IRC log for #asterisk on 20090801

00:00.18luchand if i type them in really really fast it actualy works
00:03.06luchso i have about one second to type in my extension and thats it
00:23.30*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
00:25.17*** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net)
00:37.02andresmujicadone.  :) now.. why on earth i don't have the COMPLETECALLER logged..  :/  checking...
00:39.22*** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk)
00:47.06*** join/#asterisk davevg (n=davevg-b@75.97.64.33)
00:48.17*** join/#asterisk devyll (n=paul@89.36.24.2)
00:49.31devyllare Hangup Hangup() Hungup and Hungup() all correct and do the same thing ? (terminates the connection/channel ) (ex: s,5,Hungup ) ?
00:53.17WindowsUseriono mostp eople use Hangup()
00:56.50*** join/#asterisk neoalex (n=chatzill@cpe-72-225-191-51.nyc.res.rr.com)
00:57.31neoalexhello I need some help setting up voice mail to e-mail which was working fine up until a few months ago
00:58.18neoalexthe problem I'm having is that now I'm trying to set it up to send using ssmtp
00:58.34neoalexssmtp works fine when testing from command line
00:58.43neoalexasterisk doesn't send anything
00:58.57neoalexis there any logs I could look in to see if it's even trying?
00:59.07neoalexare* there
01:00.29WindowsUserchange mailcmd to be cat > /tmp/file
01:00.44WindowsUserand see if anything makes it there?
01:00.58WindowsUseror try turning up debug really high and see if that help
01:01.25neoalexI tried core set verbose 1000 that didn't help
01:01.37neoalexlet me try the dumping the mail like you suggested
01:03.01WindowsUserif the mailcmd sends anything to STDERR it shows up when I go asterisk -d
01:07.16*** part/#asterisk ruben23 (n=RPL@122.55.48.243)
01:07.39LantiziaHey I've upgraded our phones to the latest version 7 snom firmware but now even tho all the phones are registered no calls can be made... anyone heard about that?
01:08.12Lantiziagoogle gives me less than nothing on this one, we were on 7.1.30 which worked ok, but 7.3.23 is what they're on now
01:19.55*** join/#asterisk sacitec (n=sacitec@189.129.249.246)
01:21.23*** join/#asterisk joako (n=joako@opensuse/member/joak0)
01:22.41neoalexok this is a dumb question... I have asterisk started with asterisk -f
01:22.46neoalexhow do I kill it?
01:22.55neoalexwithout it restarting
01:24.40hardwirectrl-c
01:24.46hardwiredo I win?
01:25.24neoalexno
01:25.29sacitechello, i'm testing the beta version of chan_skype with asterisk 1.4.x. I'm following the pre-configured .conf file to set up my existing(and non logged) skype account, that i want to use as test scenario. When i load try to load skype user from CLI, i get this debug(http://pastebin.com/m76f630a5), and never get logged in. Anyone has succedes using chan_skype ? All comments are welcomed =)
01:25.35hardwireneoalex: you want it to die right?
01:25.39hardwirebut you don't want it to.. restart?
01:25.47neoalexyoup
01:25.52hardwireso you want it dead?
01:25.56neoalexyoup
01:25.58hardwireinit 0?
01:26.18neoalexthanks, very helpful
01:27.02hardwirewell if you can connect to it via asterisk -r
01:27.05hardwireyou can run shutdown now
01:27.43[TK]D-Fender<PROTECTED>
01:27.51[TK]D-Fendersacitec: want another? ;)
01:27.59neoalexI can, but no matter how I stop it, it restarts the process
01:28.09sacitecque ?
01:28.10hardwire[TK]D-Fender: luser-base is an RBL right?
01:28.11hardwire:)
01:28.21[TK]D-Fenderneoalex: Sounds like you're running safe_asterisk
01:28.32sacitecy lo que quisiste decir es..?
01:28.39neoalexthe process only says asterisk -f
01:28.53neoalexI think it's the startup script that starts it when the machine first boots up
01:28.58filesacitec: you can't use your existing account, you have to use one created from the business control panel
01:29.06[TK]D-Fenderneoalex: As launched by an init using safe_asterisk no doubt.  Look for THAT process
01:29.21*** join/#asterisk Alfio (n=Amunoz@adsl-54-80.tricom.net)
01:30.10neoalexno results for ps -ax | grep safe_asterisk
01:30.22sacitecfile: you mean the control panel that comes when you first install skype on you GUI system, or it's a special version ok skype ?
01:30.24[TK]D-Fenderneoalex: Stop grepping and look at EVERYTHING
01:30.36[TK]D-Fenderneoalex: * does not restart itself.  Some other script is diong that
01:30.44filesacitec: it's a web based interface on the skype website, I believe this is in the documentation
01:30.59neoalexI know but the only asterisk process I see says asterisk -f
01:31.51sacitecfile: thanks a lot !
01:32.33[TK]D-Fenderneoalex: passtebin the entire thing
01:33.38neoalexhttp://pastebin.com/m4439e75
01:34.01sacitecfile: have you got success using this chan ?
01:34.23fileI haven't touched it yet, but others have
01:34.38hardwire:D
01:34.48WindowsUserthe skype for asterisk?
01:34.55hardwireI'm going to try it out this weekend.
01:35.02hardwirejust to miff [TK]D-Fender
01:35.06WindowsUserworks for me on 1.4.26 and svn checkout of 1.6.1
01:36.12WindowsUserhi dont pm me, i rarely notice them
01:36.15[TK]D-Fenderneoalex: Go hunt down your init script and see what its calling
01:37.05neoalextrying to find it now, that's another problem I can't remember who's calling it
01:39.10manxpowerPersonally, I'd rather not have a way for Skype users to call me.  I try to use stuff that uses open standards like SIP.
01:43.37jayteeonly thing more pesky than Skype users are those damn Philistines!
01:43.57sacitecWindowsUser: did you get a successfully connection with skype user ?
01:46.39WindowsUseryea
01:51.30sacitecuhmm, any clue why i'm getting my debug and unable to connect ? http://pastebin.com/m76f630a5
01:51.53sacitecpassword is correct
01:51.59*** part/#asterisk manxpower (n=eric@84.sub-70-222-54.myvzw.com)
01:54.11WindowsUserthats a lot of gibberish
01:55.08WindowsUserwhat debug level is that?
01:55.32WindowsUserif you do "skype show users" are you logged in or out?
01:57.10*** join/#asterisk Alfio (n=Amunoz@190.94.56.23)
01:58.15sacitecthe higgest
01:58.19saciteclogged out
01:58.35WindowsUserdid you create the account via the BCP on skype.com?
02:00.56WindowsUserskype is being very particular on that point
02:07.05*** join/#asterisk BadHAL (n=nn@70-5-70-9.pools.spcsdns.net)
02:07.27neoalex[TK]D-Fender: I finally found it
02:07.40neoalexI have it in /etc/inittab respawn
02:10.30[TK]D-FenderEW
02:10.46[TK]D-Fenderneoalex: This is not the kind of thing you spawn in there
02:11.50neoalexyeah I know, it was probably 3 AM 2 years ago and I couldn't get the debian script to run properly, so this was just a dirty suppose to be temporary fix
02:12.28neoalexanyway, I've commented it out and did an init 2 again but it's still respawning
02:13.06WindowsUserread the manpage for init, you need to -HUP it or something
02:13.09neoalexany other way besides restarting the server because I'm sure fsck is hell bent on spending an hour to check my disk
02:13.12WindowsUserkill -HUP init
02:16.05neoalexkill -HUP 2 right?
02:16.08neoalexstill respawning
02:17.14neoalexgot it... init q
02:20.08WindowsUserhorray for man pages
02:20.23neoalexok, all this was so I can start asterisk -d
02:20.30neoalexbut still nothing when I leavea a voicemail
02:20.45neoalexand it did write the voicemail to the /tmp file
02:20.54neoalexwell the email I mean
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02:23.08*** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com)
02:24.22neoalexdoes anyone tried using ssmtp for voicemail to e-mail before?
02:30.13WindowsUserssmtp < /tmp/file
02:30.35*** join/#asterisk Alfio (n=Amunoz@adsl-54-45.tricom.net)
02:30.44WindowsUserwell, sendmail -t < /tmp/file i guess
02:31.05WindowsUserdamn thing should give some kind of error, or maybe the to address is wrong?
02:33.49joakoAnyone has used Asterisk with bluetooth mobiles? What is the best way to get it running from scratch (channel driver, asterisk version, etc)?
02:34.06*** join/#asterisk anthm][ (n=anthm@CPE-72-128-94-253.wi.res.rr.com)
02:34.31*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
02:34.33WindowsUserchan_mobile is in asterisk-addons i thought
02:34.49WindowsUserget 1.6.1.1 or 1.6.1 svn if you want to try out skype as well
02:35.45WindowsUserhrm i might have a svn checkout of asterisk-addons as well, odd
02:36.12joakoWindowsUser: Ok... I will see about that I haven't tried 1.4 yet. I just recall their being various implementations
02:36.35joakoThat also reminds me has anyone been able to link the Windows Live phone service to Asterisk? I am told it is SIP
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02:36.55*** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com)
02:37.27WindowsUseri dont know anything about that actually
02:37.35WindowsUseris that part of live messenger?
02:41.13grandpapadotHey guys, I have a debian 5.02 system running asterisk 1.4.26.  When transcoding ulaw->g726 I get garbled audio, transcoding everything else seems to work fine, any suggestions?
02:41.56joakoAre you using Sipura or Linksys devices?
02:42.27WindowsUserg726 is DECT
02:42.32grandpapadotasterisk->asterisk
02:42.53grandpapadotseems better with g726aal2
02:44.04joakoNot sure what to say I've had issues with Linksys phones and ATA that do support "G726" I believe there are some settings in codecs.conf I would check that and have you tried any other "compressed" codec e.g. gsm?
02:44.17grandpapadotYea, gsm works fine.. hrm...
02:44.36WindowsUseryou're going from ulaw to g726 in asterisk
02:44.41WindowsUserbut where does the call end up?
