk when you do the co asterisk
02:18.00*** join/#asterisk ChArLeS___ (n=charles@64.35.168.55)
02:18.03ChArLeS___wow
02:18.09pbdAriel_: Not my question, but thanks anyway.
02:18.53adelasAriel_ could u help me with AMP?
02:19.01ChArLeS___Hey
02:19.05adelaslike how to get started heh
02:19.12ChArLeS___Why asterisk forks for each IAX call ? is it normal ?
02:19.48pbdMy -HEAD makefile became corrupted, and I wanted to just get a new one.
02:20.07pbdFortunately, I have more than one server with -HEAD on it, so I grabbed it from there.  but.. eeww.
02:21.57otaku_p0pevictory !
02:24.04hardwirehmm
02:24.05hardwireso
02:24.06hardwireiaxy
02:24.07hardwireI have one
02:24.19hardwireI am hoping to make it more or less auto dial when the line is picked up
02:24.57pbdCan you also use yours as a handy stove/furnace replacement?
02:25.05mago3-cnhello, how can i change the caller id, depending on what it is?
02:26.49pbdHardwire:  I believe the current quote is- your watch has more processing power than an Iaxy.
02:27.02hardwireI wasn't worried about the iaxy :)
02:27.21pbdmago- can you be more specific as to what you're trying to do?
02:27.38pbdChange the caller id based on what WHAT is?
02:27.42hardwireas soon as I pickup the phone it gives  Accepted AUTHENTICATED TBD call from 10.10.0.95
02:27.57hardwireI just want it to patch through on the asterisk side to a given number
02:28.19pbdhardwire- that will depend on the context you have the iaxy dropping into.
02:28.28hardwirelocal
02:28.38hardwirewhich has all sip phones
02:28.38*** join/#asterisk frank_sbr (n=frank_sb@modemcable175.124-70-69.mc.videotron.ca)
02:28.51pbdRight.. but what does local do on the no-match case?
02:28.51frank_sbrhi
02:29.08hardwirepbd: nothing.. and wouldn't asterisk respond if there were a no match case?
02:29.12frank_sbrdoes anyone knows how to change ilbc 30ms to 20ms encoding length
02:29.15hardwirelike
02:29.18hardwireno match found.. blah.
02:29.32hardwireactually.. what is the extension code for no match found.
02:30.45hardwirejust s
02:30.46hardwireor o?
02:31.00pbds would work.. but what's your iax.conf say?
02:32.01hardwiregood enough
02:32.06hardwirepastebin.. slow
02:32.16mepplgute nacht - good night
02:32.53hardwirehttp://pastebin.com/340417
02:32.57hardwire.ca was too slow
02:33.16hardwireiaxcompat=yes in general
02:33.23hardwireand just codec stuff after that
02:34.25nick125hmm
02:34.30pbdI've got an s,1,Answer in my gneral context..
02:34.40nick125when i try to dial 777, it says '484 address incomplete'
02:34.42pbdthat should catchall everything that winds up there.
02:34.47JerJerMXC is so goddamn funny
02:34.59hardwirepbd: yeh.. but the iaxy is waiting to interperate some dtmf
02:35.20hardwirethen it gives me the congestion.. all without talking to asterisk
02:35.24pbdJerJer- since you're awake, can you answer a quick works/noworks question for me for the nufone h323 driver?
02:35.27hardwireafter a long dely at dialtone
02:35.48JerJerdepends
02:35.49pbdhardwire- ok, so the iaxy isn't configured correctly.. for that, I'd have to have one around to mess with.
02:35.59nick125anyone know why 777 isnt working?
02:36.09JerJerno luck tonight
02:36.22hardwirepbd: there are no real options for this in the iaxy :)
02:36.38pbdJerJer: Situation is outbound (HEAD to cisco callmanager in this case, via gatekeeper) out of band dtmf.
02:36.55JerJerrfc2833
02:36.57pbdJerJer: While * is accepting oob dtmf, it doesn't seem to be sending it on.
02:37.09pbdConfigured for rfc2833 all around.
02:37.10Mycroft1has anyone ever seen this error before, i cant find any info on it and i get 400 of these messages on my console per second chan_iax2.c:2455 iax2_read: I should never be called!
02:37.38JerJerdon't call it
02:37.40pbdhardwire: have you done an iax show peers?
02:37.56hardwirethe iaxy connects to the pbx
02:38.07hardwireI can dial to any regular extension thats in my context
02:38.11hardwirevia a regular phone
02:38.16hardwireI just need a bat phone
02:38.24Mycroft1yeah iax2 show peers is fine
02:38.34hardwireoh.
02:38.34hardwireheh
02:38.37hardwirebleh
02:38.39pbd(ok, so I dropped the 2)
02:38.53Mycroft1thats cool :)
02:39.07Mycroft1its been driving me nuts because all the calls start getting audio gaps while it does this
02:39.10Mycroft1i cant find the problem
02:39.13pbdJerJer: ever heard of the dtmf problem before, or did I dig up something new?
02:39.21Mycroft1it just started when i upgraded to 1.0.8
02:39.36pbd(or worse, it's my config- which  doesn't seem likely right now, but is generally the case)
02:40.19nick125now to find some hold music :P
02:40.31drraynick - use a dialtone.mp3 :)
02:40.53pbdMycroft- I can't say that I've seen that one.. but it could be a bad compile.. you're running .8, downloaded as tarball?
02:40.57nick125lol
02:42.31pbdThere's always Allison's rendition of louie louie.. but I'll bet you want callers to wait on hold, don't you?
02:42.33hardwirewhats 'phonecore'
02:42.39nick125lol
02:42.40hardwirehttp://asterisk.gnuinter.net/files/digium/phonecore/
02:43.17pbdIt's that little place in the middle of the dial where they put the paper with your number typed on it.
02:44.03Mycroft1pbd: its running on freebsd so from the ports tree
02:44.28pbdMycroft: You like punishing yourself, don't you?
02:44.32*** join/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg)
02:44.47Mycroft1pbd: linux does not like me :)
02:45.09pbdMycroft- I'd almost guarantee that it's some sort of bad compile issue- but I don't run the bsd version, so there's not much else I can say.
02:46.37nick125i wonder why hold music crashes asterisk :/
02:46.53Mycroft1pbd: possable i dont have zaptel running, but everything seems to be working MOH and MeetMe
02:47.21Ariel_nick125, did you remove the id tags from the mpg files you uploaded to asterisk box?
02:47.26hardwirenick125: its not real music.
02:47.27Mycroft1then again i couldnt see ztdummy in the make file and since this box doesnt have a digium card it was pointless to install it
02:47.32pbdnick: likely that your zaptel is messed up, and it can't provide adequate timing.
02:47.55nick125Ariel_: oh that might help
02:48.24pbdMycroft- the ztdummy stuff is in the zaptel package makefile.. but if you don't need moh or meetme, you're right- who cares.
02:48.41*** join/#asterisk jeobjeobjeob (n=jeobjeob@pool-70-111-134-244.nwrk.east.verizon.net)
02:48.43jeobjeobjeobhey
02:48.44pbdNow, it *might* affect other timing stuff, if the kernel is off- but again, I'm no bsd expert.
02:48.54jeobjeobjeobim running into an issue with cdr and dial
02:49.04jeobjeobjeobwhen i dial another number, is it possible to have the cdr forked?
02:49.07jeobjeobjeobin a way so that
02:49.32Mycroft1pbd: zaptel timing wouldnt be causing an iax2 issue though would it
02:49.32jeobjeobjeobthe source is the system, and the destination is the phone that picks up the call (in the case of a simulring)
02:49.35pbdjeobjeob: Anything is possible, you've got the source and a c compiler, right? ;-)
02:49.52Mycroft1pbd: i remember back in the old 0.x days it was very touchy
02:50.06pbdMycroft- No, but asterisk uses zaptel as a timing source, if it can... so if that's off, little gaps could occur.
02:50.06jeobjeobjeobbetter question is: where does the channel merging occur?
02:50.43ChArLeS___ps auxwww |grep asterisk |wc
02:50.43ChArLeS___<PROTECTED>
02:50.49pbdWith modern (linux 2.6) kernels, the kernel has a high enough resolution timer to make the ztdummy stuff obsolete.
02:50.51ChArLeS___why each call is a fork ?
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02:51.16pbdBut you're running bsd- so... no clue.
02:51.23Mycroft1pbd: might explain the audio gaps sometimes
02:51.43Mycroft1i might try over the weekend, move some of the calls off to another asterisk box and see what i can do with it :)
02:51.49Sedoroxzaptel + bsd = bad voodoo
02:52.07Sedoroxtho I never had problems with it
02:52.21Mycroft1its just a pain all the other asterisk linux to freebsd moves worked seamless except the last one
02:52.22pbdMycroft- but your asterisk is sick in general, if it's giving you error messages in the iax.c code.
02:52.25Mycroft1typical :)
02:52.42Mycroft1i cant explain it
02:52.44Sedoroxyea... always the last one
02:52.49Mycroft1all the other 38 asterisk boxes are fine
02:52.58Sedoroxmaybe ya got bad memory in it.. and it messed the compile up..
02:53.08Mycroft1maybe
02:53.13pbdSuggestion?  Stick with 38 boxes. :)
02:53.23pbd39 being a multiplier of 13.
02:53.26Mycroft1i might roll the ports tree off another one thats working and the working directory
02:53.28Sedoroxlol
02:53.29Mycroft1see if its any better
02:53.49pbder.. multiple.  It's late, time for me to go- I've been beating my head against this dtmf issue too long today.
02:53.55Mycroft1pbd: this last one is for the office in melbourne
02:54.23Mycroft1not dtmf arrgg, i still feel the pain from that with the cisco 5300's
02:54.50niZonmmmm message waiting indicator
02:54.50pbdMycroft- you ran dtmf's to cisco equipment via h.323?
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02:55.15Mycroft1nope sip
02:55.35Mycroft1we also found that we had echo after we got to 100 calls
02:55.40Sedoroxfrom what I've seen in here... h323 seems like the devil...
02:55.43nick125would it be better if i put my hold music as wavs ?
02:55.44pbdMycroft- rats.  Sip to cisco for me works like a charm- if only Cisco supported compressed codecs under SIP in callmanager.
02:55.49Mycroft1we tried new, ios's firmware on the voice cards
02:55.50Mycroft1everything
02:55.56Mycroft1only a reboot gets rid of it
02:56.10pbdI've got to make these connections from Brazil to the US- no way in heck I'm trying that with ulaw/alaw.
02:56.34JerJerwhy with H.323?
02:56.56pbdCallmanager supports g.729 under h.323 trunks.
02:57.16JerJercall munger also support SIP trunks
02:57.23Mycroft1pbd: do you run into echo on the calls at all ?
02:57.33SedoroxJerJer, [22:55] pbd Mycroft- rats.  Sip to cisco for me works like a charm- if only Cisco supported compressed codecs under SIP in callmanager.
02:57.36pbdSupports them yes- but *only* for ulaw/alaw codecs.
02:58.06nick125wav would mean less stress on server then mp3, right?
02:58.44Sedoroxactually I think raw gsm format is less..
02:58.48pbdMycroft- No, no echo whatsoever.. but it's still a new app- yet to be stress tested.
02:58.53Sedoroxthats how I have my ivr...
02:58.54nick125for hold music that is
02:59.10nick125since wav doesnt have to be decoded
02:59.13Mycroft1pbd: how many calls are you doing at peak
02:59.17Sedoroxmoh.... *think* I've done mp3's... dunno what the load was tho :p know it did get too high sometimes..
02:59.24Mycroft1pbd: we ran nicely at 20 calls without issue
02:59.34nick125on mine, with one call, it skips and all
02:59.42pbdMycroft- under the -STABLE branch, I've seen 30-40 concurrent h.323 calls from CCM to Asterisk using the h.323 driver.
02:59.56pbdAnd they all wind up in meetme, to boot.
03:00.57pbdNow, I can't run calls from * to CCM under -STABLE- they have to originate on CCM. -HEAD fixes that (there's a bunch o bugnotes with my name on them in mantis to mark each step of that way)
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03:01.49pbdAnd, until yesterday, I was ready to go- then I just HAD to test it with a little app I've been working with that does some PIN authentication.. and found that the dtmf doesn't pass from * to CCM.
03:01.51Mycroft1i think we got to 45 - 50 and we started having echo
03:02.13Mycroft1pbd: no dtmf issues using h.323
03:02.27Mycroft1i might... look back into it, ive got 4 of these 5300's sitting in a store room
03:02.33pbdMycroft- it's possible- although, my users won't get to that many concurrent for quite some time.
03:02.43Mycroft1we went to all digium cards once echoing came up
03:02.53Sedoroxsend a 5300 this way :p
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03:03.26Mycroft1sedorox: they aren't mine they're my works :)
03:03.51Sedoroxdarn :/
03:03.54Sedoroxif you wanna sell them :p
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03:04.54Sedoroxmy rommate and I are putting together a small cisco lab for our room..
03:05.03Sedoroxso I'm trying to grab whatever I can :p
03:07.43adelasdoes asterisk support voice recording?
03:07.46pbdWhen we've found echo, we've always traced it to a bad device on the other end.
03:08.00pbdAnyway, I'm outa here.  Tomorrow is another DTMF day. Ick.
03:08.27Mycroft1pbd: i put asterisk on some powermac g5's to see how fast they are
03:08.34Mycroft1compaired to p4's
03:08.40Mycroft1its quiet interesting
03:08.52pbdI'm guessing I'm gonna have to start adding debugs to the h.323 channel driver again.  It acts as if it just never sees the digits- although the pbx channel debug claims it's handing them off to the channel driver.
03:09.22pbdSo.. it could be with the pbx core itself, or it could be with the h.323 driver.. or it could be my config.
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03:09.46Mycroft1pbd: so stay away from h.323 and cisco for the moment ? :)
03:09.56shido6LOL
03:10.10SwKMycroft1: finding anything interesting on your comparisons?
03:10.12pbdFor once, I think it's not the cisco side of the equation.  Not usually the case.
03:10.13Mycroft1i might take one home tonight and have a play see if i can get the h.323 stuff to work with a 5300
03:10.39pbdIf you come up with anything, I'll be around the channel for the next week or so.
03:10.43Mycroft1SwK: a G5 1.8 SP seem to be able to hold less load per volume of calls
03:10.44SwKi would just be happy if I could get t38modem to work witha  5200
03:10.46shido6erf
03:10.49shido6whats wrong ?
03:10.56Mycroft1SwK: vs a 2.8 ghz p4
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03:11.04SwKMycroft1: what OS you running on the G5?
03:11.05shido65300 and h323 works - less someone changed and fx0red something
03:11.12shido6im in on the tail end of the convo tho
03:11.20Mycroft1SwK: i tried darwin and freebsd
03:11.32SwKMycroft1: didnt try OS X?
03:11.38pbdshido- make sure you test that dtmf is working both ways through the channel driver.. that's my wall right now.
03:11.41Mycroft1SwK: darwin is osx :)
03:11.48shido6bleh
03:11.50SwKdarwin != OS X
03:11.51pbdshido- which channel driver are you using?
03:11.57shido6I smacked into that wall a while ago and fixed it
03:11.58SwKDarwin is a subset of OS X
03:12.06shido6the Secret is a single line in h323.conf
03:12.08elvisthedjanybody got a tip on echo cancellation on an iaxy2? (zap not involved.. calling out via sip, no echo on my (iaxy2) end, but the other party hears an echo
03:12.11shido6I use chan_h323.so of course <--- NuFone
03:12.15elvisthedji've seen the question on the list, but no answer
03:12.22Mycroft1Darwin reece-andersons-powerbook-g4-15.local 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc
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03:12.32shido6if you look closely at the soure
03:12.34pbdshido- and the line was?
03:12.40shido6you 'll see a line for dtmf
03:12.44shido6its hidden in the code
03:12.48shido6let me go look again....
03:13.14pbdI only saw two dtmf related lines in the source I was looking at today- dtmfcodec and dtmfmode.
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03:13.17Ferris_BEvening, all. Anyone here use sixtel?
03:13.24JerJerdtmfcodec ?
03:13.28JerJerthat is documented
03:13.41SwKDarwin bunghole 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc
03:13.41pbdCodec, to me is irrelevant, although I set it to 101 anyway- but I'm using out of band signalling.
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03:14.11pbddtmfmode I've got set to rfc2833 for the out of band stuff.. I can't use inband.
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03:14.20spacklejerjer: do you have any tips for troubleshooting line noise on a PRI?  I have static or crackles on some channels.
03:14.33pbd(I'm using g.729- inband doesn't work with compressed codecs)
03:14.44spacklejerjer: using a quad span digium card.
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03:14.50JerJerspackle: sharing interrupts?
03:15.08pbdspackle: running on any hardware that has a blue Dell on the front? ;-)
03:15.25spackleI don't think so, It's running on a proliant DL380 G2.
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03:15.48spackleEr, I don't think it is sharing interrupts, that was my first thought too.
03:15.49shido6; You may also specify on either a per-peer or per-user basis below.
03:15.49shido6;dtmfcodec=101
03:15.51shido6is the secret
03:15.57JerJerthat's not hidden
03:16.06shido6if u had 6 yagers
03:16.07shido6it is
03:16.19shido6h323.conf.sample
03:16.21JerJer; Default RTP Payload to send RFC2833 DTMF on.  This is used to
03:16.21JerJer; interoperate with broken gateways which cannot successfully
03:16.21JerJer; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
03:16.26pbdshido- it has to be commented out, too? ;-)
03:16.29JerJer; You may also specify on either a per-peer or per-user basis below.
03:16.29JerJer;dtmfcodec=101
03:16.32shido6no, pbd :)
03:17.43pbdJerJer- according to the h.323 trace, it negotiates a payload type 46, dtmf payload- and it accepts it from the other side.. but it won't send it.  Refuses- no errors, just swallows it.
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03:18.40spacklenope, the card is alone on interrupt 7
03:18.40Sedoroxniceeeee
03:18.40shido6excuse me
03:18.51pbdAhh.. Zotob is doing it's work again, wiping out parts of the net.
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03:18.54JerJerpbd:  then don't force a dtmfcodec
03:19.25twistedomg
03:19.41twistedthat was a friggin tsunami of a split
03:19.41Sedoroxlol
03:19.41twistedgood thing i had my surfboard out
03:19.48Sedoroxnot for us it wasn't :p
03:19.48Sedoroxfor you it musta been tho
03:19.56twistedwell
03:19.56Sedoroxonly dropped about 100 people..
03:20.04twisted362 in the channel rigt now
03:20.05Sedoroxfrom this chan...
03:20.05twistedit went down to 102
03:20.13Sedoroxyea...
03:20.14adelasdoes asterisk support voice recording?
03:20.22Sedoroxit want to like 285 or something
03:20.23pbdAdelas: yes.
03:20.27pbd~doc
03:20.28jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
03:20.28twistedmust have been a multi-way split
03:20.33SedoroxManxPower, its more like "the servers are splittling" (and repear)
03:20.39Sedoroxdunny
03:20.40pbdJerJer- removing it has no different effect.
03:20.41Sedoroxdunno*
03:21.06adelasdoese asterisk also support conferencing?
03:21.25ManxPoweradelas: "show application meetme"
03:21.42twistedadelas, nah.  we don't need no stinking conferencing :P
03:21.53adelasno, seriously :P?
03:22.02twistedyes
03:22.18adelasyes its supported :)?
03:22.27file[laptop]conferencing, what's that?!?
03:22.30pbdManx/Adelas: MeetMe application won't show up without a zaptel compiled (won't compile in)
03:22.44twistedi wouldn't say supported, but it's there ;)
03:22.48ManxPowerpbd: I know.
03:22.53SwK*yawn8
03:23.02twistedand it works
03:23.06twistedSwK, stfu
03:23.09ManxPowertwisted: last I saw meetme won't even build without zaptel installed.
03:23.09SwKnetsplits rule
03:23.22twistedManxPower, i didn't say it did
03:23.44pbdNow, for the real question- does Asterisk support videoconferencing?
03:23.58SwKhah
03:24.10pbdAnswer: Yes, but it isn't supported. ;-)
03:24.11file[laptop]it supports it in another universe
03:24.14ManxPowerpbd: HAHA!  That's a good one.  Here, I have one!  Does Asterisk support T.38?  HAHAHAH!
03:24.27twistedManxPower, nono, i have one better
03:24.32ManxPowerWhat other Asterisk jokes are there?
03:24.36JerJerpbd: how are you determining the dtmf doesn't go anywhere?
03:24.38twistedManxPower, can asterisk effectively replace a 5ess?
03:24.56file[laptop]or: can Asterisk make me the next Vonage?
03:25.00ManxPowertwisted: I'd not heard that one before.
03:25.04twistedManxPower, lol
03:25.09*** join/#asterisk gopherspidey (n=webb@dsl-musc-15-031.machlink.com)
03:25.11SwKasterisk can make you the next vonage
03:25.14ManxPowerfile[laptop]: Of course it can!  Just use your DSL!
03:25.18pbdJerJer: Couple of ways- observation- the application on the other end doesn't receive it.. but through h.323 trace 8, and channel debug, I see the event registered, but no pdu going out.
03:25.27ChArLeS___ManxPower: I got to see that if using g723 each call is a new fork man
03:25.42ChArLeS___ManxPower:  2.4ghz can barely handdle only 12 calls
03:25.45pbdJerJer: Yet, if the other side of the conversation (same channel) sends a dtmf tone, I see the pdu come in, and the channel debug registers it.
03:26.02ManxPowerChArLeS___: only mentally instable people use G723.1
03:26.09Qwell12 calls on 2.4ghz?
03:26.22file[laptop]G723.1 is a verrrrry CPU intensive codec
03:26.27ManxPowerQwell: Well, the ITU codec isn't optimized, and it's a cpu hog.
03:26.38pbdCharles- er.. are you doing massive amounts of transcoding on that doom server, too?
03:26.43ManxPowernot to mention being illegal in much of the world.
03:26.47SwKi love getting requests for 723
03:27.08twisted723?
03:27.14SwKtrasncoding g729 <-> g723.1 is the bestest
03:27.19file[laptop]O.O
03:27.36Netgeeksyou never know, he may have paid for a g723 license!  thats why he's running a 2.4GHz system!
03:27.55pbdWhat's a 723 license run these days, $1K/ channel?
03:27.58file[laptop]yeah and my name is Mr. Muffin Man
03:28.04Netgeeksoooh, I see some digium people here.  I have a REALLY dumb arse question
03:28.58Netgeeksis a pci express 16 slot backwards compatable so that you can plug a TE card into it?  /duck
03:29.38JerJerpbd: submit a bug with massive detail
03:29.47ChArLeS___We bought G723 not G729 I don't have G729 so I have to use what my boss got.
03:30.30pbdJerJer: That was my thought- but I wanted to make sure I wasn't just wrong in my config somehow- dtmf seems like the world's simplest setup, yet this problem is really odd- Id think someone else would have seen it.
03:30.34ChArLeS___probably G729 is also CPU intensive
03:30.48file[laptop]you bought G723... yeah...
03:30.52pbdCharles- IMHE, 729 isn't that bad.
03:31.25ChArLeS___I can handle more than 60 calls on G711 on that
03:31.27ChArLeS___box
03:31.35file[laptop]the interweb is crashing! oh no
03:31.41ChArLeS___file[laptop]:  what's the matter ?
03:31.51JerJerpbd: however, I did a quick test on the H.323 load for 7960s and DTMF worked as expected
03:32.01pbdSure, 711 has no compression at all- if you're not transcoding it, you should be able to support a *lot* of calls.
03:32.07file[laptop]Asia is really sucky right now for routing
03:32.08*** join/#asterisk nick125 (n=nick125@unaffiliated/nick125)
03:32.26file[laptop]52% average packet loss, 670ms average response time
03:32.27*** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfk24.dialup.mindspring.com)
03:32.38ManxPowerI'm a fan of G726 and Speex
03:32.42file[laptop]lots of routers are down there it seems :(
03:33.01pbdJerJer- I haven't played with the h.323 load for a 7960, I'm running * with 7960's attached via SIP.  If my channel is connected to CCM via SIP (testing, I can't run that in production), dtmf works fine.
03:33.31ManxPowerpbd: can you make your cisco send payload 101?
03:33.36pbdBut, doesn't matter if I use the 7960, or an IAX-attached softphone, outbound * via h.323 doesn't wanna talk oob dtmf.
03:33.40ChArLeS___I'm making a SIP -> IAX ->IAX -> SIP gateway
03:33.53QwellChArLeS___: why?
03:34.28Juggieif anyone uses agi extensively, could you please chime in on bug# 4854 on the bug tracker.... plz tnx.
03:34.30pbdTomorrow, I'll pull together the relevant traces and submit it.  Hard to show- there's nothing in the logs to show that it's failing- it just doesn't happen.  If there were error messages, I'd be happier.
03:34.31*** join/#asterisk oplog2 (n=oplog2@206.222.29.50)
03:34.52ManxPowerJuggie: Honestly I don't think Asterisk should work around the oddities of every language out there.
03:35.19ChArLeS___Qwell:  to save bandwith
03:35.31pbdManx: I'd have to research that one.  I will say that, if I call out from the 7960 through a Sipura to PSTN, then back in to my CCM box, all works well there, too.
03:36.22pbdManx: the 7960's dtmf options are differently defined- the choices are inband and 'avt' - avt works for rfc2833.
03:36.36JuggieManxPower, it still isnt reversed in stable...and even so, on head i dont feel a dont send sighup agi flag would be unreasonable.
03:36.50Juggiethe suggestion that i should load perl to disable sighup, and then run php inside is lame i wont do that
03:37.02Juggiei would do a custom compile of php to enable thread control first.
03:37.10pbdNet effect is, my testing has narrowed it down to use of the h.323 channel driver, with a working call, will receive but not send oob dtmf via rfc2833.
03:37.14*** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com)
03:37.28ManxPowerJuggie: I would prefer something that allowed you to specify which signals to send via SetVar before calling AGI
03:37.37pbdWhich sounds like a nice, clean bug.. if there ever is such a thing.
03:38.08Juggiemanx, that would work for me too, i dont really care how i can enable/disable it, as long as it can be done via dialplan logic.
03:38.24HogieI threw away 250k lines of bleh today, and starting over from scratch and have to the 9th to build it again.  I dont think I'll do it
03:39.22pbdHogie- sounds like you need more monkeys and typewriters.
03:39.51Hogiethat's actually kinda funny, since I had them change my business cards to "Network Monkey Boy"
03:40.10Hogiegot 500 of them printed that way, and the president of the company was pissed that I did that
03:40.26pbdWhy, did he want that on his card, too?
03:41.20Hogieshe's anal...  I was in blue jean shorts today... (I flew this morning before going into work), and before I could change, I got swamped with "this is broke, that is broke", so when lunch came around, I was like, Im not changing clothes, I dont wanna wash something for tomorrow" and she was all mad
03:41.42adelasdo i have to setup a trunk in order to get my cisco phones to work?
03:41.47twistedHogie, that's a lot of bleh
03:41.58adelasb/c i just wanna see if i get my cisco phones to call each other
03:42.55Hogietwisted: you have no idea...  I had a nice start on the project, handed it off to someone else, and they totally fucked it up, spent the last 2 weeks trying to fix it to make it work for now, but tonight I just scrapped the whole thing and am going to start over
03:43.30pc4Hogie - Blue jean shorts?  That is pretty bad though =)
03:43.42twistedHogie, yeh... that sucks
03:43.43pc4Blue jeans are -ok- here, but I don't know how well those shorts would pass.
03:43.50pc4HEhe -- but I can get away with flip flops =)
03:44.11pbdG'night all.
03:44.14HogieI wear jeans everyday except when I know someone is coming to the office that needs to be impressed, or I go out to a meeting
03:44.37twistedHogie, sounds like my attire
03:44.37Hogiebut, when I go flying, I ALWAYS wear shorts right now, we have a heat advisory where I live today (and tomorrow)
03:44.40twistedjeans and a tshirt
03:45.08HogieI put on a polo shirt or a collared shirt
03:45.14*** join/#asterisk jkk_ (i=joe@67.14.192.58)
03:45.16twistedoh, well aren't you fancy
03:45.20Hogiebut dockers dont last with me, bluejeans do
03:45.22Juggiehogie, did you ever hang out on efnet?
03:45.26Hogiehttp://gallery.cyberjunky.net/August18Solo
03:45.29HogieJuggie: yup
03:45.38Hogiethere right now infact
03:45.39Juggieyour nick is famaliar
03:45.48Juggiedings a bell
03:45.49Juggienot sure why
03:46.10pc4Camera on your solo? :P
03:46.19Hogiepc4: dont tell my instructor
03:46.41pc4tsk tsk...
03:46.42pc4hehe
03:46.50Hogiehttp://gallery.cyberjunky.net/August18Solo/P0000460
03:46.54HogieI fly models there sometimes
03:46.56pc4It's hard enough to fly the damn thing as a low time pilot... let alone photos.
03:47.14Hogiepc4: Im the record setter
03:47.15Hogie;)
03:47.16pc4hehe
03:47.19pc4Damn lucky flatlander :P
03:47.27twistedHogie, hah
03:47.29Hogieonce I get up out of traffic
03:47.41Hogie(ie, away from KGPM), I set the autopilot
03:47.43Hogieand take pics
03:47.49Hogiewhile heading to the practice area
03:47.50pc4I have to deal with 1/2 mile wide valleys at 7,500 feet densite altitude and 12,500 mountains to the west and north.  Basically, no way out :P.
03:47.53*** join/#asterisk r0m (n=SysOp@a81-84-68-51.cpe.netcabo.pt)
03:47.59twistedi love it when autopilot flys you into mountains
03:48.00pc4autopilot on a trainer?  :P
03:48.11Hogieautopilot on a c172SP
03:48.15Juggiehogie, you didnt do anything bbs wise, or hang in #celeron #woe anything like that did you?
03:48.16pc4nice plane :P
03:48.19pc4180 horses if its sp
03:48.23Hogieoh yeah
03:48.25Hogie#bbs, #mysticbbs
03:48.30Juggiebingo
03:48.33pc4160 horsse at 8,000 foot densite altitue sucks.
03:48.34Juggieiniquity?
03:48.41Hogieyeah, back in the day
03:48.42pc4You gain like... 300 feet per minute.
03:48.48twistedrenegade.
03:48.49twistedbitches.
03:48.55Juggiethats where, iniquity scene :)
03:49.06Juggierenegade was for chumps
03:49.09twistedHAH
03:49.12Juggieiniquity had no hard coded strings :)
03:49.18Juggienot one ! :)
03:49.20twistedneither did renegade
03:49.22HogieRedneck Technology was my bbs
03:49.24twistedninny
03:49.30Juggierenegade did
03:49.34Juggiebecause i remember people would hexedit
03:49.37Juggieto change them
03:49.40twistedhah
03:49.54twistedi never hexedited, and had no stock strings
03:50.06Juggieall i know is iniquity made renegade its bitch.
03:50.11twistedhahahaha
03:50.15*** join/#asterisk Grubs (n=Miranda@c220-239-96-230.eburwd5.vic.optusnet.com.au)
03:50.17twistedif you would like to think that, then be  my guest
03:50.20pc4Hogie - What do they charge/hour for that plane, wet?
03:50.22Juggie:)
03:50.33*** join/#asterisk ryandude (n=sprakens@203.131.164.162)
03:50.34Hogie$101/hr I think, I could check the site if you want
03:50.36twistedi should setup a telnet bbs
03:50.37Juggiei ran iniquity cdn fidonet hq :)
03:50.41twistedjust for shits and giggles
03:50.45ryandudehello everyone
03:50.47Juggielets play BRE :)
03:50.49HogieI was suppose to be in one today without an A/P
03:50.51twistedOOH YES
03:50.57twistedJuggie, set up a telnet bbs on your side too
03:50.58pc4Just wondering... not bad for a 98+
03:51.00Juggiebre/sre rock.
03:51.01twistedand we'll BRE that shit
03:51.07Juggiehmmmm
03:51.08Hogieomg, BRE!
03:51.11Juggiewhat was the front end mailer called.
03:51.15HogieI still have my irex
03:51.16twistedfrontdoor?
03:51.19Hogieinternet rex
03:51.19pc4It's a leaseback =)
03:51.21Hogiekey
03:51.21Juggieyah! :)
03:51.27Hogieand frontdoor sysop edition
03:51.28Juggieol frontdoor, taking mail calls.
03:51.28Hogie:)
03:51.32twistedyep
03:51.34Juggieusurper
03:51.35Juggielord
03:51.36*** join/#asterisk bmg505 (n=leon@rndf-146-53-60.telkomadsl.co.za)
03:51.37Juggielord2
03:51.38twistedhahaha
03:51.42pc4MajorMUD!
03:51.42Juggieusurper was cool
03:51.43pc4=)
03:51.44twistedi had a big ass lord board in nashville
03:51.54Juggieit was the text version of grand theft auto :)
03:51.57pc4game connection + majorbbs + doom2 was awesome back in the day.
03:52.00pc4hehe
03:52.03JerJerWWIV
03:52.05Hogiedwango
03:52.07twistedWWIV
03:52.08twistedOMFG
03:52.10pc4dwango cost too much.
03:52.13twistedthere were SO many holes in WWIV
03:52.14JuggieWWIV, didnt like that
03:52.20Juggieiniquity was sexy
03:52.33JerJeryeah it was fun hackin the hell out of wwiv bbs' back in the day
03:52.33Juggiei ran a board called "the underground" for 3 years
03:52.35twistediniquity was about as sexy as michael jackson in a corset
03:52.40tzangerwwiv!!
03:52.48tzangerI started with telegard and then renegade
03:52.51tzangerbut grabbed the wwiv source
03:52.59twistedtzanger, i ran renegade... it rocked
03:53.01Juggieoh yah telegard
03:53.09Juggiea friend of mine swore by that
03:53.09twistedtzanger, i wrote quite a few complex menu systems for renegade
03:53.13Juggiei swore by iniquity
03:53.16JerJerthen Wildcat
03:53.17Hogiepc4:  http://www.aviatorair.com/graphics/rent/n2128s.jpg
03:53.20Hogiethat's what I was flying today
03:53.22twistedtzanger, last i saw, they wound up all over the country
03:53.35Hogielike, the EXACT one
03:53.36Hogie:P
03:53.49tzanger:-)
03:53.51twistedhogie, flying yourself to astricon? :p
03:54.13Juggiehttp://bbslist.textfiles.com/709/
03:54.14Hogiehttp://www.aviatorair.com/graphics/rent/n8963v.jpg is what I usually fly, but the beacon light wasn't working when I preflighted, so they moved me up from it to 1BS
03:54.18Juggiei'm listed :)
03:54.21JerJeravgas is too goddamned expensive
03:54.28HogieJerJer: $3/gallon...
03:54.34JerJeroh i know
03:54.41Hogiecompared to $2.80/gallon for car gas
03:54.42Hogiehmmm
03:54.47JerJeri've gota C172RG
03:54.50tzangerhttp://bbslist.textfiles.com/519/ was my hood
03:54.52tzangerice-9
03:54.59tzangernew gold dream
03:55.27Hogietwisted:  I would, but im only at 17 hours flight time....
03:55.32JerJer$2.51 for regular gas here, today
03:55.33*** join/#asterisk HellAgony (i=HellAgon@200.121.236.192)
03:55.42tzanger519-888-0085
03:55.43tzangerWaterloo, ON   Ice-Nine
03:55.44tzanger(1990)
03:55.50HogieI just paid $2.89/gallon to fill up my civic
03:56.04*** join/#asterisk ZX81 (n=ZX81@222-153-118-124.jetstream.xtra.co.nz)
03:56.11JerJeri want a hydrogen fuel cell
03:56.23ZX81as a UPS?
03:56.37JerJeras a car
03:56.43ZX81<PROTECTED>
03:56.45Juggieit wasnt easy to convince my parents to get an extra phone line
03:56.53ZX81<PROTECTED>
03:57.05ZX81<PROTECTED>
03:57.06JerJerJuggie: i talked them in to 2 extra phone lines  :)
03:57.06ZX81<PROTECTED>
03:57.11Juggiehaha
03:57.13JerJerbling bling baby
03:57.13drrayzx81 - just sit on it and go look at the AMPS!
03:57.17Juggiei used to get donations
03:57.23Juggieone time someone decided instead of donations
03:57.31Juggiethey would steal road pylons and put them on my driveway
03:57.38nick125what is a good kbps for the on hold music?
03:57.39Juggiethey did that like 3-4 times over a summer
03:57.40shido6wow - I can only imagine the street mechanics working on that, cigarette in mouth dropping ash in the midst of hydrogen fumes
03:58.08brc_feel the beat
03:58.10brc_can ya
03:58.12brc_can ya feel
03:58.13nick12564kbps good?
03:58.19brc_no
03:58.28brc_what code
03:58.41brc_transcode it to native
03:58.46nick125for the hold mp3 music
03:58.47brc_so you won't have to on the fly
03:58.49shido6u know the ones...  "hows it running..." then click click...  udner the hood... messin with my damn throttle .... "sounds like its a little lean!"  revs some more... ( bitch get the fsck out from under my hood! )
03:58.58JerJerlol
03:59.15shido6guess its just a detroit thing... but u know what im talking about...
03:59.22HogieJerJer: my instructor quit on me today
03:59.23shido6$20 mechanics
03:59.32JerJerHogie:  i would too
03:59.42Hogiesomething about soloing at 11 hours:(
04:00.18Hogiehe kept trying to push me toward the commercial pilot route...
04:00.19pc4Hogie - quit?  lol -- or moved on?
04:00.21pc4Hehe.
04:00.36Hogiehe got a job at United
04:00.38Hogieor something
04:00.40pc4Hogie - Many instructors are shortimers -- its the way the industry is.  They want to move on to commercial.
04:00.42pc4Yup :P
04:00.43JerJeri don't remember how many hours i had when i soloed
04:00.44HogieI'd have to read the news again
04:00.46hellopAnyone know how to tell if a Zyxel 2000w is a v.2?  Does that just mean it supports SIP v2?
04:00.49pc4So what do you do for a living?  hehe.
04:00.56pc4Work at an isp?
04:00.59HogieIm a network Monkey Boy
04:01.06Hogiena
04:01.09JerJeri've got like 1200+ hours now
04:01.13nick125anyone here know of a good ata device for asterisk (i might get one some day)?
04:01.19Hogiedamn, I want my own plane
04:01.37loudbuy ariel's plane.
04:01.44Hogiethere's a runway by one of our offices that's 1.2 hours driving away
04:01.48Hogieor about 20 minutes flying
04:02.55*** join/#asterisk dcm_ (n=Craziman@208.3.11.172)
04:02.56Hogie(mainly cause we have to go west 20 miles on 30mph roads, then south 70ish miles on a 70mph highway)
04:02.59JerJerthere is two grass strips within 2 miles of my bat cave
04:03.15pc4There is a huge runway about 800 feet from my house.
04:03.25pc4Yes, the jets wake me up at 7 am when it opens =)
04:03.36*** join/#asterisk WilliamK (n=wkeller@c-67-172-202-228.hsd1.tx.comcast.net)
04:03.38*** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com)
04:03.39Hogieloud: how much?  I'll give $10
04:03.41JerJerthen there is a real airport (uncontrolled) like 11 miles away, which is where we have it hangered
04:04.00loud10k, his plane is around 45k.
04:04.10pc4JerJer - Ours is controlled, thankfully.  Otherwise it'd probably be suicide -- non standard traffic pattern... one way in... one way out...
04:04.11Hogiewhat is it?  experimental?
04:04.16pc41 mile wide valley :)
04:04.42pc4And yes, they put damn jets down in it
04:04.43pc4hehe
04:04.57pc4The president of flying j slammed his into the mountain next to it a year ago.
04:05.04JerJersweet
04:05.06nick125(kinda off topic): is there a way to turn a mp3 into a 64kbps mp3 using transcode or something like that?
04:05.09pc4Two years before that it was the oracle dude with one of his surplus military planes.
04:05.18nick125or maybe i could try wav again
04:05.24HogieKGPM is only 4000ft long, so we dont usually get larger things in
04:05.37Hogiebut, we have all the news heli's based there, etc
04:05.45pc4Well I'm in a resort town, so you get big stuff in that's not meant to fit :P
04:05.50JerJerpc4: i went thru my private at a class c airport
04:06.23JerJerthen commerical and everything else was at a 141 school at a class d (atc, no tracon :)
04:06.28*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
04:06.46pc4JerJer - You a commercial pilot?  Or just get it for the hell of it?
04:07.03JerJeri'm also a CFI
04:07.05Hogiewhat's the main diff between 141 and the other training?
04:07.06pc4JerJer - I love flying... the safety aspect just bugs the hell out of me :P
04:07.46drrayit's just an unforgiving hobby
04:07.54adelashey, which sip should i be using for makeing a extention with cisco phones? SIP rfc2833, inband, or info
04:07.56JerJerpc4: i wrote an essay in high school about the future of aviation and won 5th place which was the flight fees for any 141 school in Michigan
04:08.20infinity1adelas: sounds like you're talking about dtmf.
04:08.22pc4JerJer - That's not a bad deal.
04:08.25infinity1adelas: usually rfc2833
04:08.26JerJernope
04:08.28pc4I'm paying for it our of my paycheck... and it HURTS.
04:08.37pc4JerJer-  This is where I'm at:  http://www.airnav.com/airport/SUN
04:08.44adelask thx infinity1
04:08.49GrubsDare I ask an asterisk Q? - which router/firewall distros besides ClarkConnect do people install asterisk onto?
04:08.53adelaswats dtmf?
04:09.05*** part/#asterisk santiago (n=santiago@63.245.86.202)
04:09.10*** part/#asterisk dcm_ (n=Craziman@208.3.11.172)
04:09.15JerJerhttp://www.airnav.com/airport/KFNT <-- that's where i did my private
04:09.44pc4Why is it that I'd rather learn my private at a non land-locked valley-surrounded high elevation airport?
04:09.46*** join/#asterisk Beccara (n=Beccara@219.89.209.122)
04:09.52JerJerhttp://www.airnav.com/airport/KTVC <- that's where i did my instrument, commercial, multi-engine and CFI
04:09.55pc4I suppose I'll become a better pilot for doing it here :P
04:09.55GrubsIs IPCop + asterisk on the same box a possibility? What about monowall?
04:10.01*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:10.10pc4Do you think?
04:10.14drraymonowall is BSD variant
04:10.23nick125Grubs: monowall is way too small, it would take a while to get anything near asterisk on there
04:10.29pc4That's pretty (for Michigan :P)
04:10.36nick125it doesnt even have perl iirc
04:10.36drrayI've run firestarter on an asterisk box
04:10.39GrubsI see
04:10.40HogieJerJer: honestly, what's the diff between the 2 diff private pilot trainings?  Ive been up to long to think of the one besides 141
04:10.46nick125nor a compiler
04:11.02JerJerpart 141 is highly structured
04:11.05pc4Hogie - google it.  But basically 61 = at your own pace, unstructured.  141 = highly structured curricliam and checks and balances.
04:11.12pc4You can graduate with 5 hours less, too.
04:11.13pc4or so.
04:11.16GrubsThx  I know people have ClarkConnect and asterisk on the same box. - I guess its bigger :)
04:11.21pygrammerListenin' to Zeppelin through the phone :-P finally got ztdummy to work...
04:11.21Hogie141 is less hours, right?
04:11.27pc4Hogie - Yes... if you can do it in less.
04:11.29JerJeryes, because you are checked more often
04:11.36pc4Hogie - It's not much less, but you can use simulator for some of it.
04:11.41JerJerlike you do a check ride after each phase
04:11.44HogieI know im part 141
04:11.49HogieIve done 2 checkrides so far
04:11.50pc4It really doesn't matter anyways, you probably can't get it in 35 regardless.
04:12.05JerJerinstead of just at your private, or commercial or instrument
04:12.13*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:12.22adelascan someone tell me wat this means "/var/www/html/panel/safe_opserver: line 5: 1859 Terminated      ./opserver.pl"
04:12.28adelasthis is after i added an extention
04:12.30heison~seen sivana
04:12.31jbotsivana is currently on #asterisk (22h 6m 31s)
04:12.31JerJerHogie:  pop quiz...that's Vx of your aircraft?
04:12.31pc4I'm a firm believer that it really doesn't matter... one way or the other.
04:12.46nick125hmm
04:12.55Hogie64 knots
04:13.03*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
04:13.04JerJerok a tough one... Vne  :)
04:13.14Hogie160
04:13.24JerJeryay
04:13.27Hogiefor my "normal"
04:13.33HogieIm not sure about the SP's
04:13.35Hogiebut the 172M
04:13.37HogieI do know
04:13.39JerJeryeah
04:13.45adelascan someone tell me wat this means "/var/www/html/panel/safe_opserver: line 5: 1859 Terminated      ./opserver.pl"
04:13.57blitzragenick125: what OS?
