IRC log for #asterisk on 20100909

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00:27.01Micccan the sip jitter buffer help faxes over g711 and credit card machines?
00:29.08pabelangerno, but more bandwidth might help
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01:22.03pabelangerAnybody have experience with Supermicro boxes?  Looking for feedback
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01:42.08Tech_Travispabelanger: I use a SM mb in my server and it's great for the last year.
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02:13.21nickzxcvhi, so when I run fxotune -i I see /dev/dahdi/4 absent: Device busy when it comes to my fxo interface
02:13.36nickzxcvbut when I run lsof | grep dahdi it doesn't show any processes have it open
02:17.58nickzxcvi'm running fxotune as root, is there anything else i should look at to see why its busy or how fxotune can lock it?
02:18.30nickzxcvis there a lock file or something?
02:18.49shido6odd whats lsmod say?
02:19.34shido6im getting rusty now that I use crisco gateways instead of dahdi hardwar
02:19.36shido6e
02:20.39nickzxcvthis is on freebsd, http://pastebin.com/L9uFAbb3
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02:23.09shido6can u stop asterisk and reload dahdi modules?
02:27.17nickzxcvasterisk isn't running when i run fxotune, i just reloaded the modules
02:27.59nickzxcvbefore running dahdi_cfg it said /dev/dahdi/4 absent: Device not configured and then after it said /dev/dahdi/4 absent: Device busy
02:28.43nickzxcvthis is what dahdi_scan says http://pastebin.com/syb6T9pW
02:28.54nickzxcvthe fxs doesn't work because i didn't have a power plug for it in this system
02:29.17nickzxcvbut i only want the fxo to work right now anyway, could the fxs still be a problem?
02:30.04nickzxcvthis is the dahdi/system.conf http://pastebin.com/AvtuPfBB
02:30.12nickzxcvis there anything else i should set there for this?
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02:36.49shido6is the fxo card plugged in?
02:36.58shido6and powered?
02:37.06nickzxcvto the phone line you mean? or in the system?
02:37.20shido6the hardware
02:39.12nickzxcvyes, it is (its in a colo now so i can't look at it but it was the last time i was there)
02:39.32nickzxcvand dahdi_scan detects it as i would expect
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03:04.02sawgoodIf I have voicemail set to go out via email (as a .wav file) ... should I be able to 'see' this process happening from the CLI (to confirm if VM to email is working)?
03:04.58sawgoodI know the * box can send email (because I do that from outside of * using Mutt or mail)
03:09.09ChannelZno it doesn't really say anything
03:09.39sawgoodI'll check the mail logs then
03:09.41sawgoodthanks
03:11.45sawgoodshould voicemail.conf have anything special in it to make this happen (other then what I've read on the Internet to send voicemails to emails)
03:12.49ChannelZI guess that depends on what you've read
03:12.56sawgoodexactly
03:13.11ChannelZbut 'mailcmd' is important, and the mailbox needs to be setup properly with the email address argument
03:14.15sawgoodby 'mailcmd' do you mean from the CLI or just configuing mail in general on a Linux box?
03:14.32ChannelZmailcmd in voicemail.conf
03:14.40sawgoodgot it .. looking now
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03:41.26sawgoodtail -f /var/log/maillog shows this entry for every email trying to leave the box as a voicemail to email
03:41.28sawgood<PROTECTED>
03:41.47sawgoodproblem is ... I can send emails fine from the command line (using mutt or mail)
03:41.59sawgoodsendmail is resolving hostnames just fine
03:42.19sawgoodI'm not sure what Asterisk is doing different vs sending emails from the CLI in Linux
03:43.39ChannelZthey might just be calling 'mail' and not sendmail, who knows
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05:03.43BeeBuuhow can i log all the call ,even some call were NOANSWER?
05:04.14ChannelZwell CDR should log that already I thought
05:04.44BeeBuuChannelZ: i don't think so
05:06.00BeeBuuChannelZ: i make cdr & sip and queue in realtime by mysql,but i found if the queue no answer,no log in CDR
05:07.49ChannelZmaybe that's a failing of realtime then, I'm seeing NO ANSWER's in my csv log
05:07.58Kyoshbeebuu: cdr does it all for realtime
05:08.08Kyoshi have it all in my cdr's
05:08.32Kyoshanswer, busy, congestion, no answer, failed
05:08.35ChannelZmissing field then?  or something in the dialplan that cause calls not to ever end that way..