02:45.02grandpapadotpolycom(ulaw)->asterisk->g726->asterisk
02:45.40WindowsUserand the second asterisk plays to chan_oss or something?
02:46.16joakoPersonally I would use G729 on the polycoms and between asterisk machines.... I don't believe there is another common codec between the two
02:46.57grandpapadotg729 passthrough works fine
02:47.08grandpapadothrm...
02:49.00joakoAlthough I can't recall recent compatibility issues between dissimilar Asterisk versions, what versions are the 2 machines?
02:49.10grandpapadot1.4.26 on both ..
02:49.16grandpapadotOne is debian 5, the other is debian 4
02:49.28hardwirehi
02:59.58*** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica)
03:28.02*** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com)
03:38.16joakoSo chan_mobile only works with CSR chipset bluetooth, not broadcom?
03:39.40WindowsUseri guess?
03:40.04*** join/#asterisk Alfio (n=Amunoz@adsl-54-245.tricom.net)
03:40.10WindowsUserBus 002 Device 002: ID 0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode)
03:40.16WindowsUserI'll believe that
03:40.24grandpapadotHey guys, I thought ilbc wasn't included in asterisk after 1.2?  When I show codecs in 1.4.26, it's there...
03:41.04WindowsUsercore show translation
03:41.23WindowsUserprometheus*CLI> core show codecs
03:41.23WindowsUserDisclaimer: this command is for informational purposes only. It does not indicate anything about your configuration.
03:41.24grandpapadotIt's there but with a bunch of '-'
03:41.30WindowsUserthe - means its not available
03:41.35grandpapadotahh
03:42.49grandpapadotTnx.
03:45.22*** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk)
03:47.18joakoWindowsUser: Either way I don't think the mobile I intend to use will work! I was trying to pair it with the loaner car I had and it would pair fine but the car would never consider it connected.
03:47.36WindowsUserwhat phone?
03:48.25WindowsUseralso theres like 2 or more ways for a phone to connect to an audio gateway, theres the dumb headset profile that can answer a call, and a profile that allows dialing
03:48.48*** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar)
03:49.34joakoSamsung r210... it's a $50 phone for a $30/month unlimited service
03:50.16joakoThey had the nerve to ask me if I wanted to pay $5/month + $40 deductable for insurance!
03:51.05WindowsUserhahahaha
03:51.22WindowsUserwell how long is the contract for the phone?
03:51.32WindowsUseror no contract? ;)
03:51.53joakoNo contract... I could return the phone to the store and exchange it for another one if it didn't work
03:52.15joakoBut I would "have" to wait a month since you get the 1st month free when you activate a new line :)
03:52.27joakowww.metropcs.com FWIW
03:52.39WindowsUserwhat country? im in canada
03:52.54joakoCountry to your north
03:53.09WindowsUserrussia?
03:53.18WindowsUseror did you mean south? ;)
03:53.29joakoYes lol
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04:12.42*** join/#asterisk BeeBuu (n=beebuu@219.135.40.94)
04:14.05BeeBuuwhat's the command that will play my busy message before record a busy voicemail,voicemail(111@default,b) doesn't work,please help
04:15.51WindowsUserwhat does it do now?
04:16.18*** join/#asterisk joako_ (n=joako@opensuse/member/joak0)
04:16.30BeeBuuit can't play busy message before record
04:16.45BeeBuuand i had make the busy message
04:17.03grandpapadotVoiceMail(b111@default)
04:17.47BeeBuugrandpapadot: i had tried too~~
04:18.03grandpapadotThat's it.  Did you record your busy message?
04:18.19[TK]D-FenderBeeBuu: "core show application voicemail".  and show us that you made the recording and the failed attempt to enter the box playing the message
04:20.11BeeBuu[TK]D-Fender: voicemail(111@default|b)?
04:20.26[TK]D-FenderBeeBuu: "," not "|"
04:20.44[TK]D-FenderBeeBuu: and show us your failure
04:20.54[TK]D-Fendereealong with a dump of the VM folder.
04:21.17BeeBuu[TK]D-Fender: thanks.let me try
04:22.24BeeBuu-- Executing [111@sipuser:1] VoiceMail("SIP/1001-081de3f0", "111@default|b") in new stack
04:22.25BeeBuu-- <SIP/1001-081de3f0> Playing '/var/spool/asterisk/voicemail/default/111/temp' (language 'en')
04:23.10BeeBuu[TK]D-Fender: can you see "111@default|b"? but asterisk still playing temp
04:23.32BeeBuuwhat's problem now?
04:23.53grandpapadotHey TK, do you know the effective bandwidth usage of asterisk's GSM codec?  I know the codec calls for like a 13k bitrate, but I was trying to find the IP overhead.
04:23.54[TK]D-FenderBeeBuu: it will ALWAYS play the temp over any other message <-
04:24.21[TK]D-FenderBeeBuu: that made so you can override yourr normal message while on vacation and not have to re-record your orgianals upon your return
04:24.45BeeBuu[TK]D-Fender: so what can i do next?
04:24.48[TK]D-Fendergrandpapadot: 20kbps ~
04:24.53grandpapadotThanks.
04:24.56[TK]D-FenderBeeBuu: . DUH... REMOVE IT
04:25.05BeeBuuremove the temp message?
04:25.08[TK]D-FenderYES
04:27.19BeeBuu[TK]D-Fender: Done.it's OK. It mean the temp over other message! no document say this~~~
04:27.26BeeBuu[TK]D-Fender: thanks.
04:30.08grandpapadotHey! 1.4.26's moh=files now resumes where you left off, no more madplay, yah!!!!
04:31.28*** join/#asterisk joako_ (n=joako@opensuse/member/joak0)
04:36.30WindowsUserhrm
04:36.36WindowsUserdoes the spa3102 have on hold music?
04:37.25[TK]D-FenderWindowsUser: No, why would it?
04:37.36[TK]D-FenderWindowsUser: Since when does a phone generate its own music?
04:37.55WindowsUseri dunno
04:38.22jblackThe spa3102 is a phone?
04:39.08[TK]D-FenderATA
04:39.10[TK]D-FenderClose enough
04:39.34jblackOk, if it's anything like the SPA-8K, then it can do music on hold from a stream
04:40.05jayteetrue
04:40.12*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
04:41.16jblack[TK]D-Fender: Seems like a reasonable question to me...
04:41.16[TK]D-FenderBRB
04:41.32[TK]D-Fenderjblack: Again, what would the phone device generate it?
04:42.17jblackgives [TK]D-Fender a sad, but oddly comforting, look
04:42.17[TK]D-Fenderjblack: a PBX yes, but why a phone?  SIP phones & ATA's connect to more central system that are in charge of such things
04:42.17grandpapadotg711->gsm hardley has any cpu overhead, g711->ilbc is like 33% (on a 1.6GHz Atom N270)
04:42.29jblackWindowsUser: Check for streaming. It may be able to take music from a stream, and play that. Think "icecast" like stuff.
04:42.37[TK]D-Fendergrandpapadot: Low bits are not free bits :)
04:42.43grandpapadot;)
04:43.01jblackThose devices are a lot more than phones, tk. They're practically a mini-pbx.
04:43.07joako_jblack: SPA8000 is a gateway only, no? You still need some PBX between that and your other endpoints. Asterisk is a nice opensource PBX that supports MOH from streams :)
04:44.00[TK]D-Fenderjoako_: He knows these things already..
04:44.05jblackNope. Get a pile of sip accounts from various providers, and hook each one up to a different phone, with various customized ringtones, call waiting, three way calling, etc.
04:44.18WindowsUserwell i have a spa3102 and when i press flash on a call theres some moh action going on
04:44.25jblackIt's not _nearly_ as functional as *.
04:44.29WindowsUserand im not sure how to tell if its asterisk or the unit
04:44.30[TK]D-Fenderremembers jblack's start here, along with the curious fellow who recommended the SPA-8000 to him...
04:44.38jblackThat was you. :)
04:44.54[TK]D-Fenderjblack: OMG! \o/
04:45.04jblackI've had it for... maybe two years now?
04:45.17[TK]D-Fenderjblack: Easily.  How's the She-bitch doing?
04:45.46jblackIt's still a pita. fiddle in the web interface for too long, and it needs a cold start
04:46.00[TK]D-Fenderjblack: I was talking about the Ex ;)
04:46.00jblackMy bet is they cheaped out on ram
04:46.15joako_jblack: Better than grandstream... fiddle with it and you need to toss it out
04:46.30grandpapadotgrandscream sucks in every possible way
04:46.47[TK]D-FenderBRB...
04:46.52jblackLet's not play childish "let's call something by a similiar sounding derogatory name". That's so...
04:47.11grandpapadotgrandsucks
04:47.16grandpapadotgrandgarbage
04:47.24grandpapadotgrandcheapassbrokeshit
04:47.49*** join/#asterisk akant2 (n=chatzill@70-59-167-73.omah.qwest.net)
04:48.04jblackDude, you done insulting yourself?
04:48.19grandpapadotgrandjblackfanboy
04:48.22jblackCould have just said "old, fat and forgetful"
04:48.27grandpapadotlol
04:48.42jblackI'm not a granstream fan. Never had any of their stuff
04:49.09jblackI bet you picked on your brother's last name when you were a kid. :)
04:49.35*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
04:49.58WindowsUserfender i guess the SPA3102 knows to ask asterisk to put a call on hold?
04:50.18[TK]D-FenderWindowsUser: Naturally
04:54.38WindowsUserthat is so cool
04:56.41[TK]D-FenderWindowsUser: ATA is just like any other SIP phone, you just plug the handset onto it
04:59.20*** join/#asterisk kihote (n=chatzill@118.69.66.118)
05:04.12WindowsUseryea I'm slowly learning about SIP and asterisk
05:13.16*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
05:15.27thehargrandstream does in fact suck.
05:16.03theharwe use hundreds of 286's and not even grandstream can fix our bugs
05:16.11theharaltho they are bugs related to the sonus switch we use
05:23.43[TK]D-Fenderthehar: So what you're saying is you are incapable of buying a quality product?