04:13.58JerJeradelas:  we saw it the first time
04:14.02nick125blitzrage: linux
04:14.04adelaso hehe :p
04:14.33ChArLeS___wow
04:14.34ChArLeS___fuck
04:14.39JerJerHogie: had any emergencies (real) in flight yet ?
04:14.44ChArLeS___sip uses the double of the bandwidth
04:15.03nick125it doesnt show the wav up in aMP either
04:15.04*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
04:15.04adelasso can someone tell me?
04:15.04Hogiejust got freaked out today when I pulled the throttle out too fast at the end of downwind...
04:15.08nick125its chmodded 777
04:15.10Hogieit did a BANG! sound
04:15.19JerJerHogie: backfire...hell yeah
04:15.33pc4JerJer - Have you had any?
04:15.40JerJerlol yeah
04:15.40HogieI was like "Grand Prairie Tower, Cessna 121 Bravo Sierra reporting base, would like a full stop"
04:15.43JerJer2
04:15.48pc4Do explain.
04:15.50Hogiecause I didn't know what it was
04:15.54Hogienever had it happen before
04:16.13Hogieby the time I landed, I was shaking I was so freaked out, lol
04:16.38pc4I had a dust storm come on rapid approach from over the mountain.   It was scary as crap.  Luckily the instructor was there and did one hell of a short field landing.  I called ATIS after we landed and 30 seconds later it was 25 kts crosswinds.
04:16.47ChArLeS___when is cisco going to support IAX ?
04:17.01Hogiewell, I guess it was when I got out to tie down the plane I was shaking
04:17.14JerJerfirst one was pre-solo, so i was with my instructor, of course.  We were on climb-out, about 2000 feet when all of the fuses popped
04:17.22JerJerso no electric
04:17.33nick125mp3 works, but, mp3 skips and all
04:17.46pc4JerJer - Solution?
04:17.47JerJerwe just rolled back into the traffic pattern - shook our wings and got Green lighted by ATC to land
04:18.03pc4And the second one?
04:18.10JerJerfun
04:18.26JerJeri already had instrument and commercial ratings
04:18.36*** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au)
04:18.47*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
04:19.01pc4=)
04:19.15JerJerup in Traverse City and my recently acquired girlfriend finaly decided to go flying with me for a nice weekend at a casino in the UP
04:19.33pc4and subsequently released girlfriend?  hehe
04:19.38pc4Ok... go on
04:19.40JerJerabout 30 minutes into the flight i noticed the oil pressure fluxuating
04:19.43JerJerthen go to nothing
04:19.51*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
04:19.55JerJerthen fire blows out of the cowling
04:19.59pc4Oil pump went out "ohh shit" -- going to throw a rod soon.
04:20.01JerJergirl friend screeches
04:20.20JerJeri unplug her headset, turn off the gas and magnetos
04:20.38pc4So you could hear atc over her screams, right:?  lol
04:20.54pc4<JerJer> up in Traverse City and my recently acquired girlfriend finaly decided to go flying with me for a nice weekend at a casino in the UP
04:20.56pc4<JerJer> about 30 minutes into the flight i noticed the oil pressure fluxuating
04:20.56pc4<JerJer> then go to nothing
04:21.01pc4<JerJer> then fire blows out of the cowling
04:21.03pc4There you go :)
04:21.06dougheckasweet
04:21.14dougheckadidja crash?
04:21.17JerJerand get on the radio "Minneapolis center, Cessna 6396 Kilo declaring an emergency fire in engine"
04:21.53JerJerand i spiral down to an open field - gril friend now clutching the seat and still screaming
04:21.59dougheckaLOL
04:22.14dougheckaorgan music!
04:22.22pc4spiral -- as in rapid descent w/ flaps?
04:22.22JerJerthe instant the wheels stop rolling, girl friend exits the plane and falls onto the ground
04:22.33pc4You landed on the field? =)
04:22.44JerJerpc4:  no flaps - we want all the time we can get
04:22.47pc4You told her to immediately exit to the REAR, right?
04:22.50JerJerbut yes a pretty tight spiral
04:23.06pc4JerJer - Ok -- I thoguht you wanted on the ground ASAP, due to fire.
04:23.09*** join/#asterisk Cresl1n (n=Cresl1n@adsl-67-126-59-49.dsl.pltn13.pacbell.net)
04:23.12pc4I'm guessing the fire was extinguished by then.
04:23.26*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
04:23.34pc4What did the farmer say? :P
04:23.45JerJeryeah the instant i killed the gas the motor stopped and the relative wind blew out the flames
04:23.46pc4Do continue
04:23.54pc4Well that's the good news
04:24.03pc4Could have been worse.
04:24.06JerJeryeah
04:24.24JerJerthen i get back on the radio and give a postion report to minn. center
04:24.37JerJerthey dispatch the local fire department
04:24.38dougheckawhat caused the issue?
04:24.45JerJerand the mechanic from the college
04:24.59JerJerdoughecka:  oil line burst
04:25.05dougheckaah
04:25.08dougheckauser error?
04:25.09doughecka:P
04:25.26dougheckaor just component fail :P
04:25.31pc4Might have been mechanical there.
04:25.34JerJerthe mechanic proceeded to replace the oil line and said, "alrighty she's ready to fly again"
04:25.35pc4hehe.
04:25.42pc4And you said -- you fly it home, right?
04:25.43pc4hehe
04:25.48dougheckaJerJer: no more oil?
04:25.53JerJergirl friend flips out "there is no way in hell i am getting back into that plane"
04:26.02dougheckano more girlfriend?
04:26.10JerJerso girl friend drives mechanics truck back to traverse city and he and i fly home
04:26.19JerJerhome = traverse city
04:26.20dougheckalol
04:26.30pc4So.... how did the relationship work out?
04:26.37JerJersurprisingly she stuck with me for a while longer
04:26.55JerJerand even got back into a plane with me, like 3-4 months later
04:27.12*** join/#asterisk bonez41 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net)
04:27.22dougheckaJerJer: while someone pushed it into the hanger? :P
04:27.46*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net)
04:28.03JerJershe kept hinting about the mile high club, but being pilot in command and doing the nasty at the same time usually conflct
04:28.06*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
04:28.16pc4lol
04:28.19pc4Should have went for it!
04:28.37snewpyjerjer: yeha, plus you need to be houdini to join the mile high club in a Cessna :)
04:28.49pc4Damn controls to the plane aren't adjustable and too damn low :P
04:28.50pc4hehe
04:29.06JerJeryeah i can see the headline  "two coeds found naked in aircraft ruins"
04:29.09*** join/#asterisk ilustrate (i=user@200.92.29.182)
04:29.11snewpylol
04:29.19dougheckaHAH
04:29.37Hogieautopilot...
04:29.40Hogiebackseat...
04:29.56pc4Hogie - Miss the page on weight and balance? :)
04:29.56JerJerthen who's watching for traffic?  :)
04:30.07HogieMr Auto Pilot, lol
04:30.08pc4Hogie - I'll watch that plane fall right out of the sky.
04:30.24pc4JerJer - I read a ntsb report about a guy who reported on a regular basis put on auto pilot and "Set an alarm clock"
04:30.24pc4lol
04:30.27EquinoxHow can I see what codecs a call is using?
04:30.28pc4darwin award.
04:30.33pc4Apparently he had been doing it for years.
04:30.42JerJerEquinox:  show channel <foo>
04:30.54JerJerread and write format
04:30.55Hogiewould tower tell me if I hit the tail of the airplane on the runway?
04:31.17*** join/#asterisk Beccara_ (n=Beccara@219.89.209.122)
04:31.31JerJerpc4: that's hilarious
04:31.34pc4If they saw it :P
04:31.47pc4They'll tell you if your baggage door is unlatched.
04:32.20pc4JerJer - What city you in?
04:32.26dougheckapc4: should the "door ajar" light be a clue? :P
04:32.32*** join/#asterisk BrainSurg (n=paul@69.158.227.13)
04:32.52HogieI just dont remember the tie down ring under the rudder having a flat end before takeoff like it did after that last landing
04:32.54JerJerpc4: the metropolis of Mecosta, MI
04:33.05JerJerright up the street from the Mecosta International Airport
04:33.18*** join/#asterisk blake__ (n=veteran@cpe-66-25-30-132.houston.res.rr.com)
04:33.30JerJerhttp://www.airnav.com/airport/27C
04:33.40blake__is there any way to have asterisk dial multiple numbers at the same time, and whoever picks up first gets the call?
04:33.48shido6yes blake
04:33.49JerJerlol i love it when hey 'estimate' gps corrds
04:33.49pc4Should fly to idaho :P
04:33.50pc4hehe
04:33.56JerJerhey ?  they
04:34.03pc4Tiny runway.
04:34.13JerJerand elevation - great!
04:34.28JerJerSurface: turf, in poor condition
04:34.32JerJerpoor is not the word
04:34.37JerJerpiss poor maybe
04:34.37dougheckacrappy?
04:34.55JerJerNOTAM: "watch out for old tires"
04:35.00dougheckalol
04:35.01JerJerand trash
04:35.17JerJeryeah its 1022 feet with a serious displaced threshold
04:35.27JerJerso call it 800 foot
04:35.42JerJerObstructions: 45 ft. trees, 360 ft. from runway, 108 ft. left of centerline, 8:1 slope to clear
04:35.50pc4Is that your "1 aircraft based on field"?
04:35.51pc4hehe
04:36.27JerJersweet - they finally added the model aircraft notam
04:36.41JerJerwe've been bitching at the faa for years to add that
04:37.08pc4IS that your 1 plane that you rent?
04:37.10pc4hehe
04:37.12dougheckaI have a friend whos in new guinea, and the closest airport has a crazy runway
04:37.14JerJerlol - i just noticed the one aircraft on field
04:37.16JerJerthat's funny
04:37.23BrainSurgAnyone know where to find zaprtc?
04:37.32hellopDang... wireless VOIP looks like a big PITA
04:37.34dougheckagrassfield... its at an angle, so when you take off your going downwill
04:37.45dougheckaand when you take off, you go DOWN
04:37.47dougheckato pickup speed
04:37.56dougheckaand quickly go hard left
04:38.04dougheckacause the valley turns
04:38.16JerJersounds like taking off at vegas
04:38.17dougheckaat one point your BELOW the runway
04:38.33SplasPoodHrm, does asterisk have any support for "Call Waiting Deluxe" ?
04:38.39*** join/#asterisk raphWRKN (n=raph@210.15.240.107)
04:38.40JerJerdefine deluxe
04:38.44dougheckaof course the throttle is at full power till your FAR away :)
04:39.12JerJernice
04:39.16SplasPoodJerJer: Well, there's a service called Call Waiting Deluxe...   I've seen some pots handsets which support it... basically when you get the beep it gives you options on how to handle it
04:39.27SplasPoodconference, ask calling party to hold, busy...
04:39.29dougheckayea, hes a commercial pilot, but hes only flown out once :)
04:39.35SplasPoodforward..
04:39.47JerJeryeah i like to fly forward  :P
04:40.10JerJerdoughecka:  sounds like fun
04:40.21h3xits called using a voip phone
04:40.24JerJeri flew sky divers for a few months
04:40.31dougheckalol
04:40.39h3xcall waiting deluxe that is
04:40.44JerJeri would always try to beat them down
04:40.57dougheckahahaha
04:41.03dougheckaas your wings snap off
04:41.16SplasPoodh3x: heh, yes.. but I was thinking of when using standard pots phones..  Just curious, I have no application
04:41.24JerJerlast one out the door I would do a hammer head and floor the throttle
04:41.29SplasPoodsetting up some 2.4ghz portable I picked up and I see it in the menu options
04:41.31dougheckahahaha
04:42.03*** join/#asterisk BharatS (n=bharatsa@210.211.246.47)
04:42.19h3xSplasPood: well you can sort of do that with adsi phones
04:42.29dougheckaman my internet sux
04:42.41dougheckamonkey balls
04:42.43h3xbut the question is "why" when adsi phones suck and cost more than voip phones
04:42.52h3xif that is what you are looking for
04:42.57SplasPoodwell thats the thing, I wasn't thinking of ADSI phones
04:43.15SplasPoodI was thinking of providing this same type of call waiting service to pots phones
04:43.48h3xby what means
04:43.51*** join/#asterisk fad (n=fahad@202.142.189.86)
04:43.52pc4Well I'm outta here... nice talking guys =)
04:43.56JerJerl8r
04:44.02JerJeri need to get back to the code
04:44.02pc4Bye =)
04:44.04pc4hehe
04:44.15h3xif you are talking about with voip adapters for residential users
04:44.21h3xthen asterisk has nothing to do with that
04:44.22SplasPoodno
04:44.34SplasPoodnevermind...
04:44.38h3xwell
04:44.48h3xwhat practical way can you deliver a bunch of lines to people with pots lines
04:44.51h3xthat needs this feature
04:45.00SplasPoodI'm not talking about delivering service to anyone
04:45.02SplasPoodI was just asking
04:45.07SplasPoodif I plugged a normal handset
04:45.11SplasPoodthat supported this feature
04:45.12SplasPoodinto a card
04:45.13SplasPoodin a box
04:45.16SplasPoodrunning asterisk
04:45.17SplasPoodzaptel card
04:45.27SplasPoodcould that type of call waiting indicator/etc be sent.
04:45.56h3xi mean you could hack it
04:46.01SplasPoodSince a number of standard (ie, non voip, non adsi, etc) "home" phones seem to have support
04:46.03h3xhave dtmf buttons that do things with that call
04:46.13SplasPoodit was just a point of curiosity, since I just saw the option for it in a new phone
04:46.17h3xbut i cant figure out a practical purpose of it
04:46.17SplasPoodtrue
04:46.30h3xoh that stupid iQ thing?
04:46.37h3xCaller IQ?
04:46.41SplasPoodI dunno, it was referred to as "Call Waiting Deluxe"
04:46.44SplasPoodbut that could be it...
04:46.55h3xCaller IQ is a sick way to get more money out of people that bought a phone
04:46.59h3xits like a lite version of ADSI
04:47.29h3xbut i mean
04:47.44h3xi dont see a point in doing call rejection with a call waiting call when you cant do it with a normal incoming call
04:47.47h3xlike you can do with a voip phone
04:47.54nick125it says i have 0 messages, when i call *98, it says i have 5
04:48.04SplasPoodoh yea of course... like I said, just a point of curiosity...
04:48.09h3xmaybe your mailbox= string is busted
04:48.28BrainSurgh3x: Dumb question, what is ADSI?
04:48.36nick125h3x: its set for the same box num
04:48.44h3xI think that CallerIQ stuff is proprietary and isnt a bellcore spec so the precise definition of it wouldnt be implemented in asterisk
04:48.47jsmithAnybody know of a way to set the auto-answer on the Cisco 7960 via the TFTP configs?
04:48.55SplasPoodh3x: ahh, that makes sense
04:49.03h3xAnalog Display Station Interface or something
04:49.35h3xits a "smart phone" whereby it communicates with the "Telco" using a V.29 FSK 1200bps modem to receive screens and scripts
04:49.42h3xand DTMF or V.29 to transmit data
04:50.00h3xaastra.com has some example adsi phones but now they are mostly going voip
04:50.09BrainSurgh3x: Cool.
04:50.12BrainSurgthanks
04:50.37*** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au)
04:50.45Hogiejsmith: you can't...
04:51.05Hogiethat's one of the things listed as not config fileable on cisco's site
04:51.09Hogiefor sip firmware
04:51.20*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
04:51.34GoshenCan someone add to the README file the prerequisites that are needed on the system before you make?
04:51.42Goshenlike bison, openssl ect...
04:51.52Goshenthey are on the asterisk.org webpage, but not in the README file
04:52.35Goshenmpg123
04:52.47jsmithGoshen: mpg123 is not a *requirement*
04:52.51*** join/#asterisk juice (i=1000@mo-67-77-188-19.dyn.sprint-hsd.net)
04:52.55Goshenyou get the idea
04:53.00jsmithYeah, I know...
04:53.09*** join/#asterisk spoot_nick (n=julio@50.118.233.220.exetel.com.au)
04:53.20Goshenjust need someone with access to take a second and drop a paragraph in there
04:54.12*** join/#asterisk immo (n=immo@202.142.189.86)
04:54.45ilustratecan i install asterisk with regular analog phones?
04:55.09jsmithilustrate: Yes, if you have the right hardware card.
04:55.32immocan I change the sip port for a particular user in sip.conf other than 5060
04:55.39spoot_nickspeaking of which, does anybody know if USA FXS cards would work in Australia?
04:55.49BrainSurgjsmith: Need an FXS interface, right?
04:55.54jsmithBrainSurg: You got it.
04:56.22jsmithspoot_nick: Check the Digium site... I know several of their T1 cards are certified in Australia, but I'm not sure about the TDM series...
04:57.05immo<PROTECTED>
04:57.23ilustratewhat would i need to setup a 4 trunk 6 extension setup. i'm trying to do my first commercial install and a client wants to (another client with over 10lines and 6 locations) wants a way to monitor calls before the telefone bills without waiting 2 months.
04:57.41EquinoxWhat do you guys think of the sipura adapters?
04:58.05Hogiemy sipura 2100 is making a great paperweight (I haven't even taken it out of the box yet)
04:58.26EquinoxI need some sorta analog adapter
04:58.28jsmithilustrate: Check out CDR.
04:58.33EquinoxJust trying to figure out what the least painful one is.
04:58.45ilustrateCDR is that a company or what?
04:59.20jsmithilustrate: No, it's Call Detail Records....
04:59.23jsmithilustrate: Go look them up.
04:59.29ilustrateperfect. o.k.
04:59.47jsmithEquinox: IAXy is the least painful.
05:00.00BrainSurgjsmith: IAXy?
05:00.17Equinoxjsmith- Supports iax?  Nice.
05:00.19jsmithBrainSurg: Go check Digium's hardware selection...
05:00.24jsmithEquinox: You got it!
05:00.46Hogieoops, I dialed 008 from my home phone (that's hooked up to the work * box), and...  I must have just yelled "MUPPET!" into the external loud speaker at work, doh, my bad
05:00.53Equinoxjsmith- The S101?
05:01.10jsmithEquinox: Yes, I think that's the one... grey in color, right?
05:01.26*** join/#asterisk oej (n=oej@ti111210a080-0810.bb.online.no)
05:01.26adelashey, i have 2 cisco phones setup that are connected to the phx, one with extention 200 and other with 201
05:01.30adelashow can i get them to communited
05:01.35Equinoxjsmith- Looks black
05:01.35adelascommunicate*
05:01.39adelasor call eachtoerh?
05:01.40dougheckaholy water
05:01.44jsmithEquinox: Yeah, that's the one...
05:01.55adelashow can i get one of my cisco phones to call the other one?
05:02.14spoot_nickadelas: add an extension for each. check the SIP examples
05:02.19jsmithadelas: Add a dialplan entry, maybe?  exten=>101,1,Dial(SIP/102)
05:02.28hellopadelas dial its extension defined in sip.conf
05:02.34spoot_nickand that as well...
05:02.37Barmalwhy sometimes I keep getting those auto destroying call messages?
05:02.48adelasok
05:03.05BrainSurgDoes anyone know of a canadian mail order place that sells digium hardware?
05:03.58adelaswait, wat entry would i add?
05:04.06adelasthis is confusing
05:04.32jsmithadelas: Ok, here goes.
05:04.51jsmithadelas: Let's assume the Cisco phones are setup as users SIP/101 and SIP/102 in sip.conf.
05:04.55jsmithadelas: Are you with me so far?
05:05.08Hogiesip.conf?
05:05.10Hogiewhat's that?
05:05.16ilustrateare CDR records immediately available? analog?
05:05.21adelasyea
05:05.30adelaswell, i have 200 and 201
05:05.35JerJerdefine immediately
05:05.35jsmithadelas: OK, that's fine.
05:05.47ilustratewithin the hour at least
05:05.48Hogieilustrate: they are on our system, as soon as the call is taken down
05:05.59ilustrateo.k.
05:05.59jsmithadelas: And in sip.conf, you've set them to start in a certain context, let's call it [internal]
05:06.18JerJerunless you are doing cdr batching
05:06.29adelasokay
05:06.30spoot_nickadelas: as far as I configured mines, asterisk generates CDR csv logs soon as the call is completed
05:06.42HogieJerJer: you still run nufone, right?
05:06.43spoot_nickadelas: (busy, non answered, answered)
05:06.52jsmithadelas: Then, inside that context (the one I called internal), you'd add an extension.
05:06.54spoot_nickthat's the default
05:06.55JerJernope that was last weeks project
05:07.09jsmithadelas: exten=>200,1,Dial(SIP/200)
05:07.10Hogieheh
05:07.11adelasjsmit, can you pm me with an example to type in?
05:07.17jsmithadelas: exten=>201,1,Dial(SIP/201)
05:07.20adelasok
05:07.26jsmithadelas: That should do it.
05:07.35Hogienufone has saved us $162 in the last month
05:07.39Hogieyay 4 nufone
05:07.45JerJersweet
05:08.09jsmithJerJer: So what are you up to now?
05:08.26nick125_lappyi got a quick question: is it possible for, if after someone is on hold in a certain queue, that it would forward the call to my cell phone?
05:08.54adelasjsmith thx
05:09.09adelasnow i gotta figure out how to do multiple phone lines
05:09.14jsmithadelas: No problem... go buy http://www.oreilly.com/catalog/asterisk/
05:09.29BrainSurgHogie: What is nufone?
05:09.55nick125_lappyi also wonder, with a asterlink line, is it possible to have multiple in and out lines (i know asterlink supports 6, but, is there something special i have to do to asterisk?)
05:09.56jsmithBrainSurg: IAX and SIP termination...
05:10.00JerJerjsmith:  trying to finish members.nufone.net, but its not getting there very quickly
05:10.01adelasheh lol
05:10.01hellopmy asterisk is behind my firewall,  I want to use neighbor's wireles with my VOIP wifi phone.  Aparently that's not possible.
05:10.11h3xthis is so funny
05:10.18h3xhttp://www.sylantro.com/partners.html
05:10.27h3xI like how they have like the whole entire fortune 500 listed there
05:10.44h3xand then on the right side theres a place setter "Customer quotes or facts will go here"
05:10.52h3xhahahhahahahhahahha
05:11.11JerJerthat's great
05:11.16BrainSurg:)
05:11.23JerJerthis page needs to be mirrored for future reference
05:11.26hellopSo maybe I need to get an IAX wifi phone instead of a sip one?  Has anyone played with wireless VOIP and trying to roam?
05:11.35h3xand then they flipped horizontal
05:11.42nick125_lappyanyone here use asterlink?
05:11.43hellopBTW, I do have permission from 3 different wireless access points to experiment with.
05:11.44h3xthat stupid clip art
05:12.26h3xhellop: bwahaha theres no iax wireless phones
05:13.00h3xmakarios.blessed.net - - [18/Aug/2005:22:07:32 -0700] "CONNECT 66.103.132.86:11111 HTTP/1.0" 501 16 "-" "unknown"
05:13.00h3xmakarios.blessed.net - - [18/Aug/2005:22:07:32 -0700] "POST /default.rxml HTTP/1.0" 200 4874 "-" "unknown"
05:13.03h3xwhat a jackass
05:13.03JerJercorrection, there are no available iax wireless phones
05:13.17h3xwell
05:13.18helloph3x  HAHA! probably not even possible to run on a PDA..
05:13.27h3xi guess you could hook up a cordless phone to an iaxy and you have one
05:14.02BrainSurgWhat's the going rate for an IAXy?
05:14.10h3x$TOO_MUCH
05:14.10JerJeroh there is one - they are just being very lets say cautious with the release
05:14.10jsmithBrainSurg: Around $100.
05:14.27h3xfor $80 you can get a two line device from sipura that actually has good audio quality and lots of features.
05:14.35h3xoh and t.38 is in beta now
05:14.36jsmithh3x: But it's SIP.
05:14.43JerJerh3x: obviously you haven't seen the new iaxy
05:14.45h3xso? it does 10 flavors of nat traversal
05:14.57h3xits just a new case!@
05:15.00JerJerum no
05:15.02jsmithh3x: I'd still take a new IAXy...
05:15.11jsmithh3x: Speak what you know...
05:15.24h3xthe only point of iax is nat traversal
05:15.30h3xbut the sipura works anyway
05:15.35h3xi triple natted it and it still worked
05:16.05jsmithh3x: Wrong again!
05:16.11JerJeriax uses less bandwidth  :P
05:16.18h3xno it dosent
05:16.19jsmithh3x: There's much more to IAX than just NAT traversal.
05:16.23h3xyou cant run g.729 on the iaxy !
05:16.23JerJeryes it does
05:16.27JerJerthere is no RTP
05:16.29jsmithh3x: Try trunking SIP calls...
05:16.37JerJerhence less overhead
05:16.39h3xjsmith: You cant use trunking on a single line device
05:16.41JerJerhence less bandwidth
05:16.51h3xanother problem
05:16.55jsmithh3x: I realize that... I *INVENTED* IAX trunking...
05:17.00h3xtheres no way to keep someone from iaxyprov'ing your iaxy
05:17.06flewidManxPower: thanks for your suggestion earlier today of adding txgain in the zap cfg to stop the double digits :) working great
05:17.13flewidmanxpower: owe ya a beer :)
05:17.21BrainSurgWhat is T.38?
05:17.23h3xmy point is you gain nothing from getting a iax capible ATA
05:17.33JerJerits iax
05:17.36JerJernot sip
05:17.39jsmithh3x: And again, I'll say that I'd much prefer one that does IAX, and not SIP
05:17.39h3xso
05:17.47JerJerjsmith: ditto
05:17.49h3xiax will never do t.38
05:17.49jsmithh3x: For some of us, it's a big deal.
05:17.57JerJeriax doesn't need T.38
05:18.06jsmithh3x: What does T.38 have to do with the transport protocol :-)
05:18.18jsmithFax over IP is so lame anyway...
05:18.20hellopwho was the actor in the move Quicksilver?
05:18.28hellopmovie
05:18.28JerJerif you want to fax, download coppice's kick ass dsp code and fax
05:18.29jsmith... hello, twenty-first century here ...
05:18.40JerJerthen transfer it over IP using a binary transfer protocol
05:18.45SplasPoodwtf is the point of fax over IP anyway
05:18.47JerJerthen shit it out a PRI at the far end
05:18.55JerJerinstant T.37 without the bullshit
05:19.03h3xyeah well think about what ATAs are used for, not everybody wants to throw their fax machine away
05:19.05jsmithSplasPood: Amen!
05:19.22JerJernobody says they have to throw away their fax machine
05:19.22SplasPoodjsmith: no come on.. there's gotta be an application....
05:19.23jsmithh3x: Have you ever actually tried sending a lot of faxes over T.38?
05:19.45jsmithh3x: I'd rather have appendicitis.
05:19.48JerJerstore and forward is so much more efficient
05:20.02h3xyeah but then youd need like 8 megs of ram in your ata
05:20.08jsmithSplasPood: Stopgap technology?
05:20.12JerJerwho says the ata has to do shit?
05:20.28h3xlets see
05:20.28SplasPoodjsmith: I suppose...
05:20.34h3xmost people use ATAs on cable modems
05:20.36JerJerata - local asterisk box running rxfax
05:20.47h3xsince for DSL you need a analog line in 90% of the world
05:20.49h3xor more
05:20.51jsmithLater all... have a good night, and dream of starfish...
05:20.54ilustratewhat happens if the power goes out with asterisk. does it reset after you restart the computer.
05:20.58h3xand nothing else is practical in most situations
05:21.01JerJerlocal asterisk box running rxfax scp's tiff image to remote aasterisk box with txfax and pri
05:21.09h3xcable modems have packet loss out the yinyang
05:21.12JerJerremote asterisk box with txfax and pri shits it out the PRI
05:21.12jsmithilustrate: Only if you tell it to... it's just a program, like any other...
05:21.36ilustrateso it should just start where it left off?
05:21.40h3xyeah right, sell that idea to somebody doing vonage type shit
05:21.48JerJerdon't do vonage type shit
05:21.52jsmithilustrate: Does your word processor start where it left off?
05:22.03JerJerbe smarter than Citron
05:22.05JerJercome on
05:22.05jsmithilustrate: It will only restart on reboot if you set it up that way...
05:22.17ilustrateno. but i guess i meant. would i have to re-configure everything?
05:22.22hellopWe just want to be able to bust out the PDA, which can get net access from any free access point, and work with the * box specified in the softphone.  Any suggestions?  Anyone want to work with me on this?
05:22.28jsmithilustrate: No.
05:22.29JerJerilustrate: this is why we have hard drives
05:22.35ilustrateo.k.
05:22.47jsmithG'night all...
05:22.49BrainSurgAnyways, good night everyone.
05:22.50JerJernight
05:22.51hellopI got PDAs, CF wireless cards, Wireless fones, ATA's whooo wants to help?
05:22.54*** part/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com)
05:23.08Hogieokay hellop, bring them all to my house
05:23.10Hogieand I'll help
05:23.14hellopI get free access points at work, at home and at the college.
05:23.31h3xthere is a ton of shit the sipuras do
05:23.53hellopHogie, how about you fly to Hawaii and U can stay at my house by the beach?
05:23.54h3xlike, codecs besides g.721 and g.726?!??
05:24.10*** join/#asterisk tc0nn (n=tec@spyglass.timsnet.com)
05:24.15JerJercodec selection will change
05:24.18Hogiehellop: na
05:24.26h3xwith a two bit half ass processor?
05:24.33JerJerand locking down the provisioning will too
05:24.57JerJersipura's run on a DVD based MPEG microcontroller
05:25.01HogieJerJer: you mean since linksys bought sipura?
05:25.01JerJerthat's not two bit ?
05:25.13JerJerHogie:  i'm talking about the iaxy
05:25.15tc0nnLinksys..ie Cisco?
05:25.31h3xthey still have the best designed and best priced atas there is
05:25.37JerJerin your opnion
05:25.59Hogiebest ata = asterisk box with a fxs card;)
05:26.17h3xwell lets see
05:26.24tc0nnAny idea what mutes DTMF when connected to the console/dsp (via oss channel)
05:26.35JerJerthe mute function?
05:26.51tc0nnIs it the oss driver killing dtmf?
05:27.06h3x$200 for two ports of iaxy
05:27.11h3xor $85 for two ports of sipura
05:27.11tc0nnYeah, I get a quick burp of DTMF, not enough to detect though.
05:27.13h3xDUH
05:27.21JerJersipura doesn't do IAX
05:27.27JerJerat least publicly
05:27.38h3xbut theres no reason to do iax
05:27.39h3x?
05:27.40SplasPoodUnlimited NationalAccess/BroadbandAccess services cannot be used (1) for uploading, downloading or streaming of movies, music or games, (2) with server devices or with host computer applications, including, but not limited to, Web camera posts or broadcasts, automatic data feeds, Voice over IP (VoIP), automated machine-to-machine connections, or peer-to-peer (P2P) file sharing, or (3) as a substitute or backup for private lines or dedicated data c
05:27.47JerJersure there is
05:27.48SplasPoodhahaha... thats verizon's wireless broadband TOS
05:27.52h3xLike what
05:28.06JerJerbecause its not sip
05:28.09JerJeror H.323
05:28.15JerJeror mcgp
05:28.20h3xso what
05:28.20tc0nnor skinny
05:28.23HogieSplasPood: you wouldn't want to do any of those anyway
05:28.26h3xit still uses more bandwidth when you have a crappy ass codec to use
05:28.37Hogiethe latency is so freaking high on BroadbandAccess it sucks
05:28.39JerJerthat will change, so your argument is moot
05:28.42Hogieeven ssh is laggy
05:28.51h3xwithout a hardware change?
05:28.54tc0nnnot as band as starband
05:29.00SplasPoodHogie: I heard it was still better than anything else around...
05:29.12JerJeri don't work for Digium, so i duno
05:29.14h3xif digium charged the $10 per channel license for g.729
05:29.19Hogieit is... i have a pc card
05:29.20h3xthat means it would be $110 for a iaxy with g.729
05:29.22Hogiewith it
05:29.36*** join/#asterisk B4 (n=B4@202.69.48.245)
05:29.42JerJerthe $10 is for the codec implementation inside of asterisk
05:29.49JerJerand that's retail price
05:29.54JerJerwho pays retail ?
05:29.57Hogiebut there's no way teamspeak will even work on it, I dont know how you would live with 300 - 400ms latency and IAX2/SIP protocol on it
05:30.00B4hi
05:30.38HogieI even tried uploading pics from webcam on the laptop before while driving down the road, and even at 30 secs, it was having problems getting overloaded at times
05:30.50B4can anyone suggest what would be the best way to hookup a PRI (Euro) to asterisk?
05:30.53*** join/#asterisk sangee (n=rkuru@Toronto-HSE-ppp3697175.sympatico.ca)
05:31.00Hogiewith an E1?
05:31.00JerJeryeah GPRS is not friendly to anything cool, yet
05:31.02tc0nnB4- a digium card
05:31.08tc0nnTE410P
05:31.14JerJerTE411P
05:31.23JerJerits worth it
05:31.35tc0nnI'd stay away from the 411 for now
05:31.36B4what about the new dual span cards?
05:31.49JerJertc0nn:  why you say that?   we use them in production
05:31.52*** part/#asterisk ilustrate (i=user@200.92.29.182)
05:31.56tc0nnI've got two... with the echo cards removed. Box is unstable with them on.
05:32.04sangeeanyone know how to setup realtime sip registration using mysql? someone please help me?
05:32.06JerJerrock solid here
05:32.16tc0nnWhat hw/os ?
05:32.43JerJerdell with e1000 disabled
05:32.49JerJerlinux 2.4.20
05:33.01Hogie1850?
05:33.07B4any experience with TE210P?
05:33.15JerJersc 1425
05:33.19JerJeri think
05:33.25JerJersomething like that
05:33.26tc0nnWell.. I've got one Dell 1750 w/RHES4, another 1750 with RHES3, neither have the e1000 disabled. Its on a different IRQ anyway.
05:33.44JerJerget RES shit off of there
05:33.47*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:33.53JerJergo with slack or debian - good lord
05:34.10Hogiegentoo 4evah!
05:34.26JerJertc0nn:  doesn't matter...for some reason Digium's cards hate the e1000
05:34.30B4JerJer: any experience with TE210P?
05:34.33JerJerno matter what IRQ
05:34.37JerJerno we have all 4 span cards
05:35.24tc0nnWell, its been up and running ever since we removed the echo card. I may try a pci nic and put the card back in.
05:35.32JerJeror the e1000 hates digium - haven't figured it out exactly
05:35.48tc0nndigium wanted me to do that so they could get a developer in the box to see wtf was going on.
05:35.52B4I am just 1 km from telco exchange, can run the PRI straight on copper without HDSL right?
05:36.00tc0nnno
05:36.02JerJertc0nn:  yep
05:36.10JerJeri know the drill
05:36.14GoshenWhere does monitor dump the recordings?
05:36.28B4no?
05:36.34JerJer<PROTECTED>
05:36.36JerJeri'm guessing
05:36.37tc0nnB4. I think t1 only goes 384 feet on cat5, probably less on a telco bundle.
05:36.38h3xthe nice thing about having sip and rtp seperated is if you get your DIDs from 3rd parties, you can set up the RTP session directly between a consumer on the internet and that underlying provider without chewing up your own bandwidth
05:36.46h3xand keep an accurate CDR of the time logged
05:36.51JerJerdon't buy from 3rd parties
05:36.58B4E1?
05:37.19tc0nne1 probably less...
05:37.25tc0nnSince its 25% more bw.
05:37.38h3xall for just managing call setup traffic
05:37.45B4hmm ok, so would need the HDSL on both ends
05:37.49B4more cost!
05:37.57tc0nnWhy is that?
05:38.14tc0nnShould be able to find old/unused smartjacks in any office building...
05:38.25B4well, I would need to buy the equipment
05:38.31JerJerh3x:  so then how do you garuntee QoS that way?
05:38.41B4smartjacks?
05:38.41*** join/#asterisk dev2005 (n=dev2003@222.33.36.205)
05:38.42h3xhow do you guarantee QoS when your customer isnt on your network
05:38.45h3xit dsoent fucking matter
05:38.51JerJersure it does
05:38.57JerJerits called peering
05:39.13tc0nnsmartjacks/hygain unit/xDSL to T1 converting testpoint/demarc.. whatever you want to call it.
05:39.18JerJerwhen you get peering agreements you make damn sure they pass on QoS information
05:39.26h3xwell the places i get dids on voip are on better internet connections than i could ever afford
05:39.39JerJeryou hope
05:39.48Vcoheh..heh..
05:39.50tc0nnh3x - who are you using?
05:39.53tc0nnfor did's?
05:39.54h3x:P
05:39.56JerJeri don't base my business on hope
05:40.09h3xa half dozen different clecs
05:40.28syleSet option TDD MODE, value: OFF(0) on Zap/3-1
05:40.30h3xwww.carrierone.net/dids has a excel spreadsheet
05:40.32tc0nnHow much do they charge? Just curious if I'm getting ripped off.
05:40.34sylewhats this mean
05:40.36h3xalthough a lot of stuff is missing from it
05:41.08tc0nnWhat kind of circuit was on that zap channel?
05:41.09h3xthey are all different ,and its a apples to oranges comparison because what im buying them for wholesale requires a $5k-$25k a month committment
05:41.16JerJertc0nn:  if you are paying per minute, you are getting ripped off
05:41.20syleILEC line
05:41.23h3xshit
05:41.24h3xno
05:41.30h3xyou get ripped off if its flat rate
05:41.38tc0nnno, I pay $10 per hundred per month.
05:41.44h3xbecause what happens is they turn off simultaenous call capability
05:41.49JerJerum
05:41.53JerJeryou buy UNE
05:41.57*** join/#asterisk oej (n=oej@ti111210a080-0697.bb.online.no)
05:42.06h3xoh you are talking about DIDs on a PRI or what?
05:42.10tc0nnyes
05:42.15tc0nnDID on PRO
05:42.16tc0nnPRI
05:42.17h3xno im talking about vlip delivery
05:42.17JerJertrunks are separate yo
05:42.22tc0nnok
05:42.23h3xvoip
05:42.41JerJerthat's the VoIP provider versus real telco carrier play right there
05:42.43Goshenwhat file has the moh default music?
05:42.52h3xfor instance i pay KMC $0.002/minute or they want $2/mo for a flat rate did with 500 min cap
05:43.03JerJerwhich is highway robbery
05:43.11h3xanybody with a fucking calculator can figure out
05:43.13tc0nn<PROTECTED>
05:43.14h3x500 minutes is $1
05:43.14dev2005who use E400P ?
05:43.27Goshentcc0nn: what tarball has the default moh mp3 files?
05:43.41JerJerwe pay $45 a month for a OC-3 cross connect
05:43.42h3xit isnt that bad with little hidden fees
05:43.52tc0nncvs asterisk-sounds probably
05:44.05GoshenI just installed asterisk on my new system and it doesn't have any files in /var/spool/asterisk/moh
05:44.08GoshenI did asterisk-sounds
05:44.19tc0nnupload your favorite mp3 then
05:44.23tc0nnand make mpg123
05:44.32syleyou mean /var/lib/asterisk/sounds lol
05:44.40h3xconsidering their whole market coverage is tier 3 markets
05:44.41JerJersyle:  not for MOH
05:44.43tc0nnnot for moh
05:44.46h3xthat price is nothing
05:44.56h3xthats places that level 3 wants like 0.03/Minute
05:44.58h3xwell
05:45.01h3x2.something anyway
05:45.01Goshenmy last install had those nice moh files
05:45.14JerJerh3x:  we don't pay per minute for our DIDs
05:45.17JerJerperiod
05:45.22h3xand they are using telica switches
05:45.42JerJerand just because we are only offering DIDs in one state doesn't mean we don't have more states
05:45.44h3xyeah well they dont do usage sensitive when you are doing TDM for everything
05:45.58tc0nnWhats GlobalCrossing like these days? They've been cold-calling the shit out of me..
05:46.02JerJerits the right way to do it
05:46.12JerJerreselling someone elses VoIP is just asking for trouble
05:46.16h3xit dosent matter how you do it when your carrier uses VoIP anyway and double converts
05:46.28JerJerdon't buy from carriers that do that shit
05:46.38h3xall of them are using voip now.
05:46.43JerJerum no
05:47.24h3xThe larger ones using sonus gear has already replaced IMT links on the back end of their legacy switches with voip
05:47.30h3xto conserve bandwidth between sites
05:47.42JerJerwho says we have to use larger ones?
05:47.43tc0nnAnyone know if a sip.conf -> realtime upload script exists?
05:47.48h3xso its already going through a g.729 compression cycle once
05:48.21h3xBecause all the smaller ones are just dumping their traffic on qwest, mci, global crossing, sprint, at&t anyway?
05:48.31JerJerum i am talking about DID yo
05:48.41h3xUh....
05:48.55h3xDIDs transported for long distances need echo cans
05:49.02h3xon TDM
05:49.08JerJerok and the problem is ?
05:49.21h3xwell everybodys depreciating that shit including global crossing, focal, etc.
05:49.23h3xeven mci
05:49.38JerJerand who says they need to be transported long distances using TDM?
05:49.44h3xthe MRCs are rediculious
05:49.51h3xwell then you have to buy colo everywhere otherwise
05:50.02JerJerok and the problem is?
05:50.22BharatSHello there
05:50.28JerJermoo
05:50.30h3xthe point is that their million dollar sonus, telica, whatever gear is a hell of a lot better quality than your shitty asterisk boxes at doing echo cancelling and codec translation
05:50.45BharatSI am working on building a Manager interface for the Asterisk
05:50.46JerJerif you say so
05:50.53tc0nnCool. We need one of those.
05:51.09JerJeryet what kind of technical complaints can you find about us?
05:51.09h3xfor example, on a properly configured asterisk box with a t1 card in it connected to a PRI to a CLEC here
05:51.24h3xwith echo cans disabled as they should be, g.711 fax over a LAN to that box i cant get more than 9600 bps
05:51.37JerJerasterisk was not designed with data in mind
05:51.44h3xbut g.711 over the public internet, many hops away to their east coast gateways even i get 14.4kbps no problem
05:51.51h3xit aint data, its sound
05:51.51BharatScan anybody suggest me the some key features that are need to be considered in building the ASterisk Manager interface
05:52.00h3xif my fax machine can notice a sound problem across a lan
05:52.08h3xthen its not doing a very good job with voip translation to tdm
05:52.36h3xif it has to renegoate to 9600 bps
05:52.51JerJeri'm done feeding the troll
05:52.53JerJergoodnight
05:52.56h3xhahahah
05:53.05h3xwell asterisk isnt exactly quality software
05:53.13JerJerthen don't fucking use it
05:53.24h3xnor is linux either, but lots of religious nuts to go around
05:53.41JerJeryet you benifit from both on a per minute bases
05:53.48h3xim using SER
05:53.54JerJerwho isn't?
05:54.10bonez41what's SER, at the risk of sounding ignorant?
05:54.17h3xwell i have only 1 asterisk box for transporting calls, and thats to handle two local PRIs thats it
05:54.27tc0nnSip proxy
05:54.31h3xeverything else is offloaded directly to a carrier's voip gateway or to MAX TNTs
05:54.47fadopenH323 and oh323 are same? i want to install openh323 with asterisk
05:54.50h3xso im not really using it that much :P
05:55.09JerJeryou are still benefiting from it
05:55.16h3xfad save yourself some trouble by using neither
05:55.17bonez41h3x, why not writing your own, something better?
05:55.18JerJeryet you bitch at it every change you get
05:55.18tc0nnsingle point of failure
05:55.19h3xI am?
05:55.46fadi have tp installed
05:55.46h3xI was just discussing the use of it in a "ITSP call transport" type application
05:55.58tc0nntoilet paper?
05:56.00JerJercalls go in and/or our of your two local PRIs don't they?
05:56.01h3xor using it instead of big irons
05:56.12Goshenfound the sample moh files /asterisk-1.0.9/sounds/fpm-calm-river.mp3
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05:56.16Goshennot sure why they didn't install
05:56.21h3xvery few, and i am going to move those over to tnt's t1 ports when i get more dsps in a box
05:56.40JerJeruntil then, stop bitching
05:56.40syleis it stupid to set echo cancellation on a fxs port hehe
05:56.46h3xwhen my xspedius contract expires im porting everythign to KMC
05:56.56tc0nnI'm out.. Have fun guys
05:56.56h3xbecause xspedius cant do E911 MSAG records on a per DID basis
05:57.04*** part/#asterisk tc0nn (n=tec@spyglass.timsnet.com)
05:57.08h3xthen it will be completely irrelevant
05:58.09h3xI'm not really bitching about anything, other than pointing out that I don't see how taking TDM into a ESP's network and converting to VoIP with $0 software just so you can "maintain QoS"
05:58.21h3xreally achieves anything over using all the carrier's expensive gateways
05:58.30JerJerlets see- what bout application logic
05:59.09BharatShas anybody worked on the Asterisk Manager Interface?