05:08.59BeeBuuKyoush:o,really,so why it doesn't happen to my?
05:09.16Kyoshdoes your cdr's record any records?
05:10.16BeeBuuKyosh: yes,it only record the ANSWERed call.
05:10.24Kyoshonly answered?
05:10.47BeeBuuyes,so bad for me.
05:11.13Kyoshis your mysql table set up correctly?
05:11.45BeeBuuif it's incorrect,how can it record the answered call?
05:14.02Kyoshdont ask, just check
05:14.17Kyoshall columns must be entered in the order required, not random order
05:14.47BeeBuuyes,i follow the article in voip-info.org
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05:15.19Kyoshhttp://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL
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05:17.59BeeBuuKyosh:thanks,i had done like that.
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05:54.54Get_The_Fishanyone else here testing 1.8?
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06:02.14schmidtsgood morning all
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06:31.25d-bAny recommendations for a an au -> voip connection and a eu voip i> au ?
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06:51.58AndyRomanoany ideas? - got an analog telephone calling our e1-line - and all digits after the first get cutoff. if i call from a digital phone or mobile - all is working as expected - any ideas?
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06:53.14ectospasmAndyRomano: try relaxdtmf in chan_dahdi.conf
06:53.56ectospasmunfortunately that may allow the analog phones to work, but may break the voip and mobile phones
06:55.48AndyRomanok will try now ty
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06:58.23AndyRomanochanging relaxdtmf to yes doesn't change anything :(
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07:00.41ectospasmis there any difference in DTMF debug between the successful and unsuccessful DTMF navigation input?
07:01.13AndyRomanodidn't try to do so till now
07:01.15ectospasmAndyRomano: edit logger.conf, append "dtmf" to the "console" line
07:03.19ectospasmI need to go to bed, it's too late for me to be up.
07:03.49AndyRomanoty 4 your help and gn8
07:04.21AndyRomanothe extension is not dialed with dtmf - it is appended to the number direcly
07:05.03AndyRomanowhen i call the extension 40200 i get a "extenstion '4' in context …… doesn't exist
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07:21.54jkroonhi guys, dahdiras docs indicates that pppd needs to be patched to be dahdi aware - but I'm unable to locate information regarding exactly what this means - any ideas?
07:23.26TobSnydersomeone here who can give me some hints concerning configuration of isdn telephone sets
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07:27.03DNDguys is there any way i can prevent the 503 error that unregisters my soft phone when i received a busy tone?
07:34.04jkroon503 != busy.
07:34.10jkroon480 == busy.
07:34.26jkrooncheck your logs to figure out what causes the 503 to be sent to begin with.
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07:41.22DNDjkroon: based on one website: The call is going to be automatically disconnected. Try to call later.
07:41.32DNDwil lcheck on logs
07:42.49jkroonyes, 5XX => internal type errors.  you need to figure out what causes the error.  My SIP ref indicates that 503 implies Service Unavailable.
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07:53.07Diffen2Morning. Do any of you guys know an application that can show extension status if a call has been forwared using cfa or followme to a cellphone, that the extension show busy? I know that the extension are not busy when there are a call scenario like that but somehow broadsoft manage to get this working and i belive asterisk could do that too.
07:54.28DNDjkroon: here's part of the log file: http://pastebin.ca/1936373
07:56.52jkroonDND, that is a long log, and a very quick scan doesn't show anything out of the ordinary except i'm wary about those last few lines re chan_dahdi.  you're getting this on SIP though?
07:56.59jkroonperhaps enable sip debug for that peer?
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07:57.27jkroonDiffen2, you could use call groups ...
07:58.04Diffen2jkroon hmmm how do you mean? im quite new to asterisk
07:58.56DNDjkroon: that's just one call i believe. but one line there is this:"Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack"
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08:13.28TobSnyderno users with ISDN telephone sets here?
08:15.51TobSnyder<PROTECTED>
08:15.51TobSnyder<PROTECTED>
08:18.03jkroonDND, google for "isdn cause codes", first hit, look for code 3.
08:19.56jkroon3 => no route to destination.
08:20.04jkroonB number is most likely incorrectly formatted.
08:21.22jkroonCalled g0/048004931 <= not familiar with your area, but that's 9 digits - i would expect at least 10, but if that was the problem I'd also expect ISDN cause code 28.