05:23.44[TK]D-Fender:p
05:28.05theharheh
05:28.27theharwe are 78% done replacing them with pap2ts which aren't the best thing to replace them with but the cost benefit is much better
05:30.43WindowsUserwhat'd be the best thing to replace them with?
05:32.21neoalexI still can't get voicemail to e-mail to work, I've replaced ssmtp with msmtp and that works from console fine too, however mailcmd never seems to get called since there's nothing in the log
05:32.48neoalexit's exactly the problem described here: https://issues.asterisk.org/view.php?id=15199
05:33.04theharthey are a fine ATA
05:37.06WindowsUserneoalex: what user is asterisk running as?
05:41.25neoalexroot
05:43.02WindowsUserhrm it must be something simple
05:47.57neoalexcould this cause it: Prefixing the mailbox with an option is deprecated ('su8600')
05:48.02[TK]D-Fenderneoalex: Someone else's problem doesn't help us.  It went in and sat idle for months after 2 days of activity
05:48.42[TK]D-Fenderneoalex: And yes, not following instructions when they app tells you to your face should be a shoot-on-sight-offense :p
05:48.58[TK]D-Fenderreaches for his ClueBbat (tm)
05:49.13WindowsUserneoalex: but is it putting the voicemail into like /var/spool/asterisk/voicemail/context/8600/INBOX
05:50.37neoalexYes it does, and not only that, it also writes the email if I use: mailcmd=cat > /tmp/vmtest
05:51.29WindowsUserand then sendmail -t < /tmp/vmtest works?
05:51.41neoalexactually it's putting the voicemail in /var/spool/asterisk/voicemail/default/8600/unavail
05:54.17neoalextrying now
05:56.55neoalexit doesn't send the email no, but I think I see the issue
05:57.10neoalexthere's an _ in front of the recepient address in the log for msmtp
05:57.48neoalexI have no _ in front of the email address in voicemail.conf
05:58.06neoalexso I'm have no idea why asterisk is adding that
06:04.31*** join/#asterisk darksmurf (n=asdf@166.135.30.88)
06:05.52WindowsUseris the _ in front in the /tmp/vmtest?
06:06.50darksmurfack! problems. Setup: Asterisk 1.4, on DSL behind NAT with an IPMAP so all traffic seems to come from a specific public IP. Asterisk thinks it has a 192.x.x.x IP. I am trying to use externip to tell it the true pub IP, but when I 'sip debug' the peer I still see the 192.x.x.x IP listed in the INVITE. Is this correct? I am getting a 403 Forbbiden error when trying to connect to a SIP Trunk. (Velocity Networks if anyone has any suggestions about that...)
06:08.03neoalexnope, found the problem, in vmtest there was a space
06:08.10darksmurfI should mention that I think the reason for the 403 is because my provider will not try to connect to a private IP. They want to see my public IP in the invite. Sound plausable?
06:08.17neoalexin voicemail.conf I had a space after the comma
06:08.21neoalexthat was the issue
06:08.22*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
06:08.25WindowsUserhorray
06:08.28neoalexfinally works now
06:08.34neoalexthank you for all your help
06:09.03[TK]D-Fenderdarksmurf: PASTEBIN your configs and the SIP DEBUG of a failed call
06:09.05[TK]D-Fender~pb
06:09.06infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
06:09.18darksmurffender will do
06:12.13darksmurffailed call: http://pastebin.com/m73a4fba8
06:14.14darksmurfconfig is done via trixbox, what parts do you need? I have been able to use a friend's asterisk box work as a SIP trunk. The only thing I have done is add nat=yes and externip=99.x.x.x  to the general section of sip.conf (rather, trixbox's really convoluted sip.conf replacement using seemingly hundreds of #includes.)
06:15.24darksmurfusing extern IP I would expect to not see ANY references to the 192.x.x.x IP. Maybe I am wrong.
06:15.44[TK]D-Fenderdarksmurf: sip.conf and everything it INCLUDES
06:16.01[TK]D-Fenderdarksmurf: And you have done it wrong if the internal IP shows up
06:16.25darksmurfthats what I thought. stand by, this is going to take a bit.
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06:28.33darksmurfhttp://pastebin.com/m3c6c7382
06:28.35darksmurfconfigs
06:28.59darksmurfI did omit some .confs that were #included because they were empty
06:29.39drmessanoYoure not supposed to edit sip.conf in trixbox
06:30.19[TK]D-FenderDarkYou left out "localnet" <- * has no idea what is LOCAL so when would it decide what isn't?
06:30.26[TK]D-Fenderdarksmurf: You left out "localnet" <- * has no idea what is LOCAL so when would it decide what isn't?
06:30.40darksmurfdrmessano, I didn't, I edited the file sip.conf says to edit, sip_general_custom.conf
06:30.42drmessanoyup
06:30.52drmessanook
06:30.59darksmurflocalnet would be 192.168.1.1/255.0.0.0 ?
06:31.06darksmurferr..duh
06:31.14[TK]D-Fenderdarksmurf: I highly doubt that netmask
06:31.26darksmurfyeah...
06:31.30drmessanoI completely doubt the netmask
06:31.40drmessanoand the IP
06:31.43drmessanowrong and wrong
06:31.55drmessano192.168.1.0/255.255.255.0 perhaps
06:32.18darksmurfthat looks better
06:32.34drmessanoor "correct"
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06:36.46darksmurfthe 'contact' line in the invite is correct now, thanks. Now I am getting a 404 error. I think that may be because I am not sending a full 10 (11?) digit phone number.
06:40.14WindowsUserprobably
06:41.01WindowsUserI know flowroute requires 11 digit dialing for north america
06:41.29WindowsUseryou can append digits via the dialplan if you need to btw
06:42.27[TK]D-FenderWindowsUser: ....TRIXBOX
06:43.04[TK]D-Fenderdarksmurf: As of now...
06:43.06[TK]D-Fender~trixbox
06:43.07infobotextra, extra, read all about it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
06:43.37darksmurfGOT IT. Thanks everyone. Now it is time to play with the dial rules.
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06:45.08darksmurfah, sorry about that. I shall head over there from now on.
06:45.43darksmurfStill, thanks for the help all the same.
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07:35.09MiccI can't use some things because it says I don't have a timing device. Is there any way to get around that without buying special hardware?
07:35.34MiccI have the dahdi_dummy driver installed.
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08:41.59kron4egwhat is the difference between 1.4 and 1.6 branches?
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09:33.53svm_invictvsHiya
09:34.29svm_invictvsSo I'm setting up some hosted PBX service w/ asterisk.  I have a SIP client channel set up, and I'm looking to set up like 4 extensions for individual offices.
09:34.39svm_invictvsShould I enable or disable "allowguest"?
09:52.05*** join/#asterisk viq (n=viq@unaffiliated/viq)
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10:04.30*** join/#asterisk phryk (n=phryk@yggdrasil.phryk.net)
10:04.40phrykMornin'
10:05.39phrykI'm moving into my own place soon, and I thought about setting up an asterisk daemon on my miniserver, what exactly do i need? Anything besides a working linux and internet access?
10:07.50kron4egTDM card maybe?
10:08.25phrykwhat is that?
10:09.43kron4egcard to connect your asterisk server with PSTN
10:10.13phrykpstn is the "normal" telephone net?
10:10.55kron4egyep
10:11.13kron4egline from your local telco
10:11.27phrykIf I have a provider that grants me a voip flatrate, can i somehow go through that?
10:11.49kron4egsure
10:11.50phrykI mean if I get that for free, why should I pay for plain telephone services :)
10:11.56kron4egpstn jusn an option
10:12.18kron4egphryk, no reason to pay :)
10:12.21phrykSo with voip i don't need a tdm card but still can phone to the normal telephone net
10:12.35phrykkron4eg: I wouldn't have to pay, even if i would go through the ptsn?
10:14.24kron4egyou have to pay your local telco if you using it's line, but, since you not planning to use it, you don't have to warry about it
10:15.03phrykokay
10:17.23phrykI think I'll construct some weird kind of hands-free speaking system :D
10:20.50svm_invictvsYou could just get a SIP provider, too
10:23.41kron4egOMG
10:23.48svm_invictvs?
10:23.49kron4egjust discovered Adhearsion
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10:30.12phryksvm_invictvs: If my internet provider provides voip, i should have sip access, right?
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10:41.09box2sip is not for the feint of heart
10:41.54phrykwhy not?
10:42.08phrykit's similiar to http and smtp, so it can't be that hard to understand...
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10:48.35mechbangircis there anyway i can define global variables in a separte file and then use them in my dialplan?
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10:49.49LantiziaLo, since upgrading my snoms to the latest version 7 - they still register but have status UNREACHABLE... any tips on what I should check for?
10:52.09box2phryk: sip is the leading cause of stroke, diabetes, and road rage
10:52.48box2you may grow to understand it, but the DANGERS INVOLVED
10:52.51box2shudders
10:53.44phrykI have people in my im contactlists, who sent me chain mail.
10:54.03phrykI haven't turned mad, yellow and stinky yet.
10:54.07phrykI am fucking invincible.
10:56.07box2heh
10:56.37box2i will sing a nice tune for your eulogy
10:58.27phrykI know a shaman. He will revive me.
11:00.29box2shamans are crafty with their legal documents, be sure not to sign anything which leads to you being revived as a zombie slave
11:00.41box2that shit happens a lot more than one would think
11:01.13box2or at least have a roomate co-sign for you, as collateral
11:01.36phrykI already did that.
11:02.09phrykHe is my slave. I once killed him and revived him. He is not able to do harm to me.
11:06.39box2that's awesome
11:06.43box2i wish i could do that
11:07.36phrykTrust me, that was damned hard work...
11:08.08phrykKilling a Voodoo-Shaman isn't as simple as it sounds...
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11:44.09yahhHi guys
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11:46.31yahhI want to use asteriskwin32 on my windows system
11:46.35yahhwith the FXO card
11:46.57yahhAnyone have experiance with it?