05:59.10JerJeror i want to see a carrier gateway become a ham radio repeater controller
05:59.15BharatSplease reply
05:59.20h3xwhen * really costs more per channel than second hand equipment with DSPs and G.729 licenses built in
05:59.22JerJeror control X-10 devices
05:59.35JerJeror interface the telephone with the web
05:59.44Insanity5Has anyone uses didx.org?
05:59.57JerJerInsanity5:  no - you shouldn't either
05:59.59BharatSnop
06:00.00h3xwell of course its the best thing around for end user applications but that isnt what nufone nor carrier one does
06:00.12h3xi mean, objectively
06:00.16Insanity5JerJer - It seems the biggest porblem would be provider reliability and latency.
06:00.16Insanity5hehe.
06:00.17BharatSsetting up gui for the Asterisk...
06:00.43JerJerh3x: i've configured thousands of end-user asterisk boxes
06:00.56JerJerand have written hundereds of custom apps for end-user asterisk boxes
06:00.59JerJerhow is that not what we do?
06:01.05Insanity5Those are some pretty bad sample on hold music files.
06:01.11h3xshrug i guess its the chicken and the egg
06:01.30Insanity5Anyone have a good greeting/on hold .wav/.mp3 colelction? :P
06:01.40Insanity5I thought about ripping one of the tracks off the half life 2 soundtrack... hehe
06:01.52JerJerInsanity5:  have The Voice record your own custom prompts
06:02.05Insanity5the voice?
06:02.10JerJerAlison
06:02.16Hogiewe got mr moviephone to do our prompts
06:02.17h3xi don't believe voip belongs on the public internet either, but if its gonna end up there then it dosent really matter where it comes from that much
06:02.22h3xits all finger pointing from there
06:02.23Insanity5JerJer - whois alison?
06:02.30JerJerThe Voice
06:02.32h3x(back to my point about offloading RTP)
06:02.36Insanity5huh?
06:02.53Hogiehttp://www.digium.com/index.php?menu=product_detail&category=extras&product=THEVOICE
06:02.57Hogiethere Insanity5
06:03.03JerJerthe voice!
06:03.13h3xobviously if someone called and bitched that they cant reach some gateway
06:03.32h3xall youd have to do is change an option to proxy that call over
06:03.48JerJernot if your DIDs are coming from that one gateway
06:03.57h3xso in a sense you sorta have a means of redundancy workarounds
06:04.02Insanity5JerJer-  No sample files :(
06:04.18JerJershe records custom prompts
06:04.26h3xwell, most of these providers have both direct trunking into here and public ip access
06:04.35HogieInsanity5: you know the demo file?
06:04.41h3xif i change where i tell them to send the call to it will come in on a private ip circuit
06:04.42Hogieer, the demo dialplan?
06:04.45JerJeroperative word being 'most'
06:05.01Insanity5Hogie - which one?
06:05.10Insanity5JerJer - I thoguht some sexy lady would be the best bet.  hehe
06:05.14h3xmost being all but the morons like Level(3) which serves areas that nobody else does
06:05.18HogieInsanity5: when you do make samples after compiling *
06:05.20h3xso if a subscriber had a number with them
06:05.23h3xthrough anybody
06:05.27h3xthey would have the exact same problem
06:05.44Insanity5Hogie - Are tehy .wav.mp3 files somewhere?
06:05.48JerJernot if you pulled it off of the tadem
06:05.54h3xer. i forgot to mention they dont have a private IP MPLS whatever option
06:06.02Hogiegsm, in /var/lib/asterisk/sounds
06:06.12Insanity5Hogie - Gas, winamp can't play those, can it?
06:06.21Hogiethere's a gsm plugin for it
06:06.22Hogieso yes
06:06.25h3x"pulling it off the tandem" in the 9000 rate centers i can cover right now with 6 carriers is seemingly impossible
06:06.34JerJerif you say so
06:06.57*** join/#asterisk Blake0PS (n=blake@blakeops.com)
06:07.00h3xthat would be like at least 6,000 boxes that cost $1500-2000k a piece or something
06:07.07h3xer $2000
06:07.07JerJerno
06:07.47h3xpulling off the tandem huh, are you using ss7 box or what
06:07.58Blake0PSCan Asterisk detect ACTS tones (from Bell System payphones)?
06:08.17h3xoh my god
06:08.27h3xi hope you are setting up COCOTs
06:08.37Blake0PSdeath to COCOTs
06:08.41Insanity5Hogie - God, the quality of those gsm files sucks.
06:08.49h3xare you writing an article for 2600 or something?
06:09.00Insanity5Do pay phones even exist anymore?
06:09.10h3xsomebody needs to make a voip payphone
06:09.14Blake0PSI use payphones instead of a cell phone, it's cheaper.
06:09.15JerJerthey already have
06:09.18h3xor at least an ATA that does ground start
06:09.22Insanity5Blake0PS - IF there's any left.
06:09.49h3xall the payphone guys have gotten into ATMs instead
06:09.56h3xi went to this stupid payphone show here in vegas
06:10.05h3xall of them are has beens
06:10.14h3xtalking about the old days of payphones
06:10.15Insanity5I really don't know if "the voice" is that great.
06:10.23Insanity5Maybe it'd sound better if it wasn't crappy gsm.
06:10.32Blake0PSI'm a phone phreak, there are payphones here, and I want to know if Asterisk can reliably detect ACTS tones
06:10.39h3xInsanity5: theres a web site with a whole bunch of voice recording talent out there
06:10.41nick125_lappytime to redo my Customer Care/Support menu...this should be fun
06:10.45GoshenI saw a payphone the other day, I had to look twice because it was an unusual sight
06:10.46h3xallison is on it among others
06:10.53Insanity5h3x - Somebodyh as to have a repository of common items :)
06:11.00Insanity5hehe
06:11.00h3xif you just google for allison smith you will run across it
06:11.11JerJeror google john todd
06:11.25JerJerhe has a sounds repository of alison prompts
06:11.33Blake0PSHere are the payphones in my area http://www.yapl.org/list.php?list=user&show=cataloged&uid=9&perpage=1000
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06:11.36h3xblake__: So, why would anybody have ever written the code to do that
06:11.39h3xer
06:11.40Insanity5Is it better quality if you don't use gsm?
06:11.49JerJera lot of which made it into asterisk-addons, but there are some that didn't
06:11.55JerJerer asterisk-sounds
06:11.58JerJerwhatever its caleld
06:11.59JerJercalled
06:12.11h3xyou know whats funny, i accidently made a $13,500 aculab board a redbox once
06:12.30h3xspeaking of asterisk-addons
06:12.39h3xthe MYSQL() dialplan thing sucks ass
06:12.42Blake0PSACTS tone detection is useful. These phone exist and Asterisk is a PBX.
06:12.47h3xi should write a new one
06:12.59h3xbut maybe PGSQL()
06:13.09Juggiehow about just use agi
06:13.12h3xblake, how would you plug a payphone into asterisk
06:13.34Blake0PSFXS?
06:13.44h3xJuggie, well thats just it, i want to avoid doing that when its goign to get hammered with requests to fill in dialplan variables
06:13.58Insanity5Do you need a license to enter the payphone business or can you put your own outside your gas station?
06:14.01Juggieyou can set a dialplan var direct from within agi
06:14.04h3xblake, its ground start
06:14.26Insanity5Althoguh, I doubt you could net even $5/month :P
06:14.29Insanity5hehe
06:14.38h3xI know you can but calling an AGI, or using MYSQL(), forks a threaded process from hell, ergo Asterisk
06:14.47h3xto exec an external program
06:14.47Blake0PSoh, no FXS modules do ground start?
06:14.53h3xand it cant keep the mysql socket open
06:14.55h3xto reuse
06:15.06h3xwell on channel banks yes
06:15.13Juggieh3x, theres a PHP() as well
06:15.14h3xbut ive never seen a ATA that does ground start
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06:15.17Insanity5Would these voice prompts sound better if they weren't gsm?
06:15.42h3xjuggie, I just wonder what happens when fork gets called 200 times at the same time
06:15.54h3xthat would suck <TM>
06:16.03Juggieh3x, i believe that would be called bad design on your part.
06:16.22Juggieyou cant have it both ways
06:16.26Juggiefork per MYSQL
06:16.28Juggieor blocking
06:16.29Juggiepick one
06:16.38Juggieif you share a socket, then MYSQL() calls are blocking
06:16.41h3xneither.  how about app_mysql_that_works_for_shit.so
06:16.46Insanity5Here's a bunch of free voice prompts:  http://www.microsoft.com/downloads/details.aspx?FamilyID=ef62d47d-ce5a-44fa-864f-3c31c14769b7&displaylang=en
06:16.51h3xI might actually call it that
06:16.51Insanity5=)
06:16.58h3xwhen the module is loaded, it creates a pool of mysql handles
06:17.10Juggieh3x, its going to fork, or be blocking.
06:17.16h3xno it wont
06:17.21h3xmysql library is thread safe
06:17.49h3xdo the whole thing in C as an asterisk module
06:18.20Juggiebtw, how would you generate 200 calls a second
06:18.28h3xit aint that hard
06:18.30Juggieer, 200 calls to the fork
06:18.38Juggiewell, how many lines do you have
06:18.44h3xhahaha
06:18.45Juggieor whats your max active calls
06:18.49h3xsee www.carrierone.net
06:18.59h3xactually it wont be used here but
06:19.16Juggieok
06:19.20Juggieso give me a number
06:19.24Juggiemax calls on the system
06:19.24h3xasterisk can handle a couple thousand calls on one box if its all voip
06:19.33Juggiehahaha
06:19.34Juggieyou wish
06:19.37h3xit does
06:19.41h3xno codec translation
06:19.45h3xno zap channels
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06:19.59Juggieas long as * isnt in the RTP path
06:20.01Juggieyou will be ok
06:20.07Juggiemaybe
06:20.14h3xthat dosent really matter if its copying
06:20.22h3xbut yeah better without
06:20.23h3xin any case
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06:20.33Juggielinux isnt the best kernel for packet throughput
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06:20.39h3xthe point is, to make it work for everybodys applications better not just mine
06:20.41Juggieyou would do better without
06:20.41Insanity5What format should on hold music be in?
06:21.15h3xneeding to manipulate data in mysql or postgres or whatever is important in some situations to do for every single call through the system
06:21.19Juggieh3x, regardless, you could create a MYSQL with connection pooling yes.
06:21.24h3xsometimes several times for a single call
06:21.30GoshenAfter leaving a voicemail which is supposed to get mailed (using old config files on a new system) I get this error message
06:21.30Goshenpostdrop: warning: unable to look up public/pickup: No such file or directory
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06:21.41h3xsurely i hope that cdr mysql and so forth do it this way
06:21.43Goshenis that a postfix configuration problem?
06:21.45Juggieh3x, then i must ask you, why dont you do all your work from within one script
06:21.59h3xbecause, it may be at different call stages
06:22.04Juggieyour point?
06:22.07Juggieput it all in one agi.
06:22.32Juggieagi can contain an entire call flow from answer to hangup
06:22.33h3xbecause the dialplan cant move along
06:22.36h3xif you are in an agi
06:22.45Juggieyou dont need dialplan within an agi
06:22.48Juggieyou can code a dialplan
06:22.53Juggiein perl, or php or whatever
06:23.05nick125_lappyanyone here know a cheap ATA that is unlocked and works nicely with asterisk (under $50 maybe??)
06:23.14h3xi guess but some of the stuff needed may not be supported in the agi
06:23.16Insanity5What's a good cheap voip phone or ATA?  It seems like the adapter is the better way to go with more flexibility, but on hold and stuff might be an issue.
06:23.26Juggieh3x, give me an example.
06:23.34h3xor the dialplan needs to be manipulated using sql storage
06:23.40h3xby outside programs
06:23.53Juggiewhat do you mean
06:24.03h3xWell ill just make up something
06:24.10h3xlets say you were to set up a vonage wannabe co
06:24.22h3x(bad example from what iw as just talking about but whatever)
06:24.43h3xyou have an extranet that creates new shit in your dialplan via sql depending on what your users select for options
06:25.06h3xmeanwhile in your dialplan it may need to do sql queries to grab settings such as,
06:25.07Juggieuhhuh
06:25.13h3xif some feature is enabled or not
06:25.16Juggielet me show you an example
06:25.20h3xor, what your forward to phone number is
06:25.27h3xor whatever. ...
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06:25.54h3xit would just make so much more sense if you could use MYSQL() to grab that and stuff it in a variable at will
06:26.03h3xif it would scale like that
06:26.32Juggietake a look @
06:26.53h3xI realize you can do this with an AGI, but the thing is that with something that is so pivotal in any given dynamically configurable application
06:26.54Juggiewww.pastebin.ca/20399
06:27.02h3xits time well spent to make fucking sql queries inside the dialplan work
06:27.07Juggiethey do work
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06:27.55h3xi have my doubts about asterisk forking a couple dozen times in a second and not screwing up something
06:28.20Juggieso then reduce the forking
06:28.32Juggiefork once for an agi
06:28.35Juggieor fix mysql
06:28.37h3xyou cant if thats because of a couple dozen calls set up at once
06:28.59h3xthe other problem is that mysql will shit itself
06:29.03h3xif you connect to it too many times
06:29.49h3xit sure would be nice to rewrite asterisk
06:30.04h3xin an interpeted language with C modules doing the tough shit on the back end
06:30.13crash3mtalking about resperl?
06:30.14Juggieholy crap dude
06:30.17Juggiemysql will be fine
06:30.25Juggiephp connects to mysql every web hit
06:30.39h3xi thought php used a persistant connection pool
06:30.40Juggieand servers get hundreds, of hits a second... it will be ok.
06:30.48Juggienot unless you explicitally enable it
06:30.51h3xoh
06:30.59h3xhmmm
06:31.00crash3mI've seen resperl self-fuck an * box
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06:31.09h3xheh
06:31.17Juggiei did a hammer/bulkdial a few weeks ago
06:31.18Juggieusing *
06:31.22Juggiewith mysql cdr enabled
06:31.26h3xahahaha
06:31.32Juggiei crashed the pbx providing me the t1
06:31.34Juggie* was ok
06:31.37crash3mheh
06:31.43Juggiei actually locked the line card on the other end
06:31.45h3xthat is amazing
06:31.49Juggieit needed a hard reset
06:31.58h3xwhen i started using asterisk a couple years ago
06:32.08h3xi simply hooked it up back to back to an aculab card
06:32.09h3xPRI
06:32.15h3xtold the aculab card to place 23 calls at once
06:32.19h3xand the asterisk box.. well....
06:32.23h3xthe fucking kernel paniced
06:32.38h3xthings have come a ways
06:32.47h3xbut still
06:32.58h3xit could be better.
06:33.11h3xHEAD crashes a lot
06:33.18syleSetVar(loop=loop+1)
06:33.20sylehow do i do this
06:33.24h3xand that just comes with any huge ass C program
06:33.28Qwellsyle: do what?
06:33.34syleincrement var by 1
06:33.55QwellSetVar(loop=$[${loop} + 1])
06:33.56Qwellmaybe?
06:34.15h3xbut if the crazy shit went into glue modules and the logic was all written in interpeted languages
06:34.40h3xI did this exact same thing with my aculab software development, i wrapped its libraries on the C backend of an interpeted language Pike
06:35.01h3xafter i got all those bugs worked out i never had any crash problems anymore
06:35.06Juggieh3x, gdb the crashes, post patches.... people will love you
06:35.18h3xbecause Pike's garbage collection took care of a lot of things
06:35.42h3xbut it could be done on anything.. Lua, Ruby, Perl, Python, Whatever
06:36.04h3xeven better is have it hinged on one thing and then glue a bunch of interpeters together so you can code in anything
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06:37.20h3xit sounds really sick but i think it would work
06:38.19Juggieagi allows any intrepreter
06:38.30h3xi know :P
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06:39.34h3xhehe ruby would be a crazy language to write a pbx in
06:39.54Juggiefyi... * has a new lang for dialplan
06:39.55Juggieael
06:40.06h3xadv extension logic?
06:40.13Juggiemaybe
06:40.14Juggienot sure
06:40.37h3xThe point of making most of the pbx in an interpeted language is you can inherit objects and modify them
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06:41.03h3xfrom a 3rd party development standpoint anyway
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06:41.56h3xit would be a pain in the ass to decide what language to use though
06:42.18h3xpython is organized and popular now but its slow as hell
06:42.32h3xwhich normally isnt a big deal but is for this sort of thing
06:43.26MustangMattAfter all the help drray gave me last night getting asterisk setup and working I decided to writeup an article on how to setup asterisk from scratch using debian. I tried to write it from a beginner's point of view. Would anyone be interested in reading what I've got and giving me feedback?
06:44.24h3xi hate to tell you this but i think that already exists :)
06:44.27MustangMattMore specifically, it's how to setup asterisk as an answering machine.
06:44.37h3xoh
06:44.38MustangMatth3x: I was unable to find one as simple as mine.
06:46.01h3xwell i would but i dont use debian so..
06:46.03MustangMattAnyway here's what I've got. http://kaatman.com/asteriskansweringmachine.html I'd love feedback. It's still missing a giant section on extensions.conf and doesn't achieve all the goals I setup yet. Think of it as written for newbs by a newb.
06:51.24Guggemandhmm, my * installation doesnt seem to retry a sip register when it fails
06:53.29Blake0PSwhat part of a call to an asterisk box detects and decodes DTMF to a digital digit?
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06:57.33X-RobBlake0PS - None of it.
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07:03.43Blake0PSnotice i asked what part of a call, not what part of asterisk
07:04.01Blake0PSsomething has to convert my 2 frequencies making up the digit 9 to an analog form
07:07.51X-RobBlake0PS - perhaps more information. Describe, exactly, where you're pushing this nine.
07:08.57Blake0PSI want to find out where the A to D conversion occurs so that I can detect my own analog signal and convert it to something digital
07:09.14Blake0PSACTS tones to be specific
07:09.32Guggemandif you have your phone connected to some box i would think that converts it
07:09.50Guggemandif its a softphone it sends the digital when you push it
07:09.52X-Rob*picks one of the many places it could be, because you're not being anythign near 'exactly'* In the phone
07:10.31Guggemandif its a phone not connected to anything, then its magic :)
07:11.27Blake0PSACTS phone -> CO -> FXO -> Asterisk
07:12.00X-RobAhha
07:12.06X-Robso it's coming in via a X100 or TDM?
07:12.22Blake0PSsomething of that flavour
07:12.45X-RobSorry
07:13.03X-Rob'How is it coming in. Via a X100 or a TDM?'
07:13.19X-Robthe answer is 'X100' 'TDM' or 'Something else' and you then explain what the somethign else is
07:13.22X-Robwhy are you being so damn stupid?
07:13.45Blake0PSask a stupid question...
07:13.56Blake0PS"Which are you using, X100P or TDM"
07:14.13GuggemandBlake0PS if you want help you should answer in a sane way
07:14.18Blake0PSThe answer is TDM with FXO module
07:14.41X-RobOK. TDM's do DTMF detection in hardware.
07:15.14X-Robhowever, if you do care about significantly clever DSP stuff, look at spandsp, which does soft faxing via asterisk
07:15.35Blake0PSwhat FXO interfaces have software DTMF detection?
07:16.06X-RobX100
07:16.11X-Rob(as far as I know)
07:16.29X-Robsome channel banks have hardware, some channel banks don't
07:16.31X-Roblook at libpri
07:17.35X-Robno
07:17.38X-Robdon't look at libpri
07:18.00X-Roblook at dsp.c
07:18.07X-Rob(in /usr/src/asterisk)
07:18.07X-Robsorry
07:18.41Blake0PSaha
07:18.48Blake0PSthanks
07:20.28*** join/#asterisk tuxinator_linuxM (n=tuxinato@ip68-109-146-168.ph.ph.cox.net)
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07:28.23*** mode/#asterisk [+o drumkilla] by ChanServ
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07:41.28djin_ibDoes anyone know a reason for 'HDLC Abort (6)' after upgrading from 1.0.7 to 1.0.9(.1)?
07:41.47*** join/#asterisk rumba (n=ropawa@cpe-68-201-147-53.sw.res.rr.com)
07:41.59djin_ibManxPower suggested avoiding onboard NIC yesterday.
07:42.38djin_ibHowever, this is not easy for me to try (remote location)
07:43.27*** join/#asterisk Sloboda (n=sloboda@195.137.227.70)
07:46.10nighty-uhmm
07:48.00SlobodaHi! Could you tell me your preferences for linux soft phone for *. I have tried khone and linphone. Need only dialing buttons and phone book.
07:48.52SlobodaIs one with more pretty interface?
07:49.28SlobodaLike ExpressTalk, for example. (It is only for Win)
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08:03.35ukhis there any particular IRC channel where people hang out discussing non-Asterisk SIP/H.323/VoIP issues with servers, proxies, gateways and clients?
08:04.43SlobodaREGISTER 123
08:05.10crash3mukh: /msg chanserv list *voip*
08:05.54*** part/#asterisk Sloboda (n=sloboda@195.137.227.70)
08:06.22crash3mis it just me, or is voip-info.org going really slow right now?
08:07.54crash3mwait, it might just be the nightly crons starting
08:10.23gordonjcpall over the world, it's time for someone's nightly cron to start
08:12.16nighty-gordonjcp: not here
08:12.44*** join/#asterisk edwin_ (i=edwin@252-131-222-203.rev.techex.net.au)
08:13.00gordonjcpnighty-: nor here
08:13.24*** join/#asterisk wd (n=wd@pdc.investtel.cz)
08:13.25gordonjcpit's pretty slow here too
08:14.29nighty-gordonjcp: it's more because it is day time
08:14.37nighty-gordonjcp: like 10:00 AM
08:14.38nighty-:)
08:14.45gordonjcp9am here
08:15.01gordonjcpit may be 3am where the servers are, prime time for cron.daily
08:15.07nighty-gordonjcp: where ?
08:15.17nighty-gordonjcp: UK ?
08:15.42gordonjcpI'm in the UK, I don't know where the servers are
08:15.45gordonjcppresumably the US
08:16.19gordonjcp-5 would put them in eastern US, at about 3am
08:16.48*** join/#asterisk Liquefact (n=NNSCRIPT@62.116.83.61)
08:17.05nighty-gordonjcp: LA
08:17.31gordonjcpok
08:17.40gordonjcpwestern US, 1am or so?
08:17.48nighty-gordonjcp: PNAP , LAX
08:17.58nighty-gordonjcp: they are BBN customers
08:18.14nighty-gordonjcp: for IP transit
08:18.52*** join/#asterisk gaffer (n=cliff@69.36.245.165)
08:18.56nighty-gordonjcp: commpartners I mean
08:19.07gordonjcpah right
08:19.37Liquefactis there anyone there who can answer my questions about asterisk
08:19.55*** part/#asterisk gaffer (n=cliff@69.36.245.165)
08:20.20gordonjcpLiquefact: well, since you haven't told us what your questions are, there's no way of knowing
08:20.38Liquefactok, i'll tell
08:21.18*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
08:21.30*** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net)
08:21.38Liquefacti'm to set up an intercom system which will work in a simulator
08:21.38Lan16`spdy^gTphello
08:21.50Lan16`spdy^gTpi've a problem can somebody help me ?
08:21.56Liquefactcan asterisk be used in this system as an intercom
08:22.15Lan16`spdy^gTpodbc say that its connected to my mysql database
08:22.16gordonjcpLiquefact: depends what you want it to do
08:22.33Lan16`spdy^gTpextconfig.conf is set up
08:22.34dudesLiquefact - I believe you can use console/default as an intercom ... (via your soundcard.)
08:22.46Liquefacttwo way voice communication between consoles, and nothing more
08:22.54Lan16`spdy^gTpand for iax ive no registration for peer
08:22.56Lan16`spdy^gTp:s
08:23.05Lan16`spdy^gTpsomebody now the problem ?
08:23.40Liquefactbut the communication must be digital, recordable, and traceble
08:24.03dudesLiquefact - that's what the wiki is for
08:24.06dudesuse it
08:24.06*** join/#asterisk Inv_arp (i=junya@adsl-8-230-188.mia.bellsouth.net)
08:24.25Lan16`spdy^gTpdudes can you help me with the odbc support
08:24.31Lan16`spdy^gTpeverything is good
08:24.34Lan16`spdy^gTpodbc loaded
08:24.38Lan16`spdy^gTpconnected to database
08:24.45Lan16`spdy^gTpextconfig is set up
08:24.56dudesLan16`spdy^gTp - I read what you typed before
08:25.04Lan16`spdy^gTpok
08:25.09Lan16`spdy^gTpand you think ?
08:25.17Liquefactin asterisak home page this channel is considered as a support channel
08:25.23dudesI don't use odbc
08:25.35Lan16`spdy^gTpive try to use mysql only
08:25.41Lan16`spdy^gTpin result with the cvs
08:25.54Lan16`spdy^gTpthe phone don't ring etc...
08:25.55Liquefactthat's why i'm asking questions here, you may not reply me and so not bothered
08:26.08dudesLiquefact - read the fucking wiki
08:26.15dudesyou'll find what you need
08:26.16Lan16`spdy^gTpauthentification with mysql is good but everything go down when i make a call
08:26.31Lan16`spdy^gTpits wy i dont want to use the cvs
08:26.50Lan16`spdy^gTpi use it 3 times and 3 times it give me problem
08:26.51Lan16`spdy^gTp:s
08:26.53dudesLiquefact - it's your fault you're too lazy to read
08:27.01tzafrir_laptopit is a free support channel: we help if we feel like it. Mostly if the question is interesting enough
08:27.14Liquefactfuck you dudes
08:27.20dudestzafrir_laptop - hehe
08:27.30Lan16`spdy^gTp^^
08:27.34dudesLiquefact - you're not my type ...
08:27.49Lan16`spdy^gTpcan someone have got some idea with odbc support
08:28.09syleis there a digit to insert a pause is dialing DTMF tones?
08:28.32sylein
08:28.45Liquefacteverywhere in the net there are bastards like you trying to yell to people trying to learn something
08:28.46drumkilladudes: if you don't want to help, don't answer the question.
08:29.20*** join/#asterisk meppl (n=mephisto@84-245-169-70.ipool.celox.de)
08:29.58Lan16`spdy^gTpLiquefact whats your problem ?
08:30.03dudesdrumkilla - I told him how he could achieve his goal; all he has to do is read like the rest of us.
08:30.48sylehow do you insert a pause in a dial sequence? anyone know?
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08:30.59Lan16`spdy^gTpno
08:31.28dudesdigital ... (asterisk is  a software PBX?) ... traceable (of course) ... recordable (yes)
08:31.54whiskerstyle: use 'w' characters; ie Dial(Zap/1/1234www111)
08:33.25Lan16`spdy^gTpwhy odbc dont request my database however odbc is connected to :^o ?
08:34.04*** part/#asterisk ukh (n=ukh@130.226.34.10)
08:34.27tzafrir_laptopLiquefact, what do you mean by "traceable"?
08:35.43dudesLiquefact - I will not respond to your last statement (but fuck you) ... you can use console/default as a intercom.  You can use several ways to trace via asterisk.  you can use monitor to record any use of the intercom extension.  Look I helped.
08:36.10tzafrir_laptopLiquefact, Asterisk uses by default a rather low-quality "phone-line" encoding for sound
08:36.34tzafrir_laptopIs that quality good enough?
08:37.13dudesYou can use System to echo to a file via use of an extension.  Asterisk also provides other methods (cdr) and so on
08:37.30tzafrir_laptopI still don't see how you can trace (regardless of Asterisk), unless you have some other identification method there
08:38.07dudesyou can set a var via a channel (callerid) and append it to a file
08:38.26tzafrir_laptopdudes, but this is just an intercom, right?
08:38.38dudeshe wants it traceable
08:38.39tzafrir_laptopwhat generates the caller id in the first place?
08:38.41dudeslets not forget
08:38.47dudesthe calling in party
08:39.01dudeseverything coming into a asterisk PBX is identified
08:39.02tzafrir_laptopIsn't it an intercom device?
08:39.15dudesman you're high
08:39.27dudesa channel is bridged
08:39.32dudesto the intercom extension
08:39.42dudestherefore, you can trace it if you want to
08:40.18dudesa intercom can be connected to your soundcard via a PA system and have an extension set to it.  But you can trace it (if you want to)
08:41.05RoyKcypromis: ping
08:41.06Liquefactthanks all, i think i got the answer
08:41.12dudesGood
08:41.27Liquefactsorry for all bad words, and taking time
08:41.35dudesdo you want an example of how you can make it traceable via a working example
08:41.43dudesI'd be glad to PM you
08:41.56dudesIt's not like I don't do this 16-18hr's a day for a living
08:42.16Liquefacti have little irc knpwledge, and dont know what PM is
08:42.24Liquefactalso my english is not that good
08:42.26dudesPrivate Msg
08:42.34Liquefactyep, sure
08:42.43tzafrir_laptop~pm
08:42.43jbotextra, extra, read all about it, pm is private message, or perl mongers, or pathetic moron: when you see someone say pm, they're asking if you think that they're a pathetic moron, or something you don't do without asking permission
08:43.20dudesjbot knows all
08:43.20jbotand don't you forget it
08:43.21tzafrir_laptopLiquefact, you can also /msg jbot pm
08:43.39tzafrir_laptopthis will avoid the whole channel hearing about it
08:44.17tzafrir_laptop~all
08:44.18*** join/#asterisk juraj (i=juraj@smtp.rebelium.net)
08:44.31jurajhello, anyone has any idea what this means?
08:44.34jurajAug 18 21:20:09 WARNING[2179] channel.c: No channel type registered for 'zap'
08:44.34jurajAug 18 21:20:09 WARNING[2179] app_meetme.c: Unable to open pseudo channel - trying device
08:44.50jurajis it only that it uses ztdummy device?
08:45.06Lan16`spdy^gTpwhy odbc dont request my database however odbc is connected to :^o ?
08:45.08tzafrir_laptopjuraj, chan_zap not loaded? meetme needs a timing source?
08:45.09*** join/#asterisk r0m (n=SysOp@bl8-28-55.dsl.telepac.pt)
08:45.15r0ma
08:45.18jurajtzafrir_laptop: I have ztdummy
08:45.28jurajtzafrir_laptop: it works actually :)
08:45.35jurajtzafrir_laptop: I'm just curious about that error
08:45.36tzafrir_laptopjuraj, maybe chan_zap.so is not loaded?
08:45.37r0mgood morning
08:45.40jurajor warning
08:45.41tzafrir_laptopzap show channels
08:45.47jurajtzafrir_laptop: I don't want chan_zap
08:46.00jurajtzafrir_laptop: I'm a sip only pbx
08:46.07jurajtzafrir_laptop: :)
08:46.13tzafrir_laptopyou need it to use ztdummy for timing
08:46.16*** join/#asterisk esi (n=ewaldirc@simonis.xs4all.nl)
08:46.34jurajtzafrir_laptop: I actually use ztdummy and have no chan_zap, since the meetme works
08:46.39*** join/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com)
08:46.47jurajtzafrir_laptop: I have no problem with the conference, just curious about the warnings
08:47.28Lan16`spdy^gTpwhy odbc dont request my database however odbc is connected to :^o, someone can help me ?
08:47.39dudesjuraj - if you use conf you use chan_zap and the psuedo zap device
08:48.55juraj*CLI> zap show channels
08:48.55jurajNo such command 'zap' (type 'help' for help)
08:48.59juraj:)
08:49.26jurajok, so I presume it says, it tried the zap pseudochannel and if it could not load it (since chan_zap is not loaded) it tried the ztdummy pseudo device?
08:50.01tzafrir_laptopdudes, or maybe meetme reads from psedudo directly and not through chan_zap?
08:50.21jurajtzafrir_laptop: it has to, because I really have no chan_zap and it works :)
08:50.40dudesusarules1*CLI> zap show channels
08:50.41dudes<PROTECTED>
08:50.41dudes<PROTECTED>
08:50.48jurajI'm just curious, I'm trying to debug a bug that has happened to me yesterday
08:50.51jurajhttp://bugs.digium.com/view.php?id=4986
08:50.52juraj:)
08:51.29jurajdudes: it seems, that if chan_zap is not loaded, it uses the device directly
08:51.38jurajdudes: and that's what the warning is about
08:51.45*** join/#asterisk MustangMatt (n=Miranda@firewall.thoughtprocess.net)
08:51.53dudesthen use chan_zap an resovle it
08:52.08jurajno problem, so this is not a problem at all
08:52.19jurajthe device works
08:52.20dudesSince the ztdummy is a *dummy* drive for zap
08:52.46*** join/#asterisk Kernel_Core (i=Raph@217.218.94.242)
08:53.03Kernel_Corehi all , anybody familiar to configure X101P card ?!
08:53.21jurajwhat happened to me yesterday, that two people had a conference in a room and after cca 15 minutes, one got disconnected by asterisk
08:53.26jurajI'm trying to figure out why
08:53.52jurajone more question about the notice :)
08:53.53jurajAug 18 21:17:01 NOTICE[2179] chan_sip.c: stale nonce received from '<sip:1003@192.168.1.222>;tag=as3c32c77f'
08:53.55*** join/#asterisk nagl (n=nagl@137.208.4.162)
08:53.57dudesjuraj - I have 10 lines on a PBX and conferences last sometimes for hours a day ... no problems
08:53.57jurajwhat does this mean?
08:54.12jurajdudes: this is quite a complicated setup
08:54.16jurajdudes: :)
08:54.17dudeshaha
08:54.24dudesyou think you have a complicated setup
08:54.26dudesI laugh
08:54.32dudeshaha
08:54.35jurajdudes: :)
08:54.45jurajdudes: ok, this one is not that complicated on a pbx side
08:55.10jurajdudes: it goes through a checkpoint firewall in one IPSec connection and then it uses another ipsec encapsulated in it to the PBX
08:55.18jurajdudes: :-)
08:55.38Kernel_Coreis zaptel 1.0.9.1 compatible with X101P devices ? Like V.92 Intel softmodems ?
08:55.38dudesit sounds like you make things too complicated for it's own good
08:55.46jurajdudes: it took me quite a lot of time (including fixing bugs in racoon) to make this work.
08:55.55jurajdudes: it's not what I want :)
08:56.14jurajdudes: it's what a client wants. they absolutely require (by secure policy) to use checkpoint to enter the LAN
08:56.20jurajdudes: no other way to get there
08:56.35jurajdudes: and they need all clients protected by ipsec using their own certificates
08:56.44jurajdudes: nothing I can change about it
08:56.46*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
08:56.56jurajdudes: but it seems I found at least one bug in asterisk thanks to this :)
08:57.05Mimmuswhat's the exact meaning of 'overlapdial=yes'
08:57.13dudesperhaps it's a bug on your end
08:57.27juraji'm sure there's also a bug on my end
08:57.44jurajbut this one: http://bugs.digium.com/view.php?id=4986 is certainly in asterisk :)
08:58.16dudesit's not like asterisk is written is C or anything (not that I have anything against C.)
08:58.24dudesOr it's a program that is free
08:59.25*** join/#asterisk razu (n=razu@217-159-242-106-dsl.est.estpak.ee)
08:59.46jurajit's free and it's in c, i don't get what you tried to tell :)
09:00.02*** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin)
09:00.05PakiPenguinhello everyone
09:00.10jurajhello PakiPenguin
09:00.11juraj:)
09:00.24dudesjuraj - sorry ...  I'm pretty pissed because I spend the hole day today fixing shit for retards.  (and I had a date, but I had to work and fix shit.)
09:00.37PakiPenguindudes, i feel for you :(
09:01.48dudesjuraj - the fact is it's C (C is a good language for somethings.) and as good as C is it has it's flaws.  And, moreover, it's free so dont' expect the world.
09:02.09jurajdudes: np. I'm trying to finish this stupid project for two weeks and I always fix bugs. Two days ago, it was some stupid bug in racoon and Linux kernel, that has been well-known for at least half a year. Up until now, the kernel developers and the racoon developers are telling, that the bug is on the other's side, so no one fixed it. I prototyped a working fix in twenty minutes. If they at least said that on a page or something....
09:02.10dudesthat's my point
09:02.47jurajah
09:02.47juraj:)
09:03.15jurajdudes: I'm quite familiar with the source code to some extend, at least with chan_sip.c. But for some internals I am not.
09:03.24dudesjuraj - as a dev my self ... sometimes things don't get done because other things on a list are more important.  But not always =)(
09:03.58dudesjuraj - so you have an issue with conference's getting disconnected
09:04.06jurajyep
09:04.09dudesafter XX amount of time
09:04.26dudesbecause your clients are using IPSec
09:04.33jurajyep. the network connection stays alive. I can ping the asterisk machine.
09:04.40jurajI don't know if it's because of that.
09:05.29dudesis there sometype of timeout on the IPsec DB
09:06.09jurajthe issue I reported in Mantis is, that if the client disconnects, chan_sip sends the data, because in ipsec setup, you always get temporary error if there's no security association. But this one is not related, I just reported it because in the morning, I found 400MB log of "Resource temporarily unavailables" :)
09:06.17dudesI suppose I should look over your bug report
09:06.39jurajdudes: it is indeed. several timeouts. There's an SA timeout and a rekeying timeout.
09:07.00jurajdudes: anyways, when new SA is registered, the old one can be used too
09:07.09dudesI've never seen a asterisk log file over 131MB's
09:07.10dudeshaha
09:07.15jurajdudes: so you don't get even one packet lost
09:07.42dudesand that was on a Dual P4 3.06Ghz /w HT
09:07.44jurajeach SA has expiration time and before it ends, new SA is negotiated and both are valid.
09:08.32dudesso is this bug report your's?
09:08.55jurajdudes: yeah, the bug report is that when you get "Resource temporarily unavailable"s for whole night, it's not _that_ temporary, and I wonder if someone would be interested in RTP packet if the whole night lasting temporary error would get resolved :)
09:09.03jurajyep, this one is mine :)
09:09.18jurajbut that does not solve my problem, even if it's resolved :))))
09:09.30dudesare you allowing the RTP ports from IPSec to IPSec
09:10.09dudesie. 10000-20000  to communicate between the two? (bare in mind I'm hammered.)
09:10.51jurajdudes: allowing everything
09:11.18jurajdudes: (if they use ipsec, but that's not based on port numbers, it's a security policy set using setkey to the kernel:)
09:11.20dudesNot to be rude ... but I heard that line before with IPsec techs and it was horseshit
09:11.23jurajAug 18 22:18:16 WARNING[887] app_meetme.c: Unable to write frame to channel: Success
09:11.42jurajwhat does this mean? :)
09:11.51*** join/#asterisk Rowters (n=SilverDr@dsl-201-129-88-148.prod-infinitum.com.mx)
09:12.18jurajdudes: yeah, maybe, but now it works for me
09:12.32dudesI spend a day with a client resovling an issue similiar to yours ... anyway it was because they're techs were idiots
09:12.51jurajdudes: but at least this one is open-source
09:12.56dudesNot saying you are
09:13.03jurajdudes: checkpoint is similiar piece of shit, but I can't even find/fix bugs in the source
09:13.06hellopAnyone ever change the idle picture on a Polycom IP phone?
09:13.07juraj:)))
09:13.30hellopI tried to do it using a gimp created "Windows BMP"  maybe I have to use an acutal windows one?
09:14.12*** join/#asterisk secure75 (n=mic@p549A3EF7.dip0.t-ipconnect.de)
09:14.35jurajdudes: I'm pretty sure the ipsec is quite okay now. I don't get a one lost ping, nothing.
09:14.46dudesif a packet timesout
09:14.58dudesit's clearly dropping somewhere
09:15.06jurajdudes: and even if I did, I believe several lost packets must not be problem for a voip connection
09:16.07jurajaaah, I hate this stupid ipsec setup :)
09:16.13dudesno shit
09:16.15dudeshaha
09:16.22dudesjust use iptables
09:16.51jurajfor what? :)
09:17.15jurajyou mean firewall instead of encryption and signing? :)
09:18.23dudesargue with me about firewalls and encrytion when I'm sober and can type
09:18.51dudesregardless ... why encrytpe conf data
09:18.53kyooI'm trying to understand digium hardware... I can today buy a TDM01B and later upgrade it to a TDM03B or a TDM22B myself by buying modules?
09:19.50tzafrir_laptophellop, why use the gimp for conversions when use can use imagemagick's convert?
09:19.56*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
09:20.45dudesbecause gimp is better
09:21.06*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
09:21.10MmmmToopgimp is great...
09:21.16MmmmToopI use it quite a bit
09:21.17dudesyes it is
09:21.32MmmmToopnot good with big images though...but otherwise great!
09:21.34MustangMatt*ducking in corner* I still can't get used to the interface.
09:21.47helloptzafrir_laptop,  dunno
09:21.51tzafrir_laptopgimp is great for composing images and such. for simple screenshots, conversions and such it is a slow overkill
09:22.28helloptzafrir_laptop, come to think of it, I did notice quite a few millisecond lag on that 112/52 pixel image...
09:22.46hellop<giggle>
09:23.06MmmmToopha ha ;  )
09:23.34tzafrir_laptophellop, the startup time. Unless you're an artist with gimpl constantly running
09:23.41hellopI do know however, that the Polycom 500 is well worth the money compared to a Budgetone.
09:24.17tzafrir_laptophellop, I suppose you're one of those calling them bugtones?
09:25.05hellopHowever, sadly, I can imagine that many customers would have a hard time with the small buttons and could not even read the "Dial" on the screen...
09:25.24hellopThese are the customers that I get to sell 21" monitors to.
09:25.36kyooI'm trying to understand digium hardware... I can today buy a TDM01B and later convert it to a TDM03B or a TDM22B myself by buying modules?
09:25.50syleyes kyoo
09:26.54dudeskyoo - you shutdown your computer ... put a new module in ... turn on your computer ... configure your zaptel.conf for the new modules ... rum ztcfg -vv (no errors, good.) ... run asterisk and away you go\
09:27.15kyoosyle: dudes:  Thank you.
09:27.40kyoothevoice.digium.com is a brilliant service. :)
09:28.30Lan16`spdy^gTpwhy odbc dont request my database however odbc is connected to :^o, someone can help me ?
09:28.43Lan16`spdy^gTpwhen i make odbc show
09:28.55Lan16`spdy^gTpname : connect1
09:29.00Lan16`spdy^gTpdsn : asterisk
09:29.04Lan16`spdy^gTpconnected : YES
09:29.11hellopThe budgetone is nice for old people.. big buttons and a nice big LCD.
09:29.35Lan16`spdy^gTpand when i want to make registration it says no registration for peer
09:30.29*** join/#asterisk Maksim (n=max@213.142.207.20)
09:30.47jurajkyoo: indeed, if it also worked for something different than english :)
09:31.14kyooWhat is a good "multi line" sip phone?  One you'd want to put on a receptionist's desk. (dependable, efficient, easy call control, etc...)
09:31.32*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
09:31.33drraycisco 7960
09:31.47X-RobSnom 220 or 3600 with extension pad?
09:31.48*** join/#asterisk ti4 (n=tia_@82.119.194.149)
09:32.35X-RobEasy call control would be the GXP-2000, and for multi-line you'd need better firmware (does not exist yet) and the extension keypad (again, does not exist yet)
09:32.54X-Robgxp has a big 'hold' button. Receptionists like that.
09:33.03X-Robthe snom's is little
09:33.15kyooBut it only supports a single person at once?
09:33.22drraywith the 7960 you can forward and hold and confrence, and it has the headset
09:33.36X-Robdrray - so does pretty much every phone now.
09:35.20kyoogxp-2000 has two switching rj45 ports?  So it acts as a switch/hub?
09:35.45X-Robyep
09:35.58X-Robagain, most phones have that these days.
09:36.08kyooConvenient, I suppose - I with it would mention if they are 10 or 100. :)
09:36.22X-RobGXP-2000's are 100, PA1688 based phones are 10. Snoms are 100
09:37.31kyooBut the gxp-2000 can only handle a singe call at once?  Or a I thinking about this wrong, and that all takes place at the PBX?
09:38.31kyooCan a person with a gxp-2000 take  call, see another call coming through, put a on hold, talk to b and pick a back up when he's done?
09:38.47Lan16`spdy^gTpthx for help
09:38.49Lan16`spdy^gTp...
09:41.25*** join/#asterisk zigman (n=zigman@irc.zigman.de)
09:45.56*** join/#asterisk nagl (n=nagl@137.208.4.164)
09:47.59kyooSIP phones with asterisk can be made "as useful" as other PBX phones these days?  IE, you can see what lines are in use and available, lines can be parked, held, passed among phones, possibly listened in to by a manager?