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08:48.31DNDjkroon: how can i change cause 3 to cause 17? because it seems this is the one i need http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php
08:49.13jkroonDND - are you sure the user is busy?
08:49.27jkroonthen you need to fix the equipment that is sending the cause code of 3, not asterisk.
08:50.00DNDthe user complains that after ringing, it gives out busy tone if no one answered. which is correct. but after the busy tone, it unregisters the soft phone
08:50.17DNDand gives 503
08:50.44jkroonI would first enable sip debug for that exten, reproduce and investigate the actual SIP traffic before drawing conclusions.
08:50.47DNDso the user have to wait a couple of seconds to have the softphone re-register
08:51.22DNDa lot of  us are using different kinds of softphone but most of them encountered the same problem
08:51.45DNDhow can i debug the extension?
08:54.24jkroonsip set debug peer ????
08:55.06BarthezZhmm, can asterisk hold multiple registrations on 1 peer?
08:56.00BarthezZI'm having a fun challenge, provider suggests a $ 4 figures solution.. while it should be able with about 600
08:56.22DNDjkroon: ok got the problem basede on the log, he called 048004931. i found out that its a n 800 number which doesnt require area code.
08:56.41jkroonBarthezZ, no.
08:56.48DNDso when he dialed 048004931, it gives hangup cause 3
08:56.54jkroonhowever, i typically cheat my way around that.
08:57.02BarthezZas clear as you can be jkroon
08:57.18BarthezZyeah well i'm currently in college so didn't had the time to check :p
08:57.32BarthezZwas thinking about the workaround already
08:57.36DNDthen unregisters the phone
08:58.05jkroonBarthezZ, create 4 peers with the same name -1 ... -4 and then when it becomes time to dial you select one that doesn't have a call on it yet.
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08:58.26BarthezZwell, the situation is as follows:
08:58.39DNDis there a way that if the user enters a wrong number or maybe generates cause 3, can it just hang up not un-register?
08:58.44jkroonnot rocket science, but not trivial either.  not just anybody will be able to do it.
08:59.02jkroonYou can send a manual Hangup() with a different cause code.
08:59.20DNDin extension.conf?
08:59.47jkroonjip
08:59.48BarthezZI have 2 buildings, which require the same IP-DECT solution. But I can buy a 4 figured REALLY expensive multi cell solution, or 2 cheap wireless servers and assign them both handsets and "virtually" have the same extensions
08:59.59BarthezZonly handover etc won't be possible
09:00.02jkroonor just IAX/2
09:00.07jkroonwith trunking.
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09:00.36BarthezZbrb
09:00.36tgreermorning all
09:00.54jkroonnow you're starting to get the idea ... different accounts, same "number", ie, exten => ??,n,Dial(SIP/user-1&SIP/user-2) type of thing :p
09:01.57AndyRomanobist eh nicht über das taucher-sackerl geflogen im keller?
09:02.14AndyRomanosry wrong window ;)
09:03.33_schmidtslol :D
09:10.55TobSnydertest
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09:11.51ruben23guys any idea on this error log during installation----> http://pastebin.com/tTFNig3T
09:13.19kaldemarruben23: "You do not appear to have the sources for the 2.6.32-24-server kernel installed."
09:13.21AndyRomanokernel-headers missing?
09:13.57tgreerlooks like it
09:14.06tgreerand kernel-devel?
09:14.37ruben23kaldemar: -----> http://pastebin.com/F3XBtND9
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09:21.40kaldemarruben23: what does your /lib/modules/2.6.32-24-server/build point to?
09:23.22kaldemarruben23: i.e. "readlink /lib/modules/2.6.32-24-server/build"
09:25.00ruben23kaldemar:where is lib modules directory..?
09:26.09kaldemarruben23: that is the full path. / in the beginning means it's in root dir.
09:26.31kaldemarruben23: readlink is a command that does the job for you.
09:26.35ruben23kaldemar: im here now how do i check where did this point to.
09:27.06ruben23build -> /usr/src/linux-headers-2.6.32-24-server
09:27.19kaldemarruben23: does that directory exist?
09:28.01ruben23it does not exist on /usr/src/
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09:28.58ruben23what could ido..
09:29.01kaldemarruben23: does /usr/src/linux-headers-2.6.32-24 exist?