11:48.49Alfioyahh asterisk for windows its not recomended
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11:53.18yahhAlfio: why..?
11:53.44yahhi know it is made for linux
11:53.51yahhand made work with windows
11:54.02yahhthats why it is better to use in linux
11:54.35yahhbut is there any specific issue......then tell me plz...
11:54.49box2not to sound like a fanboy or anything, but windows just isn't fun to use
11:56.34PsychobillyAsteriskWin32 0.66b build from Asterisk 1.2.26.2
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11:57.21Psychobillyyahh its just outaded and judging by the look of the web page it must also be totaly unmaintained
11:58.08Psychobillylast update was on jan 2008
11:58.16yahhhmm
11:58.37yahhthat version must be working
11:59.00Psychobillysure :P
11:59.31yahhit is not updated, but upto 1.2.26.2 is should be ok
11:59.56Psychobillygo with linux thats the advice u will get here
12:00.42yahhplz give me guidance,  i wanted to use it in windows
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12:02.17Psychobillyi dont think anyone here has ever used this
12:03.08Psychobillyfrom my little experience with * running it on windows sounds just wrong, especially such and old unmaintained version
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12:17.46errotanyahh even if asterisk would work in windows the FXO card definietly won't
12:18.39yahhbut digium card should supported to asterisk-win
12:19.39errotanwhy do you want to run it on windows anyway ? if you have some programs that are made for windows you should start windows in a virtual environment (like virtualbox)
12:20.41yahhmy friend is going to use that and he is not knowing Linux
12:21.04yahhHe just knowing windows
12:23.11Psychobillyeven better, u will install it and he will never touch it
12:23.17Psychobillyit will be safe this way
12:26.59jayteedigium cards aren't supported on AsteriskWin32. the driver mods are only available for linux, not Windows
12:27.54jayteeaccording to some sources AsteriskWin32 was started as an April Fool's joke that got out of hand and now won't die
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12:29.03yahhare you sure...digium FXO cards is not suppored?
12:29.08Psychobillylol
12:30.38jayteeyahh, unless someone has taken zaptel code and ported it to the Windows driver spec then no.
12:31.26yahhokay...
12:31.56Psychobillyyahh maybe u can try some pbx software for windows, like yate or freeswitch
12:32.06Alfioor 3cx
12:32.12Psychobillydont know how good they are though
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12:38.28*** join/#asterisk seth911 (n=Seth@C-59-100-13-206.for.connect.net.au)
12:39.19seth911i have a phone line plugged into my modem i am trying to make calls from it, which works, but no sound is being sent or recieved
12:39.33seth911would anyone know the solution or problem?
12:40.45yahhthank you Psychobilly
12:40.54yahhthank you all
12:41.47yahhanyone have experiance with freeswitch or yate for windows?
12:41.47errotanfreeswitch works on windows well
12:41.51jayteeyahh, try asking in #freeswitch or #yate
12:42.54yahherrotan: with FXO card
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12:43.15errotanFXO card ?
12:43.47yahhA card for PSTN line
12:44.24errotani know what is an FXO card but on windows you can't find any working card
12:45.07yahhany specific reason?
12:45.09errotanyou need to use voip gateways if you want to work with windows
12:45.39errotanreason: they are made for linux apps :)
12:46.13yahhno single card with drivers for windows?
12:48.37seth911anyone?
12:48.45errotani never heard of any, there are not so many pci fxo cards on the market
12:49.18jayteethere's two particular reasons why the vast majority of pci FXO or FXS cards don't have drivers for Windows. One is the cost of developer tools to write device drivers using the Windows DDK and the other is LATENCY.
12:51.28jayteecommercial phones systems from companies like Nortel or Avaya don't use Windows as an OS. Nortel uses a customized kernel based on SRV5 Unix. stripped down to be minimal and run with very low latency.
12:52.52yahhi see
12:53.12jayteeWindows system developers have never learned to develop minimal stripped down versions of their OS. Some people think they even strive for bloat in order to drive people to upgrade memory or get a faster CPU so their investments in companies like Intel and Micron will reap extra revenue.
12:53.59yahh:)
12:54.17jayteeeven linux with a GUI like Gnome running on it tends to introduce too many interrupts (latency) which makes the earlier cards from Digium and Sangom have problems due to missed interrupts.
12:56.38*** join/#asterisk Katty (n=Katty@mail.copi-rite.com)
12:56.58Kattyhi! :>
12:57.24jayteemorning Katty *hugs*
12:57.38Kattyherroes
12:58.19KattyDo you know how to change nicknames using nickserv? I can't seem to use /nick with jmirc
12:58.58*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
12:59.44jayteenope, I've always just used /nick with Xchat and even with mIRC on Windows I'd use /nick
13:00.06Kattyhmmmmmk
13:00.14jayteenever used jmirc
13:00.50hescowhat application will wait for multiple digits?
13:00.57jayteeI prefer Xchat since I can use any perl or python script plugin
13:01.35KattyWell I prefer irssi, but its not blackberry friendly
13:02.21*** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk)
13:02.24KattyHesco read()?
13:02.25jayteehesco, WaitExten
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13:03.15Kattywaitexten is good
13:03.19hescothanks, will try read() then.  WaitExten did not quite work for me for some reason
13:03.36hescoand it is not an extension I am waiting for but an account number
13:03.37*** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net)
13:03.45KattyYou have to tell waitexten how long to wait
13:04.06Kattyand then what to do next, obviously
13:04.22KattyAhhh, k
13:05.33*** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar)
13:06.05jayteehesco, if  you're waiting for the caller to input an account number then use Read(). Read() will allow you to assign what the caller enters to a variable that you can then handle in your dialplan any way you choose.
13:06.49KattyI'd recommend rerouting all calls to jaytees zoo
13:07.09KattyWe can put a phone in with the monkies
13:07.13hescothat ought to take the load off our overworked staff, eh?
13:07.22hescothanks jaytee and katty
13:07.34jayteesets up a macro to route all incoming calls to Katty's blackberry
13:07.37hescoreading the show application read now
13:07.49Katty)=
13:08.55KattyNo calls today! Me and mom are out yard saleing
13:09.30jayteealways buy the hummels. good investment strategy :-)
13:09.58KattyHummels?
13:10.42jayteehttp://www.hummelsatadiscount.com/
13:11.11KattyBrb
13:12.01*** join/#asterisk Katty (n=Katty@mail.copi-rite.com)
13:12.42KattyHard to switch apps on this thing
13:26.30hescoMy NoOp is printing to the *CLI: (NoOp) Options: (The Account Number is: $ACCOUNTNUMBER . . .
13:26.53*** join/#asterisk [netman] (n=netman@107.Red-88-8-164.dynamicIP.rima-tde.net)
13:26.59hescohow do I get it to interpolate that variable before printing it out?
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13:28.13jayteeNoOp(The account number is ${ACCOUNTNUMBER})
13:29.33jayteehesco, one of these days when you're really really bored you might take a look at the channelvariables.txt file in the tarballs. full of good information
13:30.25hescoshould have mentioned, its being called from an agi script
13:31.18jayteesorry, AGI not spoken here
13:33.13hescoI found: doc/api/html/chanvars_8c.html, is that what you are talking about?
13:33.42jayteenope
13:34.22jayteewhat version are you running?
13:36.32hesco1.6.0.3-rc1
13:36.49hesco(NoOp) Options: (The Account Number is: ${ACCOUNTNUMBER})
13:37.03grandpapadotHey guys, VoiceMailMain has a parameter "p" that is suppsed to "Consider the mailbox parameter as a prefix to the mailbox that
13:37.03grandpapadot<PROTECTED>
13:37.08grandpapadotshit, sorry
13:37.09hescothat is the result whether or not I escape the { and }
13:37.58grandpapadotHey guys, VoiceMailMain has a parameter "p" that is suppsed to "Consider the mailbox parameter as a prefix to the mailbox that is entered by the caller."  However, it appears in 1.4.26 that parameter is active no matter what.  Anybody else experience this?
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13:49.17dexterukDoes anyone know where i can get some help with a2billing?
13:52.50hescojaytee: found: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+channelvariables.txt
13:53.04hescoand references in google to this file missing from 1.6.0.9
13:53.30jayteehesco, yep and in 1.6 it's in /doc/tex in the tarball and it's a .tex file not a text file.
13:55.29*** join/#asterisk Katty (n=Katty@mail.copi-rite.com)
13:56.41KattyI found a leppard print rug and two stuffed dinos for riddick so far :>
13:57.05*** join/#asterisk pointer (n=pointer@aj.catt.com)
13:58.20grandpapadotOk, when compiling on the Intel Atom N270, the i586 flags need to be set for the transcoder to work properly.
14:00.50jayteewonders how many concurrent calls a Intel Atom can handle
14:01.16pointerhrm.  I've been busy for a while and just realized that my outbound provider is gone (nufone).  Any recommendations for another provider that will let me set outbound CID (forwarding calls from another sip provider to my mobile)?
14:02.08grandpapadotI'm actually testing that, I'm able to sustain around 80 g729->g729 and about 120 g711-g711 and about 60 g711->gsm and about 10 (lol) g711-g722 ... 1.6GHz, 1GB
14:02.21pointerand by a while, it looks like I mean months :-\
14:03.09grandpapadotWith the exception of gsm, anything transcoded on the Atom N270 just kills it.
14:03.50grandpapadotBut we're using it for a endpoint aggregator so transcoding isn't necessary.
14:03.55*** join/#asterisk devyll (n=paul@89.36.24.2)
14:05.58devyllany ideea why navigating in an IVR is really crappy from mobile phone ? some times (lot of times) when I press a key it doesn't do nothing  ... i have to wait untill the end of the Background(audio) to be able to choose an option from the ivr  however this is happening randomly .. sometims it works with the mobile phone too .... from a local phone is working just fine .. am I missing something ?
14:07.41grandpapadotAll, I'm looking for a way to evaluate a string and see if it's a number.  I can do it by letting MATH() fail, but I was looking for a cleaner (non-hacker) way ...