09:48.23ltersbasically yes
09:48.43kyoo"basically" ? :)
09:48.44MustangMattFor anyone interested my Asterisk/debian answering machine guide is maturing: http://kaatman.com/asteriskansweringmachine.html
09:48.44MustangMattI'd still love to hear any feedback whether or not it's a relatively easy read, etc.
09:48.47ltersseeing via the phone, maybe a prob
09:49.09hellopyeah, seeing which lines are in use on the fone would be nice...
09:49.16ltersbut u can use FOP to see every thing
09:49.18*** join/#asterisk Tili[Dinner] (i=Tili@61.144.21.227)
09:49.21kyoolters: Can you explain a little more, please?
09:49.24kyooFOP?
09:49.34PakiPenguinhey Tili[Dinner]  :p
09:49.35hellopkyoo, you can use a web page to see whats in use
09:50.10lterssip phones can park, hold, transfer, listen in, pickup and all that good stuff.
09:50.29ltersbut to *see* who is on the phone u need the web piece.
09:50.42kyoohellop: Ok...  Is that because of a missing chunk of technology in asterisk, or missing parts of the SIP spec?
09:51.15helloplters, what about writing a program for something like the Polycom 500 to see what lines are in use..
09:51.35lterspolycom already can see it via the hint system in *
09:51.38kyoohellop: Exactly what I was wondering... the phones have the capability?
09:51.54lterssome do, some don't
09:52.07lterscisco / sip do not have the firmware support, yet
09:52.20lterscisco / sccp does. and it works now.
09:52.25hellopkyoo, I also would like to know how to write a program for my phone, but yes, you can d/l new programs to it.
09:52.39kyooSo if i buy a polycom 500, then the "lights" will "work"?
09:52.56hellopkyoo no
09:53.05X-Robhellop - yes, they well.
09:53.12X-RobThey _do_ work. just not as well as you'd expect.
09:53.23X-RobThere's no support for ringing indication, or for call pickup currently
09:53.30lterssearch for hint and polycom on the wiki.
09:53.34hellophmm  I'll have to look into that.  what I search wiki for?
09:53.36X-RobSee mantis bug 3644 for the patch to support all that
09:53.52ltershow to set it up.
09:54.02X-Roblters - if you're using Asterisk@Home it'll just work.
09:54.20X-Robbut I suggest the Snom phones rather than the polycom phones.
09:54.21kyooX-Rob: But it's all *possible* and probably on it's way sometime?  (Faster if I choose to roll up my sleeves?)
09:54.26ltersX-Rob, does sip/hints work for making and recieving calls.
09:54.39X-Robkyoo/lters - as I said, bug 3644 _implements it now_
09:54.41X-Robit's just not in CVS
09:54.48X-Robit's still aimed to go into 1.2
09:55.05kyooX-Rob: Ah, "bug" made me think it was a planned feature, not an implemented one. :)
09:55.24X-Robit's a bug ("it doesn't work") with a patch ("here you go")
09:55.37kyooX-Rob: Do the snom phones "light up" properly with A@H?  Or only the polycom?  (going to search "hint" in wiki now...)
09:55.39ltersX-Rob, fancy. Using sccp with sccp phones, ie, one sccp monitoring another u even can get callerid...
09:56.01lterskyoo, warning. u may not like the snom phone itself.
09:56.10*** join/#asterisk meppl (n=mephisto@84.245.164.92)
09:56.17lterskyoo, I got one and it ended up on the shelf :(
09:56.23X-Robkyoo - yes, they do. There's an issue or two still to get teased out of it, but it does work. Unfortunately, every fourth or fifth transfer from a snom crashes the asterisk server, but I'm working on that 8)
09:56.35X-Roblters - Send it to me!
09:56.40*** join/#asterisk Tili[Dinner] (i=Tili@61.144.21.227)
09:56.52ltersX-Rob, yeah, I should. got the side car and all :)
09:56.52X-Robhttp://www.aussievoip.com.au/wiki-Rob+Thomas
09:56.54kyoolters: What was wrong with it?  Did it feel cheap?  Slow response?  Fragile connection?
09:57.13ltersbuttons did not feel nice.
09:57.24X-Roblters - I'll even pay for postage. Seriously 8)
09:57.35ltershandset was not at all like the ciscos. ( we were spoiled )
09:58.14*** join/#asterisk RoyK (n=roy@213.160.242.93)
09:58.26ltersbasically it just felt cheap. the lights were a pail orange instead of red or grenn.
09:58.39ltersu bairly can see them light up.
09:58.52RoyK~seen cypromis
09:58.53jbotcypromis is currently on #asterisk-doc (1d 3h 52m 37s) #asterisk (1d 3h 52m 37s).  Has said a total of 21 messages.  Is idling for 21h 4m 38s
09:58.55RoyK~seen the_light
09:58.55jbotRoyK: i haven't seen 'the_light'
09:59.05ltersI do like that snom is * friendly
09:59.06X-Rob~the band
09:59.14ltersRoyK, ki
09:59.22RoyKlters: gi
09:59.31ltersRoyK, got my first sangoma :)
10:00.27kyooNeat, a phone accessory with it's own web interface. :) (snom 360)
10:00.43jurajkyoo: i believe every phone has it's web interface
10:00.47jurajvoip phone
10:00.54kyooOh, I guess i'm just new. :)
10:01.02ltersand the later fw really made em better.
10:01.14ltersX-Rob, u can msg me.
10:02.33*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
10:02.41hellopMustangMatt,  your setup looks nice and simple.. good job.
10:03.16MustangMatthellop: Thank you. It's not quite finished. Still need more info in the actual recording messages section but I'm learning as I write it :)
10:03.36*** part/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121)
10:05.27helloplters, I'm not seeing this hint stuff yet..  I searched for "Polycom hint" and "hint polycom line in use"
10:05.51kyoohttp://www.voip-info.org/tiki-index.php?page=Asterisk+presence
10:06.49*** join/#asterisk nroej (n=joern@lak-39-56.wohnheime.ruhr-uni-bochum.de)
10:07.22kyooDoes "hands free" mean speakerphone? :)
10:07.33jurajaaah, my wiki page :)
10:08.05kyoojuraj: I noticed it was yours. :)
10:08.44helloploading.... Bueler...
10:08.52kyooThe 320 does not support the sidecar, right?
10:09.15kyooOr is the sidecar completely standalone?
10:09.25kyoo(snom 360 sidecar)
10:11.18nroejhi all
10:12.18kyooThe aastra 480i' slights are *not* supported by asterisk hinting, correct?
10:12.33hellopWell I checked out presence,  isn't that for telling if a certain phone is in use, not for telling if a certain external line is in use?
10:13.47*** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl)
10:15.00*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
10:15.54*** join/#asterisk Mc_Tr (n=Mc_Tr@bacterio.knet.es)
10:15.56kyooIncidentally, if a web page can tell me this information, then I could write an application that can also "get at" this info, correct?
10:16.06kyoo(without scraping the web site.)
10:16.17Mc_Trhi!
10:16.59Mc_Trhas anybody SPA-841 and asterisk@home working?
10:17.46Mc_Trmy problem is that Sipura SPA-841 don't register in asterisk.
10:18.08Mc_Trbut with same user/pass in X-TEN all works perfectly.
10:18.11hellopkyoo, yes
10:18.35jontowmc_tr; i've got no problems with my SPA-841 registering..
10:18.35Mc_Tri apreciate yours comments ;)
10:19.11Mc_Trjontow, what you configure?, User/pass and proxy sip ?
10:19.28jontowoutbound proxy as well..
10:19.33jontowand make sure you check 'use outbound proxy'
10:19.50Mc_Trand what outbound proxy use?
10:20.03jontowyour * box's host/ip
10:20.25Mc_Trok, go to try it.
10:21.07jontowif you can't get it..lemme know, i'll plug my phone in before i leave for work and i'll duplicate the settings for you :)
10:21.20Mimmusgood morning, does anyone know the exact meaning of overlapdial=yes in zapata.conf?
10:21.23Mc_Trok, thanks jontow
10:21.34MadkissGNAR.
10:21.47*** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121)
10:21.59MadkissMy asterisk still terminates SIP VoIP-sessions for no obvious reason
10:22.07*** join/#asterisk andrebarbosa (n=andrebar@adsl-f49-s197-critical-coi.nortenet.pt)
10:22.11Mc_Trjontow.... don't work :(
10:22.12PakiPenguinMadkiss, your * hates you
10:22.34jontowalright, i've gotta be to work in about 30mins
10:22.44Mc_Trok
10:22.44jontowso i'll plug it in before i walk out the door and get you the settings then.. that ok? :)
10:22.48Madkiss== Spawn extension (default, 00216xxxxxxxx, 2) exited non-zero on 'SIP/2000-2bc8'
10:23.06Mc_Tri'm still waiting for you ;)
10:23.13MadkissIs there some way to make it tell me more about that, *without* enabling SIP debugging, which makes debugging actually impossible because it prints all kinds of useless messages to the screen?
10:23.46tzafrir_laptopMadkiss, enable debugging and edit logger.conf to not send debugging to the console
10:24.13dudesMadkiss - as useless as it is ... it helps debug the problem
10:24.52dudesMadkiss - But I don't know your issue so I won't pretend
10:24.54tzafrir_laptopdudes, Madkiss , actually the trace from 'verbose' is normally quite useful. The one from 'debug' is often too verbose.
10:25.31dudestzafrir_laptop - I never said anything about enabling verbose in logger.conf
10:26.08tzafrir_laptopIIRC, the default logger.conf doesn't send 'debug' to the CLI, which is indeed sane.
10:26.22MadkissInterestingly enough, while calling my own phone and hanging up then, the 7960 will need ~60 seconds to notice that the call is finished
10:27.02tzafrir_laptopMadkiss, the 7960 or asterisk? what does 'show channels' show?
10:27.19andrebarbosaanyone with experience with TE410P installation?
10:27.20andrebarbosa:s
10:27.59Mc_Trjontow; I solved MY problem.....
10:28.07Mc_Trin extension net=yes
10:28.15Mc_Trsorry, nat=yes and work prefectly
10:28.17MadkissIn verbose, I see
10:28.22MadkissAttempting native bridge of SIP/2000-aebc and SIP/tonline-eade
10:28.25jontow:)))
10:28.30jontowcool
10:28.31Madkiss== Spawn extension (default, 00xxxxxxxxx, 2) exited non-zero on 'SIP/2000-aebc'
10:28.36dudesandrebarbosa - I've done installs with 4 te410P's in one box
10:28.36Madkissand that's it
10:29.05tzafrir_laptopMadkiss, set verbose 3
10:29.12tzafrir_laptopto see what's going on
10:29.26MadkissVerbosity was 38 and is now 3
10:29.45tzafrir_laptophmmm..... so this wasn't it, then
10:30.27andrebarbosadudes, i'm witha  strange problem for me.. i do: modprobe wct4xxp
10:30.30andrebarbosaand nothing happens
10:30.32Mc_Trjontow, i'm goging to test to call
10:30.40andrebarbosano messages on dmesg, no error messages
10:30.41andrebarbosa:s
10:30.45tzafrir_laptopandrebarbosa, aparantly the module was loaded
10:30.56tzafrir_laptopandrebarbosa, lsmod | grep zaptel
10:30.58andrebarbosathe card is plugin with knigthrider lights
10:31.05dudesandrebarbosa - do a lsmod
10:31.19andrebarbosazaptel                229316  3 wct4xxp,wcfxs,ztdummy
10:31.29andrebarbosaok no?
10:31.33dudesif see if are c it ok
10:31.39tzafrir_laptopalready loaded. rmmod ztdummy while you're at it
10:31.47frenzyIs this theory possible; Two Asterisks boxes, BoxA & BoxB all sip-to-sip (in-network calls) to take place on BoxB while all other incoming and outgoing to take place on BoxA ?
10:32.08tzafrir_laptopandrebarbosa, there's no point in the extra overhead of the interrupts of ztdummy
10:32.22*** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it)
10:32.34*** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2)
10:32.43andrebarbosaok i removed it, i have it cause the bri cards
10:33.29frenzyany thoughts ?
10:33.32Mc_Trjontow; are yor here?
10:34.07andrebarbosabut, so now i have the module loadule but no messages on the dmesg and knightrider lights?
10:34.18andrebarbosathis is normal?
10:34.21Mc_Tri can't to do calls, first "407 Proxy Authentication Required" and second "488 Not Acceptable here"
10:34.35Mc_Trbut my extension it's already registered
10:34.38tzafrir_laptopfrenzy, sure. But are there any other limitations you want ot tell us about? (budget limitations?)
10:35.15tzafrir_laptopandrebarbosa, modprobe will simply do nothing if the module has already been loaded.
10:35.30opus_when asterisk timesout on a registry, can I make it reregister again?
10:35.51opus_like today, the network went down. and not all lines immediately re-registered
10:36.19andrebarbosabut i read somewhere, that when the module is loaded the lights stop
10:40.25*** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl)
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10:44.10*** part/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg)
10:54.03*** join/#asterisk Tili (n=Tili@61.144.21.227)
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10:56.14*** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at)
10:56.40nicoxhello, do anybody know if there is avariable in the dialplan where overlap digits are saved?
10:58.29*** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121)
11:00.48*** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue)
11:00.50surfduehi
11:01.07surfduewhat extention is the defualt menu calling from an internal SIP phone?
11:01.09nicoxhi
11:02.16surfduedoes anyone know?
11:02.41nroejsurfdue: i dont understand your question ;-)
11:02.48Jzalaesurfdue: give it half an hour for someone who knows to check this window before you give up
11:03.13Jzalae(i've seen it take an hour and twenty)
11:03.33Jzalae... and that is normal for this type of channel
11:03.44kyooWhat is a consumer-level firewall that supports QOS?
11:03.54kyoo(as it applies to AIX and SIP)
11:04.59surfduewhat extention is the defualt menu calling from an internal SIP phone, we made a menu for asterisk voice menu
11:05.06surfduewhat is the defualt extention to reach it?
11:05.22surfdueif that helps clear it up some
11:05.30nroejthe exten you gave to it?!
11:06.04Inv_arpkyoo: consumer lever firewalls?... never heard of any watchdog isnt too expensive  or justs set up a linux box
11:06.07surfdueahh
11:06.43nicox<PROTECTED>
11:07.19X-Robnicox - define 'overlap digits' - DTMF signals sent/received after the call is established?
11:07.28kyooInv_arp: Would you put asterisk and your firewall on the sam box?  Otherwise the noise, hardware cost, maintenance cost, etc doesn't seem worth it... But together is more exploit-prone...
11:07.30*** join/#asterisk Praktikant01 (n=lars@dsl-084-059-141-043.arcor-ip.net)
11:07.50Praktikant01good day
11:07.54X-Robbad night
11:08.27X-Robheh
11:09.22*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
11:09.48X-Robnicox - talk in channel
11:09.51X-Robnot privately
11:10.10Inv_arpkyoo: actually my firewall (iptables) and asterisk is on the same box... for my office (5 sip phones) (2 pots)
11:10.15nicoxno, digits recieved before the call established, my problem is, i have the extension 012798357 and i will forward it to zap/g1/016621735202 but the switch of the telco gets from my asterisk 016621735202012798357 why he is doing this?
11:10.41X-RobBecause that's what your dialplan is telling it to do.
11:10.52*** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121)
11:10.57X-Robwhy don't you paste the relevant bits of your dialplan to pastebin?
11:11.07surfdueis there a way to login to amp?
11:11.08nicoxno, i tell him to dial only the 016621735202
11:11.10X-Rob~pastebin
11:11.10jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://bzflag.pastebin.ca/
11:11.13surfduelike me im not on the server
11:11.17surfduecan i?
11:11.43X-Robnicox - I bet you're not. Paste your extensions.conf to pastebin
11:12.01surfduewhat port does it use?
11:12.12X-Robsurfdue - 80.
11:12.21surfdueport 80 is web?
11:12.24X-Robyes
11:12.26X-Robamp is web
11:12.34X-Robtherefore, amp = 80.
11:12.38PoWeRKiLLhi
11:13.14PoWeRKiLLsomeone know how to chaange sipura dial plan to allow *21*xxxxx# for call transfer ?
11:13.19surfdueoh i get it
11:13.20surfdue:P
11:13.24surfduekthx
11:13.37tzafrir_laptopjbot, pastebin is s/bzflag.//
11:13.37jbot...but pastebin is already something else...
11:15.20nicoxhttp://pastebin.com/340642
11:15.55tzafrir_laptopjbot, pastebin =~ s/bzflag.//
11:15.55jbottzafrir_laptop: OK
11:16.11tzafrir_laptop~patebin
11:16.11jboti heard patebin is your friend
11:16.31tzafrir_laptop~pb
11:16.31jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
11:19.11syleget a real job jbot!
11:19.12X-RobHrm. OK. That's wierd.
11:19.29X-Robexten => 0127983,5,dial(zap/g1/1099016621735202)
11:19.37X-Rob<nicox> no, i tell him to dial only the 016621735202
11:19.41X-RobSo who's retarded here?
11:20.18sylepeople who can;t code?
11:20.20X-RobAh, no, you did mention the 10990 bit earlier
11:20.33X-Robsyle - No, my accountant wife isn't retarded, and she can't code.
11:20.41X-Robshe can count beans better than anyone I know
11:21.08tuxinator_linuxMbean counting, sounds fun ;-)
11:21.24syleshes a woman she has an excuse lol
11:22.02syleif they spread their legs and cook you food what more do you need
11:22.10tuxinator_linuxMSun is going to be up soon, maybe I should get some sleep?
11:22.14rikstabean flicking more fun
11:22.25tuxinator_linuxMsyle: Not married, are you?
11:22.44nicoxyes, but the 1099 i need for the telco
11:22.45syleactually i am lol
11:22.52X-RobI cannot believe I don't have a functional DVD rom here.
11:23.45nicoxbut the 1099 is okay, the problem is the telco get 0166217352020127983
11:23.51tuxinator_linuxMMost of my stuff is packed, the wife and I are moving to California next week. (I was looking for me external DVD burner)
11:24.17syleget a dvd 9
11:25.17tuxinator_linuxMsyle: It's a http://www.plextor.com/english/products/716uf.htm
11:26.22syleidk , depends what your budget is , personally i only go with sony for dvdrom,cdrom parts
11:26.23X-Robnicox - ok, that's just wierd. How are you connecting, and how do you know the telco's getting it?
11:26.46tuxinator_linuxMsyle: never had a sony, good stuff?
11:27.16X-RobGah
11:27.19syleyeah man, sony makes their shit good, i;ve had alot of my sony stuff for 10 years and still going
11:27.20nicoxi see it in the CDR's and i'm connected with 2 E1 one incomming one outgoing
11:27.21X-RobI just looked _on top_ of my PC
11:27.25X-Robwhat's there? A DVD Rom.
11:27.26X-Rob*sigh*
11:28.08sylebut only for dvd, cd , vcr's and dvd players
11:28.30tuxinator_linuxMX-Rob: You need a cookie!
11:28.44tuxinator_linuxMX-Rob: or I need a cookie
11:28.49X-Robnicox - OK. Paste you /etc/zaptel.conf and your /etc/asterisk/zapata.conf to pastebin
11:28.50X-Rob~pb
11:28.50jbotwell, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
11:28.51X-Rob8)
11:29.02sylei think the dvd9;s are still expensive
11:29.04X-Robtuxinator_linuxM - don't you need to be in bed or something? 8)
11:29.14syleof course prob only sony making it again
11:29.19tuxinator_linuxMX-Rob: Aye, that I do
11:29.35tuxinator_linuxMX-Rob: Wife is sleeping next to me
11:29.51X-RobOoh Ooh
11:30.00X-RobYou know what's fun with _pregnant_ sleeping wifes?
11:30.08X-Robwake up blobby by poking it a couple of times
11:30.09tuxinator_linuxMX-Rob: Cow tipping
11:30.14X-Robit'll wriggle around and wake her up
11:30.21X-RobChicks dig it. Really they do.
11:32.08nicoxhttp://pastebin.com/340651
11:32.51*** join/#asterisk nagl (n=nagl@137.208.4.169)
11:33.09X-Robnicox - the 'channel =>' is the 'end' of the section.
11:33.20X-Robso you've got 'overlapdial=yes' only applied to group=3
11:33.24X-Roboh
11:33.25X-Robdoh
11:33.33X-Robsorry, didn't see the line above group=1
11:34.11ful|workdoes asterisk and openssi works nicely?
11:34.13nicoxits general, overlapdial=yes is for all channels
11:34.18X-Robif I was you, I'd be messing with the 'pridialplan' and 'switchtype' options myself.
11:34.27X-Robnote, you can shrink that down to:
11:34.31X-Robgroup=1
11:34.40X-Robsignalling=pri_net
11:34.47X-Robchannel =>32-46,48-62
11:34.55X-Robgroup=3
11:34.59X-Robsignalling (etc etc)
11:35.03nicoxhttp://pastebin.com/340652
11:35.08*** join/#asterisk folsson (n=filip@h82n1fls32o985.telia.com)
11:35.10X-Robbecuase everything inherits from above
11:35.54X-Robswitchtype=euroisdn would be a good start
11:36.00*** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net)
11:36.07nicoxthe configuration i think is okay, so pri_net is working fine and also overlapdial is working, but my problem....
11:36.11X-Rob(being that you're on european isdn)
11:36.19X-Robno, the configuration is wrong.
11:36.26nicoxno, with switchtype=euroisdn the connection is not working.
11:36.44nicoxbelieve, i have traffic on the lines....
11:36.50nicoxso it must be working
11:37.11nicoxand with switchtype=euroisdn i get no link to the telco
11:37.12X-Robthe protocol isn't right
11:37.34X-RobI'd ask on asterisk-users@lists.digium.com
11:37.58nicoxokay thanks, i will do so
11:38.03X-Robexplain your problem, say you're from germany, mention your ISDN provider, explain what's going on, say 'help'
11:38.19X-Robpost links to the pastebin.com dumps of your config files
11:38.29nicoxokay, i will do so thanks
11:39.22tuxinator_linuxMI woke up my wife when I got into the cookies.  It's all X-Rob's fault. He made me want a cookie.
11:39.47*** join/#asterisk r0m (n=SysOp@bl8-28-55.dsl.telepac.pt)
11:40.45X-RobEek!
11:41.13X-RobTell her the strange australian geek is sorry.
11:42.40tuxinator_linuxMShe is a very forgiving wife.
11:43.09*** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin)
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11:43.34tuxinator_linuxMWe are early twenty's, we can take the lack of sleep ;-)
11:43.55nicoxI have another POroblem, i will compile asterisk-CVS-HEAD but after 20 seconds i only get this message 1000 time or so on http://pastebin.com/340654
11:44.14*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
11:44.49*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
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11:45.55*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
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11:47.13*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
11:48.47X-Robnicox - Don't know about that one. Tried 'rm -rf /usr/src/asterisk' and started afresh?
11:50.53nicoxyes i tried but also not working, the myth is that on this machine i compiled it 3 weeks ago and this version is running...
11:51.46tuxinator_linuxMmake clean?
11:52.20nicoxi made it, but if the problem occures also the make clean has the same problem
11:53.14*** part/#asterisk secure75 (n=mic@p549A3EF7.dip0.t-ipconnect.de)
11:54.14*** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com)
11:55.04faa_Where asterisk is looking for pgsql libs when compiling?
11:55.19faa_If id add cdr_pgsql.so to MODS
11:56.10*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
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11:59.08tuxinator_linuxMi.think.napoleon.dynamiteblows.com .... interesting
11:59.49tuxinator_linuxMNight guys
12:07.31nicoxI have ar Problem, i will compile asterisk-CVS-HEAD but after 20 seconds i only get this message 1000 time or so on http://pastebin.com/340654
12:09.32*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:09.45kyooIs the flash call manager the only one out there?
12:10.12drraykyoo - gastman
12:11.02kyoodrray: Well, that doesn't really appear to be an improvement. :)
12:11.07drrayyes
12:12.16kyooIs OSX popular with the asterisk community?  Would an Aqua tool be appreciated?
12:13.19drrayI'm sure it would be appreciated
12:15.26macTijnkyoo: that would be appreciated. What kind of tool did you have in mind ?
12:16.20kyoomacTijn: I'm not sure, I'm noodling design right now.  Something with a heavy emphasis on speed of operation and usability.
12:16.23*** join/#asterisk cliffton_2005 (i=cliffton@adsl-69-105-116-203.dsl.pltn13.pacbell.net)
12:16.51drrayI've been kinda (ie, not at all) working on a gkrellm plugin for asterisk
12:18.00*** join/#asterisk klapzin (n=klap@201-0-66-216.dsl.telesp.net.br)
12:18.54klapzinwhere i can view a modem list ( generic ) whith asterisk suport?
12:19.23*** join/#asterisk ttyp0 (n=ttyp0@83.Red-83-55-35.pooles.rima-tde.net)
12:21.52drray~wiki
12:22.00*** join/#asterisk trasschaert (n=karl@212.68.197.226.brutele.be)
12:22.11trasschaertHello everybody
12:22.51*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
12:23.24*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
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12:26.12trasschaertDid someone can help me?
12:26.24nicoxask and we will se
12:26.27nicoxsee
12:26.41trasschaertI just install a digium X100P card into my asterisk, i can make outbound call but i cannot receive call
12:26.49trasschaertI use asterisk@home
12:26.54opus_read the manual on zapta.conf
12:27.12trasschaertif i call 7777, th ephone ring
12:28.55cliffton_2005Hello Trasshaert, how much is digium X100P card cost?
12:29.02*** join/#asterisk zoo (i=nobody@ip-54-16.travedsl.de)
12:29.05*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
12:29.18nicoxwhat happened if the call knocks on the asterisk?
12:29.47trasschaertI've buy a OEM card for only 6,5$
12:29.52nicoxdid anybody installed the current asterisk-cvs-head version?
12:30.19*** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com)
12:30.21trasschaertit only ring on the calling phone, nothing append on the sip phone
12:30.30klapzinhttp://www.laptops4me.com/product_info.php/modem/all-56k-modems/p/v-92-pci-intel-w-voice-s-w-modem/cPath/176_239/products_id/1444
12:32.35lathos42Good Morning/Afternoon/Evening/Night everyone
12:32.36nicoxhm, behind a nat?
12:32.55*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
12:33.03trasschaertno
12:33.34trasschaerti can make call from sip to pstn but not from pstn to sip...
12:34.07trasschaertIn amp, i configure incoming call to ring to a extension
12:34.10nicoxdo you see the call in asterisk console?
12:34.23trasschaerthow can i check it?
12:34.32trasschaertwhere can i see it
12:34.40nicoxlogin via ssh
12:34.45nicoxtype asterisk -r
12:34.52nicoxand see what happens
12:35.37*** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net)
12:35.38trasschaerti'v ethe asterisk cli, when a make a call nothing appen
12:37.06spdyhi everybody
12:37.11nicoxhi
12:37.17nicoxtype set verbose 9
12:37.21nicoxand set debug 9
12:37.37spdyi have a core dump when i try au authenticate a user using mysql
12:37.37nicoxif there also nothing happened, the call does not come to the asterisk
12:38.10trasschaertnothing append...
12:38.17spdysomeone now this bug ?
12:38.26trasschaerti already put a analog phone on the line and it ring
12:38.34spdycore dump with mysql iax authentification
12:38.39trasschaertso the line is ok, i already exchange the card
12:39.16trasschaertsomething special to configure in the zapata.conf?
12:39.27nicoxpaste it and let us see
12:39.47kyooHow often do the digium cards fail?  (Is replacement warranty worth it?)
12:40.07jontowkyoo; i've yet to have one fail, but i've heard of others having it happen
12:40.20jontowwe always keep 1 spare of everything we buy thats in a critical box
12:40.29jontowie. for every 1 T100P in service, we have a spare sitting in hotswap box
12:40.37spdyi have a core dump on asterisk when iax try to authenticate a user using realtime mysql
12:40.38*** join/#asterisk docelm0 (n=docelm0@67.106.194.90.ptr.us.xo.net)
12:40.42spdyno one knows ?
12:40.52jontowconfigs and data are tar zcvf'd nightly and then scp'd across a link to the other, where they're untarred
12:40.57kyooAnd how necessary is technical support, considering I already have an asterisk server up and working via SIP?
12:41.03jontowso all we have to do is start asterisk and swap cables
12:41.11kyooIs it more complicated than I expect?
12:41.13jontowits low-cost redundancy
12:41.22opus_spdy - me too
12:41.24jontowyou get free installation support anyway.
12:41.35opus_spdy - i saw that.. its weird
12:41.45jontowif you need help after the install and this channel can't provide it.. good luck, at that point you're gonna pay for support and it'll be worth it if you're on a deadline :)
12:42.00spdyopus - arf
12:42.03jontowthe digium folks know their shit and will get you what you need :)
12:42.05tzafrir_laptopjontow, sounds lame;-) tar czf - | ssh other server 'cat >archive.tgz'
12:42.08spdyopus - you have the last tarball ?
12:42.16opus_yeah
12:42.19opus_cvs from yesterday
12:42.33opus_spdy - i don't even use IAX but have iaxpeer/iaxusers pointed to my sip table
12:42.43BeirdoFUCK
12:42.46opus_i mean my 'users' table
12:42.50tzafrir_laptop(and my last ssh command was actually highly lame)
12:42.55jontowtzafrir; very.. :)
12:43.02jontowi gave the idea of it.. its actually a pipeline
12:43.03Beirdosomething seemingly toasted my 10/100 switch
12:43.06opus_spdy - the backtrace saids it something with strlen
12:43.13Beirdonow my asterisk box can't reach the world
12:43.15spdyopus - ok
12:43.18Beirdo:(
12:43.29jontowtar zcvf - | ssh otherserver 'tar zxvf -' <-- thats more of the idea ;)
12:43.40spdyopus - i know that sip has got the same probleme one time and they fix it
12:43.40opus_spdy - nobody seems to want to work on it. are you a programmer? i'd like to work on it
12:44.03kyooHow do the polycom 300 and 301 differ?
12:44.22BeirdoI guess I'll be buying a new one
12:44.24BeirdoPOS
12:45.25spdyopus - its a problem in the chan_iax2 ?
12:45.32opus_no, realtime
12:46.08spdyok
12:46.10spdy:s
12:46.37trasschaertnicox: sorry, i'm not a linux user, how can i copy all the file?
12:46.44trasschaertwhats the vi command?
12:47.10nicoxsorry, i don't know asterisk@home so i can't answer
12:47.11opus_trass you should really learn unix and come back :(
12:47.40opus_some ppl might get offended by the questions. 'ask a smart question and get a smart reply' is the general guideline
12:47.58opus_spdy - do you use realtime mysql or realtime odbc?
12:48.05nicox<PROTECTED>
12:48.07spdyopus - realtime mysql
12:48.12nicoxcan anybody help me?
12:48.21opus_nicox - yeah, like an infinite loop?
12:48.21spdyopus - with the last tarball version on the digium site
12:48.32opus_spdy - yeah. did you ever try odbc?
12:48.53spdyopus - odbc is no longer supported in the last tarball
12:48.57spdyopus - :(
12:49.01trasschaert; Zapata telephony interface
12:49.02trasschaert;
12:49.02trasschaert; Configuration file
12:49.02trasschaert[trunkgroups]
12:49.02trasschaert[channels]
12:49.04nicoxand why?
12:49.11*** join/#asterisk Katty (n=ladykatr@68.112.15.110)
12:49.23nicoxnot paste here!
12:49.26opus_nicox - make clean, get the latest source again, make libpri and zapata again.
12:49.38opus_that seems to fix it but i couldn't figure out why
12:49.38nicoxi did so
12:49.48nicoxlibpri and zaptel is okay
12:50.12opus_you need to rebuild it. my only guess is that there is a makefile command in /asterisk that checks the date on include files from ../libpri and ../zaptel
12:50.16nicoxbut if i try to compile asterisk i have this problem
12:50.21*** join/#asterisk nagl (n=nagl@137.208.4.182)
12:50.35*** join/#asterisk PakiPenguin_ (i=uppal@unaffiliated/pakipenguin)
12:50.35nicoxi recompiled it!
12:50.53opus_spdy - it sucks. odbc isn't supported. i really would like to use it. i hate mysql.
12:50.58nicoxbut i will try to delete all  download from cvs and recompile again
12:51.01opus_spdy - what do you use for voicemail users?
12:51.11*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
12:51.34Kattymorning.
12:51.38spdyopus - now i use static but in a few times i will use mysql
12:52.02spdyopus - yes odbc seem easy to use its idiot :(
12:52.08opus_spdy - i made the mistake of using mysql for voicemail users.. arge.. i had to patch the code to change the voicemail password
12:52.31spdyopus - :$
12:52.39spdyopus - not really cool :s
12:53.02opus_i really really dislike mysql. it corrupts my data
12:53.09spdyopus - i want them to fix the ****** iax core dump
12:53.21spdyopus - i use pgsql sometimes
12:53.40*** join/#asterisk morfe (n=morfe@tor/session/x-e805691422953e2d)
12:53.42opus_do you have valgrind installed?
12:53.53spdytar xvfz aster.....
12:53.54spdymake
12:53.57spdymake valgrind
12:53.59spdymake install
12:54.00spdy:)
12:54.02opus_spdy can you load up asterisk with gdb, load the core file, and type 'bt' cp to pastebin i want to see it
12:54.37opus_http://pastebin.ca/20364
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12:54.54spdyno pb
12:54.58spdyi do it now
12:54.59spdywait
12:55.14nicoxcompile is running, so i hope now its working
12:55.37opus_nicox you are the only other person i know thats had this problem.
12:55.46opus_nicox from a brand new install on a box I get that problem.
12:56.21morfehello. someone can say me where I can buy a hfc-s card?
12:56.43spdyopus - tu es francais :p ?
12:56.47nicoxthe problem is there again
12:57.02*** join/#asterisk uma62 (n=sudhir@pool-71-114-89-44.washdc.dsl-w.verizon.net)
12:57.07opus_<--- only knows english and C
12:57.08opus_:)
12:57.15nicoxbut, the miracle is that 3 weeks ago i compiled it without this problems
12:57.18spdyopus - ok
12:57.21morfespdy> moi oui..
12:57.23*** join/#asterisk tuyan (n=tuyan@81.214.255.57)
12:57.26spdyhello morfe
12:57.27spdy:)
12:57.33spdyca fait du bien des frenchies ici :D
12:57.46morfespdy> :p
12:57.57opus_nicox fuck. try this. cd /usr/src ; find ./ -exec touch {} \;
12:59.20spdy:)
12:59.36*** join/#asterisk dsfr (n=dsfr@digium.com)
13:00.27spdyopus - gdb display no stack when i load asterisk
13:00.53opus_not even 'bt'?
13:02.07spdyin shell : gdb asterisk
13:02.11*** part/#asterisk B4 (n=B4@202.69.48.245)
13:02.11uma62I am using a 3 Polycom phones. From time to time, the calls will go to voicemail straight away. I can always make a call from the phones though. Any solution for that
13:02.14spdy(gdb) bt
13:02.19*** part/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121)
13:02.19spdyNo Stack
13:02.22spdy:s
13:02.30opus_oh
13:02.36opus_set your ulmit -c unlmited
13:02.43zedkatufI have set up a VOIP to PSTN gateway, and I can ring my landline number from my computer, but I can hardly hear myself on my landline..I'm wondering if thr rtp stream might not be getting through.....is there any way to diagnose this at all?
13:02.44opus_then, gdb> load core.90210
13:02.56opus_well, create the core first
13:03.53*** join/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com)
13:03.56spdyok
13:04.09funxionzedkatuf are you using a digium card for pstn gateway?
13:04.24DaPrivateergot disconnected dunno if that went through. if it did im sorry for repeating myself:
13:04.25DaPrivateerquestion. I am using a Digium X100P; when transferring a call I need to send a hook flash (doing it through an AGI script). ive tried a few different things that did not work. anyone have any ideas how I can do this?
13:05.37zedkatuffunxion: No, I'm using a softphone atm, so in other words I'm doing VOIP ---> PSTN
13:05.52*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
13:06.10kyooIs it not possible to purchase an unlocked RT31P2 ?
13:07.13*** join/#asterisk trasschaert (n=karl@212.68.197.226.brutele.be)
13:07.33*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
13:09.38kyooIs anyone a service provider that will sell me one with 1 day's service?
13:09.40jontowhmm.. is that the vonage box?
13:09.44*** join/#asterisk jiro5281 (n=jiro5281@203.131.137.76)
13:10.01kyoojontow: It's a linksys router firewall with QOS and two FXS ports.
13:10.09jontowhmm
13:10.12Kattymrow?
13:10.17kyooAnd I can't understand why the general public can't buy them...
13:10.27jontowhttp://support.sipphone.com/index.php?_a=knowledgebase&_j=rate&_i=61&type=yes
13:10.34jontowif thats the box..
13:10.36jontowthen thats your answer
13:10.46jontowi suspect it is not, however.
13:11.14kyooI don't think so - no clue, actually...
13:11.49kyooI'm on hold with viopsupply - they are checking if I provide voip service, am I voip service provider...
13:12.02*** join/#asterisk epablo (n=epablo@200.75.139.188)
13:12.10epabloHello People!
13:12.15opus_<PROTECTED>
13:12.19opus_http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging
13:13.10kyooNo, but I can buy the exact same product from Draytec for two hundred dollars more....
13:13.50kyooWTF is up with the linksys product? I don't understand why I can't buy one - why would linksys care who I am?  Do they get kickbacks from vonage or something?
13:14.46opus_yes from the VOIP mafia
13:15.49*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
13:16.33kyooWTH, same problem everywhere I go - why is this device not for sale to the public?
13:16.42fenlanderkyoo: the general public can buy them in the uk...
13:16.49epabloI've just pluggin my first E1 on a TE110P.  I can get calls to work both ways with iBLC.. but if a use g729.. the is a translation problem
13:17.00kyoofenlander: I wonder how much the shipping would be.
13:17.01*** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com)
13:17.11*** part/#asterisk Maksim (n=max@213.142.207.20)
13:17.22olivier_<epablo> do you have g729 licences ?
13:17.34*** join/#asterisk eminence_ (n=achin@cpe-24-198-66-186.maine.res.rr.com)
13:17.41jurajquestion about connecting FXO a FXS ports.... if i want to replace a pbx, I can get a digium card with T1/E1 interface. But it actually has only two plugs. What do I need to actually have 50 plugs for ports? :)
13:17.54jurajI'm completely new to telephony, I was just doing voip until now :)
13:18.14jontowjuraj; they call 'em a channel bank
13:18.24jontow:)  there is much info on the wiki about them
13:18.26*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
13:18.27epabloolivier_: not at this time.  I thought that the passthough would work
13:18.31DrWho17is there a mysql only version of the odbc voicemail storage?
13:18.36jurajjontow: thanks, I'll look at the wiki
13:18.39jontowits pretty much ethernet<-->RJ11*many
13:18.42fenlanderkyoo: no idea, but you are probably better off with an spa 2002
13:19.01kyoofenlander: Does the spa2002 do firewall, NAT and QOS?
13:19.33jontowno
13:19.37jontowits an ATA
13:19.43RoyKkyoo: not only that, it makes coffee as well
13:19.52RoyKjontow: an ATA can do that as well
13:19.58jontowyeah but not the SPA2002
13:20.00RoyKjontow: perhaps apart from the coffee
13:20.10jontowif it can.. im interested
13:20.11jontowi've got one :)
13:21.05kyoothe 2002 is basically a 2000 with a 10BT hub built in, right?
13:21.05fenlander:)
13:21.42*** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com)
13:22.04RoyKkyoo: no, it's more like a 2001
13:22.12RoyK2000 doesn't have QoS, no T.38 etc
13:22.53morfesomeone can say me where I can buy a hfc-s card?
13:23.08RoyKanywhere
13:23.20kyooSo, is anyonein here a service provider that can sell me a RTP31P2-NA?
13:23.51*** join/#asterisk BoDePlOt (i=user@pool-68-162-3-185.nwrk.east.verizon.net)
13:23.59opus_spdy yuck. realtime_peer (peername=0x81134c0 "4\021\b4\021\b") at chan_iax2.c:2593
13:24.55spdyopus - so whats the problem ?
13:25.15opus_is your peername in like unicode or something?
13:25.21opus_thats fucked
13:25.29opus_i think your problem is different then mine
13:25.54spdyby peername you think var name ?
13:26.02opus_The value yes
13:26.16spdy601
13:26.20spdyits the number
13:26.21kyooDid vonage underwrite the cost of development on this device?
13:26.39spdyand in realtime mysql name = number
13:27.01opus_dude something completely fucked realtime
13:27.14*** part/#asterisk morfe (n=morfe@tor/session/x-e805691422953e2d)
13:27.16spdyi think so
13:27.35opus_i'm not at my dev site, but, what if you check out an older version of asterisk / asterisk_addons and rebuild?
13:27.55spdyits what we are going to do
13:28.02opus_cvs co -D "Aug 1 2005" asterisk asterisk-addons
13:28.22spdyok
13:28.28spdyi take it
13:28.30spdyand test
13:29.19*** part/#asterisk juraj (i=juraj@smtp.rebelium.net)
13:29.42opus_does asterisk-cvs include changes to asterisk-addons?
13:30.14*** join/#asterisk tuyan (n=tuyan@81.214.255.57)
13:32.16nick125_lappyanyone got any links to a cheap ATA device that works with asterisk?
13:32.24MmmmToopexit
13:32.28jontowvoipsupply.com has a Sipura SPA-2002
13:32.35jontowvery nice device, i think i paid $69.95 for it
13:32.40jontow2 FXS ports
13:32.57nick125_lappyhow hard to config?
13:33.06jontowits got a myriad of web-config options
13:33.24jontowreally, you only need to touch the "User 1" and "User 2" pages, changing.. iirc, 4-5 things per page
13:33.28nick125_lappy202 has two ports?
13:33.35nick125_lappy2002 has two ports?
13:33.40jontow(ip address twice, user, pass, checkbox to say use outbound proxy)
13:33.42jontowyes
13:34.44nick125_lappyis there a difference other thenthe one port between the 1001 and the 2002
13:36.18*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
13:36.19nick125_lappyis the iaxy better for asterisk?
13:37.07jontowthe iaxy supports digium
13:37.12*** join/#asterisk teddey (i=electro@never.needs.a.reasontodrink.com)
13:37.15jontowits a little more pricy, a little smaller, and much cooler looking
13:37.21drrayand it works through nat
13:37.22jontowit does work a bit better being based on IAX2 but it has limitations.
13:37.30*** join/#asterisk Chotaire (i=chotaire@chotaire.net)
13:37.45drraythe Iaxy does not have a web configure util
13:37.59teddeyhey does anybody know that the theoretical max users/conferences are with meetme?
13:38.01jontowie. only supports a small subset of codecs, you provision it from the asterisk machine with a command line tool and a text config file (but its easy.)
13:38.13jontowand it doesn't do DNS.. so you have to input the IP address of your * server, not the hostname
13:38.26drrayyeah, if you can make head from cvs you can provision an Iaxy
13:38.30nick125_lappythats usually no problem, because i have to enter the ip anyways
13:38.30jontowagreed..
13:38.31jontow:)
13:38.39jontowit does ulaw and adpcm, iirc
13:38.46jontowso if you have low-bandwidth links, it may not be for you.
13:38.55drrayan Iaxy is a perfect device for use with wrtg54
13:38.59jontowagreed
13:39.09*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
13:39.11drrayyou can provision it from the wrt I believe
13:39.18nick125_lappyit wont be used over the internet, just wireless
13:39.22jontowlets just say.. you can tape the thing to a doorjam its so small
13:39.40drrayit does get warm, and the blue light is pretty damn bright in my darkened living room
13:39.54drraybut it sounds great
13:40.11*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
13:41.00teddeyso far with a quad t1 card and a single xenon 3 ghz 2G of ram i can get 96 people on and in conferences the quality is fine
13:41.01DaPrivateerI am using a Digium X100P; when transferring a call I need to send a hook flash (doing it through an AGI script). ive tried a few different things that did not work. anyone have any ideas how I can do this? please?
13:41.03drrayif I was going to hand an idiot an ATA and tell him to go home and hook it up by himself, an Iaxy may not be right
13:41.17*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
13:42.07drrayI've also not been able to get a fax to go over teh Iaxy
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13:43.16nicox<PROTECTED>
13:44.13*** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net)
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13:47.27*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:48.21Ariel_hello everyone
13:49.01teddeyhey does anybody know that the theoretical max users/conferences with meetme?
13:49.08*** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl)
13:49.26Hmmhesaysoh about 5
13:49.44Ariel_LOL 5...