09:29.15ruben23it does not exist
09:30.00Azrael808Hi guys, I'm looking at purchasing some additional phones for our Asterisk server. Our regular supplier doesn't have the best collection of phones, but two in particular caught my eye.
09:30.23kaldemarruben23: apt-get install --reinstall linux-headers-2.6.32-24-server
09:31.53Azrael808The Cisco SPA504G
09:32.01ruben23kaldemar: i got it now,what do i do next.
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09:32.36Azrael808and a Draytek VPH350
09:32.52Azrael808Does anyone have any experience with these two devices? should I avoid them like the plague?
09:33.25kaldemarruben23: back to the "make all" step
09:34.58ruben23i still got the error--> http://pastebin.com/YkjtdUuK
09:35.08skyionSIP 180 ringing is handled in signalling, where is the responsibility to actually generate the tone, is it up to the handset or else in the event of a PRI card is it up to the card?
09:35.51jkroon180 is handled by the handset, 183 means premature media (handled by whichever device decided to use 183 instead of 180.
09:36.24jkroonalso, if channel is already up you will generate tones locally (often), in which case it's most likely up to the indications subsytem to generate the tones.
09:38.58kaldemarruben23: does /usr/src/linux-headers-2.6.32-24-server exist?
09:40.52ruben23yes iy did exist now
09:41.51kaldemaryet make says "no such file or directory" for /usr/src/linux-headers-2.6.32-24-server
09:41.54kaldemarinteresting.
09:42.15ruben23<PROTECTED>
09:45.24kaldemarruben23: pastebin output for "ls -ld /usr/src/linux*"
09:46.40ruben23http://pastebin.com/6s4GAq3k
09:48.51kaldemarruben23: apt-get install --reinstall linux-headers-2.6.32-24
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09:49.30kaldemarruben23: if you install something with apt-get, don't remove it by hand.
09:50.49ruben23ok
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09:52.43skyionin the event of a SIP 180 response reaching an asterisk system and being destined for a BRI card, what would happen?
09:56.38ruben23kaldemar: thanks you so mcuh
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10:05.12Azrael808If no one can confirm or deny that those two phones are any good, does anyone have any particular handset recommendations?
10:09.18_schmidts@azrael we use the cisco spa504
10:09.36Azrael808Oh, cool
10:10.07_schmidtsatleast we have around 2k of the spa941 out at our customers and the 504 is the new type of this phone
10:10.35Azrael808Excellent, the phone on offer by our supplier is the 504G... I assume this is essentially the same thing.
10:10.59_schmidtsyes, the old spa941/942/962 are EOL by now
10:11.26AndyRomanogood to know ;)
10:11.31*** part/#asterisk AndyRomano (~Adium@mail-gw.helwacht.at)
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10:12.34_schmidtsbut whats real nice about the 504 you could use the side modul to use blf
10:12.50Azrael808blf?
10:13.14_schmidtsBusy Lamp Field to see the activity of other extensions if they are talking, ringing or idle
10:13.20_schmidtsand also do a pickup on a ringing phone
10:15.40*** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no)
10:15.45Azrael808Cool! That's pretty useful
10:16.03hrhrhrthat may not work out of the box, however
10:16.33Azrael808Hey, baby steps... I'll be happy just to get some additional handsets for users
10:16.39Azrael808We can look at fancy features later!
10:16.41Azrael808:)
10:16.44hrhrhr:D
10:17.56_schmidtswhat do you mean? the blf would work, but pickup is a little work to do ;)
10:19.20hrhrhrindeed
10:19.34hrhrhrblf with pickup worked out of the box with my test handsets
10:19.39hrhrhrwhen i went and bought snoms
10:19.42hrhrhrit just refused to work
10:19.58hrhrhrthat was about 2 years ago tho
10:20.02hrhrhrmaybe it's easier now
10:20.31_schmidts;)
10:20.33Azrael808Thanks so much for your advice guys.
10:20.52Azrael808I'm off to spend some company money now! :)
10:22.24ruben23hi guys i been installing this version but first time to see this ---> http://pastebin.com/RZWy6DEh
10:24.03kaldemarruben23: did you have a previous install of asterisk on the machine? if so, you probably didn't do a make uninstall for it before installing the new one.
10:24.57ruben23now this is the problem all are manually remove..