14:07.45grandpapadotIn 1.4 dp
14:08.13KattyPointer, we use bandwidth.com
14:08.31pointerKatty: thanks!
14:08.36*** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be)
14:09.32pointerKatty: do they have a pay as you go or do you use the prepaid/monthly plans?
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15:09.54LantiziaAnyone had issues with snoms behind a NAT and asterisk since firmware 7.1.33?
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15:23.46hescojaytee: thanks, found it.
15:24.23hescohow was it again I convert .tex to .pdf?  Was it dvips file.tex; ps2pdf file.dvi ???
15:24.54hescoI seem to think there was a third step along the way, but I can't for the life of me remember what that might have been.
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15:53.54SkramXanyone know the voicexml syntax to have the system say "first", "second", "third" for 1,2,3?
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16:13.56VladTheImpaledNeed some advice on dial plan strategy.  I have one ITSP but multiple DIDs (4).  I need to control incoming calls.  I've put the full details of the problem at http://pastebin.com/m1e1eaea1
16:14.34VladTheImpaledThis is with Asterisk 1.4.26
16:16.48WindowsUsergoto() a different context
16:17.02PsychobillyVladTheImpaled make seperate contexts for each line
16:17.55VladTheImpaledPsychobilly:  Thanks.  Yes, that is what I thought.  But my problem is that how do I handle that in sip.conf?  The ITSP is telling me to have one sip.conf for their service, which ties to one DID
16:18.06WindowsUsergoto() a different context <---
16:18.12Psychobillywhat WindowsUser said
16:18.35WindowsUserexten => 5555551212,1,Goto(wakeupcall,797,1)
16:19.05VladTheImpaledAahhh.. I think I see.  So basically the incoming sip.conf is singular, and I do some sort of IF statement on $EXTEN to work out where the call was supposed to go?
16:19.29WindowsUserif you want
16:19.46WindowsUserbut I'd goto right at the start
16:20.42WindowsUserexten => 2125551111,1,Goto(CompanyA,${EXTENSION},1)
16:20.48WindowsUserexten => 2125551112,1,Goto(CompanyB,${EXTENSION},1)
16:20.58WindowsUserand then write up CompanyA and B contexts
16:21.15VladTheImpaledthat's exactly what I was looking for
16:21.32VladTheImpaledThanks so much for this.  I'm still trying to get my head around it all, but gradually its sinking in
16:22.50[TK]D-FenderVladTheImpaled: exten => 2125551111,1,Goto(CompanyA,s,1) <- should run IVR's off non numeric extens so they can't be dialed recursively.  Thats what "s" is for
16:23.14[TK]D-FenderVladTheImpaled: So no need to make variable in your Goto.  then just make your "s" exten in [CompanyA]
16:24.41VladTheImpaled[TK]D-Fender:  So if I use s for incoming DID reference, I just reference s as the exten in the other contexts?
16:26.39[TK]D-FenderVladTheImpaled: No, in your case your ITSP dials an extension matching the DID that arrived to them.  You match on this, and YOU jump to separate places to process as you wish inside your dialplan
16:26.57[TK]D-Fender~stdextens
16:26.58infobot[~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf.  "s" is also used to make IVRs & macros.
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16:31.46[TK]D-FenderVladTheImpaled: Contexts are import security separations that control what can be done when a device points to one, or when you jump into one.  In a multi-level IVR you'd have each sub-menu in its own, etc.
16:32.39[TK]D-FenderVladTheImpaled: For many purposes you'd put the extens that would dial phones, etc often in 1 context, and INCLUDE it in IVR's, and in "collector" contexts that also include contexts that have the extens that let you dial out, etc
16:33.03VladTheImpaled[TK]D-Fender:  Yes, that is what I thought.  I can definitely see how it separates the calls into their appropriate areas.
16:35.23VladTheImpaled[TK]D-Fender:  However for some reason my incoming calls are not being handled and put to their appropriate context.  I've pasted what I've done at:  http://pastebin.com/m621b5612
16:35.45VladTheImpaledam I missing something in the 'incoming' context for this?
16:36.03[TK]D-FenderVladTheImpaled: Yeah, thats good for each DID that you can ID that way.  Now the real work occurs in those other contexts
16:37.43VladTheImpaled[TK]D-Fender:  right, but its not working.  I'm getting an error of:  [Aug  1 09:30:13] WARNING[30507] pbx.c: Channel 'SIP/sip.broadvoice.com-089cc4c8
16:37.43VladTheImpaled' sent into invalid extension 's' in context 'edgeneering', but no invalid handl
16:37.44VladTheImpaleder
16:38.37VladTheImpaledIn the context that receives the call, I'm just doing stuff like this:  exten => s,1,Answer(100)
16:39.12VladTheImpaledDo I have to be specific to the DID in the 'child' context that it is being sent into?
16:45.26VladTheImpaledhang on, fixed it
16:45.42VladTheImpaledmy bad....  I had some legacy mess in there from past attempts but its working now
16:45.48VladTheImpaledThank you all for this!!!!!!
16:46.53[TK]D-FenderVladTheImpaled: You're welcome, and good to hear
16:50.15[TK]D-Fenderis off to enjoy the sun... while it lasts...
16:50.17[TK]D-FenderBBL
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16:56.41LantiziaHey, I'm having a weird issue where if my snom phone is on 7.1.33 version or higher... it registers... but on the asterisk CLI you see it retransmitting 4 times and then "really destroying SIP dialog" over and over
16:57.11Lantiziaany phone on this version or higher can't make or receive calls - but they are registered... the pbx is not behind a nat but the phone is ... but it's never been a problem till now
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17:01.31grandpapadotHey guys is there a way to "see" the generic jitter buffer in action?
17:02.16carrarIf you're Neo
17:03.12grandpapadotOr at least verify it's actually doing something, lol
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17:13.04devyllany reason for which trying to choose an option from an ivr ex:   3,1,Goto(blabla) does not always work when calling from mobile phones ? this problem occurs only with mobile phones ...
17:13.08devyll?
17:14.00devylland when it does ...  it seems i still can choose option number 3 but only after the Background file is finished playing or .. only after a certain amount of time from when the play started
17:14.11devyllany help on diagnosing this problem ? don't know where to start ...
17:15.50carrardevyll, it's a mobile, they don't always have great audio
17:16.05carraror they don't press the DTMF long enough
17:16.24grandpapadotWill jbenabled=yes not show up in sip show peer output?
17:16.44carrardevyll, use background instead of playback
17:19.11grandpapadotIs there any way to verify a jitter buffer is working in 1.4.26?
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17:29.41LantiziaOK I've got a system running 1.4.19.1 and I want to upgrade... which file (is it that makeelect thing?) do I keep so that when I recompile from new source it does the same config as the last time?
17:34.55WindowsUsertry makeopts
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17:39.21svm_invictvsphryk: Yeah.  If not SIP they're providing *some* type of VoIP.  The thing is, some ISPs don't like you hooking up a PBX to their channel.
17:40.05svm_invictvsphryk: Likely it's SIP.  But, once again, they may somehow block you using asterisk because they want to charge for hosted PBX
17:40.18florzwell, and some isps don't like you actually using their bandwidth ... so what? =:-)
17:40.56florzthat someone wants to take advantage of you doesn't really oblige you to much, does it?
17:43.41grandpapadotlol
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17:50.13hescoI'm working on an AGI script which includes these two lines:
17:50.16hescoprint "EXEC Read ACCOUNTNUMBER \"\", 6, \"\", 1, 4 \n";
17:50.25hescoprint "EXEC NoOp \"The Account Number is: \${ACCOUNTNUMBER} \" \n";
17:50.43hescowritten in perl, in case that was not obvious.
17:51.45hescohowever, the *CLI reports:
17:51.48hesco-- User entered '33333'
17:52.19hesco-- AGI Script Executing Application: (NoOp) Options: (The Account Number is: ${ACCOUNTNUMBER} )
17:53.00hescowhat am I missing here that is preventing the proper interpolation of my user defined variable.
17:53.11hesco???
17:53.32Qwell\
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18:04.07hesco~paste
18:04.07infobotextra, extra, read all about it, paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/
18:07.01hescoI have an AGI question, pasted at: http://bin.cakephp.org/view/1745668161  Can anyone here please offer some guidance?
18:08.20WindowsUserwhy are you reading a number into a variable asterisk side and then trying to get it?
18:08.32WindowsUsertry READ instead of EXEC READ
18:08.46hescook, trying that now
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18:10.29hescoprint "READ ACCOUNTNUMBER \"\", 6, \"\", 1, 4 \n";
18:10.40hescodid not even pause to have me enter the number
18:11.13hescowith EXEC READ it did slow down to do the READ function
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18:11.44hesconone of my attempts to print AGI functions directly to STDOUT have succeeded, unless wrapped in the EXEC.
18:11.45WindowsUserwell, you're trying to write an AGI like you're in the dialplan
18:12.04WindowsUserare you reading in stdin?
18:12.29hescoIn a dialplan I would have written that as: Read(ACCOUNTNUMBER,'',6,'',1,4)
18:12.31wwalkerI've looked around a lot and can't find a decent comparison of app_conference and app_meetme.  Does anyone have a pointer to a comparison of the two?
18:12.35LantiziaIf I'm building an asterisk server that doesn't have ANY cards (so it's purely sip and iax only) do I need dahdi?  or indeed anything but asterisk?
18:12.50WindowsUser~agi
18:12.51infobotmethinks agi is the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI
18:13.26hescoyes, I've got code starting: while(<STDIN>){ copied character for character from the sample
18:13.49WindowsUserokay then EXEC read to ACCOUNTNUMBER
18:13.58WindowsUserand then GET VARIABLE ACCOUNTNUMBER
18:13.58hescook, trying that then
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18:14.26WindowsUserokay then EXEC read to ACCOUNTNUMBER <-- i meant the syntax that worked before
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18:15.35hescois 'to' a part of that syntax?