13:51.09bjohnsonteddey: I think it all depends on hardware power
13:51.15Ariel_morning Katty
13:51.21nicox<PROTECTED>
13:51.41teddeyheh ive already had 96 so i dont believe its 5
13:51.47Ariel_the meetme depends on hardware, codec and much more. So there is no real number
13:52.00teddeythere is no hard limit is what your saying?
13:52.10drraynot built in to the software
13:52.14*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
13:52.20Ariel_teddey, as far as I know no.
13:52.44frenzyAug 19 13:51:29 WARNING[24526]: chan_sip.c:4832 check_auth: Stale nonce received from '<sip:7777777@xx.xx.xx.xx>'
13:53.14teddeywell then you can put it in the books that a xenon 3G with 2M of cache and 2G of ram will do 96 users
13:53.16frenzywhat does that mean ? my UA connects after a while I get that
13:53.25teddeywhen the ds3000p is released ill let you know what the max is
13:53.55Ariel_teddey, it really depends on the codec as well which one were you using for all the channels
13:54.01*** join/#asterisk Hmm-work (i=negative@66.173.103.108)
13:54.13Ariel_frenzy, check your qualify=60 or less
13:54.57teddeyG.711 ulaw all of my calls are from the PSTN
13:55.15*** join/#asterisk likwid-- (n=likwid@nc-69-68-67-119.dyn.sprint-hsd.net)
13:55.31Ariel_teddey, that is why no transcoding.
13:57.26kyooDoes anyone know how to crack one of these RT31P2 routers?  If I can't buy one unlocked, can I just unlock one that I've bought?
13:58.21opus_kyoo
13:58.22teddeywhy would i want to transcode?
13:58.28opus_you need to intercept the dhcp request
13:58.47opus_then change the traffic to download your own rom
13:58.49teddeythat just adds more load
13:58.52opus_EVERYTIME it reboots
13:59.27kyooopus_: Can't I just make a DNS change?
13:59.41Ariel_teddey, that is what I mean if you do transcoding it will be less connections.
13:59.48Ariel_kyoo, no
14:00.18kyooHrm.. So stupid - why are they sold like this?  Did vonage pay for their development?
14:00.27opus_yup
14:00.36Ariel_kyoo, they paid yes lots of money
14:00.44teddeyright but i have 96 people dialed in from the pstn transcoding would kill me
14:00.50kyooI hate corporations...
14:01.05opus_hey. open source hippy meets wall street
14:01.13Ariel_teddey, that was my point you asked how many can I get. I posted that it depends on transcoding.
14:01.30teddeyah sorry i didnt understand what you were saying
14:01.31kyooMore like Open Source Hippy tries to beat wall street...
14:02.01kyooIs there a similar device for a reasonable price? NAT FW, ATA, QOS router
14:02.02Kattyyay for hippys
14:02.10opus_smoke weed everyday
14:03.45Ariel_Big Biz is the reason we have high oil cost... The Future market is driving the oil cost higher then it should be. (There is no actuall shortage)...
14:04.37mutno
14:04.39file[laptop]blame Canada
14:04.42mutit's all the hippies fault
14:04.50file[laptop]it was in a song, so it must be true
14:05.10zedkatufnewb Question: If I set my softphone up to use a stun server, will that mean it will attempt to bypass my asterisk box?
14:05.45olivier_stun server just says your softphone your external I
14:05.49olivier_IP
14:05.52muti can't believe Zafi.B is still roaming around
14:06.03muten-mass
14:06.19zedkatufolivier: does that mean I'd still be going through my asterisk box, though?
14:06.32EquinoxAny issues with the grandstream HT-286?
14:06.52olivier_yes. your soft phone use your external IP in the packet it send asterisk instead of local ip
14:07.03Ariel_zedkatuf, yes it does unless you use canreinvite=yes
14:07.48zedkatufolivier_ - ok that's useful to know, tnx
14:08.24DaPrivateerlemme ask a different question. can anyone tell me how, in AGI, to dial on the zaptel channel that is in use?
14:08.53zedkatufAriel - tnx..I've been using AMP to set this up, and afaik it doesn't add canreinvite=yes by default (Icould be wrong abt this though!!)
14:10.47*** join/#asterisk secure75 (n=mic@ppp-82-135-1-232.mnet-online.de)
14:10.59Hmmhesaysno it doesn't
14:11.24moverany one here use cosini ss7 stack?
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14:21.07kyooDoe anyone have an actual phone number for verilan?
14:21.13kyoos/Doe/Does/
14:22.37ManxPower<PROTECTED>
14:22.37*** join/#asterisk Dublin_drupaller (n=dub@83-70-36-157.b-ras1.prp.dublin.eircom.net)
14:22.43ManxPowerthat's funny
14:22.52Dublin_drupallerhiya guys...
14:22.56|dennis|Question: We presently have a nortel pbx system that we rent fom a local phone company. We have three pots lines that we use to make calls with and receive calls on. I wish to move to an asterisk system. I intented to use IP Phones. So as far as i can tell all i need is an asterisk box, a 4port fxo for the three + maybe 1 lines and i am good to go. Am i correct?? Please Help...
14:23.21DrWho17sound fine
14:23.24Faithfulanyone know much about the debian version of Zaptel?
14:23.34Dublin_drupalleris this an okay place to ask a (probably stupid) question about asterisk?
14:23.44*** join/#asterisk santiago (n=santiago@63.245.87.180)
14:23.49NivexFaithful: I know a little from when I had to get ztdummy built.  What ya need?
14:24.12jontowdublin; we get many of those a day.. just give it a shot
14:24.28FaithfulOh I want to know if the debian version of 1.0.9 has the bristuff patched against it
14:24.33nick125_lappyDublin_drupaller, i do it all the time, so, you shouldnt have a problem :)
14:24.38|dennis|DrWho17..is there any particular 4port fxo card that you recommend?? I was looking at the Digium TDM04B
14:24.42NivexFaithful: ah, there I cannot help you.
14:24.43tzafrir_laptopFaithful, any problems?
14:25.15tzafrir_laptopFaithful, http://updates.xorcom.com/iso/ , there's an apt source attached at the end
14:25.17DrWho17dennis: nope, never used one, I've only used t1's
14:25.34Dublin_drupallerlol..thanks jontow..I'm using drupal (php based CMS) and I was thinking of bolting asterisk onto a site I'm working on...is that a ridiculous idea? Has anyone integrated asterisk into a website community?
14:25.37jontowdennis; that is the one we recommend.
14:25.38FaithfulWell I am trying to get zaphfc built against the debian version
14:25.46DrWho17a didium card is probably going to use the least tinkering to get working properly
14:25.54tzafrir_laptopit's only bristuff RC8h (actually equivalent to RC8j), but should be good enough
14:26.04Faithfulthey give you the module source no instructions and they don't compile
14:26.06nick125_lappyDublin_drupaller, you could use asterisk for conferencing, so, it could work like that
14:26.06*** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
14:26.09|dennis|jontow and DrWho17: thanks
14:26.10DaPrivateerManxPower still here?
14:26.16Dublin_drupallerthanks nick
14:26.29Dublin_drupallerany pointers for more info. on that front?
14:26.34Faithfulbut then... the asterisk source must be patched for the bristuff to work
14:26.53*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
14:27.11nick125_lappyDublin_drupaller, well, for actually setting up asterisk, voip-info.org is good
14:27.25DrWho17Dublin_drupaller: user roled interfaces for asterisk are pretty sparce
14:27.29Dublin_drupallerokay..thanks nick..will check it out....appeciate it.
14:27.37FaithfulWhy bother... ? because the debian implementtion of asterisk is nice and the packeges are pretty complete
14:27.47tzafrir_laptopFaithful, so I gave you a link to Sarge packages with bristuff patched in
14:28.00Nuggetwhat did debian change?
14:28.02tzafrir_laptopRebuild them
14:28.18*** part/#asterisk Dublin_drupaller (n=dub@83-70-36-157.b-ras1.prp.dublin.eircom.net)
14:28.30tzafrir_laptopNugget,  for once, buildinng debs out of zaptel
14:28.42|dennis|Aother Question: The T1..is an fxo card as well i assume and the only and big difference between the t1 and the regular fxo cards is the on board echo cancellation and hence the huge cost difference? Am I correct?
14:28.45tzafrir_laptopA number of fixes for other architectures
14:28.50Nugget*nod*
14:28.59tzafrir_laptopAnd some different file locations
14:29.28*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
14:30.08tzafrir_laptop|dennis|, T1 and FXO are totally different
14:30.09ManxPower|dennis|: you are wrong.
14:30.28ManxPowerand FXO card supports 1 channel.  A T-1 card supports 24 channels.
14:30.42DaPrivateerManxPower - i tried what you said. the line flashes but when it tries to dial i get a message that hte card is congested. any ideas?
14:30.45ManxPowerA T-1 card also support both FXO and FXS (since it's all digital)
14:30.46tzafrir_laptopManxPower, 24? 23?
14:31.01file[laptop]23 for a PRI
14:31.03|dennis|ah ok....thank you ppl
14:31.04ManxPowertzafrir_home: 24 channels.  On a PRI one of the channels is used for signaling.
14:31.10*** join/#asterisk slak- (i=slak@ircdz.com)
14:31.16slak-wa-zup
14:31.17DrWho17dennis: I carry 24 channels per t1, more efficient, you might even get your 4 lines carried in cheaper
14:31.21Faithfultzafrir_laptop: Thanks that's a winner
14:31.34slak-so yea we're thinking about junking asterisk here ;(
14:31.41slak-because of echo
14:31.52slak-they cant deal with the 5second echo period
14:31.54FaithfulI think I should scrub my box and start again now.
14:31.55DrWho17what does asterisk have to do with echo?
14:32.02slak-zaptel
14:32.51|dennis|talking about echo....if i get the four port fxo TDM04B, use IP phones ......will echo be a problem?
14:33.04kyooIf I purchase a g.729 license from digium - can a remote user with a SPA-2000 or a SNOM phone take advantage of it? (IE, Use it to connect to the asterisk box)
14:33.15kyooI don't understand where the codec actually *is*...
14:33.18DrWho17never noticed any echo issues myself, generally it's somewhere else besides asterisk anyway
14:33.23DaPrivateer9
14:33.27DaPrivateeroops
14:33.42ManxPower|dennis|: echo is ALWAYS a problem
14:33.54drrayeither copper wire or lag
14:33.56ManxPowerslak-: So fix the echo.
14:34.05DrWho17I don't have any analog hops
14:34.07*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
14:34.07*** mode/#asterisk [+o drumkilla] by ChanServ
14:34.14file[laptop]darumkilla!
14:34.15DrWho17tdm/sip or iax
14:34.21ManxPowerThere are several ways to do it.  The easiest is to drop $1,000 on a hardware echo canceler, as someone talked about YESTERDAY on the mailing lists.
14:34.23slak-ManxPower its not working out too well
14:34.31drumkillahey file[laptop] :)
14:34.37ManxPowerDrWho17: So you never call someone on an analog phone?
14:34.37|dennis|ManxPower: Which is best way to deal with it? because if we are to move to asterisk, admin will want to make sure that voice quality and line condition is same or almost equal to what we presently have....so what do you suggest doing?
14:34.52slak-suck a dong
14:34.59slak-forget about phone conversation
14:35.02ManxPowerslak-: What hardware echo can are you using?
14:35.17ManxPower|dennis|: the best way to deal with it is a hardware echo canceler.
14:35.18slak-uhm none?
14:35.23ManxPowerNot Digium, however.
14:35.27slak-i use the quad fxo module
14:35.29slak-digium
14:35.34slak-and i use software echo cancellers
14:36.37|dennis|slak-: it seems that you have a setup quite like what i want to have.....so how bad is the echo really?
14:37.05ManxPowerslak-: Ah.  I use echocancel=yes  echotraining=yes, lower my txgain until it's as alow as it can go and still have the far end hear us.  I also use the ECHO_CAN_MARK3 echo can
14:37.09drrayI used 4 FXO ports and got very little echo
14:37.47slak-well
14:37.50ManxPowerHOWEVER, we are experimenting with tellabs hardware echo canceler.
14:37.57slak-i just replac ed my * box with a more powerful machine
14:38.00slak-and i think it got better
14:38.08slak-i set my tx gain to 8.0
14:38.10slak-and it got beter
14:38.11slak-better
14:38.22ManxPowerslak-: and if you set your txgain to -8?
14:38.27slak-never tried
14:38.42slak-can i set it and restart * without killing any current phone conversations
14:38.43ManxPowerwell, the lower your outgoing volume the less echo you'll have.
14:38.52ManxPowerslak-: nol
14:39.01slak-echocancel=128
14:39.01slak-echocancelwhenbridged=yes
14:39.01slak-echotraining=500
14:39.01slak-relaxdtmf=no
14:39.01slak-rxgain=8.0
14:39.01slak-txgain=2.5
14:39.26ManxPowerslak-: I already gave you my settings.  Don't flood the channel
14:39.33*** join/#asterisk nagl (n=nagl@137.208.4.178)
14:39.35ManxPowerand your txgain is 2.5 not 8
14:39.36slak-alritey
14:39.42slak-oh i mean rx
14:39.55ManxPowerYou don't care what the rxgain is on the FXO ports.
14:40.09ManxPower(for echo can issues, that is)
14:40.13drray:)
14:40.26slak-well some echo cancel guide i read on voipinfo suggested that most peopple benefit from around 8.0
14:40.53drrayyour txgain is what you need to change
14:40.56ManxPowerslak-: 1) the TRANSMITTED audio is what the echo is, so playing with rxgain does nothing for echo.
14:41.03*** part/#asterisk bkw_ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net)
14:41.13slak-ManxPower alrite so you want me to change the tx to -8?
14:41.14ManxPower2) the value needed for txgain is unique to each location
14:41.18slak-how will that effect the volume
14:41.22drrayand each ear
14:41.34ManxPowerslak-: you have to play with it while the system is not in production
14:42.00slak-fuck it no ones using it
14:42.10mishehubah.
14:42.36ManxPoweryou want to keep lowering your txgain as much as you can but still allowing callers to hear you.
14:42.43slak-haha okay its at -8
14:43.06slak-crap * crapped out
14:43.10ManxPowerthen either stop and start asterisk or load/unload chan_zap.so or (CVS-HEAD only) reload chan_zap.so
14:43.15slak-wont even dial with -8
14:43.29ManxPowerslak-: well your volume is prolly too low.
14:43.41ManxPowerThere is no magic fix, you have to keep experimenting
14:43.44*** join/#asterisk Goshen (n=Goshen@67-40-107-29.slkc.qwest.net)
14:44.52*** part/#asterisk Equinox (n=secret@star.l93.com)
14:44.53slak-txgain=-5.0
14:44.55slak-works okay
14:45.14*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com)
14:45.15slak-no difference interms of echo tho
14:45.29ManxPowerslak-: There won't be unless you set everything right.
14:45.50slak-it went away 100% after like 15 seconds
14:45.51ManxPoweri.e. changing the echo canceler in zconfig.h, rebuild zaptel, change the other echo cancel options
14:47.43ManxPowerI have 9 asterisk servers and don't have echo after applying the settings I gave you earlier.
14:47.52ManxPowerslak-: Ah.  I use echocancel=yes  echotraining=yes, lower my txgain until it's as alow as it can go and still have the far end hear us.  I also use the ECHO_CAN_MARK3 echo can
14:48.01ManxPowersorry, echotraining=900
14:48.09*** join/#asterisk blitzrage (n=leif@asterisk/documenteur-extraordinaire/blitzrage)
14:48.17slak-900?
14:48.20slak-damn
14:49.16slak-u found that echo_can_mark3 works best?
14:49.26ManxPowerslak-: for OUR setups, yes.
14:50.06*** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net)
14:50.30slak-can i call u i want to test echo :D
14:50.33tzangerI use MARK2 and optimize zaptel for the proc I'm working on
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14:50.48slak-whats the default one
14:50.51tzangerMARK2
14:51.25coppiceall the echo cancellers in * suck. you just have to find the lesser of several evils
14:51.57nroejall echo cancellation sucks, but this one sucks less *g*
14:52.59coppicenot really true. I've done echo cancellers that left everyone deeply impressed. when we did the early ones for *, though, the main goal was low compute rather than good results
14:53.03slak-whats the difference between echotraining 500 and 900
14:53.07slak-what does it really mean
14:54.31drrayso my channel bank would have echo cancelation on it?
14:55.08tzangerdrray: I've never heard of htat
14:55.26tzangercoppice: when you die can I have your brain?
14:55.54tzangerslak-: I suggest you do some rudimentary reading, you're asking very basic questions that are very well documented in the wiki and in the doc/ directory of the asterisk source tree
14:56.17coppicedrray: channel banks are mostly designed for PSTN connection. they don't need echo can. they need a low price tag
14:57.04tzangerdrray: channel banks generally spend their money on good hybrid design which eliminates a lot of their designed echo opportunities
14:57.24tzangerdrray: when you throw the T1 into a computer and introduce PCI bus latencies and other overhead you create more opportunities for echo
14:57.37coppicetzanger: no they don't. echo just isn't that important in a channel bank
14:58.47tzangercoppice: I've never had bad echo problems due to poor hybrids in channel banks -- the adit600s I tend to prefer also have marketing mojo which suggests dynamic impedance matching which isn't too far-fetched, although the silabs parts that the TDM modules uses are supposed to do something similar
14:58.54coppicethere is only one driver in channel bank design - <$
14:59.30tzangercoppice: I dunno about that -- there are cheap channel banks and expensive ones, and the featureset runs the gamut between the two ranges
14:59.47coppicethe silabs parts don't do dynamic balancing, but they are tunable
15:00.34coppicetzanger: I used to run a team doing line cards. there was only *one* driver, and that was for public exchanges, not some grotty box
15:00.36tzangerthat's what I was getting at -- the tunability... perhaps not automatic but being able to tune to me means it's dynamic.  :-)
15:00.44mishehubah.
15:00.47bkw_I hear lots of complaints about the digium echo can board
15:00.49DrWho17drray: you probably need a echo cancellation card for your channel bank
15:01.00drraywell, I have very little echo
15:01.03bkw_mishehu, you having fun with OPAL too?
15:01.04DrWho17ok
15:01.04bkw_hehe
15:01.13mishehubkw_: You've Got Mail<tm>
15:01.23DrWho17carrier access and adtran both have them
15:01.24mishehuheh
15:01.34coppicesomeone has a tuning program for the tdm400. dunno ho good it is. he asked me how to do it. dunno if he took my advice.
15:01.36drrayI just read here and the mailing list and get a creeping fear that one day I will wake up with echo
15:01.51mishehubkw_: we're having pwlib fun at the moment.  I give kudos to craig, it's actually a pretty nice API
15:02.12bkw_yes even I understood it
15:02.17*** join/#asterisk luba (n=chatzill@altom.net1.nerim.net)
15:02.23bkw_but I don't understand C++ as well as I should
15:02.33bkw_its like 5 lines of code to setup an IAX2 endpoint
15:02.34coppicebkw_ what do you hear about the digium ec board?
15:02.36bkw_like 12 for SIP
15:02.57bkw_coppice, I just see on the lists people are saying they hear clicks and pops and all kinds of weirdness with the hardware echo can
15:02.57tzangeryeah fxotune... that's for tuning the FIR filter
15:03.06kyooWhen one person calls into my Asterisk system (calls BV number and gets sent into asterisk via SIP) his keys make no tones - he cannot choose any menu options.  His dtmf works on other calls (but he says this does happen to specific numbers) and other are able to use my menus just fine ... Is there anything I can do?
15:03.13mishehubkw_: I can be of help, of course.  c++ is my main favorite language.  tony j. (guy who works with me) knows c++ and java also.
15:03.27coppicepwlib seems to have been the biggest problem with openh323. it kept breaking things
15:03.31*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
15:03.42*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:03.47*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
15:03.48mishehuI was thinking of writing a c++ application and mentioning a greets to kram in it, just to raz him a bit...  since he so despises c++.
15:04.02bkw_coppice, but pwlib has been around for almost 20 years hasn't it?
15:04.04ManxPowermishehu: most sane people do.
15:04.17coppicetzanger: i think it changed name. isn't called not the fxotune or something?
15:04.39tzangercoppice: that's what I said.  :-)
15:04.49mishehuManxPower: most sane ppl do what?
15:04.50tzangeror are you saying it's not called fxotune anymore?  I don't have any FXO port so I don't really know
15:05.02coppicei said "not the fxotune" :-)
15:05.10ManxPowermishehu: most sane people hate C++ for realtime or OS usage.
15:05.13bkw_coppice, does pre19 fix anything major?
15:06.29DaPrivateerGrrr... I am trying to preform a PBX transfer with a digium X100P (sip phone). anyone have any suggestions? please? i need to get this done
15:06.29tzangerheh
15:06.32tzanger"Multiply in your head" (ordered the compassionate Dr. Adams)
15:06.32tzanger"365,365,365,365,365,365 by 365,365,365,365,365,365.  He [ten-year-old
15:06.32tzangerTruman Henry Safford] flew around the room like a top, pulled his
15:06.32tzangerpantaloons over the tops of his boots, bit his hands, rolled his eyes
15:06.32tzangerin their sockets, sometimes smiling and talking, and then seeming to be
15:06.35tzangerin an agony, until, in not more than one minute, said he,
15:06.38tzanger133,491,850,208,566,925,016,658,299,941,583,255!"  An electronic
15:06.40tzangercomputer might do the job a little faster but it wouldn't be as much
15:06.43tzangerfun to watch.
15:06.44mishehuManxPower: well, I don't even suppose that c++ is an end-all-be-all, it has its place...
15:06.51*** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net)
15:06.55*** join/#asterisk punker- (n=asdas@200.62.182.109)
15:06.56tzafrir_laptopregardless of C++, it just seems that pwlib tries to do too many things
15:07.24tzafrir_laptopSomewhat like QT
15:07.27mishehutzanger: that a dave barry quote?
15:07.35tzangermishehu: no it's from my fortune file
15:07.44*** join/#asterisk santiago (n=santiago@63.245.87.180)
15:07.59*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmv6.dialup.mindspring.com)
15:08.04mishehutzafrir_laptop: pwlib is more or less just an abstraction layer.  do with it what you want to do with it, basically.
15:08.08mishehuignore the rest.
15:08.29bkw_its like libc
15:08.32bkw_just a lib
15:08.51mishehutzanger: and dave barry is in fortune a lot...  the style seemed like him.
15:09.05bkw_ok now to go buy a book c++
15:09.17tzafrir_laptopmishehu, look at what recently happened to QT
15:09.43mishehutzafrir_home: what recently happened to QT ?
15:10.01mishehuall I know is they released a new ver, and have an open source win32 build for it as well...
15:10.23tzafrir_laptopmishehu, and last time I heard, you don't even have to link all of libc in. -lm sounds familiar?
15:11.27tzangermishehu: I know, but this wasn't him, I forget the attribution
15:12.04tzafrir_laptopmishehu, in QT4 they have finally followed gtk in that they split their library into several sub-libs. So now yo can sanely build console QT apps
15:12.43mishehutzafrir_laptop: how many sub packages did they split to?
15:13.36mishehubkw_: ok, so you Don't Have Mail...  it bounced on me heh
15:13.49tzafrir_laptopNot sure exactly. I'm more familiar with the gtk side. I know quite a few programs that use glib, pango etc. and not gtk
15:14.25*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
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15:15.02coppicebkw_ if the bug that fixed is one hurting you its always major :-)
15:15.05jaikeanyone using polycom 301? were looking for headsets that match those
15:16.02coppicebkw_ pre19 builds cleanly on x86_64 machines. It tolerates FAX machines where the timing is a big off somewhere better
15:16.18bkw_oh I need to update that
15:16.29*** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
15:16.46*** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net)
15:17.26tzafrir_laptopspandsp pre19?
15:18.15*** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net)
15:19.03*** part/#asterisk zedkatuf (n=audela@82-46-92-246.cable.ubr08.azte.blueyonder.co.uk)
15:20.28ManxPowerI really hope I don't have pneumonia
15:20.35slak-smoke a cig
15:20.38slak-and some ganja
15:20.40slak-best cure
15:20.55ManxPowerganja only treats the symptoms
15:20.56coppiceManxPower: don't worry. you can't pass it on to us
15:21.04slak-man
15:21.04*** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer)
15:21.17ManxPowercoppice: but I want to share!
15:21.17slak-i have to setup a 64bit chroot env on this linux box
15:21.21slak-anyone have experience with that
15:22.08*** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
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15:24.54*** join/#asterisk pbxbart (n=pbxbart@p54B008D9.dip0.t-ipconnect.de)
15:25.01*** part/#asterisk pbxbart (n=pbxbart@p54B008D9.dip0.t-ipconnect.de)
15:25.22JerJersorry i've only done 8 bit
15:25.29tzafrir_laptopslak-, on debian there is debootstrap for that...
15:26.06slak-yea what do you suggest, install the 64bit port right off the bat or install it ontop of the 32bit
15:26.16slak-like with a chroot
15:26.22tzafrir_laptopslak-, chroot for what? on what distro?
15:26.33slak-debian
15:26.42slak-pure64
15:27.03tzafrir_laptopslak-, why not ask on #debian ?
15:27.28*** join/#asterisk nagl (n=nagl@137.208.4.166)
15:28.02*** join/#asterisk Luba_ (n=chatzill@altom.net1.nerim.net)
15:28.02olivier_and with your nick you shuld install a slackware ;-)
15:28.25*** join/#asterisk blake__ (n=veteran@cpe-66-25-30-132.houston.res.rr.com)
15:28.35bkw_Å“
15:28.48Faithfultzafrir_laptop: I'm giving that rapid cd a whirl see what I think of it ... if it detects my HFC Bri card I will be impressed
15:29.18*** join/#asterisk audela (n=audela@82-46-92-246.cable.ubr08.azte.blueyonder.co.uk)
15:29.44Faithfulwhat is pure64 as opposed to amd64?
15:30.00tzafrir_laptopisn't amd64 pure64?
15:30.07QwellFaithful: pure64 as opposed to a hybrid 32/64
15:30.28tzafrir_laptopthat is: not s hybrid 64/32 port like the ones of most distros
15:30.32*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
15:31.09FaithfulHmmm.  so debian-amd64 is not pure?
15:31.24Faithfuland when you say hybrid what are you meaning?
15:31.39lathos42Anyone have an idea why i'm not getting ringback calling out through my SPA-3000 with my Polycom IP501, when I get ringback from my Sipura SPA-841?
15:32.31*** join/#asterisk heison (n=heison@ns.somanetworks.com)
15:32.37heison~seen sivana
15:32.39jbotsivana is currently on #asterisk (1d 9h 26m 39s)
15:35.29lathos42Whoops, I'm not getting any audio back at all on my IP 501
15:37.30*** join/#asterisk nroej (n=joern@lak-39-56.wohnheime.ruhr-uni-bochum.de)
15:37.34nroejre
15:38.38tzafrir_laptopFaithful, debian's amd64 port is "pure". That is: it does not contain ia32 compatibility libs
15:38.50tzafrir_laptopthere is no /lib and /lib64
15:39.02coppicethat must suck rather badly
15:39.15coppicemost people have lib and lib32
15:39.52coppiceFC4 has a mix, though :-\
15:39.55tzafrir_laptopcoppice, maybe. I have no idea. I only played with an amd64 system for a short while about two years ago
15:40.13*** join/#asterisk nroej (n=joern@lak-39-56.wohnheime.ruhr-uni-bochum.de)
15:40.14nroejre
15:40.15coppicemy X2 machine is very nice, and very fast
15:40.45tzangerX2?
15:40.49Qwelldual core?
15:40.58coppicethe dual core AMD64
15:41.07*** join/#asterisk Juxt (n=Juxt@64.135.20.202)
15:41.12*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
15:41.21tzangerah
15:42.04*** join/#asterisk gclark (n=gclark@66-193-73-162.gen.twtelecom.net)
15:42.06coppiceI have an X2 4200+ machine, and a dual xeon 2.4GHz machine. spandsp runs several times as fast on the X2, but I am not yet clear why it is so much better
15:43.11tzafrir_laptopcoppice, a related topic, which I could not get answers to on the mailing list
15:43.31shido6christ
15:43.34shido64200+
15:43.35tzafrir_laptopDEbian had a habbit of building everything with 386 compatibility.
15:43.36Qwellyes?
15:43.37gclarkHello all.
15:44.19tzafrir_laptopI was wondering what are the minimal parts I need to rebuild optimized to gain maximum extra performance.
15:44.22Juxthello
15:44.29gclarkI am having a problem with CVS-Head.  I installed it but when I try to start asterisk via the init script I receive permission denied.  I am running this as root?  Any thoughts?
15:44.38tzafrir_laptopI was also wondering what could be a useful benchmark
15:45.16tzafrir_laptop"show translations" is a useful benchmark because it is easily accessible on every asterisk installation. Not sure about the value of the numbers
15:45.21Qwelltzafrir_home: glibc and the kernel would probably give the most noticable boost
15:45.39tzafrir_laptopgclark, permissions denied to where, exactly?
15:45.42jaikeis possible to set the recording filename for queue calls to include the extension number that will receive the call?
15:45.52tzafrir_laptopQwell, what about codecs?
15:46.02Qwellall of them :p
15:46.16gclarkI am trying to run '/etc/init.d/asterisk start'
15:47.44*** join/#asterisk rkvarala (n=rkvarala@59.93.96.168)
15:47.52gclarkThe exact error message is 'Starting asterisk: execvp: Permission denied'
15:48.21tzafrir_laptopalso: libgsm uses (dynamically) libgsm. how much of the work is done in the asterisk module itself and how much in libgsm? anybody experimented in that?
15:48.38tzafrir_laptopgclark, /usr/sbin/asterisk is not executable?
15:48.51*** join/#asterisk aVaLaNcHe (n=stecnic@m85-94-161-25.andorpac.ad)
15:48.59Kattyhmm.
15:49.06aVaLaNcHebzzz
15:49.11Kattyi have dumb question - no surprise there
15:49.16tzafrir_laptop'bash -x /etc/init.d/asterisk start' will give you a trace
15:49.30QwellKatty: no such thing as a stupid question
15:49.47KattyQwell: there is :P
15:49.49gclarkNo.  the init script that is created from make config and it is put into /etc/init.d/asterisk and a flag is start|stop|restart
15:49.56Qwellnah, just stupid people :P
15:50.08ManxPowerQwell: I was going to say that.
15:50.09Kattythis thing i'm quoting is going to go Broadband -> Firewall -> Router or Switch -> Asterisk -> Phones
15:50.23KattyBut i don't know if I need a router or a switch.
15:50.35Qwelldoes your firewall route?
15:50.35Kattyi get that a router..uhh...routes things.
15:50.39jaikegclark: y not just run asterisk -vvvvc?
15:50.41tzafrir_laptopgclark, something is wrong here. Do you want to isolate the problem?
15:50.45Kattybut in terms of the firewall routing....
15:50.48Kattyi don't know what i'm looking for
15:50.54Kattyso, how do i know if my firewall is also a router?
15:51.13*** join/#asterisk charles___ (n=charles@64.35.168.55)
15:51.25charles___you
15:51.26QwellI guess if it forwards packets to a switch, it would be a router, right?
15:51.26ManxPowerKatty: MOST firewalls also route, especially the cheap ones.
15:51.38charles___hey guys
15:51.43KattyManxPower: that doesn't help me determine anything.
15:51.48aVaLaNcHeWhen I try to make a external call my debug says ..
15:51.51KattyQwell: i have a model number of a firewall i want to use.
15:51.57*** join/#asterisk tAURUS (n=_enver_@213.227.207.45)
15:51.58*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
15:51.58Qwelloh, you don't have it yet...
15:52.05Juxthas anyone ran asterisk in daemontool?
15:52.11charles___is there a way to decrease the ammount of rtp on SIP ?
15:52.18aVaLaNcHequit
15:52.32tzafrir_laptopJuxt, does daemontools work for UDP?
15:52.48Qwelltzafrir_laptop: daemontools is an init replacement thing, afaik
15:53.01*** part/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk)
15:53.19*** join/#asterisk aVaLaNcHe (n=stecnic@m85-94-161-25.andorpac.ad)
15:53.24aVaLaNcHebzzz
15:53.53aVaLaNcHeWhen I try to make a external call my debug says ...
15:53.54aVaLaNcHeAug 18 12:24:01 WARNING[3507]: chan_zap.c:7138 zt_request: Unable to determine channel for data {Zap/g1/840064
15:53.58aVaLaNcHeAug 18 12:24:01 NOTICE[3507]: app_dial.c:777 dial_exec: Unable to create channel of type 'Zap'
15:54.01aVaLaNcHeAug 18 12:24:11 WARNING[3507]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't'
15:54.04aVaLaNcHein context 'default'
15:54.06aVaLaNcHe..
15:54.12aVaLaNcHeanyone can help me ? please
15:54.13*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk)
15:54.15*** join/#asterisk jdg (n=jdg@CA03F8ED.adsl.mana.pf)
15:54.32ManxPoweraVaLaNcHe: You have a { where a ( should be
15:54.48funxiongood cal ManxPower
15:54.58aVaLaNcHeManxPower: where ?
15:55.05ManxPowerdata {Zap/g1
15:55.07funxionwhere you dial your zap channel
15:55.20ManxPoweryou ONLY use { and } for variable names
15:55.27gclarkasterisk -vvvvvvvvvc
15:57.03punker-does anybody knows any site where i can find some asterisk with mysql documentation?
15:57.12Qwellpunker-: the wiki
15:57.13Qwell~docs
15:57.13jbot[docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
15:57.20*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:57.20*** mode/#asterisk [+o anthm] by ChanServ
15:57.33ManxPower~mailinglist
15:57.33jbotmailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:58.58rkvaralawhat is the newextension in Agentcallbacklogin
15:59.09rkvaraladoes any body have any idea on this?
15:59.56punker-thanx
15:59.59funxionnewextension?
16:00.26*** part/#asterisk jdg (n=jdg@CA03F8ED.adsl.mana.pf)
16:01.17aVaLaNcHeManxPower: ok, thx that's write now ... but it continue saying Aug 18 12:24:01 WARNING[3507]: chan_zap.c:7138 zt_request: Unable to determine channel for data {Zap/g1/840064
16:01.20aVaLaNcHeAug 18 12:24:01 NOTICE[3507]: app_dial.c:777 dial_exec: Unable to create channel of type 'Zap'
16:01.23aVaLaNcHeAug 18 12:24:11 WARNING[3507]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't'
16:01.26aVaLaNcHein context 'default'
16:01.28funxionwhen you use the agentcallbacklogin you specify the callacbk extension during login
16:01.30ManxPoweraVaLaNcHe: no, that's not right.
16:01.31aVaLaNcHesorry
16:01.34ManxPoweryou still have a { in there.
16:01.34aVaLaNcHesorry
16:01.50funxionaVaLaNcHe did you reload?
16:02.05aVaLaNcHeonly de line pbx.c 1952 ast_pbx_run Timeout no rule 't'
16:02.21ManxPowerwell but a hangup after your dial then
16:02.38tzafrir_laptopBTW: isn't there a better way to serch the list archives? e.g: what do you think about gmane?
16:02.46ManxPower~mailginlist
16:02.49ManxPower~mailinglist
16:02.49jbotextra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
16:03.10tzafrir_laptopManxPower, yes, I've just erad that text above
16:03.10gclarktzafrir_laptop, do you have any ideas on the asterisk init script?
16:03.12Juxtdoes anyone have a list of all area codes and city, state ?
16:03.20ManxPowertzafrir_home: never heard of is.  Is it the GNU Lion Hair or something like that?
16:03.35ManxPowerJuxt: no.  Area codes span miltiple cities and states.
16:03.51tzafrir_laptopgclark, does asterisk start when you run it manually? asterisk -cvvvv
16:03.55funxionjuxt you want bellcoe v and h tables
16:04.07tzafrir_laptop(someone already asked you. I probably missed the answer)
16:04.07Juxtok where do i find htem
16:04.14gclarktzafrir_laptop, yes it runs fine
16:04.22funxionthey usually cost money
16:04.24ManxPowerJuxt: you purchase them for large amounts of money.
16:04.38*** join/#asterisk [kwoot] (n=kvirc@i2rs-son.xs4all.nl)
16:04.47citatsheh, its only a few grand for a one time copy :)
16:04.54Juxtheh ok
16:04.57tzafrir_laptopgclark, so with what parameters is it run in the init script?
16:05.23DaPrivateerManxPower - any chance you can help me a little more with the pbx transfer problem?
16:05.26funxionI remember it being 600 bux last time I updated my list
16:05.34tzafrir_laptopgclark, use script to get the output of that bash -x' line to a file
16:05.36Juxthttp://www.telcodata.us/telcodata/downloads
16:05.43Juxtdoesn't seem all that pricey
16:05.48tzafrir_laptopand pastebin it
16:05.56charles___ManxPower:  is there a way to decrease the amount of samples per second over SIP ?
16:05.58Juxtlata-npa-nxx-state-city-zip.csv
16:06.12charles___ManxPower:  I can see here that it's overkilling
16:06.17*** join/#asterisk Defraz (n=t0tal@tim.ibccom.net)
16:06.19gclarkstart() {
16:06.19gclark<PROTECTED>
16:06.19gclark<PROTECTED>
16:06.19gclark<PROTECTED>
16:06.19gclark<PROTECTED>
16:06.19ManxPowercharles___: Why in the world would you ever want to do that?
16:06.19gclark<PROTECTED>
16:06.21gclark<PROTECTED>
16:06.23gclark<PROTECTED>
16:06.25gclark<PROTECTED>
16:06.26tzafrir_laptop~pb
16:06.26jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
16:06.27gclark<PROTECTED>
16:06.29gclark<PROTECTED>
16:06.31gclark<PROTECTED>
16:06.33gclark<PROTECTED>
16:06.34charles___ManxPower: because 8 sip channels over G723 is using 1MB
16:06.35gclark<PROTECTED>
16:06.36ManxPowergclark: DON'T FLOOD THE CHANNELS!!!!!!!
16:06.37Qwell~pastebin
16:06.37jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
16:06.37gclark<PROTECTED>
16:06.38Qwellasshat
16:06.39charles___1Mb
16:06.39gclark<PROTECTED>
16:06.40citatsnanpa has some text files as well that you can download but they have limited info
16:06.41Juxtgee yesterday i got kicked for that!
16:06.41gclark<PROTECTED>
16:06.43gclark}
16:06.44ManxPowercharles___: so switch to a better codec.
16:06.54funxionkewl
16:06.58aVaLaNcHeManxPower: asterisk -vvvvvc says ... Called g1/800246 , Channel 0/1 span 1 got hangup, Hungup 'Zap/1-1', No one is available to answer at this time
16:07.03ManxPowerJuxt: can we jick him again
16:07.04*** join/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
16:07.11ManxPowerDaPrivateer: no.
16:07.25charles___ManxPower: man, the same codec over IAX is doing 80Kbps
16:07.40ManxPoweraVaLaNcHe: sounds like you are on a PRI and you are not checking the result of the dial
16:07.47tzafrir_laptopgclark, run: script logfile
16:08.01ManxPowercharles___: That is correct.
16:08.03rkvaralaManxPower do you have any idea on this Agentcallback login new location
16:08.09tzafrir_laptopgclark, then run bash -x /etc/init.d/asterisk start
16:08.14[kwoot]anyone able to help a newbie? want to connect asterisk to isdn/bri using modem. version is cvs-head from 13 aug 04.
16:08.18ManxPowercharles___: and you are not running G723
16:08.33ManxPowercharles___: want to try describing your situation accuratly this time?
16:08.35aVaLaNcHeManxPower: yes, but I don't know the solution ... can you help me ?
16:08.37tzafrir_laptoprun exit. and paste the file logfile in http://pastebin.ca
16:09.03tzafrir_laptop~docs
16:09.03jbotsomebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:09.16ManxPoweraVaLaNcHe: I can do hand holding for $120/hr.
16:09.27aVaLaNcHethx
16:09.28aVaLaNcHexDD
16:09.35[kwoot]got to go.
16:09.35ManxPoweraVaLaNcHe: otherwise read the wiki about PRI and look at the std-exten in extensions.conf.sample
16:09.43aVaLaNcHeok, thx
16:09.54tzafrir_laptop[kwoot], read the wiki (woip-info) about modems. which modem exactly? I believe chan_capi and misdn are supported with HEAD
16:10.12tzafrir_laptopbah
16:10.12ManxPowerand understand that when you dial out of a PRI you will NEVER hear a busy siglal from the telco, Dial will end the call and set DIALSTATUS = BUSY
16:10.51ManxPowerthen you have to generate the correct tones to the caller using the dialplan,  an example of all this is in std-exten
16:10.52bkw_ManxPower, that also depends
16:11.06ManxPowerbkw_: in 1.0.x I can't get priindication=inband to work
16:11.24bkw_as I said it depends on how the PRI is provisioned also
16:11.34ManxPower*nod*
16:11.36punker-which is the best linux for asterisk?
16:11.46ManxPowerpunker-: stop trolling
16:12.00nicoxHello, do anybody know how to cut a dtmf-stream in asterisk?
16:12.01ManxPowerpunker-: the best linux for ANYTHING is "what you are familiar with"
16:12.07*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:12.28ManxPowerI'm joking.  The best Linux is Mandrake/Mandriva linux 9.2
16:12.30bkw_ManxPower, stop being a fucking prick
16:12.40bkw_punker-, use what you're comfortable with
16:12.42brad_msswwhew, just went live today with asterisk ... finally get to trash that POS phone system we did have ...
16:12.49punker-oohhh thanx :)
16:12.51*** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com)
16:13.18Ariel_wow best linux question
16:13.20MikeJ[Laptop]brad_mssw, what did you have?
16:13.21punker-but i asked that cos some linux are more vulnerables to hackers right??
16:13.33bkw_punker-, WRONG
16:13.35brad_msswMikeJ[Laptop]: talkswitch PBX ...
16:13.39Ariel_punker-, which one do you like to use?
16:13.39tzafrir_laptoppunker-, why is that?
16:13.42bkw_linux is only as secure as the admin makes it
16:13.47punker-well.. i used fedora
16:13.58MikeJ[Laptop]well, that is probably less secure ;)
16:14.03Ariel_ok then use fedora or RH or CentOS they all work with Asterisk.
16:14.04punker-i installed asterisk in fedora core 3
16:14.08blitzragepunker-: if you like RH style - use CentOS, Fedora is just a bitch
16:14.11nicoxanybody how can help?
16:14.15Qwellbkw_: unless you use lindows, or whatever the hell they call it now
16:14.19QwellLindora?  heh
16:14.27nicoxHello, do anybody know how to cut a dtmf-stream in asterisk?
16:14.33punker-but i was reading a documentation of asterisk on devian
16:14.43tzafrir_laptopproblem with fedora is that it will not stay supported long enough. Unless fedoralegacy starts improving real-soon-now
16:14.44Ariel_devian.....
16:15.09Ariel_if you like Fedora and want something stable that works great use CentOS
16:15.19coppicenicox: tried a carving knife?
16:15.22punker-:O
16:15.22tzafrir_laptopnicox, we're busy in a distro fight. But ask your Q anyway
16:15.38punker-CentOS????
16:15.53blitzragepunker-: www.centos.org
16:15.55Beirdotzafrir_home: don't hold your breath about fedoralegacy :)
16:15.56punker-yeah
16:15.59coppicethe best linux to use in *not* the one you are familiar with
16:16.00nicoxHello, do anybody know how to cut a dtmf-stream in asterisk?
16:16.03punker-im right ther
16:16.11tzafrir_laptopnicox, can you get the stream in a var and use Cut?
16:16.20coppiceits the one some other sucker is familiar with, and you get them, to do all the work
16:16.38nicoxi don't know which var the dtmf stream is... thats my problem
16:16.57tzafrir_laptopBeirdo, which is why I don't understand why people base their Asterisk servers (that should remain running for long) on it
16:17.06punker-ok... i have several options... CentOS... Mandrake... Devian....
16:17.17Ariel_coppice, nice
16:17.18tzafrir_laptopeven worse: some are even installing new asterisk installations on RH9 still
16:17.23sylego fedora core 4 or freebsd
16:17.25tzafrir_laptopDebian!
16:17.29nicoxi get a dtmf-stream which comes from the e1 as overlap digits... but with this stream at the end ob the number my telco says nono
16:17.31tzafrir_laptopnot Devian
16:17.38punker-ahh ok... no devian
16:17.50Beirdoas I wanted to strip shit down and only compile what I wanted
16:18.07nicoxdebian is okay, all my asterisks are running on debian *g*
16:18.09BeirdoI use Ubuntu for all my other new installs
16:18.10punker-some friend told me that freebsd is a good linux
16:18.20jaikegood linux?