10:25.11ruben23:-(
10:26.42ruben23kaldemar: do i got options for it..? to resolve this.
10:26.44*** join/#asterisk BANSAL (~bansal@117.207.84.86)
10:28.53kaldemarruben23: "rm /usr/lib/asterisk/modules/* && make install"
10:34.23ruben23http://pastebin.com/GHQaR3wS
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11:17.46hayhi all... I would like to have asterisk running on FreeBSD and on an ISDN line (1 BRI)... what PCI ISDN cards are tested and working in this combination... of course, as cheap as possible :-) thx
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11:56.08hrhrhrhay: take a look at the b410p. dunno about fbsd support tho
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12:31.01geemeeI made an assumption that creating a conference room would be the same almost as an extension, however I can dial the confernce room number from an IVR by dialing the "extension" of the conference room. Is this behaviour correct?
12:31.18geemeeI cant dial the conference room sorry - typo
12:32.59*** join/#asterisk coppice (~chatzilla@m121-203-224-75.smartone-vodafone.com)
12:33.53TobSnyderwhere can I change default numbers like 411 for phone and so on?
12:34.43drmessanodefault numbers?
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12:37.46TobSnyderfeature codes
12:38.13TobSnyder*98 voicemail
12:38.17TobSnyder411 phonebook
12:38.20TobSnyderand so on
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12:43.26geemeeDifferent wording to my question: Hi there, When creating conferneces can these be direct dialed in IVRs? Example make a conference room 250 and have extension 230 and 240. At the IVR someone could dial extension 230 or dial 250 to get the conference room?
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12:50.53hayhrhrhr: thanks... I am sure that there are some fbsd + asterisk + isdn users here and if anyone could recommend some 1 BRI ISDN cards, that would be excellent
12:51.19*** part/#asterisk rossand (~aross@95.215.120.8)
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12:53.47TobSnyderfound it
12:54.40TobSnyderjunghanns
12:54.40*** part/#asterisk AndyRomano (~Adium@mail-gw.helwacht.at)
12:54.54TobSnyderseem to be supported very well ?
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13:05.08_schmidtsdoes anyone know a way to send AOC infos via sip?
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13:08.35_schmidtsi have found sirrix but it looks like this is only able to send it via pri
13:09.24_schmidtspatton told me they support AOC via sip in a application/QSIG body
13:11.59*** join/#asterisk n0tk (~n0tk@216.160.42.30)
13:12.44_schmidtsoh its allready in 1.8 itself
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13:16.04*** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net)
13:16.27Kattygreat maker, i am exhausted
13:17.12t_dot_zillasomehow we've been hacked, the hacker is calling a DID -> extension -> to outbound international call
13:17.17t_dot_zillahow can this happen?
13:17.18*** join/#asterisk LordZ (~lordz@217.12.113.114)
13:17.56upblol
13:18.25*** part/#asterisk LordZ (~lordz@217.12.113.114)
13:19.16*** join/#asterisk LordZ (~lordz@217.12.113.114)
13:19.20t_dot_zillahas anyone heard of this happening?
13:19.42ectospasmt_dot_zilla: that's usually because you allow folks dialing into the IVR to get to an outside line
13:19.57ectospasmlike including from-internal in from-pstn in extensions.*
13:20.03t_dot_zillano is no IVR though, DID goes to extension
13:20.31t_dot_zillaonce the extension is reached, that extension makes the outbound call
13:20.39*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)
13:20.49_schmidts302 redirect from this extension?
13:20.50*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
13:20.55ectospasmt_dot_zilla: I'd have to see the CLI trace of the call
13:20.55Kattyhi fender.
13:20.57Kattybender.
13:21.02krionanyone with a link on how to write extension ?
13:21.04Kattyhugs fender bender
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13:21.25beekhugs Katty
13:21.34Katty:>
13:21.36Kattyhugs beek
13:21.42beekHow are you today katty?
13:21.48[TK]D-Fenderkrion: http://dictionary.reference.com/
13:21.54[TK]D-Fenderkrion: I'm sure they'd know
13:22.05beeknotes [TK]D-Fender is in great form today.
13:22.08Kattybeek: exhausted, actually :< two nights in a row up till midnight.
13:22.15ectospasmbad, [TK]D-Fender
13:22.16krionsorry if you don't understand my frenchglish
13:22.24beekKatty: Drinking and carousing?