18:16.22WindowsUserokay then EXEC read to ACCOUNTNUMBER <-- i meant the syntax that worked before <-- this is me trying to say dont read what i say as syntaxically correct
18:17.01hescook, thanks for that clarification
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18:23.23hescoprint "EXEC GET ACCOUNTNUMBER \n";
18:23.40hescoWARNING[25307]: res_agi.c:1494 handle_exec: Could not find application (GET)
18:23.52WindowsUserwhat about GET VARIABLE ACCOUNTNUMBER
18:24.18hescoI had tried that first and got this:
18:25.07phryksvm_invictvs: It would be a flatrate
18:25.07hescoWARNING[25364]: res_agi.c:1494 handle_exec: Could not find application (GET)
18:25.21WindowsUserodd
18:26.17hescoI thought so too, b/c *CLI> agi show get variable gives me expected output
18:26.56hescoI thought so too, b/c `*CLI> agi show get variable` gives me expected output
18:29.26hescoLantizia: I'm pretty sure you need dahdi for MeetMe().  Not sure what else.
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18:29.56hescoLantizia: make menuconfig will show you dependencies
18:30.28phrykMh, I thought only the kernel had menuconfig
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18:30.59WindowsUserits actually makeopts but theres an alias
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18:31.24hescoactually perhaps that should have been `menuselect`
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18:33.17Lantiziahesco, both do the same thing - but I don't see any dependancy things
18:33.51hescoas you use the arrow keys to go through the options, look at the bottom of the screen
18:34.34Lantiziayeah meetme depends on it (well zaptel but same thing)
18:34.50Lantiziahesco, I was more wondering about something I heard where zaptel is needed to provide a timing source
18:34.57Lantiziaeven if theres no card in the system
18:35.48hescoI've tried my AGI call using both EXEC and simply GET VARIABLE (var_name>.  Both produce the same output, none.
18:36.42hescoLantizia: I had heard that as well.  And that it was critical to MeetMe().  Not sure what else requires it.
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18:43.07LantiziaDo I need the linux kernel sources if I'm just compiling dahdi and asterisk?  Is the linux kernel sources only needed for libpri?
18:44.03hescoI believe that each of those packages require kernel headers to compile against
18:44.06Psychobillykernel source or just kernel headers
18:44.31Lantiziaso I don't need the source and exrtact it and all that
18:44.36*** part/#asterisk rwong (n=ricky@www.roflwaffle.com)
18:44.47Lantiziajust installing the headers will do for dahdi and * ?
18:45.23exc_essHey all, quick question. I have a dialplan that, when you call in externally, plays a greeting message and then transfers you to the extension you dial. Unfortunately I don't hear a ringtone while it's ringing the extension. I do get ringtones when calling extensions internally. I'm pretty sure that I used to get ringtones when calling from outside, before upgrading to 1.6
18:45.39exc_essI tried adding the "r" flag to the Dial() command, but it didn't help
18:46.45exc_essThe relevant section of my dialplan is here: http://pastebin.com/m21df4664
18:47.02hescoLantizia: I think that is right.  Test it, if that does not work, grab the entire source instead
18:47.09Lantiziaok
18:52.50MiccAre there any unified communications apps for asterisk? I know asterisk is one in itself in a way, but I mean something that ties it all together using asterisk?
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18:55.08lolo2I have followed this tut: http://www.757.org/~joat/wiki/index.php/Inbound_PSTN_Calls_to_Asterisk,_via_Gizmo5_and_GoogleVoice and am wondering if i can have two callers on thepbxc at the same time or if the way i set it up does not allow me two?
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19:02.39exc_essNo one has any idea how to get a ringtone when dialing an extension through an IVR? I feel like this is fairly standard stuff, and I'm just missing some small point
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19:04.15WindowsUserwhat are you calling?
19:04.49exc_essWindowsUser: me? Well the relevant part of my dialplan is here http://pastebin.com/m21df4664
19:05.20exc_essIt plays a greeting, and asks you to dial an extension. When you dial that extension you can't hear a ringtone over the line.
19:05.55WindowsUserso are you calling SIP/someone or DAHDI/someone? or Skype/something?
19:06.15exc_essSIP/someone
19:06.47Miccexc_ess, if the cli shows it is ringing, the caller usually hears it ringing.
19:07.07exc_essThat's what I would have thought, also that's the behaviour I used to have
19:07.12exc_essI've just migrated the system to a new computer
19:07.33exc_essThe CLI does show it ringing, btw
19:07.44Miccexc_ess, what is your progressinband set to?
19:07.47Miccin your sip.conf
19:08.20MiccWhat kind of device are you calling in from?
19:08.21exc_essNot set at all
19:08.28Micctry progressinband=yes
19:09.24exc_essMicc: progressinband didn't seem to help
19:09.37Micchttp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband
19:10.01exc_essAlso, it doesn't work if I call in from my cellphone externally, or if I dial a special internal extension which puts me into the [external] context
19:10.04Miccexc_ess, your calling device should receive the ringing sip packet and generate the tone itself.
19:10.40Miccexc_ess, it may be a nat/firewall issue eating the ring packet.
19:11.03exc_essNo NAT internally, though
19:11.07exc_essnor firewalls
19:11.16exc_essso I could see that being a problem when I actually call from outside
19:11.34exc_essbut when I dial my special "make me external" extension you wouldn't think it would be
19:11.50Miccprogressinband=yes should really do it. Do you have an answer in the dialplan first?
19:12.26MiccNo nat? that is extremely rare these days.
19:12.36MiccIs it a dedicated or virtual server?
19:13.06Miccyou should put progressinband=yes on the sip trunk as well as the phone you are dialing.
19:13.19exc_essIt's a dedicated sever
19:13.20Miccbut it probably doesn't need it on the phone you are dialing.
19:13.22exc_essIn the office
19:13.27exc_essthat's why there's no NAT internally
19:13.33Miccexc_ess, oh, I see what you mean.
19:13.53Miccexc_ess, what is your incoming line device?
19:13.58exc_essI had an Answer() originally, which I thought might be the problem, so I took it out, but it didn't seem to make a difference
19:14.22MiccYou should have an answer in before playing audio.
19:14.25exc_essOur internal asterisk server is hooked up only via the internet to our external SIP provider
19:14.33Miccbut I think background and playback do an answer if you didn't do it.
19:14.38exc_essI guess they must
19:14.44exc_essbut I can put it back in to be explicit
19:15.00Miccexc_ess, then you probably have nat between you and your sip provider.
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19:15.15Miccexc_ess, I like to be explicit.
19:15.18exc_essYeah, except that that doesn't explain why I don't get ringback when I do the call internally
19:15.31exc_essI set up ext 198 to GoTo(exposed,s,1)
19:15.59exc_essSo if I call that from an internal phone it should be equivalent to calling in externally, but not have NAT issues
19:16.03Miccexc_ess, do you have an extension that calls Dial?
19:16.36exc_essYeah, that's how the extensions work
19:16.53Miccexc_ess, what parameters are you passing to the dial command?
19:17.11Miccexc_ess, also, what type of phones are you using internally?
19:18.00exc_essAnalogue phones hooked up to Linksys SPA-2102s
19:18.14exc_essMy whole dialplan is here: http://pastebin.com/m103c86a7
19:19.01exc_essThe only flag I pass to Dial right now is the 't' flag. I tried passing the 'r' flag but it didn't make a difference
19:28.11exc_essMicc: Is it possible that it's a bug in my Asterisk installation? I'm on a Mac now, and was also having issues where if I enabled converting voicemail to "wav" it would play a static burst instead of the soundfile. So there might be flakinesses tehre
19:30.04WindowsUsermaybe your wav player doesn't support 8khz sample rate
19:30.41Miccexc_ess, what order did you build asterisk and dahdi in?
19:31.49exc_essI used macports, which did the install
19:31.55exc_essso I think it must have done dahdi first
19:32.07exc_essI remember that the last thing it compiled was asterisk
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19:32.56exc_essBut why should that matter? I don't have any "hardware device interfaces"
19:33.27Miccexc_ess, It matters a lot. Thats the main problem I find with new installations and reinstallations.
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19:33.42Miccexc_ess, make sure dahdi and dahdi_tools are installed first.
19:33.59Miccexc_ess, you don't need most of it, but for some reason it is needed for some basic asterisk functionality.
19:34.21Miccesc_ess, and its good to install dahdi_dummy too.
19:34.25WindowsUserhow does moh determine which file to play?
19:34.28Miccwhat version of asterisk is this?
19:34.38WindowsUserevery time i use hold its fpm-world-mix
19:34.48exc_ess1.6.1.1
19:35.03MiccWindowsUser, from the config file for moh and the musicclass defined in your dialplan.
19:35.17Miccor in sip.conf for an account.
19:35.46MiccWindowsUser, I haven't played with that in a long time though. Just look at the music on hold page on voip-info.org
19:36.06exc_essMicc: The only files in my install that have dahdi in their names are "chan_dahdi.conf.sample" and "spy-dahdi.gsm". I'm guessing that means it's not installed
19:36.19MiccWindowsUser, there is a command like "moh show" that will show you which classes and the files you have defined.
19:37.06Miccexc_ess, if you didn't build it and install it yourself then its probably not installed. Although there could be a package to install it.
19:37.10WindowsUseroh i probably just need to turn random on
19:37.22MiccBut its a pretty particular thing that needs to compile with your kernel sources.
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19:37.54MiccWindowsUser, moh show classes
19:37.55exc_essMicc: Looks like they don't have a dahdi package in macports, so I'd have to do that by hand
19:37.59exc_essugh
19:38.09WindowsUseraye
19:38.10Miccand moh show files
19:38.25WindowsUseryea i can tab surf the console
19:38.32WindowsUserthx tho
19:38.42Miccexc_ess, yeah its a bit of a pain, you'll need kernel-devel and kernel-src too I think.
19:38.51exc_essMicc: Ah well, I think what this means is that I won't be fixing anything tonight. Damn I was hoping it was something trivial. Thanks Micc.
19:39.00exc_essWhy is nothing in life simple?