16:18.22Ariel_argh
16:18.22Beirdofreebsd != linux
16:18.28*** join/#asterisk nagl (n=nagl@137.208.4.161)
16:18.28Beirdofreebsd = blech
16:18.28Ariel_freebsd is not linux
16:18.33sylefreebsd > linux
16:18.37blitzrageHardware won't work on FreeBSD
16:18.38Beirdothey are totally different
16:18.38jaikepunker...try winxp
16:18.39rkvaralaiam getting this error in Agentcallbacklogin chan_agent.c:1298 __login_exec: Extension '1000' is not valid for automatic login of agent '1000'
16:18.44jaike:P
16:18.49sylehardware works fine on fbsd
16:18.54Ariel_ROFL winxp
16:18.54punker-FREESB
16:18.55blitzrageAsterisk is developed on, and for Linux. If you're going to use FreeBSD, you better be good :)
16:18.59Beirdofreebsd can eat me
16:19.00Beirdo:)
16:19.01Beirdohehe
16:19.12punker-Linux Freesb
16:19.38*** join/#asterisk Nukemizer (n=Nuke@67.137.28.165)
16:19.56Ariel_<PROTECTED>
16:19.56*** join/#asterisk Blake0PS (n=blake@blakeops.com)
16:20.06QwellYou hit enter instead of r?
16:20.12QwellAriel_: You were just a tad off there
16:20.16Beirdodo SIP->IAX :)
16:20.21Ariel_Qwell, yes
16:20.43Ariel_Qwell, new ergonomic stupid keyboard. Need my old flat one back.
16:21.16Blake0PSWith 2 TDM400Ps installed, which one will be ports 1-4 and which one will be ports 5-8?
16:21.21lathos42Beirdo: I have * running on my WRT54G at home that i've successfully used to route a call to my * server here at work
16:21.29Beirdocool
16:21.55Beirdomy use would be to hook up Sipura devices, and to trunk over the internet via IAX
16:22.04BeirdoI don't like SIP particularly
16:22.05*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
16:22.05Beirdo:)
16:22.38Beirdoif I'm careful, I won't need to translate codecs
16:22.48nicoxdo anybody know if there is a variable where i can find dtmf-digits?
16:22.48Beirdoas I'm sure that will suffer on a wrt
16:22.59lathos42Beirdo: That's what I did.. I have a SPA-841 that I registered with the WRT, and then used IAX to call between *
16:23.05Beirdo:)
16:23.16lathos42Beirdo: It worked fine for me using ulaw all the way through the call
16:23.24Beirdoyeah, that was my plan
16:24.18nicoxsomeone who can help?
16:24.32jaikelathos: ur spa841 working fine? we bought around 20 and 5 are defective now
16:24.52Beirdoexcept I'll be using the SPA3000
16:25.03lathos42jaike: Yeah, it seems to be working fine.. I only have one at the moment
16:26.03jaikelathos: were using ip301s right now...they working fine
16:26.48nicoxhelllo?
16:27.02lathos42I've got a strange problem at the moment where the IP501 can't hear any audio when I try to call out through the PSTN of the SPA-3000.. but I can get audio just fine if I call any of my phones
16:27.49brad_msswlathos42: are you going through NAT, I had that issue yesterday
16:28.14lathos42brad_mssw: Nope, they're both on the same network.. plugged into the same switch on my desk actually
16:28.26nicoxdo anybody know if there is a variable where i can find dtmf-digits?
16:28.41brad_msswlathos42: wow, i don't know what the ip501 is, is it also a sip device ?
16:28.46brad_msswif so, that doesn't make sense
16:28.58lathos42brad_mssw: Yeah, its a Polycom SIP phone
16:29.33*** join/#asterisk J[SS] (i=ph33r@smartserv.ipv6.smart-serv.net)
16:29.47DaPrivateerlet me ask a different question. if im in an AGI that just answered a call on a zaptel FXO card, how can i dial a number on that card?
16:30.06DaPrivateerie just play DTMF tones to the caller
16:30.47coppicehow come Dell charge so much less in .us than in asia? damned annoying
16:31.02*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:31.08*** join/#asterisk DannyF (n=wizard@c-6d4fe353.24-0099-74657210.cust.bredbandsbolaget.se)
16:31.13*** join/#asterisk uma62 (n=sudhir@pool-71-114-93-22.washdc.dsl-w.verizon.net)
16:31.38nicoxthere is a dtmf stream from the incomming E1 and i will forward the call, but the dtmf-stream is also forwarded, and i don't know hot to stop this stream
16:32.05shido6whatever you do
16:32.09shido6dont cross the streams
16:32.31coppicethat would be bad, wouldn't it?
16:32.48nicoxi have to cut the stream, but i don't know how
16:32.57mutpinch really hard
16:33.05Qwellsharp knife
16:33.09coppice"I am the GPG keymaster"
16:33.11coppice"I am the GNU Gatekeeper"
16:33.24citatslol
16:33.26*** join/#asterisk cypromis (n=michael@83.149.70.59)
16:33.30shido6I am Zuul.
16:33.48Hmmhesaysanyone using serweb in here? the install file says nothing about database setup
16:33.57nicoxnobody an idea?
16:34.59Beirdobah humbug
16:35.08*** join/#asterisk illvm_ (n=illumina@cpe-65-185-103-95.woh.res.rr.com)
16:35.45illvm_can anyone help me get a polycom IP500 to connect to my * box?
16:36.32blitzrageillvm_: you should ask specific questions to help you get it setup. People can't hold your hand.
16:36.47*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
16:37.04nicoxhow can i stop a dtmf-stream?
16:37.42coppicepull the plug out
16:37.44illvm_Ok... The phone is behind a NAT, I used the web portal to configure the external IP address on the phone and that seems to be working correctly. However, even though the password and username are correct, I still get a 401 message back from asterisk.
16:38.30*** join/#asterisk sbnelson (n=sbn@pm3-10stl5.xtraport.net)
16:38.39nicoxgood joke coppic
16:38.49illvm_http://pastebin.com/340897
16:39.30|dennis|which ip phone does one recommend to be best used in a SOHO( a small community school actually) setting..
16:40.23shido6PAP2-NA's and analog phones
16:40.31shido6IAXy's and analog phones
16:41.39shido6SPA-841
16:41.50sbnelsonAsterisk crashes every so often; is this the place to discuss?
16:41.59shido6yes
16:41.59sbnelsonmy asterisk server, I should say.
16:42.15|dennis|shido6: so with the PAP2-Na's i can use my existing phones ..hmm..right?
16:42.23shido6yes
16:42.23sbnelsonIt always happens after getting chan_zap.c: Unable to get index, and nullok is not asserted
16:42.25shido6and it has 2 ports
16:42.31shido6so 2 users can use it simultaneous
16:42.36shido6around $70 each
16:42.43illvm_so does anyone have any ideas why the Polycom SoundPoint IP 501 doesn't register?
16:42.50shido6yes
16:42.52shido6check your pm
16:43.01shido6its your sip.conf
16:43.11nicoxis there a chance to change a dtmf stream?
16:43.20Derkommissarfrom sip.conf , the file
16:43.20Derkommissarhow can tell it to open res_odbc ?
16:43.27shido6and can u get me a screncapture of your config for the polycom or make it accessible from teh pub net
16:43.52*** join/#asterisk JonR800 (i=jon@p1mp.org)
16:45.39blopwhat would u recommand as rackable hardware for asterisk (used with a tdm400p inside of it) ?
16:45.54*** join/#asterisk JonR800 (i=jon@p1mp.org)
16:46.08*** join/#asterisk luke-jr__ (n=luke-jr@user-0c938q3.cable.mindspring.com)
16:46.15|dennis|shido6: thanks..
16:46.18jaikeillvm: i can dcc you a config doc for polycoms if u like
16:46.22blopsth with the pci port in the front in place of the back would be nice (to patch with the zaps)
16:47.04*** join/#asterisk JonR800 (i=jon@p1mp.org)
16:47.55sbnelsonAsterisk (CVS-HEAD-04/16/05-14:27:21) dies with chan_zap.c: Unable to get index, and nullok is not asserted -- does anyone have a suggestion?
16:48.17*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
16:48.20sbnelsonactually dies sometime afterwards.
16:48.29jaikemy only problem with polycoms is that volume resets after each call...looking for a way to make it permanent
16:48.35coppice"Dear Asia-Pacific customer. Dell thinks its bedtime in Asia, and doesn't want to sell you anything"
16:48.56*** join/#asterisk JonR800 (i=jon@p1mp.org)
16:49.00coppicejaike: the lack of an echo can in the handset is a common complaint about polycoms
16:49.18*** join/#asterisk Gamentine (i=WinNT@d66-222-224-51.abhsia.telus.net)
16:49.49jaikecoppice: those are the ones good enough for our budget..ciscos are expensive
16:50.12coppicei wouldn't call polycoms cheap
16:50.48nicoxis there a chance to change a dtmf stream?
16:51.36*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:51.55*** join/#asterisk secure75 (n=mic@85.233.35.180)
16:52.19coppicejaike: it could be the echo is why they keep reseting the volume
16:52.47jaikeTo conform to regulatory requirments,
16:52.48jaikehandset and headset volume
16:52.48jaikewill return to a preset level after each
16:52.48jaikecall. Hands-free volume settings will
16:52.48jaikebe maintained across calls.
16:53.26jaikecrap
16:53.53*** join/#asterisk clyrrad (n=ddd@CPE0050bae8d02c-CM0011aea484a4.cpe.net.cable.rogers.com)
16:53.56coppicedunno any regulation which calls for that. it will lessen the impact of echo if you don't make every call loud, though
16:54.20movercan i get the echocancel board thats on TE406P for my TE405P from digium?
16:54.30nicoxis there a chance to change a dtmf stream?
16:55.05moveri cant get any infos. support not answered yet.
16:55.35drumkillamover: yes, but you would have to ship your board back in to have it upgraded
16:55.51coppicejaike: unless the mean the approval for minimum return loss. that would be a nasty twist on true meanings. the sort of thing to make a marketer proud :-)
16:55.59moverdrumkilla really?
16:56.04drumkillayes
16:56.09*** join/#asterisk tzanger_ (n=tzanger@mixdown.ca)
16:56.21moverargh...
16:56.33coppicemy guess is they need to put the V2 FPGA code in at the same time
16:56.37jaikecoppice: theyre using regulations to hide echo problems i guess
16:57.33sbnelsonasterisk is dying; only clue is "chan_zap.c: Unable to get index, and nullok is not asserted" sometime before it crashes.  Does anyone have an idea on what I can look for?
16:57.34movercoppice ... u are familar with ss7. i got desperate echo on one ss7 machine with libisup. i see this in IAM: [___0____] Octet 1: Echo control device indicator (0=outgoing echo control not included, 1=is  included
16:57.49moveris this the cause where echo come?
16:58.00moveron gsm calls now echos happened
16:58.15nicoxis there a chance to change a dtmf stream?
16:58.53jaikesbnelson: install asterisk from scratch...thats what i do
16:59.00*** join/#asterisk docelm0_ (n=docelm0@67.106.194.90.ptr.us.xo.net)
16:59.16sbnelsonjaike: I did!
16:59.24coppicemover: strange the GSM system should have removed the echo before the audio gets to you. what ss7 kit are you using?
16:59.49*** join/#asterisk sivana_ (n=sivana@mixdown.ca)
17:00.14*** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net)
17:00.15SpaceBasshowdy
17:00.18moverlibisup TE405P on a Dual Xeon 2.8GHz (32bit SuSE 9.3)
17:00.59movercoppice echo is only on caller side never on called side
17:01.03*** part/#asterisk esi (n=ewaldirc@simonis.xs4all.nl)
17:01.25*** join/#asterisk _blop (n=blop@213-193-176-119.adsl.easynet.be)
17:01.25coppicewhich side is the caller? the GSM or the PSTN?
17:01.40movercaller is IP Side
17:02.01moversip->*->pstn = SIP hear echo
17:02.16coppiceso, the caller hears his voice echoing back from the GSM phone?
17:02.22moversip->*->pstn = SIP hear echo from himself exactly
17:02.52movercoppice no on sip->GSM no echoes on SIP side only on SIP->PSTN
17:03.14movercalls
17:03.17mover:)
17:03.39coppiceit makes no sense to turn off the echo can on the GSM system. I didn't think any ISUP selection would do that. maybe i'm wrong
17:04.02nicoxHEEEEEEEEEEELLLLLLLLLLPPPPPPPPPP
17:04.16movercoppice u dont understand me
17:04.38jontowquestion..
17:04.45coppicemover: you mean the GSM calls are OK. its only SIP<->PSTN that has trouble
17:04.50jontowasterisk seems to be doing ANI, not caller ID on my PRI going to my voicemail system
17:04.52*** join/#asterisk cypromis (n=michael@83.149.70.59)
17:04.58moveri have desperate echos from my owhn voice when i call a pstn phone via a snom over libisup
17:05.07nicoxis there a chance to change a dtmf stream?
17:05.10jontowwhich is all fine and well except im gonna have to disable it, i think..  the public service commission will not be dealing with my insolence there
17:05.25jontowa number with blocked caller ID rings in with the correct number when they leave a voicemail
17:05.32jontow.. and apparently that is VeryBad(tm)
17:05.49jontowhas anyone had to deal with this?
17:06.00sbnelsonnicox: what do you mean?
17:06.13coppicemover: OK. without an echo can you probably will have that problem.
17:06.31movercoppice echocan is on for calls even are bridged
17:07.12coppicemover: it sounds like that is not really the case
17:07.48nicoxi have a e1 with incomming calls, and i will forward a call to another E1, but asterisk do not only dial the number which i say, asterisk dial also the exten as dtmf digits after the correct number
17:07.57movercpatry Echo Cancellation: 128 taps, currently OFF
17:08.20nicoxso. exten => 123,1,dial(zap/g1/456) is dialed as 456123
17:08.31*** part/#asterisk tAURUS (n=_enver_@213.227.207.45)
17:08.33nicoxand 123 are diald as dtmf digits
17:08.53movercoppice what idea you have????
17:09.21*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
17:09.31coppiceyou said ec is on, and the said currently off. which is it?
17:09.58movercurrenty off is because channel is ofhook
17:10.10moverits onhook echocanel is on
17:10.13sbnelsonnicox: what are the exact lines from extensions.conf?  I do the same thing on a T1 and don't have ur troubles.
17:10.20moverwait i paste an onhook channel
17:10.57nicoxdo you have overlapdial=yes in zapata.conf?
17:11.02moverEcho Cancellation: 128 taps, currently ON
17:11.40coppiceit sounds like something is not right in the EC config. maybe when the message says is not what is really happening
17:11.48nicoxbecause i need it to get most of the calls, because our POTS send the digits not in one block
17:12.41moverwait i tell you ec conf
17:12.52sbnelsonnicox: no
17:13.27sbnelsonnicox: I don't think I can help you then.
17:13.33*** join/#asterisk cypromis (n=michael@83.149.70.59)
17:13.41moverechocanel=yes echocanelwhenbr.=yes echotraining=yes thats all
17:14.11moverbut i have tried training=800 and 600 all the same
17:14.59*** part/#asterisk sbnelson (n=sbn@pm3-10stl5.xtraport.net)
17:15.06moveronly mark3 and agreessive supression kill the f.. echo. but calls are totally unnatural and to clean an filtered
17:15.37*** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com)
17:16.02coppiceah, so there is some way to kill the echo. at least that means it is being applied. now to figure out why it is normally ineffective
17:16.24movercoppice i dunno
17:16.30moverall the logs are the same
17:16.59moverall the calls are to the same pstn phone and all tries are echoed
17:17.39*** join/#asterisk jsmith (n=jsmith@64.50.35.114.2O7.net)
17:17.43coppicewhat happens if you disable EC? is it the same, better, worse?
17:17.54moversame
17:18.11moverbut echo a little clearer
17:18.37moverbtw: i use all straight 1-0-9 libpri,zaptel and *
17:18.44Kattyhewwo.
17:18.47coppiceah, better quality echo :-)
17:19.05movercoppice exact
17:19.37moveri have played with rx,txgain no success all the same
17:19.52*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
17:20.02nicoxhas anyone an idea to my problem?
17:20.08charles___Hey, do you guys know any reason to my asterisk be forking in 10 process on start ?
17:20.08moverso what went wrong... §%$§$%@@!###
17:20.38Qwellcharles___: 10 asterisk processes?
17:20.47charles___Qwell: yes, on start
17:20.55charles___Qwell: my other asterisk is solitaire
17:21.04anthmsome linux show threads as processes
17:21.25charles___Qwell:  but this new gets 15 children
17:21.33tzafrir_laptopthe latest from Michael Robertson: http://www.michaelrobertson.com/archive.php?minute_id=186
17:21.50*** join/#asterisk w0w0 (n=w0w0@125.Red-83-46-188.pooles.rima-tde.net)
17:21.57movercoppice u thinking or clueless?
17:21.59charles___anthm: no way man, if it has a PID it's a proccess!
17:22.06tzafrir_laptopSo does an * server that should serve a whole campus and includes a T1 card costs 600$?
17:22.45Qwellkram: evening
17:22.52tzafrir_laptopif they have exactly the same memory sizes: threads of the same process
17:22.58kramhi qwell
17:23.05Darwin35Katty did it I saw here . she used wire cutters and went right to it in the network closet
17:23.15Katty...
17:23.16coppicemover: dunno. kind of strange. there should be some substantial effect when EC is on
17:23.34Kattykram: feeling better today?
17:24.33Darwin35man if this softclient copiles and works we will have a sip softphone worth using
17:24.43Darwin35compiles even
17:24.53anthmcharles___ if you are the expert why are you asking, I can assure you that gentoo for instance has it set so threads have pids and show up as processes but since you seem to know better I gues I have no idea.
17:25.37movercoppice change of entire box havent solve the echoes
17:25.53kramuhm, i don't think anything here has changed since yesterday.  i have a friend visiting this weekend so i'll just try to spend some time away for a bit
17:26.06Kattykram: good plan (=
17:26.07moveron another box with a TE110P singlespan the same echoes
17:26.18coppicemover: that was pretty unlikely to be a cure
17:27.15*** join/#asterisk _PiGreco_ (n=a@adsl-ull-87-69.47-151.net24.it)
17:27.18_PiGreco_hello
17:27.43coppicethreads are like a loving family - they share the same address space :-)
17:27.47_PiGreco_is it possible to REGISTER to a server different from the one in sip:user@server register line in sip.conf ?
17:28.09mishehuanthm: howdy.
17:28.17anthmallo
17:28.46_PiGreco_i mean i am user@server but i want to REGISTER to server2 instead, same username
17:28.54_PiGreco_(i have a kind of dns problem)
17:28.58movercoppice i am despaired ...
17:29.00mountieanthm: Re threads/pids: Linux 2.4 each thread has a PID - in 2.6, they've separated threads/PIDs.
17:29.32mountieanthm: but of course, some distros (like RH,SUSE) have backported parts of 2.6 to 2.4 (making threads share PID)
17:29.43moveri cant reach markku
17:30.09coppicemover: markku might be on vacation
17:30.16charles___anthm: it is pid's man
17:30.19movercoppice i know
17:30.22*** part/#asterisk jluk (n=jluk@82-32-208-99.cable.ubr07.newt.blueyonder.co.uk)
17:30.52anthmyah, your asterisk is rebellious and forks magicly
17:30.59charles___anthm: root@pc01:/etc/asterisk# ps -ef |grep asterisk |wc
17:30.59*** join/#asterisk pbxbart_ (n=pbxbart@p54B02FFE.dip0.t-ipconnect.de)
17:30.59charles___<PROTECTED>
17:31.10bkw_mountie, thats Linux threads vs NPTL
17:31.14moverso nevertheless i need to fix it asap
17:31.21coppiceanthm speak with forked tongue
17:31.25*** part/#asterisk pbxbart_ (n=pbxbart@p54B02FFE.dip0.t-ipconnect.de)
17:31.27charles___anthm: my asterisk is like a bunny, 1 minute and 20 children
17:31.35anthmps -ef |grep asterisk |wc
17:31.35anthm<PROTECTED>
17:31.38anthmme too
17:31.51mountiebkw_: Right.  NPTL doesn't work on 2.4 (exept on some suse/rh kernels with the kernel support backported)
17:31.51anthmcos i have a gentoo box that shows threads as pids
17:31.57tzafrir_laptopanthm, is that 2.4 or 2.6?
17:32.26charles___anthm:  I have 2.6 here
17:32.27*** join/#asterisk [hC] (n=hardcore@8.10.2.5)
17:32.28anthmand on redhat9 there is 1
17:32.32tzafrir_laptopon 2.6 there is /protc/<PID>/task with the threads data . And ps indeed shows just one entry for asterisk
17:32.34charles___anthm: 2.6.12
17:32.38anthmit depends on the distro
17:32.48coppice2.4 or 2.6 make no difference. its the display software that either groups them or doesn't
17:32.53tzafrir_laptopRH backported it from 2.6
17:32.55bkw_root@shinzon [Fri Aug 19 12:32 PM]  ~/opal/samples/simple
17:32.55bkw_<8>:ps aux | grep asterisk | grep -v safe_asterisk | grep -v grep | wc -l
17:32.55bkw_1
17:32.56anthmand the THREADS_HAVE_PIDS or some flag
17:32.59movercoppice so what u were do in my situationÃ?
17:33.03Qwellps -ef | grep asterisk | wc
17:33.03Qwell<PROTECTED>
17:33.10tzafrir_laptopBut then again, 2.6 is becoming more and more common
17:33.11[hC]hey guys.. Any suggestions for what i can do (other than nat=yes and qualify=yes) to keep sip phones that are behind nats from going unreachable every now and then?
17:33.22bkw_linux threads suck
17:33.30bkw_nptl is 400% faster
17:33.52bkw_but can be harder to debug
17:34.07*** join/#asterisk Tili (i=Tili@218.19.67.91)
17:34.13charles___which os uses NPTL ? freebsd ?
17:34.25Qwelllinux 2.6 (and backport 2.4?)
17:34.29bkw_WRONG
17:34.30tzafrir_laptopcharles___, linux 2.6
17:34.51Qwellbkw_: wrong which?
17:34.54bkw_its glibc that needs to be compiled with NPTL
17:34.56bkw_not really the kerenl
17:34.58bkw_er kernel
17:35.09Qwelldunno, thats how gentoo makes it seem
17:35.20bkw_export USE=nptl
17:35.22bkw_emerge glibc
17:35.29Qwellexport?
17:35.35QwellUSE="nptl" emerge glibc
17:35.36Qwellboi
17:35.39bkw_same thing
17:35.40tzafrir_laptopbkw_, but it needs kernel-level support as well
17:35.46Qwelleasier :p
17:35.48anthmthere is no way for asterisk to accidently fork so you can be positive you are seeing threads that is the jist
17:35.53tzafrir_laptopAs I have just showed, /proc looks different
17:36.16anthmthere are only like 3 places that fork in the whole dist of asterisk
17:36.18bkw_tzafrir_laptop, I don't think you enable anythign special in the kernel ata ll
17:36.20bkw_for NPTL
17:36.26anthm1 to go in the bg for the console
17:36.35anthm1 in the mpg123 style moh
17:36.44tzafrir_laptopbkw_, does it work with a vanilla 2.4? 2.2?
17:36.52bkw_2.6 kernels yes
17:36.56bkw_lower kernels NO
17:37.02tzafrir_laptop2.6 already have it
17:37.02anthmand a few more here and there in some apps
17:37.14bkw_I don't even consider NPTL for 2.4 and lower
17:37.27anthmagi of corse
17:37.41*** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca)
17:37.47Derkommissarbkw_, im having a problem with res_odbc i dont know if its known.... afther a couple of hours when i reload... it doesnt slurp the config..
17:37.59bkw_Derkommissar, what country are you in?
17:38.11DerkommissarUSA at this momen :-)
17:38.19bkw_but where are you from normally
17:38.31DerkommissarIm Cuban, but i work in ecuador
17:38.38bkw_I talked to you on the phone
17:38.40nicoxis there anyone who have a good heart?
17:38.45DerkommissarYes you did
17:38.50Derkommissarim peters friend
17:38.56Derkommissarwe talked before though
17:39.00bkw_yes I put two and two together.. i'm quick like that ;)
17:39.00Qwellnicox: I will when I get my transplant
17:39.03Derkommissaryou consulted for me before too
17:39.10Derkommissar:-)
17:39.12DerkommissarLOl
17:39.35nicoxplease help me... i need help with overlapdial and dtmf digits
17:39.38bkw_Derkommissar, ok someone has been dinking with the config stuffs
17:39.59bkw_let me look
17:40.02Derkommissar:-/ what do you mean
17:40.10Derkommissari cant duplicate it always
17:40.12Derkommissarbut sure
17:40.26bkw_let me look
17:40.28bkw_hrm
17:40.34*** join/#asterisk sigmounte (n=sigmount@85.201.48.109)
17:41.00Qwellugh, don't msg me
17:41.04*** part/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com)
17:41.15shido6?
17:41.21coppicei can't always duplicate, but I so love to practice ;-)
17:42.11charles___anthm: thanks man, I will look at ntpl
17:42.12Qwellcoppice: I think you're thinking of another word. :p
17:42.15nicox<PROTECTED>
17:42.59Qwellcoppice: propagate maybe
17:42.59Derkommissarnicox, i like falco, but i dont speak the lenguage
17:43.01coppiceQwell: dunno. the successful duplication efforts produces reasonable replicas
17:43.12moverDerkommissar hehehe
17:45.24Beirdohow about Vienna Calling?
17:45.36charles___Do you guys know if I NOLOAD all the unnecessary modules will make any performance effect ?
17:45.38BeirdoFalco++
17:46.53blitzragecharles___: it could a bit. I prefer to use the autoload=no option and specifically load the modules I need. I made a file to do it - its on the wiki at http://www.voip-info.org/wiki-Asterisk+Slimming
17:47.54charles___blitzrage:  thanks
17:48.02moverwelche asterisk version ist das?
17:48.16*** part/#asterisk Juxt (n=Juxt@64.135.20.202)
17:49.17kyooI have Asterisk running through a sipura spa-2000 ... can I put people on hold, transfer, etc using an analog phone through this adapter?
17:49.24shido6yes
17:49.25shido6you can
17:49.30Qwellcall parking, and # transfer
17:49.31moverhmm ich würede mal versuchen 123,1,trnsfer(234)
17:50.05kyooshido6: Qwell: Where would I find docs for that?
17:50.10Qwellwiki
17:50.16shido6wiki
17:50.31nicoxtransfer, is vielleicht ne idee, mahc ich gleich mal
17:50.36shido6or or hs1wkwtftad
17:50.43Qwellshido6: eh?
17:50.50bjohnsonsmartass
17:51.02bjohnsonheineken
17:51.03shido6hire some 1 who knows what the fsck they are doing
17:51.14shido6wyw
17:51.16bjohnsonstrudel
17:51.18shido6while you watch :)
17:51.22bjohnsonfraulein
17:51.28bjohnsoncombine and enjoy
17:51.29Qwellal (and learn)
17:51.36shido6wsc
17:51.39shido6with screen capture
17:51.44charles___is there a way to have multiple bindaddr ?
17:51.56shido60.0.0.0 for all
17:52.13shido6if u have 1 out of 9 devces u dont wanna lsiten on
17:52.19shido6tis ok to listen
17:52.36Qwelljust do 0.0.0.0 and setup iptables
17:52.40charles___shido6:  wanted to listen on 2 only
17:52.40shido6yep
17:52.42*** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2)
17:52.51shido6what Qwell said
17:53.00charles___shido6: is 1 or all the possibilities?
17:53.04Qwell--dport 5060 --d 192.168...
17:53.18charles___Qwell: yes
17:53.26charles___-j DROPP
17:54.41charles___bindaddr=IP, IP;  doesn't work with space also doesn't. I'm going to assume 1 or ALL
17:54.44charles___is it?
17:55.05*** join/#asterisk coppice_ (n=chatzill@226.193.17.210.dyn.pacific.net.hk)
17:57.11Hmmhesaysfile oh file where art thou
17:57.34file[laptop]uh oh
17:57.50*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
17:57.51file[laptop]what do you want? :P
17:58.20rkingheh, me?
17:58.31Hmmhesaysbeing an SER n00b myself, can you tell me if there is a way for ser to tell you why the config file is bad? i'm just getting ERROR: bond config file (2 errors)
17:58.35Hmmhesays*bad even
17:58.42file[laptop]oh, yes ;)
17:58.51file[laptop]ser -c
17:58.51blitzrageasteriskdocs.org totally restored!
17:59.13Hmmhesayswonderful, thanks file[laptop]
17:59.17file[laptop]yw
17:59.46blitzragehttp://ask.slashdot.org/askslashdot/05/08/19/1428233.shtml?tid=215&tid=222&tid=4
18:00.15rkingi have asterisk running, and ohphone handy, and i'd like to get either in a state where i can present them to my company as an alternative to skype, but as far as i know i'd need another person on the other end.  actually, is there an equivalent to skype's echo123 service?  a bot running H323 that will echo sounds back?
18:01.06bjohnsondo you have h323 compiled for asterisk?
18:01.19bjohnsonthat will likely be more of a problem
18:01.44bjohnsonthere are examples in the demo of playing back a file, time and date, etc
18:02.09bjohnsonwhy use h323 if you don't have to .. use a iax or sip softphone instead
18:02.14rkingbjohnson: hrm, i was told "asterisk is voip, with pots if you have the hardware"
18:02.26bjohnsonyes that is true
18:02.46bjohnsonbut the h323 channel driver isn't included with it, just the sip and iax ones
18:02.53rkingbjohnson: i completely apologize for my total ignorance of h323 vs iax vs sip.  i'm in a greenfield area of exploration on all this.  prior to 3 months ago i didn't even own a headset. =)
18:02.56bjohnsonyou have to compile h323 separately
18:03.08bjohnsonit has lots of dependancies and is difficult to compile correctly
18:03.15rkingok, cool
18:03.23rkingso linphone?
18:03.37bjohnsoneasier to use sip or iax unless you already have a h323 system you have to tie intop
18:03.40*** join/#asterisk coldfeet (n=cold@dsl-80-46-109-145.access.as9105.com)
18:03.43bjohnsonlinphone is a sip one
18:03.51bjohnsoniaxcomm is a iax one
18:03.54bjohnsonthere are other
18:03.57bjohnsonthere are others too
18:04.32rkingbjohnson: we do have to support windows and mac users, so maybe sip is the choice?
18:04.51Qwelliaxcomm is cross platform
18:05.16tzafrir_laptoprking, in fact, there are about 4 separate h323 channels right now
18:05.37tzafrir_laptopsip is a protocol, just like IAX and h323.
18:05.48tzafrir_laptopAnd just as "cross-platform"
18:06.25rkingtzafrir_laptop: right... but my [very cursory] search for info to compare iax vs sip seems that sip is more established, and i'm assuming would have better windows/mac support (?)
18:07.00tzafrir_laptoprking, are you sure you want a soft phone?
18:07.33rkingtzafrir: we use skype all the time, and it funamentally works great, but skype crashes a lot, and has a 5-person conference limit, and is not open
18:07.43tzafrir_laptoprking, anyway, there are nice phones for both protocols for windows and for mac "as well"
18:08.05Ethonrking: You can plug an asterisk between your trunk and your normal pbx to handle voip with your normal phones
18:08.06rkingtzafrir: so you recommend iax?
18:08.55tzafrir_laptoprking, iax would do a much better job if you want to traverse NAT.
18:08.57rkingEthon: yeah, that's totally a goal for the future stuff - but right now we'd be happy to have 100% voip going on.  we use freeconference.com for when people have to be POTS'd in, currently
18:09.12tzafrir_laptopIf it's inside the same LAN there isn't that much a difference
18:09.21rkingtzafrir: awesome, yes - that would be a constant problem.  i was figuring siproxy would be in the mix
18:10.05Ethonrking: Was just a solution we made for many of our customers which did not replace their old telephone hardware and use voip
18:10.17charles___Cpu(s): 67.9% us,  3.6% sy,  0.0% ni, 24.6% id,  0.0% wa,  0.0% hi,  3.9% si
18:10.28charles___hehe 20 channels over G723 on a 2.4Ghz
18:11.37rkingi am so grateful for your help.  i do have one area where i am unclear, but feel free to ignore me at this point:
18:12.37heison~seen sivana
18:12.40jbotsivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 20h 3m 7s ago, saying: 'ya'.
18:12.45rkingi'd like to test iaxcomm in isolation for call-out, then i'd like to use iaxcomm+asterisk to make a conference, and then i'd like to show it to someone i work with.  is there anyone here i can call out to for just ~30 seconds, and then i'll start fiddling with the asterisk side on my own after that.
18:13.26rkingbjohnson mentioned the demo's for playing back files, but i'd first like to eliminate iaxclient variables if possible
18:13.42charles___Qwell: hey man do you really have 720 threads ?
18:13.46Qwellcharles___: no
18:13.51charles___Qwell:  hehehe
18:14.10charles___Qwell: the machine here is at 70% cpu and it's pretty stable
18:14.59rkingis there an online iaxbot somewhere?
18:15.09Qwellrking: to do what?
18:15.10Ethonrking: Sorry, I'm at home.. my internet connection is not fast enough for a demonstration
18:15.15charles___maybe with NPTL I can compress 1 T1
18:15.25*** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com)
18:15.48rkingQwell: so i can use iaxclient to call it and verify that everything is working
18:15.59Qwellrking: guest@24.50.66.194/s
18:16.25Qwellit'll be lagged, but sound is sound
18:17.04rkingright... now, if i could figure out iaxcomm... 1 minute
18:17.18Qwellrking: in the dial box, just put what I said
18:18.07rkingawesome!
18:18.12rkingmusic to my ears.
18:19.25drumkillawtf is an iaxbot :)
18:19.32Qwell~iaxbot
18:19.46rkingdrumkilla: guest@24.50.66.194/s is definitely what i'd call an iaxbot.
18:19.53drumkillaha
18:20.44rkingnow to config asterisk to host a conference call.
18:20.49drumkillaI have heard Qwell's shinanigan's before
18:21.11kyooWhat would I search for specifically in the wiki to learn about call control from an adapted analog phone?
18:21.53*** join/#asterisk oej (n=oej@apollo.webway.se)
18:23.30PupenoLHello.
18:24.11bjohnsonkyoo: you have a supira?  search for sipura
18:26.06jarrodhey anyone know of an agi or the likes that can accept a key sequence and set a flag that would notify the extension to forward a call instead of ring the device?
18:26.24*** join/#asterisk mag_um (i=cvx@219.95.202.108)
18:26.37shido6can I dd over a partition without formatting it?
18:26.58Qwellshido6: if you dd from one partition to another, a format isn't needed.  it copies the FS
18:27.15Qwellsame with disk to disk, but replace fs with partition table (and fs..)
18:27.37Qwellor, do you mean, format it before dd?
18:28.52*** join/#asterisk zoo (i=nobody@ip-54-16.travedsl.de)
18:29.31brad_msswi've got 2 TDM400Ps, one with 4 FXO's one with 4 FXS's ... everything seems to mostly work fine, but it seems that sometimes the phone system misreads the extension numbers entered (especially if someone calls from a cell phone) ... for instance, someone typing 114, might be transferred to 111 instead ... I've verified via the asterisk logs that it says it got 111 ... are there any tweaking settings for these?
18:29.49JerJershido6:  dd is simply a low level copy
18:30.01JerJerit doesn't care, just just does what you tell it to do
18:30.07brad_msswi've also verified 114 was actually entered as per the history in the cell phone
18:30.50ManxPowerbrad_mssw: set relaxdtmf=no and play around with your txgain and txgain for the FXO ports.  too loud or too soft would cause these problems
18:31.05ManxPower..er..txgain and rxgain
18:31.07brad_msswManxPower: thanks, I'll try it
18:31.26greg_workbrad_mssw: ztmonitor can help you set gain
18:31.31*** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
18:31.45*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
18:31.56netnameusHi... my box is hanging during bootup on "loading zaptel hardware modules"... how can I get past this point?
18:33.35*** join/#asterisk brookshire (n=matt@c-67-190-190-32.hsd1.co.comcast.net)
18:35.02bkw_netnameus, boot up and remove the zaptel init
18:35.13bkw_could be any number of things
18:36.29netnameushow do i bootup and remove the zaptel init?
18:36.35netnameus(i'm new to linux)
18:36.42bkw_take the card out
18:36.51netnameusoh, i dont have a car
18:36.52netnameuscard
18:36.59brad_msswManxPower: is relaxdtmf default to on or off ?
18:37.01bkw_then at the boot prompt do linux single
18:37.32netnameusi don't have a zaptel card
18:38.03ManxPowerbrad_mssw: no idea.
18:39.45*** join/#asterisk r0d3nt (n=RatMan@wsip-24-234-241-84.lv.lv.cox.net)
18:39.52brad_msswManxPower: default appears to be 'no' ... perhaps I should turn it on instead ?
18:40.42JerJeryou won't know until you try it
18:40.45JerJertest
18:40.48JerJertest again
18:40.55JerJertest more
18:41.09brad_msswyeah, that's the hard part ... it only happens sometimes :/
18:41.11ManxPowerbrad_mssw: no, don't do that
18:41.18ManxPowerbrad_mssw: you have a volume problem then
18:41.28brad_msswok, I'll bark up that tree then, thanks
18:44.18Kattyhmm.
18:44.59*** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com)
18:45.58netnameushow can i stop it from trying to load the "zaptel hardware moduels"?
18:46.13Qwellnetnameus: <bkw_> then at the boot prompt do linux single
18:46.39bkw_Qwell, thanks for taking care of my light work
18:46.42blitzrageanyone know what a .snp file is?
18:46.49Qwellbkw_: I'll bill you later
18:46.53bkw_thanks
18:46.58bkw_;)
18:47.08jarrodanyone know of a good archive of useful agi scripts?
18:47.18Kattyanyone want to sell me 60 phones?
18:47.32netnameus"Error 27: Unrecognized command"
18:47.34netnameusis what i get
18:47.42blitzrageKatty: sure, do I have to actually send you phones, or can I just take the money and run? :)
18:47.53Kattyblitzrage: ;P
18:48.13jaikewhere do u guys buy ur equip? the only site i know is voipsupply.com
18:48.16zoai will sell you 60 freeware phones :)
18:48.19zoahey ho all
18:48.31Qwelljaike: there is also that was...umm...starts with an
18:48.31Qwellx
18:48.36blitzrageKatty: :D
18:48.43puowvipKatty, Western Electric model 500?
18:48.53*** join/#asterisk Defraz (n=t0tal@24-119-12-238.cpe.cableone.net)
18:48.57Katty60 phones, cheaper than 54 bucks a piece.
18:49.01Kattythose are the stipulations.
18:49.03netnameusi get "Error 27: Unrecognized command"
18:49.07QwellKatty: any type of phone?
18:49.09Kattyand voIP, obviously
18:49.14QwellI've got these awesome fisher price phones...
18:49.17Qwellwireless
18:49.18KattyQwell: hardware, not software (=
18:49.29Katty;>
18:49.36zoaworking phones ?
18:49.37ManxPowerKatty: if those are the requirements then tell them that you cannot provide a system at that price and find another vendor.
18:49.45KattyManxPower: shoo (=
18:49.49ManxPowerYou don't want to work on a project where they are so cheap.
18:49.52KattyManxPower: you obviously don't see my sarcasm.
18:50.02zoai dont think you can do it for that price
18:50.02*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
18:50.05ManxPowerKatty: I'm 1/2 out of it today.  Sick.
18:50.08jaikewe might be buying couple of hundred soon...still have to decide between polycoms and snoms
18:50.19zoasnom snom snom
18:50.23Kattysnom :<
18:50.25Kattypolycom :>
18:50.25ManxPowerpolycom polycom polycom
18:50.27jarrodi like polycoms
18:50.29Qwellcisco
18:50.30puowvipWestern Electric model 500.
18:50.35Ariel_polycom polycom
18:50.40Qwelljaike: http://www.voip-info.org/wiki-VOIP+sites  Use at your own risk
18:50.45jaikepolycom 5, snoms 1
18:50.57Kattywe have 13 polycoms and 1 snom
18:51.08brad_msswwhat should I dial while using ztmonitor to test the gain, etc?
18:51.21MikeJ[Laptop]ALASKA!
18:51.22zoaoh whatever you do, stay away from thomson
18:51.27ManxPowerwe have 60 polycoms
18:51.39jaikeyah..were leaning on getting polycoms
18:51.41zoawe were interested ins several thousand thomsons, and i ask for latest firmware
18:51.44KattyMikeJ[Laptop]: no, mexico.
18:51.53zoaand the reply is: you cannot do software upgrades on that thing
18:51.54zoaWTF ?
18:52.03zoahow come they have UK firmware in the uk
18:52.06Qwelljaike: ahh, voxilla, thats the one I was thinking of
18:52.07ManxPowerzoa: they are not owned by Digium, are you?
18:52.10zoaand they have a web interface for that
18:52.11ManxPower..., are they?
18:52.17zoanopez
18:52.36jaikecool..thanks
18:52.50zoai gave a negative advise for that thing, based oa on that reply
18:53.08*** join/#asterisk jorgito (n=jorgito@snat2.arachne.czfree.net)
18:53.09jorgitohi
18:53.40*** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
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18:53.58*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
18:53.58*** mode/#asterisk [+o drumkilla] by ChanServ
18:55.27*** join/#asterisk Grumbly (n=Mantacul@209.151.244.129)
18:55.43Grumblyhello.  I was wondering if I could use asterisk as a replacement for panalog
18:55.49Grumblyif anyone knows what that is
18:55.55JerJerdefine panalog
18:56.25GrumblyPanalog is a Call Management system / pbx logger.  Works with panasonic pbx stuff
18:56.40JerJerlike a serial port thing?
18:56.40jaikelathos: thought u were having problems with ur ip 501 :P
18:56.47Grumblyum... lemme look...
18:56.58lathos42jaike: Its a love hate relationship that I have with it
18:57.18Grumblyindeed... like a serial port thing
18:57.32Grumblyheh... I never looked at the box it was running on...
18:58.04Grumblyit's got 2 serial port devices.
18:58.19JerJerso what do you want to replace?
18:58.23JerJeror perhaps how?
18:58.55zoayou could do it
18:58.56GrumblyI'd Ideally love to replace this P.O.S. software they call a call managment suite...
18:59.00zoabut it will take time
18:59.07zoabut its not asterisk that will do it
18:59.15zoayou need perl
18:59.16zoathats all
18:59.28GrumblyWe're not a voip system though
18:59.40JerJerasterisk can do analog telephones
18:59.45zoabut for the cost of making such a thing, you might as well replace everything with asterisk
18:59.51JerJerit all depends on exactly what you are looking to replace
19:00.08*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
19:00.15zoao unless its a 100000 phones setup, i'd replace it with asterisk
19:00.22zoa*so
19:00.23jaikeanyone using POE on the polycoms? nice feature
19:00.35ManxPoweryes
19:00.47zoaasterisk on slashdot again
19:00.51zoaits almost every week now
19:00.55jaikePOE switches seem expensive though
19:00.56GrumblyWell JerJer, I'd like to replace the system that logs the inbound and outbound call times, numbers, etc. with a reliable one.  It also should be able to automatically route calls based upon time of day
19:01.16zoayou need only perl
19:01.17jorgitolisten
19:01.19zoano asterisk
19:01.20jorgitolisten
19:01.32jorgitodid anybody tried to compile asterisk for some embeded system.?
19:01.40zoai did something like that before, its not fun to program
19:01.41jorgitoGrumbly, ha ha ha
19:01.55jorgitozoa, this was to me?
19:01.58QwellWhy would you want asterisk on an embedded system?
19:01.59zoajorgito, listen, listen
19:02.01zoalisten
19:02.02zoalisten
19:02.03zoalisten
19:02.10MikeJ[Laptop]Grumbly, you are talkking about a serial port logger, then you are wanting one that controls calls..
19:02.14MikeJ[Laptop]we are all missing somthing
19:02.29zoahe wants a 2 way communication over serial port
19:02.47jorgitozoa, i tried to compile it against uclibc and not not working.....
19:02.48GrumblyMike9:  the suite we've got logs calls and controlls the routing
19:02.49zoabrrr
19:02.57*** join/#asterisk greenok (n=greenr21@port0006-abm-adsl.cwjamaica.com)
19:02.58lathos42jaike: We're going to go with PoE when we roll out in production
19:02.58zoajorgito, i know people did it before
19:03.00jorgitozoa, what library exactly does asterisk needs?
19:03.01Grumblybased on time of day... though I may be mistaken about that
19:03.07zoajorgito: dunno
19:03.08jorgitozoa, tell me one pleas. .
19:03.09GrumblyIt might be a term into the pbx
19:03.16lathos42jaike: If I can get this problem with the spa-3000 fixed that is =)
19:03.16Grumblylemme ask
19:03.26zoagoogle for astlinux
19:03.46jorgitozoa to me?