13:22.28Kattyoooh it's ectospasm
13:22.35Kattybeek: movie watching, actually
13:22.54ectospasmkrion: http://astbook.asteriskdocs.org
13:22.58*** join/#asterisk [Jasper] (~jverberk@82-171-125-58.ip.telfort.nl)
13:23.09[TK]D-Fenderkrion: Then you might be better off picking one language :)
13:23.10krioni mean, i use stuff like exten => 5,1,Goto(ivr_stuff,s,1)
13:23.16[Jasper]hej guys, is there anyone here who knows why status of my sip peer shows unreachable..but I can just call out
13:23.25Kattynaps on beek's shoulder
13:23.45krionbut i'm not sure on what's every row comma separated are meaning
13:23.56*** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu)
13:24.01beekKatty: we'd be snoozing on each others' shoulders.  I'm a bit sleep deprived myself.
13:24.23Kattys'all good
13:24.34Kattythis vault is helping, but caffeine is a lie
13:24.44krionectospasm: thanks, hope i quickly find what i'm looking for
13:25.19beekI'm working on some serious french-press, freshly roasted and ground French Roast.   Mmmmmmmmm.
13:25.19[TK]D-Fenderkrion: "core show application goto" <----------
13:25.21beardyHave some cake.
13:25.49*** join/#asterisk rgsteele (~rgsteele@apo.aweber.com)
13:25.52Kattyit's too early for cake
13:26.02Kattyunless it is cake from the matrix.
13:26.02beardyLies!
13:26.15[Jasper]how can i see which sip peers are actually connected
13:26.29[TK]D-Fenderkrion: Go read your app's INSTRUCTIONS
13:26.46[Jasper]my phone is connected to asterisk..but in asterisks it reports unmonitored after I removed qualify= yes and other wise it shows unreachable...
13:26.49[Jasper]but I  know the phone is connected since I can call out using the asterisk box
13:26.50beardyKatty: Well it is I guess.
13:27.17Kattybeardy: do sweets make you sick on an empty stomach?
13:27.26[TK]D-Fender[Jasper]: You have a networking issue if qualify  times out and makes them unreachable
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13:29.07beardyKatty: Hmm, no, I don't think so. Being very hungry can though I guess. How about you?
13:29.49Kattybeardy: it induces teh ill
13:30.04ectospasm[Jasper]: is it a user, peer, or friend?
13:30.12beardyKatty: Coffee on an empty stomach then? :)
13:30.22*** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk)
13:30.41Kattybeardy: soda, atm...but foodables soon.
13:31.26beardyKatty: Afternoon coffee in around an hour here probably.
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13:31.46[Jasper]hmm ectospasm type=friend
13:31.48Kattyafternoon coffee?
13:31.57Kattyyou mean tea?
13:32.26beardyKatty: No.. coffee :)
13:32.27Kattybeardy: uk?
13:32.35beardyKatty: .se
13:32.37ectospasm[Jasper]: [TK]D-Fender was right then, it's either not registering (and you're allowing insecure calls), or there's a network issue
13:32.57Kattybeardy: ohah, k'then
13:33.26Kattyswedish accent is omnomnomnom.
13:33.27[TK]D-Fender[Jasper]: You should be looking at SIP DEBUG to see what is happening
13:33.45beardyHehe
13:33.56*** join/#asterisk theHub (~karl@69.177.93.21)
13:33.56[Jasper]hhmm let kme check with my laptop if it says unregistered too
13:34.19Kattywhat is the significance of Afternoon Tea
13:34.20*** join/#asterisk adyn (~adyn@unaffiliated/adyn)
13:34.40Kattyis it basically snack time?
13:34.46[Jasper]hmm [TK]D-Fender my laptop also fails
13:34.52[Jasper]what could be wron with the network then?
13:34.56[Jasper]nat?
13:35.16[TK]D-Fender[Jasper]: You tell us.  Because... you haven't actually told us ANYTHING yet.
13:35.45beardyKatty: 5PM, or 17:00 in sane time.. late snack time I guess?
13:35.50*** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk)
13:36.13Kattythat sounds like dinner.
13:36.58beardyKatty: (I don't always have coffee this time, but used to it from "home", so.. sometimes. at 16:15 )
13:37.00Kattybeardy: breakfast, lunch, tea?