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19:39.07MiccWindowsUser, if your file isn't listed in moh show files then it won't play it.
19:39.47Miccexc_ess, sex is simple. You could do it even if you didn't know how.
19:40.36[TK]D-FenderMicc: Probably just fuck it up... and then you're screwed ;)
19:40.45exc_essGood point. I should probably be testing that theory out instead of being in the office on a Saturday
19:41.06exc_essOn that note, bye all. Thanks for all the help
19:41.27Miccexc_ess, good luck with your simple task. ;)
19:42.02*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
19:42.16MiccTKD-Fender, I did get that client's polycom's working with the sonicwall.
19:42.24MiccThey are very happy now.
19:43.01MiccI had to turn off all the SIP settings that it had. I guess it just screws things up more for some reason.
19:43.35[TK]D-Fenderexc_ess: Micc Funny, that sounds just like the very first thing I told you ;)
19:43.59*** join/#asterisk sah-work (n=Bawbatos@99-40-7-14.lightspeed.sntcca.sbcglobal.net)
19:44.34MiccYup, I had to turn off the NAT setting in there too.
19:44.40Miccnot just transforms.
19:45.24MiccNow their only problem is the dsl modem is in bridge mode which means it won't use the QoS settings.
19:45.45MiccIt'll cost them another 300$ for the sonicwall premium that does QoS.
19:46.02[TK]D-FenderSonicWALL = pricey toaster
19:46.21MiccTKD-Fender, do you have any experience with the sonicwall QoS? Is it worth it for them to get that or should they just double NAT it?
19:47.43MiccI'm working on talking points to present to 3 VC's on tuesday.
19:48.29MiccBusiness is good, but it could be better with a quick boost of capital.
19:49.21MiccI'm not sure how much detail I should go into on VoIP.
19:49.36[TK]D-FenderMicc: Never really dealt with QoS personally
19:50.17MiccTKD, I would have though you use it all the time. Its a must for our customers.
19:50.44MiccI suppose it wouldn't be a problem if your devices are all internal.
19:51.01*** join/#asterisk lolo2 (n=lolo@c-69-180-160-4.hsd1.mn.comcast.net)
19:51.58[TK]D-FenderMicc: Most of mine are very low volume or use hardware for PSTN
19:52.17lolo2can someone pls help. If i have two simultaneous callers the first connects just fine but the second one just keeps ringing the odd things is that the log tells me its playing the sounds the the 2nd caller
19:52.38lolo2and the first caller
19:53.54Micclolo2, pastebin the appropriate section of your dialplan and your sip.conf accounts involved(trunks and devices)
19:59.55lolo2my sip http://pastebin.com/m17312fd6
20:00.25lolo2i am going google voice to gizmo5 and asterisk to gizmo5
20:01.57[TK]D-Fenderlolo2: pastebin the failed call with SIP DEBUG enabled
20:08.27lolo2http://pastebin.com/d6e2e1ef1
20:09.29*** join/#asterisk SuPrSluG (n=SuPrSluG@96.243.10.231)
20:11.05[TK]D-Fenderlolo2: I don't see an outgoing call.
20:11.16lolo2they are 2 incomming calls
20:11.26lolo2i removed one number as it is not mine
20:11.55lolo2one the first caller gets in
20:12.12[TK]D-Fenderlolo2: is your * on a public IP?
20:12.46lolo2yes
20:14.26[TK]D-Fenderlolo2: I'd still specify nat=no for your peer, and "canreinvite=no" under [general]
20:14.31[TK]D-Fenderlolo2: Just to be sure.
20:14.46lolo2ok
20:20.03lolo2ok made the changed but still same thing
20:20.51Micclolo2, how are the two calls being made and from what device? cell phones?
20:21.06lolo2one from cell one from land line
20:21.30Micclolo2, does gizmo support multiple calls per account?
20:22.02lolo2i belive so. If it did not would i still gets the messages thats its playing to both users?
20:22.53Micclolo2, try adding defaultuser=17473118234
20:23.06Micclolo2, if your using 1.6 I think username has changed to defaultuser
20:23.27lolo2im useing 1.4
20:25.45lolo2i have username=17474749813
20:26.26Miccok, thats fine then.
20:26.43MiccI would make sure gizmo supports multiple calls at once using the same account.
20:26.47MiccIt could be them that is misconfigured.
20:28.06lolo2i belize they do becouse by random chance i got it to work but then it never worked again
20:28.51MiccThats not really a confirmation.
20:28.58[TK]D-Fenderlolo2: so #2 simply never works?
20:28.58lolo2lol sorry
20:29.20lolo2never -1 time
20:29.48lolo2and yes allways the 2nd caller
20:32.57lolo2i had the same thought as micc but then i was confused as to how the log would show that the second caller is getting the sounds... if gizmo5 only allows one at a time when the log would not show that right?
20:37.29*** join/#asterisk Tagor (n=none@s55928c6d.adsl.wanadoo.nl)
20:38.07TagorI'm using APF + Asterisk. The Asterisk manual says I only have to open port 5060 for sip. But I noticed my hardware phone is connected on port 5062. Which ports do I need to configure in APF?
20:39.00[TK]D-FenderAPF?
20:39.30TagorAPF firewall
20:39.43TagorIt's a firewall that works with iptables
20:42.28[TK]D-FenderTagor: * only needs 5060 +your RTP range
20:42.40lolo2d_fender: if i use a softphone and dial my gizmo number directly and a cell and dial the did it works??
20:43.05[TK]D-Fenderlolo2: .... I can't tell the question part of that...
20:43.10Tagor[TK]D-Fender >> What's the RTP range? Is that the range used for the clients?
20:43.25[TK]D-FenderTagor: SIP = signalling, RTP = VOICE
20:43.34[TK]D-FenderTagor: * typically uses 10000-20000
20:43.43lolo2why does it work with one soft phone and one cell but not one land line and 1 cell?
20:45.04[TK]D-Fenderlolo2: does the cell fail in scenario #2?
20:46.31lolo2they both get connected to the pbx if its cell + softphone or landline + softphone
20:46.44lolo2just not land line + cell
20:47.25Xetrov`anyone know of a good stable windows mobile sip client?
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20:53.08ethicxim trying to compile asterisk but when I run make I get this error make:Warning: File 'Makefile' has modification time 2e+08 s in the future ...is this something to do with my clock?
20:56.00ethicxyup it does.\
20:56.16*** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net)
21:00.14TagorIs it possible to set maxmessage of the voicemailbox to 0 for unlimited? Or how can I remove the limitation?
21:00.36WindowsUserset it to 9999?
21:00.56WindowsUserit saves msg0000 and up, i think it'd peak at msg9999
21:01.00*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
21:01.14Tagorhmm, oke thanks :)
21:11.18*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-186-211-209.bflony.east.verizon.net)
21:12.49ethicxcan anyone help me...I finished installing asterisk but when I go to /etc/asterisk/ I have nothing there!! why is this?
21:13.07*** join/#asterisk catojo (n=catojo@189.24.3.107)
21:15.27ethicxanyone?
21:18.39WindowsUserso copy them from somewhere?
21:19.11WindowsUsertheres a configs directory in the source tarball
21:20.40ethicxk
21:20.44[TK]D-Fenderethicx: because you didn't do "make samples" like the giant advertisement at the end of "make install" told you
21:21.08[TK]D-Fenderethicx: Which would have copied them over
21:21.21WindowsUseri was wondering about that
21:21.25ethicxohhh =( damn it..
21:21.34ethicxthx.
21:21.38WindowsUserI compiled asterisk twice yesterday and didn't notice that
21:21.40WindowsUserbut hey
21:21.44WindowsUser<-- :)
21:21.44ethicxyup
21:23.18ethicxi got them now..thx
21:26.42AlmightyOatmeal[Aug  1 16:25:33] WARNING[47787]: chan_sip.c:12696 handle_response_invite: Received response: "Forbidden" from '"Jamie Ivanov"  <-- i get that when i try to dial the 3 digit extension for another SIP user.. any info/advice would be great
21:29.28AlmightyOatmealany ideas?
21:30.01[TK]D-FenderAlmightyOatmeal: Yeah, where is the failed call with SIP DEBUG in a pastebin for us to look at?
21:30.56AlmightyOatmeal2 sec
21:33.36AlmightyOatmealhttp://pastebin.ca/1515174
21:35.18[TK]D-FenderAlmightyOatmeal: THE ENTIRE CALL
21:35.44AlmightyOatmeali'm trying but i can't just dump the debug text to a file can i?
21:35.55AlmightyOatmeali can only paste whats in the ssh buffer
21:36.26[TK]D-FenderAlmightyOatmeal: Get a bigger buffer
21:38.22AlmightyOatmealhttp://pastebin.ca/1515184   is that any better?
21:39.24[TK]D-FenderAlmightyOatmeal: No.  Go get the entire call
21:40.25AlmightyOatmealhttp://pastebin.ca/1515187
21:40.31AlmightyOatmealhad putty log the entire output to a file heh
21:41.04[TK]D-FenderAlmightyOatmeal: Go into your putty config and just enlarge the scrollback to 2000 lines or so.
21:41.30AlmightyOatmealor just tell putty to dump everything to a log file like i did
21:41.39AlmightyOatmealthat is *everything*
21:44.30AlmightyOatmealor try http://pastebin.ca/1515189
21:44.36[TK]D-FenderAlmightyOatmeal: Contact: <sip:s@192.168.1.50> <-- you have not set your system up properly to work behind NAT
21:44.38[TK]D-Fender~sipnat
21:44.38infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:44.39[TK]D-Fender^^^^^^
21:44.52AlmightyOatmeal[TK]D-Fender: yes i have, everything else works just fine
21:45.06[TK]D-FenderAlmightyOatmeal: That contact begs to differ
21:45.06AlmightyOatmealthe user i'm trying to call is outside my network though connecting in via sip
21:46.18AlmightyOatmealthe user, 'deshi' that i'm trying to call is behind a nat on a remote network
21:46.46AlmightyOatmealotherwise all of my other inbound/outbound calls and extensions work just fine, its just his extension
21:47.09[TK]D-FenderAlmightyOatmeal: Well the remote side says "forbidden" and appears to be from an * box
21:47.18[TK]D-FenderAlmightyOatmeal: we'd have to see their configs too
21:47.39AlmightyOatmealhmm
21:47.48AlmightyOatmealafaik he's using a linksys spa phone adapter
21:48.16[TK]D-FenderAlmightyOatmeal: lines 217 & 230
21:48.25AlmightyOatmealof my latest paste?