19:04.01zoayes
19:04.25*** part/#asterisk greenok (n=greenr21@port0006-abm-adsl.cwjamaica.com)
19:04.27lathos42Has anyone successfully called out through the PSTN with an SPA-3000, calling from a Polycom phone?
19:04.43zoahow do you do that ?
19:04.48zoalathos ?
19:04.58zoapolycom to spa3000 to pstn ???
19:05.09jorgitowhat is spa-3000 ?
19:05.10Qwellspa3000 is an fxo, no?
19:05.18zoaah it might have fxo yes
19:05.29zoaand fxs
19:05.36lathos42Yeah, its both
19:05.37zoabut how do you connect a polycom to that ?
19:05.42zoapolycom is voip
19:05.51Qwellzoa: so is the spa3000
19:05.55zoayes i know
19:05.59*** join/#asterisk darkskiez (n=darkskie@host86-133-151-115.range86-133.btcentralplus.com)
19:06.01Qwellthrough asterisk, of course
19:06.05*** join/#asterisk sigmounte (n=sigmount@85.201.48.109)
19:06.12zoaalthough it might be possible
19:06.12lathos42SPA3000 --SIP--> * <--SIP-- Polycom
19:06.16zoaaaah
19:06.19zoathat has no pstn
19:06.19Qwellsee? :p
19:06.24zoaaaaaah
19:06.28Qwellits more like
19:06.35zoayou mean phone - fxs on spa3000
19:06.38zoayes i get it now
19:06.40brad_msswwhat's the best way to set the gains using ztmonitor ?
19:06.42Qwellpolycom(sip) > * > spa(fxo-pstn)
19:06.42zoathat should work just fine
19:06.53Qwellpolycom(sip) > * > (sip)spa(fxo-pstn)
19:06.54Qwellratehr
19:06.56brad_msswand how do you know when it's right, or close to right?
19:07.04zoabrad_mssw: trial and error
19:07.07zoalisten to it
19:07.20brad_msswzoa: from what i've read, it says not to trust your ears
19:07.28Qwellwell...
19:07.29brad_msswzoa: supposed to do some quantitative thing
19:07.34zoaits hard
19:07.36brad_msswdocs aren't that great though
19:07.37Qwellif there is echo, and it thinks its done
19:07.41zoaas one phone might be loader than something else
19:07.43QwellI'm gonna say "uhh...no.  There is still echo"
19:07.44zoalouder
19:07.52brad_msswQwell: echo is not the issue
19:07.59Qwells/echo/volume/ then
19:08.04brad_msswQwell: it's misreading the tones
19:08.10brad_msswQwell: sometimes ....
19:08.14zoaaaah
19:08.17lathos42The crazy thing is that I can call out just fine when I use my Sipura SPA-841, or the SPA-3000's fxs port.. but for some reason when I call with the Polycom it can send but not receive audio..  but if I put the call on hold, it suddenly starts working when the call comes off hold
19:08.21brad_msswlike it'll mistake 114 for 111
19:08.24zoaare you using the dtmf relax ?
19:08.29brad_msswno
19:08.30jorgitoso nobody here compiled the asterisk to embeded system?
19:08.41zoajj has it on wrt
19:08.43Qwelljorgito: What purpose would it serve, exactly?
19:09.10zoalets all pm jj
19:09.40darkskiezso, i'm behind a nat, and I can sign on with a softphone+stun to sipgate, but not with asterisk. It connects and I can make and receive calls, but the RTP doesnt come back. How should I proceed? Can I use SER somehow to do some weird STUN thing?
19:10.04*** join/#asterisk Grumbly (n=Mantacul@209.151.244.129)
19:10.07Grumblyeek
19:10.20lathos42I've got * running on my WRT54G at home
19:10.51lathos42It makes a nice 1 phone SIP to IAX gateway
19:11.27Qwellno transcoding, heh
19:11.32Qwelland it can't route at the same time
19:11.36Qwellbut hey, it works, right?
19:11.45*** join/#asterisk tessier (n=treed@se0-0.ar1.tma1.loudpacket.net)
19:11.58jorgitoQwell, well asterisk for embeded system.. exactly i am working in ISP and i would like to port asterisk to our routers, running with uclibc library
19:12.12jorgitoQwell, so crosscompiling the asterisk..
19:12.36tessierjorgito: Easier to port your routing to a standard asterisk platform.
19:13.15jorgitotessier, maybe
19:13.22tzafrir_laptopjorgito, on what CPU?
19:13.42jorgitotzafrir_laptop, normal i386 family
19:13.54tzafrir_laptopjorgito, if you want to use voip over the internet, you will need a decent CPU
19:14.15Qwellcombining routing and asterisk on one machine is kinda silly
19:14.26lathos42Qwell: I didnt try to transcode, but I was still able to access the internet from my iBook using WPA just fine :)
19:14.27blitzragehey, I'm running to the store for some food, you guys want anything?
19:14.27jorgitotzafrir_laptop, realy? i dont thing so
19:14.37jorgitoQwell, why not?
19:14.43Qwelllathos42: I was being facetious
19:14.50ManxPowerblitzrage: How about a bottle of high strength antibiotic?
19:15.00*** join/#asterisk secure75 (n=mic@gfwlan.cablesurf.de)
19:15.03Qwelljorgito: because it shouldn't work that way
19:15.22Qwelljorgito: routers route, PBXs ..PBX
19:15.26lathos42Qwell: I'd be interested to see which had a higher CPU load during the call.. my WRT54g, or the Polycom phone
19:15.33QwellYes, I DID just use PBX as a verb.
19:15.37zoawrt
19:15.37jorgitoQwell, i dont see a reasont why it should do that...
19:15.40zoaprobably
19:15.48jorgitohmmm
19:15.51jorgitoi will try..
19:15.57QwellManxPower: mind explaining this better?
19:16.09ManxPowerQwell: not today
19:16.33Qwelljorgito: if I may use an analogy..
19:16.54lathos42Qwell: You should submit PBX as a verb to one of the Open Dictionaries on the web
19:17.09Qwelljorgito: Do you have any TVs that toast bread?  How about a microwave that can download the latest sport scores?
19:17.17mishehubah.
19:17.26jorgitoQwell, pssst
19:18.01Qwellor maybe a cellphone that can make you a 7 course meal
19:18.25Qwella car that also does dishes?
19:18.35Qwella washing machine that is also a web server?
19:18.43QwellI can come up with these all day
19:18.48RaYmAn-Bxthat would rock though
19:18.53jorgitoQwell, hmmm . you blind?
19:18.55QwellRaYmAn-Bx: it might, but its useless
19:19.05Qwelljorgito: routing and calling are as different as day and orange
19:19.14Qwells
19:19.36jorgitoQwell, well linux machine with asterisk must route also ...
19:19.38jorgitoanyway
19:19.56Qwelllet me guess, it'll also have a GUI?
19:20.08heison~seen sivana
19:20.09jbotsivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 21h 10m 36s ago, saying: 'ya'.
19:21.07jorgitoQwell, or the asterisk box doesnt route?
19:21.12*** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar)
19:21.24Qwelljorgito: nope, it doesn't need to
19:22.21jorgitoQwell, what ?
19:22.26tzafrir_laptopjorgito, what prevents a linus box from routing?
19:22.35QwellWhy would the asterisk box need to be a router also?
19:22.43tzafrir_laptopjorgito, did you try building asterisk with uclibc?
19:22.53jorgitotzafrir_laptop, yes
19:22.58tzafrir_laptopand?
19:23.08jorgitotzafrir_laptop, did you see Makefile ? horrible
19:23.29tzafrir_laptopI'm willing to help thee
19:23.51tessierAnyone remember the single-fxs card that Digium used to sell with their developer kit? That is not a TDM400P card right? I think they had something different before. But still used the little daughterboards.
19:23.51jorgitorealy?
19:24.29blitzragetessier: nope - daughter boards only on TDM400P - previous card was a modem (X101P)
19:24.30tzafrir_laptopjorgito, I want to sanitize it
19:24.35blitzragetessier: and in the IAXy.
19:24.40jorgitotzafrir_laptop, what is sanitize?
19:24.49tessierblitzrage: hmm....
19:24.51tzafrir_laptopmake it saner
19:25.20tzafrir_laptopjorgito, but do you have a specific error?
19:25.43jorgitotzafrir_laptop, wait i will modify Makefile and then i will try to compile..
19:25.47tzafrir_laptopdid you try building asterisk in a uclibc build chroot?
19:26.04tzafrir_laptopwhy do you need to modify the makefile?
19:26.26jorgitotzafrir_laptop, to change includes and cc and pthreads etc...
19:27.42tzafrir_laptopjorgito, why not build it in a uclibc chroot?
19:27.44*** join/#asterisk Cybertank (n=cybertan@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com)
19:28.20jorgitotzafrir_laptop, you dont need chroot..
19:28.25*** join/#asterisk RoyK (n=roy@ti211210a080-0158.bb.online.no)
19:28.40jorgitotzafrir_laptop, just changing the cc, includes, librarys is enaugh
19:28.50tzafrir_laptopjorgito, but it will make things simpler. It means no cross-compilation settings
19:29.15tzafrir_laptopAlso: all the libs you build against are built with glibc, right?
19:29.27jorgitoyes , but you know why make it simple if you can make it hard..
19:30.03*** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
19:30.13jorgitotzafrir_laptop, no , the libc are compiled against uclibc
19:30.16*** part/#asterisk jorgito (n=jorgito@snat2.arachne.czfree.net)
19:30.22*** join/#asterisk jorge_bla (n=jorge@snat2.arachne.czfree.net)
19:30.24jorge_blahi
19:30.34jorge_blait s me jorgito
19:30.44jorge_bla/home/LRP/buildroot/build_i386/staging_dir/bin/i386-linux-uclibc-gcc -g  -o asterisk -Wl,-E  io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o  editline/libe
19:30.44jorge_bladit.a db1-ast/libdb1.a stdtime/libtime.a -L/home/LRP/buildroot/build_i386/staging_dir/lib -ldl -lpthread -lncurses -lm -lresolv   -lssl
19:30.44jorge_bladns.o(.text+0x37d): In function `ast_search_dns':
19:30.44jorge_bla/home/LRP/asterisk-1.0.9/dns.c:174: undefined reference to `__res_ninit'
19:30.46jorge_bladns.o(.text+0x3b2):/home/LRP/asterisk-1.0.9/dns.c:175: undefined reference to `__res_nsearch'
19:30.48jorge_bladns.o(.text+0x499):/home/LRP/asterisk-1.0.9/dns.c:194: undefined reference to `__res_nclose'
19:30.50jorge_blacollect2: ld returned 1 exit status
19:30.51tzafrir_laptop~pb
19:30.51jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
19:30.52jorge_blamake: *** [asterisk] Error 1
19:31.02jorge_blasorry
19:31.49jorge_blaseems like i didnt defined linker..
19:32.35jorge_blahttp://pastebin.ca/20445
19:33.18moverwhere i can se a list what free codecs will supported by asterisk (possibly with a download site)
19:33.37bkw_mover, the ones listed when you do show translations
19:33.43*** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar)
19:33.45bkw_with the exception of g723.1
19:33.48jorge_blatzafrir_laptop, do you see?
19:33.59*** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net)
19:34.54tzafrir_laptopjorge_bla, are openssl and ncurses linked with uclibc? Though this is not the error here
19:35.07jorge_blayes
19:35.14moverbkw_ thanks
19:35.30jorge_blaroot@lrp:/home/LRP/buildroot/build_i386/staging_dir/lib# ls | grep resol
19:35.30jorge_blalibresolv-0.9.27.so
19:35.36clyrradHi, can someone please clairfy for me when I put configuration changes into sip.conf and iax.conf?  From the samples I have found online they seem to contain the same sorts of items, and a few posts said you can make your configurations in either of the 2 files.  Was wondering if someone can please clarify that for me.
19:35.58Nuggethttp://lnk.nu/westpress.co.uk/3lq.jsp - cessna flies for two hours with half a wing missing
19:36.13Nugget"But despite two of the three passengers being top flight engineers on their way to fix a Boeing 767, no one noticed that half the left wing, containing one fuel tank, was missing."
19:36.17Nugget(with pic)
19:36.44*** join/#asterisk iCEBrkr_ (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
19:37.04*** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca)
19:37.12moveranyone here get ibeam from xten work with h26x through *?
19:37.26moveris this possible?
19:37.42bkw_mover, the best way to know is try it
19:37.53bkw_there are only a few ways things work or don't work
19:37.59moveri cant i dont have ibeam :-(
19:38.01bkw_just start trying stuff till it works or doesn't work
19:38.01Ariel_Nugget, seems that he needs some glasses
19:38.23bkw_mover, thats how I solve problems.. just jump in and try shit till I get debug that proves it won't or I get it working
19:38.47NuggetI have eyebeam, but I've never tried h26x.
19:38.50moverif i had it i still have tried
19:38.58tzafrir_laptopjorge_bla, can't think of anything solid right now
19:39.21jorge_blatzafrir_home, me either
19:39.38jorge_bladoes anybody know somebody who compiled asterisk against uclibc
19:39.57moveri think its the proto that Ciscos Vidphone use or i am wrong?
19:40.17*** join/#asterisk pbd (n=pbdavids@12.144.118.36)
19:41.32*** join/#asterisk Lan16`spdy^gTp (n=spdy@lns-th2-15-poi-82-64-230-87.adsl.proxad.net)
19:42.06jorge_blatzafrir_laptop, seems that lots of people have the saim problem          http://lists.digium.com/pipermail/asterisk-dev/2004-December/007831.html
19:42.28*** part/#asterisk Lan16`spdy^gTp (n=spdy@lns-th2-15-poi-82-64-230-87.adsl.proxad.net)
19:42.41bkw_jorge_bla, thats old
19:42.50bkw_coming up on a year almost
19:43.30jorge_blabkw_, but still the problem...
19:43.33docelm0_Anyone know what voltage the TDM400 cards are?
19:43.36docelm0_3.3v?
19:43.52jorge_blai  think lots of people doesnt compile asterisk against uclibc
19:44.10darkskiezdocelm: dual
19:44.22docelm0_shweet..   :)   thanks!
19:44.32Grumblyok...
19:44.55Grumblycan asterisk be used solely to LOG call data?
19:45.05Grumblyfrom a serial device attached to a pbx?
19:45.10jsmithGrumbly: Sure, if you set it up that way...
19:45.19*** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net)
19:45.20GrumblyOK
19:45.21bkw_Grumbly, um search for cdr_serial.c
19:45.22Grumblysweet
19:45.42bkw_http://www.dynx.net/ASTERISK/diff-patches/cdr_serial.c
19:45.44bkw_I wrote that
19:45.45bkw_as a test
19:45.46jsmithOr just pass the calls from one port to another, and log along the way...
19:45.52Grumblycan asterisk be used to manage a pbx?
19:45.59bkw_asterisk is a PBX
19:46.02Grumblyi kno
19:46.18Grumblybut we've got a pbx already, we just need a log reader
19:46.28Nuggetasterisk is not what you want, then.
19:46.38jsmithGrumbly: As a log *reader*?  No....
19:46.56Grumblynot reader... sorry... I'm doin several things and having several conversations
19:46.57Grumblyheh
19:47.10jsmithGrumbly: As a lot writer, then yes...
19:47.13jsmiths/lot/log/
19:47.20*** join/#asterisk Defraz (n=t0tal@tim.ibccom.net)
19:47.42*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
19:48.45GrumblyI'm looking to replace a call data analyser with the specific function of using serial ports to a pbxswitch to retrieve and write log files of inbound and outbound calls from any extention on the system.
19:48.50Grumblycan * do that
19:49.27Grumbly?
19:49.44drumkillathat would have nothing to do with asterisk ...
19:49.48Nuggetasterisk is a pbx, not an activity monitor.
19:49.48Qwelljust write a simple app, no need for a full PBX to do that
19:50.07GrumblyQwell:  I am not so much good with the coding.
19:50.59QwellYou'll need to code with anything you use
19:51.20GrumblyI mean app coding
19:51.22Grumblynot good
19:51.24QwellYou'll need to code with anything you use
19:51.28Grumblyscripting= ok
19:51.31rabelaisin sip.conf, in my register line, if my username has an @ symbol in it, how do I let asterisk know about that so it doesn't confuse the symbol with the host of the sip registrar?
19:51.40*** join/#asterisk Cybertank_ (n=cybertan@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com)
19:52.02Hmmhesaysbetter than drippy genitals
19:52.53*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
19:52.56Grumblybbiam
19:53.44nick125_lappyhmm
19:53.59nick125_lappyi wonder if theres a problem with my asterisk or xlite phone
19:54.21nick125_lappywhen i go to 7777, and press 1 (which goes in a queue), the hold music plays for a moment, then stops
19:55.12jaikenick: whats ur moh player? mpg123?
19:55.16nick125_lappyyeah
19:55.22Ariel_nick125, what does the cli says.
19:55.32jaikemake sure its the right one for ur version of asterisk
19:55.41nick125_lappyAriel_, hold on
19:55.49Qwelljaike: version of asterisk doesn't matter.  0.59r period
19:56.05Ariel_nick125, did you replace the default moh that comes with the system?
19:56.09jaikerun top....sometimes it eats up 99% or resources
19:56.30jaikeasterisk 1.0.7 has problems with mp3s with id3
19:56.38nick125_lappyAriel_, yeah, i put in some different music, 64kbps, no id3, mono
19:56.40nick125_lappy<PROTECTED>
19:56.40nick125_lappy<PROTECTED>
19:56.51jorge_blacan i paste three lines ?
19:56.57Nuggetyeah, the current state of MOH is pretty awful
19:57.09Ariel_nick125, if you change it back to the one supplie does it still do it?
19:57.15jorge_blacan i paste three lines ?
19:57.15Hmmhesayswhat the hell does 'kute' mean in a text message
19:57.23QwellHmmhesays: cute?
19:57.28jorge_bla/home/LRP/buildroot/build_i386/staging_dir/bin/i386-linux-uclibc-gcc -o gentone gentone.c -lm
19:57.28jorge_bla./gentone busy 480 620
19:57.28jorge_blamake[1]: ./gentone: Command not found
19:57.30jaikei run a script in crond to stop mpg123 every 30 mins
19:57.35NuggetI'd settle for IRC that could handle unicode.
19:57.37jaiketo make sure
19:57.39jorge_blawhat is gentone ?? ble
19:57.45Hmmhesaysit says 'hey kute'
19:57.49Qwellcutey?
19:57.50NuggetCutie.
19:57.51Hmmhesaysgeebus i hate text messages
19:57.55Qwellhowever you spell it
19:58.20pbdDon't get me started about IRC.  I was there for it's inception- a bad solution to a bad problem.
19:58.22nick125_lappythe highest thing in top is top
19:58.27Hmmhesaysits like you are stuck in aol chat hell
19:58.47Nuggetlike aol chat, but more expensive.  :)
19:58.55rkingNugget: recent xchat's (mine's 2.4.4) with properly configured locale's work great with UTF-8.  Â¿Â¡see!?
19:58.57jaikenick...try copying back original moh
19:59.04Nuggetrking: that's irrelevant.
19:59.13pbd15 years later, and it's gotten no better, despite thousands of programmers attempts to 'fix it'.
19:59.21Nuggetuntil you convince the rest of the planet to switch to xchat, it's worthless.
19:59.37rkingNugget: nah, my attitude with utf-8 is "spew it and screw 'em"
19:59.40Nuggetutf-8 is just one of a dozen "standards" for using unicode over irc.
19:59.42jorge_blawhat is gentone?
19:59.47Nuggetso none work, really
20:00.07Beirdohey cool, there's actually an ipkg for asterisk for the wrt54g
20:00.09Beirdoheh
20:00.11rkingNugget: imho utf-8 is going to emerge at the top of the pile.  what other unicode makes sense?  ucs-2?
20:00.17nick125i think my wireless might be going to junk
20:00.18*** join/#asterisk FuriousGeorge (n=furious@pool-70-111-20-125.nwrk.east.verizon.net)
20:00.22FuriousGeorgehey all
20:00.28nick125it works pretty ok on here, with a skip here and there
20:00.33NuggetI don't think we'll ever have a top of the pile.
20:00.34Beirdothis weather is lovely
20:00.35FuriousGeorgewhats that port range im supposed to foreward on rtp to get sip to traverse nat?
20:00.44NuggetI think irc is and forever will be a 7-bit clean medium
20:00.46FuriousGeorge3000-4000?  4000-5000?
20:00.57jaike10000 - 20000?
20:00.59pbdGeorge- check your sip.conf.. the range should be specified.
20:01.04nick125so now i dont think its asterisk
20:01.10FuriousGeorgejaike:  that sounds right
20:01.21FuriousGeorgethanks pbd
20:01.59pbdYou can also set the range there.. but be careful- first timers typically set it too small, and not all endpoints can deal with 'nonstandard' ranges.
20:02.04Hmmhesaysif you start ser in debug mode are all the modules supposed to echo 'initializing'?
20:02.41BeirdoI gave up on SER so quick :)
20:02.47Hmmhesayscause nathelper, tm, sl, usrloc and registrar don't. where textops stateless and maxfwd do
20:02.56rkingNugget: why 7-bits?  i've been outputting utf-8 for a while now and it definitely seems like the servers do fine with it.  word is next version of windows is going to have much better utf-8 support (and Microsoft has never broken a promise)
20:03.07Nuggetrking: the servers don't care.
20:03.14Nuggetbut most people aren't seeing what you're typing.
20:03.21Nuggetwhich is the crux of the problem
20:03.25jaikei think 10000-20000 is the usual range...ive set iptables to allow all udp on those ports...is there a security risk?
20:03.25rkingNugget: right - so it's just the clients.  exactly how many clients matter?  3?
20:03.28FuriousGeorgedefinately not in sip.conf but i believe 10k-20k is correct
20:03.41Nuggetrking: maybe 30 I'd guess.
20:03.48pbdCareful, now- don't mix client and server issues- that's one of the 'broken' bits of IRC- the protocol is fuzzy.
20:03.52Beirdortp.conf, I think
20:03.58Nuggetincluding quite a few which are console based where any sort of unicode is a tremendous hemmorhoid.
20:04.25jaikebeirdo: yup .. thats it
20:04.31BeirdoIRC and UTF-8 are not compatible.  not until the servers start supporting languages properly
20:04.31*** join/#asterisk ChibaPet (n=mason@acheron.hsd1.ma.comcast.net)
20:04.50Beirdoso the clients can actually negotiate support
20:04.54ChibaPetHey, all. Anyone familiar with SPA-3000 want to render some insight for me?
20:05.01pbdGeorge- sorry.. I should have said RTP.conf.
20:05.01rkingBeirdo: what sort of data chokes any IRC server?  (sorry, this is off-topic)
20:05.01Beirdoand that's not likely to ever happen
20:05.04queuetueI've got a A@H server running and I just converted it to be the firewall.  I can connect to the server fine from my sipura, but my bv sip connection has stopped registering properly...
20:05.14QwellChibaPet: ask away
20:05.16Beirdothe servers don't care
20:05.19nick125i wonder if it would be better if i used wavs, but, i cant get wavs to work which sucks
20:05.21Beirdothe clients do
20:05.28pbdChibaPet- I've got one, but I'm sort of a noob with it- I got it to work, if that helps. :)
20:05.46darkskiezi think its rubbish that xlite can talk to sipgate but asterisk cant thru a nat, I'm looking at ethereal dumps to see why the rtp streams get lost with *. Any ideas?
20:05.46Beirdoand there's no way for the clients to tell the server that they don't want UTF-8
20:06.04Qwelldarkskiez: firewall
20:06.22ChibaPetI'm new to *, and I'm trying to get to the FXO on the SPA-3000, and, indeed, I have an extension to get me to the FXO, and I hear the dial tone the FXO gives me, but each digit says:
20:06.24ChibaPet<PROTECTED>
20:06.27heison~seen sivana
20:06.29jbotsivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 21h 56m 56s ago, saying: 'ya'.
20:06.37ChibaPetDoesn't seem to actually be passing through digits.
20:07.00darkskiezQwell: its a nat'd connection. I can make and send calls, but the RTP stream doesnt make it back with asterisk. for some reason it does with xlite.
20:07.14Qwelldarkskiez: is asterisk set to nat=yes?
20:07.27nick125anyone here know why wav moh isnt working?
20:07.31ManxPower~docs
20:07.31jbotsomebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
20:07.36darkskiezQwell: i;ve tried yes and no
20:07.43nick125i put a wav in there, it doesnt show up in AMP, and it doesnt play
20:07.56ManxPowerdarkskiez: If you use asterisk's nat=yes then you must DISABLE NAT SUPPORT IN THE CLIENT
20:08.05ChibaPetI'm also unsure of why the FXO is not picking up on inbound calls to the PSTN, so I'm game for help with either, depending on what is more interesting for folks to help with. :)
20:08.07rkingBeirdo: that's a simple matter of converting on the client side.  Channels have encodings described in their /topic's, and clients just need enough support for encoding-per-channel, and that's all assuming that all clients shouldn't just move to UTF-8.  as long as everyone on every OS can view UTF-8, the clients that don't support it should just wither away
20:08.14Nugget"simple"?  heh
20:08.18Beirdonot supported
20:08.30Beirdoand never likely will be
20:08.47darkskiezManxpower: i am using asterisk as a client to sign on to sipgate in this scenario
20:08.51Nuggetconverting irc to utf-8 is about as easy as boiling the ocean.
20:08.57bjohnsonChibaPet: check wiki for spa 3000 forwarding trick
20:09.09nick125anyone even know if asterisk supports .wavs?
20:09.13BeirdoNugget: that's a quotable one :)
20:09.46ManxPowerdarkskiez: Ah, and you did the normal localnet= externip= figured out what RTP ports your SIP gatewway is using and forwarding them?
20:09.54ChibaPetbjohnson - read that, and it seemed interesting, but I don't mind if the SPA-3000 picks up before passing to *. I think I have the normal case configured... But of course I don't have it configured correctly!
20:09.56ManxPowernick125: it does.
20:10.46ChibaPetI have a dial-plan it should follow that should hook me into *'s start extension. But, I don't see any message from * indicating a connection attempt.
20:10.56*** join/#asterisk darkskie1 (n=darkskie@host86-133-151-115.range86-133.btcentralplus.com)
20:10.58nick125ok, so why when i put a .wav in /var/lib/asterisk/mohmp3, it doesnt play
20:11.01darkskie1ManxPower: asterisk is behind the nat. I can use xlite to talk to sipgate, both receive the calls, but no audio with asterisk
20:11.13nick125is there something special i have to do to make it work with wavs?
20:11.22ChibaPetI am, obviously, very new to all of this, so there are any number of things I might have gotten wrong.
20:11.43jorge_blajesus to crosscompile asterisk is bad dream
20:11.57ChibaPetI *do* have connections to the outside world working, and connections between various internal extensions.
20:13.01ManxPowerdarkskie1: you did not answer my question.
20:13.12ManxPowernick125: no.
20:13.15darkskie1ManxPower: i got disconnectred think i missed it
20:13.18ManxPowernick125: with is your actual ISSUE?
20:13.21ManxPowerdarkskiez: Ah, and you did the normal localnet= externip= figured out what RTP ports your SIP gatewway is using and forwarding them?
20:13.49darkskie1ManxPower: I did set them, u
20:13.52ChibaPetAnyway, can someone point me to information about this "Attempting native bridge" stuff, and what it might mean? Web searches haven't helped as yet, and reading the source isn't my preferred solution here, although I'll dig into it if I can't get clues elsewhere.
20:13.53ManxPowernick125: Mosic on Hold does not support wav, only MP3
20:13.54*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
20:13.57darkskie1i cannot forward the ports unfortunately.
20:14.10ManxPowerdarkskie1: If you cannot forward ports then you cannot use Asteirsk behind nat
20:14.21darkskie1ManxPower: how does xlite do it ?
20:14.35ChibaPetMusic On Hold can support anything that can pump out the right bits... There's a voip-info page that describes supporting different formats...
20:14.38nick125ManxPower: ok then, what would be the way to best settings to make mpg123 do less (and make the moh sound less skippy and such)?
20:14.39ManxPowerdarkskie1: I'm sure they do something horrible.
20:14.48*** join/#asterisk santiago (n=santiago@63.245.86.192)
20:14.53ManxPowernick125: no idea?
20:15.22ChibaPetnick - skippy means the machine playing your MP3s is too stressed, or your network is congested...
20:15.27*** join/#asterisk tubecat_ (n=ksoze@tidy.obscurity.org)
20:15.28rkingnick125: you could use a format that doesn't require decoding
20:15.42ChibaPetnick - drop your bitrate. The output is going over a phone, after all.
20:15.59nick125ChibaPet: its already 64kbps, how low should i go?
20:16.02nick12532kbps?
20:16.17ChibaPetis bandwidth an issue, then? voice connections are clear, or no?
20:16.20ManxPowernick125: you DO realize that all codecs except for ulaw and alaw distort music, right?
20:16.34*** join/#asterisk Koshatul (n=evangeli@CPE-138-217-190-164.qld.bigpond.net.au)
20:16.43nick125ManxPower: i figured there would be some distortion
20:16.46ManxPowernick125: 64kbps is fine
20:17.00ManxPowernick125: depending on the codec, there can be a lot of distortion
20:17.12nick125im trying GSM atm
20:17.14ManxPowerthings like G726 are heavily optimized for voice
20:17.19ManxPowerGSM is pretty bad.
20:17.23ManxPowertry ulaw if you can
20:17.27tubecat_i'm having trouble setting up an iax2 channel. on my server box asterisk does not listen on any port
20:17.31ChibaPetnick: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+faking+it
20:17.46nick125ManxPower: i dont see a ulaw setting on x-lite :/
20:17.56ManxPowernick125: aka G711
20:18.02ManxPoweror g711u
20:18.20nick125oh ok
20:18.37*** join/#asterisk the_devil_dont_s (n=Adam@195.26.12.229)
20:18.50*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
20:18.52nick125how much b/w does G711/G711U/ULAW use?
20:18.52tubecat_so my client box can't connect. any tricky reasons why asterisk won't listen?
20:19.01*** part/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
20:19.01*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
20:19.17ManxPowernick125: 64k plus overhead, about 80k total
20:19.37ChibaPetSo, anyway, anyone have clues about my FXO issues? I'll muddle it out eventually, but would love to save some pain and aggravation. :)
20:20.04nick125ManxPower: ok, so, what codec would you suggest for over the internet (pretty much with a max 1MB upload line)?
20:20.08jorge_blabrrrr, why asterisk doesnt have autoconf . is anybody working on it?
20:20.18ManxPowernick125: make it work with g711/ulaw first.
20:20.25ChibaPetjoege - mm, Debian binary packages
20:20.37ManxPowerjorge_bla: because nobody will comit to maintainging it forever
20:21.07jorge_blaManxPower, anyway it would be nice to have it..
20:21.08pbdjorge- you're asking a religious question here.
20:21.15jorge_blapbd realy?
20:21.21jorge_blaha ha ha
20:21.26*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
20:21.38pbdMost of the developers of asterisk seem to have an allergy (for good reasons- I'm neutral on this one) to autoconf.
20:21.47ChibaPetNo help to be had. The channel is much like the docs. :P Oh well. Maybe I'll pay someone for help at some point. Later, all!
20:22.03ManxPowerSince Asterisk only officially supports Linux, autoconf is rather silly
20:22.11jorge_blapbd, i am trying to compile asterisk agains uclibc but ...
20:22.21pbdFrankly, aside from a very short list of dependencies, I've never had much problem compiling it- autoconf is kind of overkill.
20:22.45brad_msswwhat should DTMF Tx method be set for, for SIP devices?  I'm having an issue where if I have a command that starts with *, it is not recognized by asterisk from a certain SIP device
20:23.06pbdjorge- See, now you're trying to make it work outside of it's small, well defined box- no one's warranteed * for that.
20:23.20brad_msswi've got options like inband,avt,info,inband+info,avt+info, and auto ... auto doesn't seem to work properly
20:23.34pbdbrad- Sounds like you're working with a Cisco phone. :)
20:23.44brad_msswpbd: Sipura 841 :/
20:23.58jorge_blado i need sound card to use asterisk ?
20:24.07Beirdono
20:24.11pbdWell, on the cisco, I set it for avt.. and set asterisk for rfc2833.
20:24.20pbdThat gives me out of band dtmf.
20:25.05*** join/#asterisk Broom (i=Broom@jescobar.ayustar.net)
20:25.06pbdI also specifically disable inband on the phone.
20:25.07Broomhey all
20:25.12*** join/#asterisk Lordy (i=steve@d040099.adsl.hansenet.de)
20:25.29Lordyhi all
20:25.36brad_msswpbd: thanks, let me give that a try
20:25.40pbdbrb
20:25.43Broomi'm having problems with incoming calls using a TE110P card, i only get this: Starting simple switch on 'Zap/12-1'
20:25.45Broomthen the hangup
20:25.50Broomany ideas?
20:26.01QwellBroom: turn verbose up and debug on
20:26.05Broomi did
20:26.06Lordyi would like to know if it is possible to have a ascend max dial out on a specified line for asterisk
20:26.07Broomthat's all i get
20:26.11Broomi did set verbose 10
20:26.16Broomset debug 10
20:26.24Qwelldebug isn't an int...
20:26.25*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
20:26.58BroomCore debug was 0 and is now 10
20:27.02Broomgot that with that command
20:27.04Lordycorrection: lucent max, i wanted to say :)
20:29.18Broomanything else i can do to verify this?
20:30.24nick125i tried that faking thing, and it didnt work :/
20:34.05Lordyasterisk with lucent max ? anyone ?
20:34.45pbdSorry, lordy- don't have one..
20:36.09heison~seen sivana
20:36.10jbotsivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 22h 26m 37s ago, saying: 'ya'.
20:36.46DaPrivateercan someone please assist me with this... how can i make the AGI dial something on the line its currently on? IE someone calls, and when it answers the phone if they press a certain option it dials back dtmf tones on the same line.
20:37.33DaPrivateer(this is on a digium X100P using zaptel)
20:38.45jsmithDaPrivateer: SAY DIGITS 1234567890
20:38.50DaPrivateerthank you
20:39.21jsmithOh, wait... DTMF.
20:39.29jsmithThat'll probably just read off the numbers....
20:39.45Broomnothing on my prob?
20:39.58dudesSendDTMF(${EXTEN}#)
20:40.22pbdBroom- I don't run any Digium hardware myself.. I'd guess no one does, from the channel today.
20:40.23jeroanyone ever integrated asterisk with a toshiba pbx using a pri ?
20:40.24shido6Zzzzz
20:40.25brad_msswanyone have a sipura 841 phone ?
20:40.28shido6I do
20:40.30shido6I run digium gear
20:40.33shido6whats the problem?
20:40.39shido6thats all I run is digium gear
20:40.40jsmithpbd: I use Digium hardware...
20:41.11jeropbd: its being done
20:41.22jeropbd: but first we have to deal with both
20:41.24pbdOk, so someone talk to Broom and walk him through the zaptel config. :)
20:41.27brad_msswcan't dial *112  or similar because the phone tries to interpret it .... trying to figure out what setting disables that
20:41.43Broomjsmith: you have digium hardware for incoming calls?
20:41.53DaPrivateerdudes - you are my hear
20:41.55DaPrivateerhero*
20:41.57DaPrivateerthank you so much
20:42.27Ariel_brad_mssw, go into the advaced setting in the phone and you will see * functions delete the one you don't want and it will pass that opiton through.
20:42.27jsmithBroom: Yes...
20:42.54pbdjero- such a conservative place.  Let me guess- asterisk for voicemail first, then buy handsets and migrate over?  Seems to be a common path... you might find some help on the lists, a lot of people ask about it.
20:43.09brad_msswAriel_: tried that, doesn't seem to work ... especially if the command is '**'
20:43.17Broomjsmith: well, my problem is that when i place a call i only get this:  Starting simple switch on 'Zap/16-1'
20:43.19Broomthen the hangup
20:43.27Broomi have verbose to 10
20:43.29Broomand debug on
20:43.33Broomthats the only thing i get
20:43.37jeropbd: not really, all VMs on *, toshiba for analog lines and * for the PRI link. then move everything to *
20:43.41brad_msswAriel_: it pretty much immediately gives a busy signal ... works from zap phones, and a UTStarcom F1000 just fine
20:43.55*** join/#asterisk supaigtr (n=yurplsl@152.53.16.10)
20:44.07Ariel_Broom, post your setting for the zaptel and zapata.conf on pastebin.ca and your dial string so we can see what you have setup.
20:44.35Ariel_brad_mssw, if you look at the dialing rules you will see it only allow one *
20:45.04brad_msswconfigurable dialing rules ... or hard rules for the phone ?
20:45.55nick125anyone here know of x-lite problems when you go to press a button (such as 1) during a call?
20:46.05pbdjero- Unfortunately, the last time I integrated a Toshiba, it was with the help of a toshiba vendor and using FXOs for the interconnect.
20:46.16jerookay
20:46.18BroomAriel: dialstring?
20:46.24jerothanks anyway pbd
20:46.25pbdjero- but what's the issue you're seeing?
20:46.41jerono issue yet, will start in a few days
20:47.05jeroI just fear we cant replace the analog links by a pri
20:47.35brad_msswoh, there's the dialplan in advanced ... heh
20:48.38pbdjero- you mean, you're afraid the pri won't integrate between the toshiba and *?
20:48.41*** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue)
20:48.42surfduehi
20:48.46jeropbd: yes
20:49.00surfduei have x-lite sip phone, i dial one in our menu and i cant get the menu to go up
20:49.05jeropbd: I'll probably get a toshiba vendor to
20:49.06surfduethe sip phone dosnt show i dialed 1
20:49.09surfduebut makes a noice
20:49.11surfdueanyone know how to help
20:49.23pbdJero- Well, the Toshiba was a little funky on FXO/FXS timing, etc- but PRI is PRI.. assuming your signalling is matched, it really should be no problem.
20:49.58jeropbd: how will we  tell the toshiba to send the voicemails to the pri ?
20:50.20pbdJero- you're planning on using a multiport digium PRI card in the * box, receive PRI from the carrier there, and pass it on to the toshiba, correct?
20:50.25Ariel_jero, you have to chnage the rules on the toshiba pbx for that
20:51.01pbdjero- Voicemail integration has a number of pitfalls- haven't seen them personally, but the MWI stuff has had more than a few askers on asterisk-users.
20:52.17Ariel_you will not have any mwi nor any way to do call back via there menu. But phones kinda work.
20:52.23*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
20:52.43luke-jr__How can I determine why a call file is being ignored?
20:52.52luke-jr__there doesn't seem to be anything with -vvvvvvvvvvvv
20:53.03jeroariel: what happens when the toshiba phone user wants to read its mail ?
20:53.24jsmithluke-jr__: Does it have the correct extension.  Are you writing the file somewhere else, and then moving it to the outgoing directory?
20:53.27jerohe will dial an extension that the toshiba has to forward to *
20:53.45*** join/#asterisk RoyK (n=roy@ti211210a080-0158.bb.online.no)
20:53.50*** join/#asterisk zotz (n=zotz@24.231.36.100)
20:54.16Ariel_you setup a dial rule in the toshiba to send the call via the pri then asterisk will take that and send them to voiemail.
20:54.20RoyKhi
20:54.22RoyKany idea what this is about?
20:54.23RoyKAug 20 00:53:37 NOTICE[5965]: rtp.c:514 ast_rtp_read: Unknown RTP codec 105 received
20:54.27Ariel_you need to be able to configure both systems
20:54.31BroomAriel: posted
20:54.36bkw_http://www.news24.com/News24/Backpage/Crime_Court/0,,2-1343-1345_1755475,00.html
20:54.49pbdRoyk: The other end is sending a codec that Asterisk doesn't like?
20:55.00luke-jr__jsmith: yes
20:55.27bkw_RoyK, I think it might be dtmf trying to use 105 instead of 101?
20:55.31bkw_whats the SDP say?
20:56.04pbdbkw- let's hear it for little old ladies!
20:56.08jsmithluke-jr__: Strange...
20:56.37jerothanks for the infos guys
20:57.19luke-jr__jsmith: http://pastebin.ca/20462
20:58.10BroomAriel: http://pastebin.ca/20461
20:58.11Broomsorry
20:58.39nick125any ideas on why when someone calls 7777 and presses 1, it doesnt go the the queue?
20:58.40RoyKbkw_: is that related to info/rfc2833-something?
20:59.01bkw_where is the sdp?
20:59.18RoyKer... how can i find that out?
20:59.22bkw_sip debug
20:59.24bkw_and post it
20:59.27bkw_on pastebin
20:59.37*** join/#asterisk Equinox (n=secret@star.l93.com)
20:59.44RoyKbkw_: ok
20:59.47jsmithluke-jr__: Did you try taking out the blank lines?
20:59.56jsmithluke-jr__: It's just a guess
21:00.00luke-jr__jsmith: no... I've had it work w/ blank lines before, tho
21:00.14luke-jr__didn't help :(
21:00.41luke-jr__interestingly, an old call file that used to work is also ignored
21:01.16jsmithMaybe you changed your outgoing directory in asterisk.conf?
21:01.35luke-jr__nope
21:01.45luke-jr__the files are disappearing once moved, too
21:02.23EquinoxThe "N" in NXX is 1-9?
21:02.42pbdIgnored as in doesn't run, or ignored as in left there, luke?
21:02.48*** join/#asterisk jaike (n=a@203.131.137.76)
21:03.12zedkatufI'm using kphone (under linux) softphone & am trying to get incoming calls working, the CLI shows me that calls are getting as far as the asterisk box from the internet.....but I keep getting "THe number you have dialled has not been recognised" on my landline....does anyone know about kphone's settings that could perhaps be borking things...?
21:03.23pbdIs it possible your system is having timing issues, or permissions issues between asterisk and that outgoing directory?
21:04.10zedkatuf(Ican make outgoing calls with no probs)
21:04.59pbdzed: Sounds like a dialplan issue.. can you reach that kphone from another softclient via *?
21:05.01luke-jr__aha!
21:05.13luke-jr__jsmith: my asterisk doesn't run as root anymore ^^;;
21:05.44*** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net)
21:05.54pbdLuke- Glad to be of help, that will be $50. ;-)
21:06.16surfduedoes anyone use x-lite
21:06.18surfdueor used or know how to useit
21:06.22surfdueIm having a problem, when i dial an extention after im on the phone xlite dosnt seem to send it through
21:06.30zedkatufpbd: good idea re another client.......I'll try that
21:06.31surfdueits not sending keys
21:07.05SpaceBasssurfdue: try changing your dmtf settings
21:07.28bkw_Equinox, N is 2-9
21:07.32surfduehow
21:07.32bkw_Z = 1-0
21:07.35surfduewhats dmtf?
21:07.36bkw_X = 0-9
21:07.36*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
21:07.50nick125surfdue: in the advance settings menu
21:07.52bkw_er z=1-9
21:07.53nick125then dmtf
21:07.57surfduewhere?
21:07.58surfduehow
21:08.12surfduek
21:08.17nick125surfdue: go into the settings, then go to the advanced settings menu
21:08.18Qwellbkw_: are there any that match ABCD?
21:08.21nick125tell me what it has
21:08.22surfduechange 2 what?
21:08.26surfdueforce bands is no
21:08.32bkw_Qwell, A B C or D ninny :P
21:08.34RoyKgrr
21:08.34RoyKAug 20 01:08:56 WARNING[5965]: chan_zap.c:2131 pri_find_dchan: No D-channels available!  Using Primary on channel anyway 202!
21:08.38RoyKany idea what that might be?
21:08.39Qwellbkw_: I mean any and all
21:08.39surfdue2800 and 110
21:08.42surfdue101*
21:08.46nick125hmm
21:08.52bkw_RoyK, I get that message all the time
21:08.53Qwell(besides [abcd])
21:08.56bkw_when I first start the box
21:09.04nick125thats the same as my settings
21:09.12*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
21:09.22SpaceBasssurfdue:  you may want to check the voip-info wiki and search for dmtf to learn about the options
21:09.23RoyKbkw_: this is repeating. B chans 1 through, say, 12 comes up and then i get that
21:09.52bkw_RoyK, its more retarted code
21:10.06EquinoxAnyone here familiar with Polycom IP500 phones?  I'm trying to set up my FTP directory and have a question:  Bootrom 2.6.2 has a bootrom.ld file.. But no bootrom.ver .. The 2.6.1 file has both.. Do I not need the .ver?
21:10.12bkw_:P
21:10.16brad_msswok, fixed the dialplan on my sipura 841 phone.  One issue though, whenever Authenticate() is called, it does not appear to work from this phone ... the # sign does nothing ... is this a dtmf code thing I need to trial and error ?  extension numbers work fine
21:10.31RoyKbkw_: but then it doesn't mean much?