13:37.40Kattybeardy: breakfast, second breakfast, elevenses, lunch, afternoon tea, dinner, supper
13:37.40beardyKatty: Yeah.. plus evening meal.
13:37.59beardyKatty: Haha, yes, more like that from where I come ;)
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13:38.23Kattybeardy: what are the 3 typical meals you guys have called?
13:38.57beardyKatty: frukost, middag(lunch), kvällsmat
13:39.22[Jasper][TK]D-Fender I have 2 phones installed and 1 phone number
13:39.26Kattybeardy: and how do you pronounce those in english?
13:39.36[Jasper]1 phone works and the telephone number works
13:39.57beardyKatty: middag is dinner, but literally means "mid day" so.. many call "supper" middag.
13:40.22[Jasper]but the other phone shows unreachable...the only thing I can see is that it's using a weird port for that phone: 21984
13:40.23beardyKatty: You mean phonetically, or translated?
13:40.28[Jasper]while the rest is on 5060
13:40.35Kattybeardy: yes
13:40.56Kattybeardy: phonetically
13:40.56beardyKatty: Which? :)
13:41.36[TK]D-FenderKatty: You don't have enough phlegm for it ;)
13:41.45Katty[TK]D-Fender: you hush.
13:42.08beardyKatty: frewcost, middaaag, qwellsmaaat, sort of ;)
13:42.23[TK]D-Fender[Jasper]: that description is useless.  what KIND of "phone" and EXACTLY how are they networked in relation to * <---------
13:42.27Kattybeardy: the aaa as in cat?
13:42.55beardyKatty: More like the a in aww, or awful
13:43.14[Jasper]it's hard to explain [TK]D-Fender ...I got 1 working phone which is in my other house...that's connected via the modem being a p-2602-r-d1a modem....so basically sip on the modem connects it..
13:43.32Kattybeardy: you should post an audio recording
13:43.39[Jasper]at this location I'm BEHIND a modem supporting sip....but I'm not using the native sip of the modem...I'm trying to conect my laptop via port 5060 to the server
13:43.49[Jasper]my laptop or my mobile phone..
13:43.51Kattybeardy: and talk slowly, not like a german
13:44.04[Jasper]port 5060 is open...since I can connect to the server...could it be that the modem blocks traffic on port 5060
13:44.07Kattyand definately not like junky
13:44.10[Jasper]incoming traffic?
13:44.15Kattydear lord that man is hard to understand.
13:44.38[TK]D-Fender[09:43]<[Jasper]>at this location I'm BEHIND a modem supporting sip....but I'm not using the native sip of the modem...I'm trying to conect my laptop via port 5060 to the server <- this modem could screw you over
13:44.49*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
13:44.49*** mode/#asterisk [+o putnopvut] by ChanServ
13:44.55krionwhat i want to do is playback something, if timeout, playback again, maximum 3 times
13:45.03beardyKatty: I might. Or when we talk sometime.
13:45.03[Jasper]i can try another port [TK]D-Fender
13:45.13Katty[TK]D-Fender: i think i'm going to learn russian after i get the french down a bit better
13:45.23krioncould i use something like $i++
13:45.23Kattybeardy: mkay
13:45.49[TK]D-Fenderkrion: No.  * != C
13:45.55[Jasper][TK]D-Fender I used utorrent to check if the port was actually open
13:46.03[Jasper]and it reports it's open port 5060 on my laptop
13:46.09[TK]D-Fenderkrion: GotoIf, Set, etc, these are the tools you'll be using
13:46.13Katty[TK]D-Fender: tho perhaps spanish might be a better idea considering my location
13:46.13[Jasper]so I don't think the modem is messing with me
13:46.19[Jasper]could it be something related to NAT?
13:46.41[TK]D-Fender[Jasper]: Yes, what did you do to set * up to work from behind it?
13:47.14[Jasper]what do you mean with * ?
13:47.32[Jasper]the asterisk box itself is in a datacenter and has a dedicated ip to the internet...so no nat there
13:47.36beardyAsterisk
13:48.00[Jasper]the other phone connects via the modem..so no nat there either...but HERE I'm behind the modem...so only location where I use NAT I guess
13:48.27*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
13:49.29Kattyohai Chainsaw
13:49.31[TK]D-Fender[Jasper]: your peer setup should be "nat=yes", and "qualify=yes".  You shuold not need forwarding on your home side.