21:48.30[TK]D-FenderAlmightyOatmeal: AlmightyOatmeal Yes
21:48.35*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
21:48.38[TK]D-FenderAlmightyOatmeal: Where is shows the UA rather blatantly
21:50.06AlmightyOatmeali dont see anything on likes 217 and 230 of my last 2 pastebins
21:50.10AlmightyOatmeallines*
21:50.11*** join/#asterisk ingenius (n=alektro@188-213-17-190.fibertel.com.ar)
21:50.20[TK]D-FenderAlmightyOatmeal: http://pastebin.ca/1515187
21:50.26[TK]D-FenderAlmightyOatmeal: Not your last it seems
21:50.42AlmightyOatmeal230 says #
21:50.43AlmightyOatmealCall-ID: 7b9c43457644faf253ae515257bd984d@192.168.1.50
21:50.52AlmightyOatmealoh 234
21:50.58AlmightyOatmeali'll have to talk to him about that then
21:51.11AlmightyOatmealthanks for the heads up
21:51.26[TK]D-FenderAlmightyOatmeal: 83 & 90 on the latest
21:51.51AlmightyOatmealty
21:52.23AlmightyOatmealthats not *my* asterisk box is it?
21:52.37[TK]D-FenderAlmightyOatmeal: the To: header should also have been a hint
21:52.51[TK]D-FenderAlmightyOatmeal: To: <sip:s@71.90.85.228> <- only * tells another system to contact them at "s"
21:52.52AlmightyOatmealshrugs
21:53.04AlmightyOatmealum
21:53.19AlmightyOatmealhmm
21:53.21[TK]D-FenderAlmightyOatmeal: there's a reason I gave you 2 line #'s.  Becase the first showed it is a READ
21:53.50AlmightyOatmealdo you think thats a config issue with my box or his then?
21:53.58AlmightyOatmeali didn't realize he was using asterisk to connect to me
21:54.02[TK]D-FenderAlmightyOatmeal: He's refusing you.  God knows why.
21:54.10AlmightyOatmealthat bastard *cries*
21:54.11AlmightyOatmealhehe
21:54.14[TK]D-FenderAlmightyOatmeal: As bad as you can screw up your system, he can match you...
21:54.42AlmightyOatmealwell my system isn't screwed up as far as i can tell
21:55.11AlmightyOatmeali'll have to work with him on that
21:55.18[TK]D-FenderAlmightyOatmeal: I have no idea what your proxy is doing to the packets I'm seeing so I'm still skepticle
21:55.29[TK]D-FenderAlmightyOatmeal: it LOOKS bad.
21:55.41AlmightyOatmealoh?
21:56.00AlmightyOatmealhow does it look bad?
21:56.08AlmightyOatmealpouts
21:56.12[TK]D-FenderAlmightyOatmeal: AlmightyOatmeal bad contact headers, etc, the fact it considers local IP's "NAT", etc
21:56.26AlmightyOatmealthat is odd
21:57.18AlmightyOatmeali have localnet=192.168.1.0/24 which should tell it my local ip's are not nat, right?
21:59.16[TK]D-FenderAlmightyOatmeal: depends on the peer definition as well.... what are you running that proxy for?
22:00.05AlmightyOatmealthe proxy is my sip provider?
22:00.16[TK]D-FenderAlmightyOatmeal: the sipproxd you run though
22:01.02AlmightyOatmealah that, siproxd is running on my router becuase i've been having trouble with registering with my sip provider and etc
22:01.39[TK]D-FenderAlmightyOatmeal: Probably for doing the rest of the NAT config wrong.  Have * do its own job.  it obfuscates your debugging
22:02.55AlmightyOatmealpf seems to be hurting SIP and RTP packets
22:04.05AlmightyOatmeali'll try it without the proxy and see what happens again
22:04.18AlmightyOatmealgf wants to go to the store for potato salad :P
22:04.54[TK]D-FenderAlmightyOatmeal: Words can barely describe how tragically white that sounds....
22:05.10AlmightyOatmealyou wouldn't believe how white i am :P
22:05.29[TK]D-FenderAlmightyOatmeal: Oh, I wouldn't bet on that ;)
22:05.48AlmightyOatmeali'm so white i glow in the dark :P
22:05.58AlmightyOatmealbut potato salad beacons
22:06.12AlmightyOatmealafk(god+i_hate_my_life);
22:07.40[TK]D-Fenderbeckons  <-
22:07.50[TK]D-FenderAlmightyOatmeal: Watch out for that flashing light!
22:21.24*** join/#asterisk Katty (n=Katty@mail.copi-rite.com)
22:21.34Kattymew
22:24.19[TK]D-FenderkattMew.
22:29.17TagorI have one context [voicemail] in my voicemail.conf. I changed the emailbody and serveremail but it won't use it for some reason. Anybody who knows what I'm doing wrong?
22:29.25TagorAnd yes I did reload/restart Asterisk
22:34.05*** join/#asterisk shido6 (n=shido6@dsl-67-212-24-92.acanac.net)
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22:35.47*** part/#asterisk sjobeck (n=Adium@pool-98-108-149-215.ptldor.fios.verizon.net)
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22:37.38AlmightyOatmeal[TK]D-Fender: har
22:37.41AlmightyOatmeal:P
22:38.03marl_scothi folks, can anyone shed some light on why the following keeps failing? exten => 01415312345,n,,Set(CALLERID(num)=${IF($["${DB(cidlookup/${CALLERID(num)})}" = ""]?${CALLERID(num)}:${DB(cidlookup/${CALLERID(num)})})})       * just bombs out with 'No application '' for extension' :(
22:39.18marl_scotlol, its ok, just spotted it after i hit return!!! i have 2 commas after the 'n' :(
22:39.24marl_scotsorry to bother u
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23:02.40TagorI have a file mailbox.gsm in /var/lib/asterisk/sounds. But I keep getting this error: file.c:602 ast_openstream_full: File mailbox does not exist in any format
23:02.51TagorDoes anyone know what I did wrong?
23:03.41WindowsUseris that the right dir?
23:03.57TagorI thought so. The wiki says
23:05.21WindowsUsertry using it more specifically Playback(/var/lib/asterisk/sounds/mailbox)
23:07.57TagorHmm that works
23:08.18TagorHave you got any idea where I can set the default location?
23:12.33*** join/#asterisk manxpower (n=eric@134.sub-70-221-216.myvzw.com)
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23:42.46keebler64Does anyone run Asterisk on an EEE PC?\
23:45.15*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
23:45.26Qwellkeebler64: there's no reason it wouldn't run
23:46.17keebler64I know it would runn
23:46.21keebler64Im installing it now.
23:46.33keebler64Just want to know how many concurrent calls it can support.
23:46.34WindowsUserwhy are you running asterisk on an eeepc?
23:46.45Qwellkeebler64: just like any other platform - the answer is "it depends"
23:46.47keebler64Because I want to?
23:46.48WindowsUser4? 10?
23:47.10keebler64WindowsUser: Because I don't feel like setting up my desktop to test my new network.
23:47.16*** join/#asterisk Alfio (n=Amunoz@adsl-54-41.tricom.net)
23:47.22WindowsUserthe only people that run things soley because they want to are OS/2 and Haiku users
23:47.36WindowsUsereveryone else has some sort of method to thier madness
23:47.39keebler64Originally I had it running on my WindPC server.
23:47.49WindowsUserah
23:47.49keebler64But again, don't feel like setting it up.
23:48.05WindowsUserwhat model of eeepc?
23:48.11keebler64since the EEEPc and WIND are similar setups. I figured it would be suitable to make comparative tests.
23:48.14keebler64900
23:48.19keebler64It actually have the celery
23:48.21keebler64not hte atom
23:48.24keebler64has
23:48.32keebler64Running FreeBSD
23:48.46WindowsUsercool
23:48.51keebler64Typing on the floor, so my grammar is a bit skewed.
23:49.11keebler64had asterisk running fine on the Wind the past year.
23:49.32keebler64Supporting 6 concurrent calls. (never tried more than that.)
23:52.15*** join/#asterisk PaulAviles (n=paviles@dsl-7-36.cofs.net)
23:52.25PaulAvileshello all
23:52.29PaulAvilesgood day
23:52.53*** join/#asterisk afink (n=afink@204.26.87.226)
23:53.26PaulAvilescan anyone suggest a good open source browser based soft phone?
23:54.03PaulAvilesif such thing exist...
23:54.38keebler64also, anyone on the asterisk+skype beta? Just got the invite the other day.
23:54.51afinknever heard of anything like that
23:55.12afinkPaul have u tried xlite?
23:55.14*** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de)
23:55.34PaulAvilesafink: : yes, but we are looking for a web based one...
23:55.44PaulAvilesxlite works great though..
23:57.48afinkcool idea.  I'm sure someone has tried something similar before
23:58.07PaulAvileskeebler64 : we did get the beta the other day but got dissapointed as you will need a commercial skype account
23:58.26PaulAvilesafink : that I what I thought...
23:58.37keebler64ah. Well, its a shame I got it so late. I just quit my job last week.
23:58.49WindowsUseris it possible to see which file is being played by the music on hold system?
23:59.04PaulAvilesI have not installed yet
23:59.16afinkjust watch the cli on verbose 3 or so
23:59.16PaulAvilesis an rpm though
23:59.33PaulAvileswhy do you care on the moh?
23:59.37WindowsUserthe rpm is intended for asterisknow only btw
23:59.50afinkshould say somthing like playing moh class default

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