21:10.49surfdueomg
21:10.55surfduecan someone just tell me what to do.
21:10.57surfduelol
21:10.58RoyK<PROTECTED>
21:10.58RoyKAug 20 01:11:17 WARNING[5965]: chan_zap.c:2131 pri_find_dchan: No D-channels available!  Using Primary on channel anyway 202!
21:10.58RoyK<PROTECTED>
21:11.01RoyKthat sort of thing
21:11.08SpaceBasssurfdue: read
21:11.11surfdueit works on other clients
21:11.16surfduejust not mine
21:11.19surfdueon nicks it works
21:11.21surfduenot mine?
21:11.29bkw_RoyK, I see that all the time.. the span goes down and back up
21:11.30surfduenick what does ure dfm say
21:11.37RoyKspan == dchan?
21:11.41bkw_but only on start/restart
21:11.45pbdbrad: sounds definitely like a dtmf issue- I've been in that particular ring of hell lately.
21:11.53bkw_what is channel 202?
21:11.58bkw_does it line up with a dchan?
21:11.58RoyKbkw_: this happens all the time, not while restarting
21:12.03pbdbrad: make sure that your dtmf settings are matched all around.
21:12.13nick125surfdue: same
21:12.20RoyKbkw_: dhcan on span 7
21:12.47*** part/#asterisk darkskie1 (n=darkskie@host86-133-151-115.range86-133.btcentralplus.com)
21:12.54PupenoLcan't asterisk be 'configured' te use a prefix (like /usr/local) when building it ?
21:12.54surfduewhats magic numbers
21:12.59brad_msswpbd: k, thanks.
21:12.59surfdueim in gsm config
21:13.08surfduenick get on skype
21:13.12SpaceBassno magic, just needs to match
21:13.33surfdueso the gsm number needs to match the dtfm?
21:13.51SpaceBasshttp://www.google.com/search?client=safari&rls=en&q=xlite+dtmf+asterisk&ie=UTF-8&oe=UTF-8
21:14.32surfduewhat do i do
21:14.51SpaceBassanyone see that slashdot article today about GSM phone integration?
21:14.56gambolputtyywa
21:15.02gambolputtyarticle didnt mention asterisk
21:15.06gambolputtyjust the slashdot thing
21:15.26*** join/#asterisk nick125 (n=nick125@unaffiliated/nick125)
21:15.26jaikefinally found out how to make polycom volume permanent! yes! dont need to buy new headsets
21:15.53SpaceBassthere is specifically an article on asterisk and gsm integration
21:16.03surfduehttp://www.google.com/search?client=safari&rls=en&q=xlite+dtmf+asterisk&ie=UTF-8&oe=UTF-8
21:16.05Qwellnot much "article" to id
21:16.07Qwellit*
21:16.27RoyKbkw_: how should i debug this? will a pri debug give much hunch?
21:16.28surfdueanyone?
21:16.29surfdueplease
21:16.37surfduehow do i fix this sip x-lite problem
21:16.52surfduenot sending the keys i click one calling someone
21:17.14SpaceBassdude, seriously... you need to research a little... there is no simple answer, it depends on how your extension is set up and what the settings in xlite are, etc
21:18.04Nivexjaike: How?  My friend just got some Polycoms and is having that problem.
21:18.31opus_yeah has ANYBODY built a gsm cellphone network to asterisk intergration
21:18.37opus_what a load of fucking crap
21:18.41SpaceBasslol
21:18.59*** join/#asterisk nitram (i=foo@superblob.com)
21:19.01zoaprobably someone did
21:19.09zoabut it would have been a cellphone carrier
21:19.18DrmCtchrwhere is the simple switch context defined at ?
21:19.23surfdueSpaceBass,  maybe u can go over it with me
21:19.25SpaceBassthere are mini-cells out there....
21:19.29surfduei relly just want to get xlite wokring
21:19.32opus_i think that the transporter from star trek is a great idea to
21:19.40opus_wheres my flying fucking car
21:19.42surfduein amp does this matter dtmfmode:
21:19.50DrmCtchri need to have dialtone played there and preferably wait a bitlonger for digits from the t1 E&M channels
21:19.54Qwellsurfdue: very much so, yes
21:20.03SpaceBasssurfdue:  absloutly... thats the point :)
21:20.11surfdueok well 2 extentions
21:20.18surfduehave the same crap in it dtmfmode: rfc2833
21:20.19SpaceBasssurfdue:  what ever it says in AMP for your extension, is what xlite should be set to
21:20.25nick125surfdue: the extensions are the same other then the password and username
21:20.27SpaceBasstry changing it to inband
21:20.41surfduehow
21:20.49surfduenvm
21:20.57*** part/#asterisk jsmith (n=jsmith@64.50.35.114.2O7.net)
21:21.12brad_msswpbd: yep, setting the dtmf equal on both sides fixed it ... also set it to use ulaw instead of gsm ... thanks
21:21.38DrmCtchranyone know where is the simple switch context defined at ?
21:21.41zoaout of band should be better if you ask me
21:21.48Broomhey, now i'm getting this error: Spawn extension (from-trunk, s, 2) exited non-zero on 'Zap/7-1'
21:21.54*** join/#asterisk Gronker__ (n=Gronker2@70.152.167.143)
21:22.25opus_zoe unless you type in your bank account number
21:22.33surfduenope
21:22.34*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:22.37surfduedosnt work
21:22.56surfduedammiot
21:23.24SpaceBasstry another client
21:23.33SpaceBassfirefly can use IAX2 and works well
21:23.40surfduelink me please
21:23.46SpaceBassiaxcomm on linux or os x...
21:23.48Qwell~google firefly iax2
21:23.49surfdueanyclient that works, works for me
21:25.45EquinoxDo the polycom config files contain the sip user & password?
21:26.23pbdbrad- glad to be of assistance.
21:27.23pbdzoa- If you can't use Ulaw, out of band is the *only* way to go.
21:28.28*** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl)
21:30.29*** join/#asterisk ryansc (n=ryansc@c-67-164-188-180.hsd1.co.comcast.net)
21:30.30adelashey, i have my asterisk@home setup, and cisco phones setup already, what kind of info would i need to add a trunk (i have a mediatrix 1204 setup)
21:31.18Broomanyone can help me with this error:  Spawn extension (from-pstn-reghours, s, 3) exited non-zero on 'Zap/12-1'
21:31.20adelaswell, i'm kinda loat in amp
21:31.24adelaslost*
21:32.05adelasi have the mediatrix ip and so on, just don't know how to set it up
21:32.08pbdErr.. check priority 3 in your s extension in the fron-pstn-reghours context? ;-)
21:32.31opus_does anyone have the problem wioth the IP 501 when a user dials *98 it actually modifies it and makes it 9*8?
21:32.51opus_whats there to change? I forget what the correct dial pattern modification is..
21:32.58opus_if somebody could cut and paste it, would be cool
21:34.36surfduex-lite not sending numbers i dial
21:34.39surfduecan anyone help
21:36.48*** join/#asterisk santiago (n=santiago@63.245.87.180)
21:37.02bkw_WHO IS THAT AT MARKS DESK
21:37.16filebkw_: guess who just called sales
21:37.26DrmCtchrplease, anyone know where is the simple switch context defined at ?
21:37.29MikeJ[Laptop]bill gates?
21:37.44surfduei cant find a sollution
21:37.47surfdueive searched every
21:37.48surfduewehre
21:37.58surfduei cant find a reason why it wont send numbers once im on the phone
21:38.12surfdueif im in voicemail it sayspush one to say voicemail, i push one nothing happens
21:38.18surfdueit acts as if it was never pushed
21:38.20surfduecan anyone help
21:38.31MikeJ[Laptop]file, well who already?
21:38.41fileMikeJ[Laptop]: it was some woman who was answering an ad for a job that pays $25/hour
21:38.44*** join/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com)
21:39.00filehow the hell she got me I have no clue
21:39.30DaPrivateerOk, general question directed towards anyone in this channel running asterisk in a production environment. I am trying to convince my boss to let me transfer our phone system over to VoIP. What can you tell me about your uptime percentage?
21:39.34MikeJ[Laptop]heh!
21:39.51MikeJ[Laptop]DaPrivateer, depends.
21:40.04puowvip100%, but I only have 7 extensions.
21:40.13puowvipsince April 2005.
21:40.20DaPrivateerya, we have only 4 extensions (all off-site) and 3 PSTN lines
21:40.25opus_daprivateer if you loose registration sometimes it doesn't connect back.
21:40.56DaPrivateerMikeJ[Laptop] what does it depend on?
21:40.59opus_daprivateer its about the same quaility of a cell phone. sometimes doesn't work. i say 1/20 calls won't go through just because its so complicated.
21:41.06*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
21:41.08MikeJ[Laptop]depends on what you are doing...
21:41.28surfduesomeone?
21:41.28*** join/#asterisk nowork (n=jfu2808@216.254.141.97)
21:41.30surfduePLEASE!
21:41.34surfdueif anyone can help
21:41.44Qwellif anybody knew, they'd help
21:41.49MikeJ[Laptop]asterisk in loads like that, doing only tdm to voip and nothing else should be pretty solid
21:41.49DaPrivateerMikeJ[Laptop] - just giving offsite personnel access to PSTN lines in the office
21:41.52QwellOR, you could pay for help
21:42.02*** join/#asterisk blitzrage (n=leif@asterisk/documenteur-extraordinaire/blitzrage)
21:42.03DaPrivateerok; when does it cause problems
21:42.06Qwelldon't, however, beg.  it makes you look stupid
21:42.08MikeJ[Laptop]but your uptime will be very dependant on your lines.
21:42.36surfduewell i am stupid
21:42.40surfdueelse i would have fixedit.
21:42.42DaPrivateerya, clearly. i meant the server itself though. clearly if the internet goes down, thats not asterisk's fault
21:42.43Qwellthen RTFM
21:42.47Qwell~docs
21:42.47jbotmethinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
21:43.01opus_get a brain moran
21:43.10Qwella brain, or some cash
21:43.12MikeJ[Laptop]DaPrivateer, like any software, the more complex the functions you use, the more problems you have.. it depends... people using complex proxy solutions, or using rarely used functions have the best luck finding bugs
21:43.23MikeJ[Laptop]but for what you are talking about, asterisk should be very solid
21:43.24DaPrivateerhehehehe
21:43.53DaPrivateerthe only other thing in it will be IAX to FWD for toll free calls
21:44.13MikeJ[Laptop]yeah, you are talking about the easiest of functionality...
21:44.27blitzrageQwell: do you know if Asterisk accepts registrations from multiple interfaces on the same box? (I think it should... but apparently not on one of my boxes - running, 1.0.9)
21:44.35MikeJ[Laptop]with no transcoding, you could run that on a wrt
21:45.06MikeJ[Laptop]blitzrage, iax?  sip?
21:45.27blitzrageMikeJ[Laptop]: IAX2
21:45.29MikeJ[Laptop]the issue to my understanding is on virtual interfaces
21:45.39MikeJ[Laptop]but seperate physical is supposed to be ok
21:45.41blitzrageMikeJ[Laptop]: yah, these aren't virtual - physical
21:45.44Qwellvirtual interfaces are pointless anyhow
21:45.52blitzrageMikeJ[Laptop]: hrmmm... I have a device that won't register for some reason
21:45.53Qwelljust give your interface two IPs, heh
21:46.00MikeJ[Laptop]what's it do?
21:46.09MikeJ[Laptop]bindaddr set?
21:46.16blitzrageMikeJ[Laptop]: not entirely sure... bindaddr not set
21:46.28Qwellblitzrage: check netstat, make sure its listening
21:46.30blitzrageMikeJ[Laptop]: by default does it just bind to the first address if 0.0.0.0 is not set?
21:46.46MikeJ[Laptop]I don't recall.. I don't multihome.
21:46.49*** join/#asterisk durak (n=durak196@85.98.97.122)
21:46.54DaPrivateerlol.. anyone else? my boss wants more people saying it works well lol
21:47.06blitzrageDaPrivateer: it works well
21:47.10Qwellasterisk?
21:47.11Qwellno, it sucks
21:47.13Qwell:P
21:47.15DaPrivateerhehe
21:47.16blitzrageyah, its shit
21:47.23DaPrivateeruptime percentage?
21:47.29blitzrage2%
21:47.29Qwelllike 3%?
21:47.37Qwellyeah, its about 2.5
21:47.42blitzrageQwell: wow! thats awesome
21:47.45DaPrivateerheh, seriously plz
21:47.45Qwell(blitzrage: nice one, btw)
21:48.06surfdueWe are interconnected with the PSTN with a CISCO and we use RFC 2833 for DTMF because most part of the GWs in the network work much better than In Band.
21:48.07surfdueTherefore customers that use XPRO have to configure DTMF force send in Band as No.
21:48.07surfdueCISCO recognizes XPRO is not sending DTMFs in band but for some reason ignores all DTMFs sent via XPRO.
21:48.07surfdueAll the rest, (PAP, Micronet, ZOOM, etc) gateways work fine on RFC 2833, but XPRO.
21:48.07surfdueIs this a bug?
21:48.14surfdueTHis guy has the same problem as me?
21:48.58surfduei tryed dtmf to yes and no on force still dosnt seem to work
21:49.12DaPrivateersurfdue - i would say contact xpro
21:49.19*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
21:49.26blitzrageDaPrivateer: with Asterisk, anyone can tell you all sorts of things, but you REALLY have to go and do it yourself and see if it'll work for you - it requires testing, it requires knowledge. Asterisk is for your TELEPHONES - you only get as much out of them as you put into them.
21:49.44Qwelleww
21:49.45surfduei have x-lite
21:49.48Qwellperv
21:50.02Qwellbetter
21:50.06blitzragelol
21:50.09DaPrivateerblitzrage - ya i have a PBX set up right now and i love it. thanks to help from some people here i even got around the problems i was having
21:50.21DaPrivateerhe just wants to see other people's success stories basically
21:50.21blitzrageback latah
21:50.44MikeJ[Laptop]DaPrivateer, the amount of people in here should say somthing
21:50.50MikeJ[Laptop]there are lots of people using it
21:50.52DaPrivateerya, thats what i said
21:50.54QwellMikeJ[Laptop]: heh
21:50.58Qwellshould say something, which way? ;]
21:51.08Qwellalot of people == alot of people that need help
21:51.15Qwellthe question is, whats the ratio?
21:51.20Hmmhesaysonly luv you gets from a tv set and it makes you crazay
21:52.05*** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net)
21:52.25DrmCtchrwhere is the simple switch configuration defined at ?
21:53.14DaPrivateerthanks guys
21:53.18fileeh?
21:54.51*** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net)
21:55.09puowvipok
21:55.16Kravenhrm it seems that noone is looking on their bugtracker @ inaccess
21:55.26Kraven:(
21:55.40Drukenfile: could have been worse, you could be encrypted too :)
21:56.14*** part/#asterisk secure75 (n=mic@gfwlan.cablesurf.de)
21:57.02vonageow
21:57.05vonageso i have to ask
21:57.11vonagewhy does everyone want to be me
21:57.12vonageim me
21:57.14vonageknock it off
21:57.20Qwellnobody wants to be vonage
21:57.26Drukenhell no
21:57.39vonagen
21:57.51EquinoxAnyone here offer some advise on polycom config files?
21:57.55Hmmhesaysvonage does have a decent network put together
21:57.59Drukeni think i'd agree with Qwell
21:58.09vonagethat i do
21:58.10brad_msswnope, we get all circuits busy messages on vonage all the time
21:58.11Hmmhesaysanyone who says vO naaaaajjjjj should be shot
21:58.18vonageyeah im a busy guy
21:58.23brad_msswand we can't pull the lines into asterisk without going through these damn linksys boxes first
21:58.27Hmmhesaysi never get all circuits busy on the business plan
21:58.34Hmmhesaysbrad_mssw: you are mistaken
21:58.37brad_msswthis is a business plan
21:58.38vonageyeah pay more call more i like to say
21:58.44*** join/#asterisk wpbrown (i=wpbrown@66.0.163.142)
21:58.46Hmmhesaysbecause I have all my lines going through asterisk
21:58.47brad_msswwe've got 4 business lines
21:58.48surfduehmm
21:58.54brad_msswHmmhesays: all of them? how?
21:58.59surfduei wish tehre was a quik fix to this DTMF settings error thing
21:58.59brad_msswHmmhesays: via softphone stuff?
21:59.09Hmmhesayshell no, i have my asterisk registered directly with vonage
21:59.09*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
21:59.13brad_msswHmmhesays: vonage says the softphone accounts can't do call hunt which we need
21:59.25vonagehey give me time give me time
21:59.27brad_msswHmmhesays: ok, i'm confused, how? what login info do you have to use?
21:59.34Hmmhesaysthe login info they provided me
21:59.42brad_msswHmmhesays: are you in the US?
21:59.44Hmmhesaysindeed
21:59.52filehot'n'sexy Hmmhesays!
21:59.54brad_msswHmmhesays: hmm, and you're on the standard business phones?
21:59.55Hmmhesaysyo
22:00.00Hmmhesaysbusiness plus
22:00.05puowvipnice nick, Hmmhesays
22:00.08rabelaishas anyone sucessfully connected to earthlink's new "free calling online" sip service? despite all my attempts, the server still rejects my login...I can't register with the sip server and I have no idea why not
22:00.12Hmmhesaysyo file wassup
22:00.16brad_msswHmmhesays: yeah, ok ... hmm ... do you have a contact with vonage to get that info?
22:00.26fileoh nothing, getting zaptel setup on a box
22:00.27brad_msswHmmhesays: or did you just call their standard support lines ?
22:00.33Hmmhesaysif you order the business plus plan they will give it to you
22:00.37vonageyou could ask me ^_^
22:00.40vonageexcept not
22:00.58Hmmhesaysbecause if you order it in the midwest you will probably be talking to me
22:01.20*** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m)
22:01.22Hmmhesaysi dunno bout their other business plans yet
22:01.34Hmmhesaysresidential definately not
22:01.38brad_msswHmmhesays: we're on the Small Business Unlimited
22:01.44brad_msswHmmhesays: is that the same
22:01.48Hmmhesaysnope
22:02.13Hmmhesayshow much you paying a month for that?
22:02.24brad_msswHmmhesays: $50/mo each line
22:02.26Hmmhesaysouch
22:02.33brad_msswHmmhesays: so like $200 total + our 800 number
22:02.48Qwelljust get a per minute account with a real provider, heh
22:02.49brad_msswHmmhesays: so they have different business plans then ??
22:02.56Hmmhesaysthe smallest business plus plan is 4 simultaneous calls with 5000k minutes us/canada for $150 a month
22:03.02Hmmhesaysand you can use asterisk
22:03.19brad_msswHmmhesays: that's not bad
22:03.25Hmmhesaysnope, works great too
22:03.26brad_msswHmmhesays: is this something you have to call about ?
22:03.32brad_msswHmmhesays: or do they have a business website ?
22:03.34Hmmhesaysofficial launch is in 2 weeks
22:03.41brad_msswah hah
22:03.48Hmmhesayssoft launch was 2-3 months ago
22:03.51brad_msswok, we'll probably be switching to that
22:03.54EquinoxHmmhesays- 5 million minutes?
22:03.55Hmmhesaysyou can get it right now
22:03.58Hmmhesaysoops
22:03.59Hmmhesays5k
22:04.05Hmmhesays5000 minutes is what I meant
22:04.09Qwell5k minutes maximum?
22:04.10EquinoxAhh :)
22:04.14EquinoxThat's 3c/minute
22:04.23brad_msswyeah, converting from digital -> analog -> digital probably isn't the best thing here ;)
22:04.46Hmmhesaysjust call them and ask for the business plus info
22:04.53brad_msswcool, thanks
22:04.56Hmmhesaysif you're in the midwest you'll get sent here
22:05.00Qwelldamn thats expensive, heh
22:05.24HmmhesaysQwell that would be expensive if you didn't get 4 did's with it
22:05.24brad_msswnow, when you say 4 simultaneous calls, how does that get configured in asterisk ... as 4 individual register lines ?
22:05.31HmmhesaysI use one
22:05.33*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com)
22:05.37DrmCtchranyone have a sounds file of dialtone?
22:05.40Hmmhesaysand have the phones in a ring group
22:06.02moverwhen i compile head from cvs i got a depend error in order i copied it from one server to another and try to do a make
22:06.06moverwhat happen here?
22:06.16moverneed i do a make xxxx ?
22:06.17brad_msswHmmhesays: are they fax capable by any chance?
22:06.22DrmCtchror know of anyway to have asterisk play dialtone ?
22:06.27*** join/#asterisk santiago (n=santiago@63.245.87.180)
22:06.36QwellDrmCtchr: playtones(dialtone)
22:06.38Hmmhesaysyou can try to fax on it, but like any voip provider chances are you'll get fax failure
22:06.38Qwellerm, dial
22:06.42Qwelland, playtone
22:06.54DrmCtchrsweet thanks
22:07.03Qwellno, it is playtones
22:07.06brad_msswHmmhesays: well, they advertise the fax stuff on their website (and we do use it via the linksys device) ... seems to work fine
22:07.15brad_msswHmmhesays: what codec do they use, btw ?
22:07.17Hmmhesaysnod, if you got the bandwidth
22:07.20brad_msswHmmhesays: ulaw ?
22:07.29Hmmhesaysthey'll use whatever you want, but preferred is g711 on their network
22:07.44brad_msswok, cool
22:07.59Hmmhesayslet me rephrase, they'll use g711 alaw/ulaw and g729
22:08.11Hmmhesaysbasically whatever cisco does they'll use
22:08.12Equinoxhat is g729?
22:08.14brad_msswis g729 lossy ?
22:08.16Hmmhesaysyes
22:08.22Qwellaren't all codecs?
22:08.27Hmmhesaysyes
22:08.32Hmmhesaysg711 is uncompressed though
22:08.32Equinoxflac isn't lossy ;)
22:08.34movercan anyone gimme e hint please?
22:08.38EquinoxNOt that it's used in voip
22:08.40QwellEquinox: nor is flac a codec ;p
22:08.43brad_msswHmmhesays: g711 is uncompressed ?
22:08.44adelashey, why cant i make a phone call on my cisco phone, its connected to the aah box i know
22:08.48Hmmhesaysbrad_mssw: yes
22:08.48Qwella telephony codec anyhow
22:08.49EquinoxQwell- Well it could be ;)
22:08.51adelasi have the trunk setup, and so is outbond setup
22:08.54brad_msswdidn't realize that ... hmm
22:09.10Hmmhesaysalthough the sampling rate on it is shit
22:09.13adelasam i missing something here?
22:09.53adelasits like my asteirks box is not communicating with my phone/internet line
22:10.03adelasi did a fdw setup to test it out first
22:10.16Hmmhesaysdrop me a line if you want any more info brad_mssw i'm out of here for now
22:10.20moveri got all times a make: *** [.depend] Error 2
22:10.38brad_msswHmmhesays: thanks ... I may take you up on that when we switch over ;)
22:10.42*** join/#asterisk jkitchen (i=kitchen@tarnishing.microsofts.name)
22:10.46Hmmhesayslater
22:10.56jkitchenanyone having issues with broadvoice today?
22:11.12jkitchenmy dialplan was working fine with them yesterday.. and today my asterisk server keeps sending 404
22:12.39rabelaisjkitchen: today and almost everyday...just doesn't work
22:12.51jkitchenit's like they changed something though
22:13.06jkitchenbecause if I create an extension in my default context with my phone number.. it works
22:13.21rabelaisagreed...calls used to actually terminate to my box, today I'm just getting a busy signal if I dial in
22:13.24jkitchenbut i have been using this thing for the past several days without such a thing
22:13.49jkitchenso your machine worked yesterday, but not today?
22:14.00jkitchenyou should sip debug and see if you're getting a 404 as well
22:14.03rabelaiswell, I don't know...I don't think I got any calls in yesterday from that number
22:14.06jkitcheni keep getting sent to voicemail
22:14.14jkitchen'busy' voicemail
22:14.19jkitchenmaybe they did screw something up
22:14.24rabelaisI have voicemail turned off...so I don't know
22:14.26jkitchenyea
22:14.59jkitchentheir website is really slow today as well
22:15.14SkramXhello all.
22:15.37jkitchenrabelais: can you do me a favor?
22:15.40rabelaisI take it back...that call actually does get to my * box...
22:15.46jkitchenhrm
22:15.49jkitchendammit
22:15.53rabelaisbut after I send the ack, it dies
22:15.58rabelaisno response from them
22:17.23rabelaisjkitchen: do yourself a favor and look elsewhere for a provider...I struggled with them for over 8 months before I finally said screw it
22:17.49rabelaisthe only reason I still have my account is to hang onto my number...I'm in the last stages of a number transfer and I'll finally be free of them
22:17.58jkitchenLooking for 7144632069 in from-bv
22:17.58jkitchenReliably Transmitting (no NAT):
22:17.58jkitchenSIP/2.0 404 Not Found
22:18.06jkitchenheh
22:18.21jkitchenknow of any asterisk-friendly providers?
22:18.35rabelaisas in iax?
22:18.42rabelaisor just asterisk friendly?
22:18.45jkitcheniax would be nice, but not necessary
22:18.51jkitchenjust... more asterisk friendly than say vonage
22:18.51rabelaisI went to telasip
22:18.52jkitchenheh
22:19.25rabelaisseems to be only one guy running the show, name's gene...he talks alot, but he's nice...and the service is stable
22:19.47mishehubah.
22:19.56rabelaisthe only problem I have is that I'm on the west coast, and a quirk about media path redirection hasn't been worked out yet...
22:20.34jkitcheni'm on the west coast as well
22:20.35*** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net)
22:20.36rabelaisso sometimes call quality gets weird during heavy traffic...we've been working on figuring it out, I'll be really happy when that issue is solved
22:20.59jkitchengrrr
22:21.25jkitchenthis thing is pissing me off... i'm new to asterisk.. here I think I have it almost working.. and now I can't get it working at all
22:21.33zedkatufjkitchen
22:21.35zedkatufHi
22:21.41jkitchenhi.
22:21.45zedkatufI found myself typing grrr on another channel at same time
22:21.50jkitchenheh
22:21.51zedkatufI'm new to * as well
22:21.53zedkatuf:)
22:22.06zedkatufI'm just an ebd user who wants to ditch his landline
22:22.23zedkatufso my issues are somewhat simpler than a lot of folks
22:22.28jkitchenmy boss wants automated call handling
22:22.33jkitchenso i'm setting up an asterisk box
22:22.38rabelaisjkitchen: if you want to test things out, setup a fwd account for yourself as well, that'll help you determine where things are funky with your configs versus broadvoice's general flakyness
22:22.44zedkatuf..yeah, that's more tricky...
22:22.55jkitchenzedkatuf: i had the dialplan working
22:22.59zedkatufre testing: I've got outbound calling working
22:23.01jkitchenbut today.. I called .... and got nothing
22:23.05zedkatufwierd
22:23.16*** join/#asterisk trash_ (i=trash@databerlin.org)
22:23.18zedkatuftry another ITSP mebbe
22:23.20jkitchenyea outbound calling still works for me
22:23.30pc4How do you fix the good old they can hear me but I can't hear them syndrome?
22:23.32pc4=)
22:23.47zedkatufpossibly oopen up more ports on ur firewall
22:23.49jkitchenoutbound calling works perfectly, in fact
22:23.59jkitcheninbound calling was working, perfectly, yesterday
22:24.00pc4zedkatuf - What am I missing?  I opened 5060 incoming.
22:24.08jkitchentoday my * keeps giving a 404
22:24.17zedkatuf5060 & 5061 UDP
22:24.25*** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net)
22:24.25*** mode/#asterisk [+o bkw__] by ChanServ
22:24.28zedkatuf10000 - 20000 UDP
22:24.34zedkatufpossibly also
22:24.42nick125_lappyanyone here know of a good sip client for linux other then x-lite that works with asterisk?
22:24.43zedkatuf3478 - 3479 UDP & TCP
22:24.51zedkatufkphone works gr8
22:24.55pc4zedkatuf - Incoming or only outgoing?
22:24.57DrmCtchrthere is no tones called dialtone
22:24.58jkitchennick125_lappy: kphone seems to work.. but x-lite works awesome
22:25.25SpaceBassnick125:  iaxcomm
22:25.28rabelaisnick125: I've heard good things about sflphone...
22:25.29nick125_lappywell, my friend is having a problem with x-lite on his computer, and we cant seem to get it working right :/
22:25.36rabelaisI just can't get it to compile   ;P
22:25.38SpaceBassgnomconf or what ever works too
22:25.45jkitchengnome-meeting ?>
22:25.48SpaceBassthats the onew
22:25.51jkitchen:)
22:25.54*** join/#asterisk tmx (n=tomx@dot.snat.baz.cz)
22:25.56jkitchenyea, I don't run gnome though
22:26.05jkitchenkde 4tw
22:26.14SpaceBassiaxcomm
22:26.20nick125_lappyi think its dtmf problems, but, his settings are exactly the same as mine for dtmf
22:26.38zedkatufnick125_lappy:
22:26.45zedkatufxlite probs under linux or winblows?
22:26.50nick125_lappyhes in linux
22:27.55nick125_lappyso am me
22:28.00nick125_lappyso am i
22:28.01nick125_lappybleh
22:28.03SpaceBassis he on the same lan as the * box?
22:28.09nick125_lappyno, its over the internet
22:28.22SpaceBassswitch to iaxcomm use iax2... make your lives easier :)
22:28.30nick125_lappyit connects right tothe asterisk, voice works, it just wont send when he presses a button
22:28.33nick125_lappyis that a gui app?
22:28.43SpaceBassyeah
22:29.14nick125_lappylooks good
22:29.38rabelaisdoes gnomemeeting support sip?
22:29.44SpaceBassnot sure why my IRC client decided to make that an action....
22:29.52SpaceBassrabelais:  i believe so
22:29.55*** join/#asterisk jsaunders (i=js@70.70.74.152)
22:30.05zedkatufnick125_lappy: I tried xlite under linux....the sound is not quite working properly atm....using kphone works gr8 however
22:30.07SpaceBassrabelais:  i see it supported a lot
22:30.18jsaundersWhat are they main ways of passing dtmf...  inband, rfc2833, any others?
22:30.19SpaceBasskphone... forgot about that one
22:30.22nick125_lappyzedkatuf, i cant find out how to put the password in there
22:30.31zedkatufin kphone?
22:30.38nick125_lappyyeah
22:30.53zedkatufare u trying to register through asterisk?
22:31.08nick125_lappyim trying to register kphone on my asterisk box, yeah
22:31.29SpaceBassoutband
22:31.50zedkatufnick125_lappy: ok, username: is extension number (eg 500)
22:31.59zedkatufwhen u click on Register button
22:32.10zedkatufit'll pop up with a box for u 2 type pword
22:32.14nick125_lappyit will ask for the pass then?
22:32.16nick125_lappyok
22:32.46nick125_lappywhat about gnome meeting, anyone use that?
22:32.47ardqwlekjqwe
22:32.51ardsorry
22:34.04SpaceBassnick125_lappy i have
22:34.07SpaceBassbeen a while
22:34.18SpaceBassis your name nick?
22:34.28nick125_lappyyes
22:34.34jkitchenmy name is kitchen ;(
22:34.38jkitchensome foo has my nick tho
22:34.50pc4sip:1234@asterisk -- which one is the username/pass?  sip/1234?
22:36.15SpaceBassactually the computer is osx2.nsnet.com but the user that auto loggs in is kitchen
22:37.38EquinoxAnyone using polycom sip 1.5.x?
22:38.30nick125there we go
22:38.34pc4Has anyone hooked a cell phone to their * box for free calls with mobile to mobile minutes?
22:39.01*** join/#asterisk Himeko (n=himeko@S01060040ca128fc3.ed.shawcable.net)
22:39.07*** join/#asterisk nicox (n=nicox@h082218027030.host.wavenet.at)
22:39.32nick125pc4: that would be interesting to see
22:39.38nicoxhello guys, is there anybody in the channel who compiled asterisk-cvs-head in the last week?
22:39.50SpaceBasspc4:  www.slashdot.org
22:39.54SpaceBassarticle on that right now
22:40.23pbdNo, I think he's talking something slightly different.
22:40.31pc4SpaceBass - similar article, but he was talking about setting up his own cell tower =)
22:40.35pbdYou want to use your cell phone as an outgoing trunk from asterisk..
22:40.39jkitchenhrm.
22:40.48pc4pbd - Incoming trunk, out through cheap-o voip
22:40.52SpaceBasspc4:  yeah, but read the first coment... there is a thing called voiceblue or something that does what you are talking about
22:41.06pc4Well, it is a device that interfaces any cell phone into a pots, yes
22:41.08pbdin the slashdot article, they're setting up a gsm cell local to your house, and gatewaying the cell station out through VoIP so you don't use the minutes.
22:41.12pc4so I suppose you could make your own dialplan and do that.
22:41.19SpaceBassi've got a sprint pcs connection card... thought about trying to integrate it once... but im not ready to write linux voice drivers for it :)
22:41.20pc4But I was wondering if anyone has ever done it... and how bad is it.
22:41.30pc4SpaceBass - Are they free voice minutes?
22:41.31SwK[Work]hah
22:41.45SpaceBasspc4:  mine is unlimited voice and data
22:41.46pbdI've not looked at it much, but chan_bluetooth may do what you need.
22:41.58SpaceBassits not a bluetooth device :(
22:42.01SwK[Work]"we're not getting your dtmf" which "c=IN IP4 255.255.255.255" in the sdp for the rtp host
22:42.33nicoxhas anyone a compiling problem with cvs-head also?
22:42.36pbdWell then, there are some makers that allow you to plug your cell phone in and use it as an outbound device for a regular cordless phone.. something could probably be built from that with an FXS card.
22:42.45SpaceBassyeah
22:42.53pbdnico- I did last night, until I realized I had a munged Makefile.
22:43.35nicoxhow can i solve the problem?
22:44.01nicoxeverytime i try to compile it hangs
22:44.04pbdDepends on your problem, exactly.  You could try re-checking out head to a new directory, and copying over the Makefile- but my problem isn't necessarily yours.
22:44.26wunderkinnicox: check ps
22:44.31pbdI blew up my compile because the makefile I had somehow didn't have a reference to one of the more critical and new source files.
22:45.08nicoxhm, i heared the fourth time that cvs-head makes troubles
22:46.04*** join/#asterisk schwank (n=schwank@tidy.obscurity.org)
22:46.14nicoxi tried to download it i think 20 time ...
22:46.21schwankhow do I make a caller enter into voicemail after a timeout on a queue?
22:47.14wunderkinnicox: oh i just saw your post on the ml, maybe see if it gave you any  warnings about your version of bison?
22:47.32fileZX81 is a crazy nut
22:47.33schwankany takers?
22:47.35shido6dialplan logic
22:47.37shido6t,1
22:48.06shido6u have a s,1 in the context there?
22:48.12schwankshido6, then how to I have it go to different voicemails?
22:48.14schwankyes
22:48.20schwankI have 5 different queues
22:48.21schwankthough
22:48.29schwankin the same context
22:48.30nick125_lappyanyone know of a voip place that offers something maybe like 10 minutes free pstn or something so i can test my asterisk?
22:48.38schwankand I'd like them to go to different voicemails on fail
22:48.47schwank(I require internationalization)
22:48.53wunderkinnick125, voipjet gives you 0.25
22:49.29*** join/#asterisk ZX81 (n=ZX81@222-153-118-124.jetstream.xtra.co.nz)
22:49.44ZX81ok it's running again now
22:49.47ZX81call me back
22:49.48ZX81:)
22:50.01fileZX81: :P
22:50.10ZX81very strange - that registration was supposed to be to fwd
22:51.38schwankhmm?
22:52.02nick125wunderkin: thanks
22:52.37Drukenfwd... i have one of those... i think....
22:52.59*** join/#asterisk Blissex (n=Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
22:53.36hardwireI should maybe donate b/w-cpu to voip-info.org
22:53.39*** part/#asterisk ZX81 (n=ZX81@222-153-118-124.jetstream.xtra.co.nz)
22:54.10nick125ok i got a voipjet account, now how do i set it up with amp?
22:54.19Drukenhardwire: you use the wiki that much ?
22:54.24hardwireyes
22:54.26hardwireits like.. on me
22:54.33nick125is it a IAX trunk?
22:54.33nick125lol
22:54.41nicoxdoes anybody know about a problem to compile cvs-head?
22:55.05Drukeni must admit... i use the wiki often myself... not so much as before
22:56.06hardwirewell
22:56.08hardwireits my reference
22:56.23Drukenand what a reference it is....
22:57.30Drukenhardwire: you have any experince with the international calls?
22:59.18pc4Can anyone test a sip server for me?
22:59.29*** join/#asterisk buddho (n=buddho@host86-133-210-171.range86-133.btcentralplus.com)
23:01.38*** join/#asterisk surfdue_ (n=surfdue@user-0c6t1g9.cable.mindspring.com)
23:02.48*** join/#asterisk postel (n=jp@unaffiliated/postel)
23:04.26pc4Can anyone test a sip server for me?  please? =)
23:08.10Drukenpc4: for?
23:08.21hardwireI should learn to just offline it
23:08.21hardwiremaybe thats what I will do
23:08.21hardwirewget ahoy
23:08.21hardwiremut: mutilator: hows the life?
23:09.13hardwireDruken: newp
23:09.43*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
23:09.43*** mode/#asterisk [+o drumkilla] by ChanServ
23:10.01zoahey drunken sailer
23:11.28pc4Druken - Just make a phone call on it.
23:11.36*** join/#asterisk alexhopper (i=Alex@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com)
23:11.46pc4Druken - I'm not receiving audio from it.
23:11.57hardwireI had sushi
23:11.59hardwiremy mind is racing
23:12.04hardwireI think its the coffee.
23:12.26Drukenpc4 check your rtp
23:12.49schwankoh.
23:13.00schwankit wasn't working because modules.conf had noload app_voicemail...
23:13.00schwank:\
23:13.30pc4Druken - I don't know where to start :( -- I think it might be my firewall at work -- if I pm you some credentials, can you test it?
23:14.08hardwirewhats the sip fromuser?
23:14.10hardwirevs CID?
23:14.45Drukenpc4: i'll guarentee it's your firewall at work :)
23:14.54pc4Druken - IT could be the firewall on the server too =)
23:14.57pc4Druken - CAn you test it? :)
23:14.58zedkatufre modules.conf, should Icomemtn one of these out.....:
23:14.58zedkatufnoload => chan_alsa.so
23:14.59zedkatufnoload => chan_oss.s
23:15.15Drukenpc4: well... it could also be BOTH :)
23:15.15zedkatufIcomomtn = I comment
23:15.35zedkatufpc4: has Druken managed to test?
23:16.11surfdue_does anyone here know alot of HTMF, using the X-Lite client?
23:16.32zedkatufsoz no idea :(
23:16.34pc4zedkatuf - Nope :(
23:16.38nick125grr
23:16.43pc4Can anyone test a sip server for me?  please? =)
23:16.48zedkatufpsate me the settings again pc4
23:17.02nick125every time i try to call out, i get 'all circuits busy, try again later'
23:17.27zedkatufposs firewall prob nick125 (?)
23:18.20nick125how would i check if the firewall is blocking it?
23:18.24*** part/#asterisk santiago (n=santiago@63.245.87.180)
23:18.38nick125ok
23:18.46zedkatufnick125: does ur firewall have logs?
23:18.48*** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue)
23:18.49surfduesorry
23:18.57nick125i need someone to help me set up the truck and outbound in AMp for voipjet
23:18.58surfdueIf anyone replied please repeat
23:19.02nick125zedkatuf: hold on
23:20.30nick125zedkatuf: ok
23:20.33nick125what should i look for?
23:20.58DrmCtchrhow come there is such a big delay between:  -- Starting simple switch on 'Zap/48-1'
23:21.01DrmCtchrand : -- Executing Answer("Zap/48-1", "") in new stack
23:21.10surfdueAnyone, if they know dtmf or X-LIte please tell me i really need help getting dtmf working
23:22.24DrmCtchris there a configuration for simple swithc im missing?
23:24.24*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
23:25.43zedkatufnick125: um, not sure really..depends on urfwall I guess
23:27.13*** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net)
23:27.18*** join/#asterisk Moc (n=mochouin@modemcable203.101-70-69.mc.videotron.ca)
23:27.20Drukenif you bring a call in on a E1, do you have to do anything special to toss it back out the E1 ?
23:27.26Mocvacation are over ...
23:27.45MikeJ[Laptop]Druken, dial?
23:28.03Drukenwell, aside from that...
23:28.18Drukendoes it carry the original DID inside the stream somewhere?
23:28.29MikeJ[Laptop]not sure what your asking
23:28.36nick125will this work as the dial plan: 9|1.
23:28.50*** join/#asterisk Jzalae (n=sk@bb-205-209-93-139.gwi.net)
23:28.52nick125i want to dial out like this: 9<area><num>
23:29.17*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:29.23nick125without dialing the 1
23:31.51*** join/#asterisk SexyKen (n=ksandell@c-67-161-5-149.hsd1.ca.comcast.net)
23:31.56SexyKenAnyone know what this error means: 0819171852|sip |4|00|Registration failed User: 1207, Error Code:480 Temporarily not available
23:32.12MikeJ[Laptop]nick125, _9XXXXXXXXXX,n,Dial(1${EXTEN:1})
23:34.45nick125anyone have voipjet setup with AMP?
23:36.10Ariel_nick125, yes
23:36.24Ariel_works
23:36.47nick125like to lend a litle assistance? :)
23:37.00nick125i setup the trunk like this pretty much:
23:37.43*** part/#asterisk tubecat_ (n=ksoze@tidy.obscurity.org)
23:37.48nick125IAX2 trunk
23:38.15nick125outbound caller id: "sometexthere" <1234>, Maximum Channels: Blank
23:38.44nick125oh i think i might have found the issue
23:38.58*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
23:39.16nick125nope
23:39.17nick125ok
23:39.37nick125Dialing rules: 1XXXXXXXXXX
23:39.49nick125Outbound Dialing Prefix: blank
23:39.58nick125trunk name: voipjet
23:40.12nick125peer details: (the long thing voipjet gives out)
23:40.20nick125incoming is blank
23:40.29nick125register string: blank
23:40.36nick125whats wrong with that?
23:40.49*** join/#asterisk bjohnson (n=bjohnson@i216-58-59-6.cybersurf.com)
23:41.36Drukennick125: prefix would be 9
23:41.44Drukendialing rules would be XXXXXXXXXX
23:42.10Drukenwould allow 97055551212
23:42.28jarrodanyone gotten reinvite working with SER?
23:42.39DrmCtchrhow come there is such a big delay between:  -- Starting simple switch on 'Zap/48-1'
23:42.40pc4How can I see what password an incoming sip user is sending?
23:42.54DrmCtchrand the first thing in the icoming context
23:43.11nick125Druken: but, in the outbound rules, it takes the 9 as the outgoing prefix
23:43.37Drukennick125: oh
23:43.55Drukenthen yeah, blank, and 9NXXXXXXXXX
23:44.31Ariel_nick125, don't put nothing but the phone number with voipjet they don't deal well with names
23:45.17jarrodbah.. ser is not allowing reinvite to occur between this sipura and my cisco gateway
23:45.53MikeJ[Laptop]pc4, sip debug
23:46.11file[laptop]oh say can you "Segmentation fault."
23:46.28MikeJ[Laptop]yes
23:46.30MikeJ[Laptop]I can
23:46.52MikeJ[Laptop]I just did...
23:47.00file[laptop]I seg faulted a few times today
23:47.01file[laptop]silly code
23:47.17MikeJ[Laptop]your code, or other stuff?
23:47.21*** join/#asterisk roche (n=roche@obiwan.inalambrica.net)
23:47.23file[laptop]other stuff
23:47.29MikeJ[Laptop]what'd you find?
23:47.36file[laptop]ICD
23:47.39MikeJ[Laptop]hehe
23:48.33nick125Ariel_: ok
23:49.27nick125it still says all circuits are busy
23:49.50rocheHi People, sometimes  my asterisk show this error "chan_zap.c:5830 ss_thread: CallerID returned with error on channel "  what could be ?
23:49.53*** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com)
23:51.28nick125Ariel_: what did you put for register string?
23:51.39pc4MikeJ[Laptop] - Where does it show the password though?
23:51.40adelasanyone here use a mediatrix?
23:52.41*** join/#asterisk apardo (n=w0w0@125.Red-83-46-188.pooles.rima-tde.net)
23:52.54*** join/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com)
23:59.09surfduedoes anyone know dtmf
23:59.14surfdueor x-lite sip
23:59.57MikeJ[Laptop]dual tone multi frequency?

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