13:49.36ChainsawHello Katty :)
13:50.02[Jasper][TK]D-Fender hmm I see...that's why it uses the weird port also because of the NAT
13:50.06[TK]D-Fender[Jasper]: And you may have to change the SIP port and specify "port=XXXX" for what you use to bypass the router.
13:50.15[TK]D-Fender[Jasper]: Which you might NOT be able to do.
13:50.37[Jasper]yea hI can change the port
13:50.43[Jasper]but I wonder how this is gonna solve the problem..
13:50.59[TK]D-Fender[09:45]<[Jasper]>and it reports it's open port 5060 on my laptop <-- make it NOT use this port
13:52.03[Jasper]but [TK]D-Fender with NAT on the client side...asterisks actually uses other ports to communicate with the client then 5060 right?
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13:52.45[TK]D-Fender[Jasper]: * uses what it is told to use.
13:52.54[TK]D-Fender[Jasper]: Run your sip client on a DIFFERENT PORT
13:53.04[TK]D-Fender[Jasper]: And inform *
13:53.13[Jasper]can I do this per sip peer?
13:53.15[Jasper]the port?
13:53.25[TK]D-Fender[Jasper]: Yes
13:53.27[Jasper]guess I can
13:53.28[Jasper]:p
13:53.28[Jasper]ok
13:53.38krion[TK]D-Fender: ok thanks, i'll try something...
13:53.48[Jasper]so let's say ...port 3000 then
13:54.28[TK]D-Fender[Jasper]: I'd say 5070
13:56.07beardy[Jasper]: 5061, 5062, and so on.
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13:57.28*** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek)
13:59.29[Jasper]not working [TK]D-Fender
13:59.48[Jasper]I get registered sip 'jasper' at ***.***.***.*** port 22907
13:59.53[Jasper]saved useragent bladiebla
14:00.05[Jasper]peer 'jasper' is now UNREACHABLE! last qualify: 0
14:01.10[TK]D-Fender[Jasper]: PASTEBIN the attempts with SIP DEBUG enabled
14:01.27[TK]D-Fender[Jasper]: "not working" doesn't help
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14:05.19*** join/#asterisk speedlnx (9709b650@gateway/web/freenode/ip.151.9.182.80)
14:05.23speedlnxhello
14:06.28speedlnxcan someone tell me which are the alternative for a CTI solution for windows and leopard that work with asterisk?
14:07.19[TK]D-Fenderspeedlnx: what is a "CTI solution"?  You you be a little more vague?
14:08.12[Jasper][TK]D-Fender http://www.pastebin.org/834523
14:09.18speedlnxsomething to call from outlook or from a web page...
14:09.24Kattyhmm. stomach is eating me alive from the inside >.<
14:10.24n3hxsspeedlnx We don't know what "something" is...
14:10.50Kattyn3hxs: ohai!
14:11.05n3hxsYo Katty
14:11.11Kattyn3hxs: let's hug
14:11.13[TK]D-Fenderspeedlnx: Anything that calls via SIP should work.
14:11.16speedlnxmmm.. a mean a software that communicate with asterisk
14:11.36n3hxs(H)
14:12.03Kattyhugs n3hxs
14:12.26speedlnxok, i will try to be more specific
14:12.33n3hxsWait, your wrinkling my flowered Carribean shirt!
14:12.50Kattyyou can't wrinkle those shirts
14:13.06n3hxsOh, yeah, you are right.
14:13.16Kattyjeesh, boys.
14:13.16n3hxsTry again.
14:13.25*** join/#asterisk UQlev (~yuriy@212.50.99.8)
14:13.51n3hxsspeedlnx, do you want to dial from a number you see on a webpage?
14:14.05speedlnxyes
14:14.06n3hxsis guessing here...;)
14:14.08speedlnxand from outlook
14:14.16*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
14:14.39Kattyhi puzzled
14:14.43n3hxshttp://www.google.com/search?q=dial+asterisk+from+Outlook&ie=utf-8&oe=utf-8&aq=t&client=firefox-a&rlz=1R1GGGL_en___US356
14:14.49[TK]D-Fenderspeedlnx: http://www.google.ca/#hl=en&source=hp&q=ms+outlook+SIP+dialer&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=2dcae7cea7739227
14:14.59n3hxsLOL
14:15.28[TK]D-FenderJFGI <----- :p
14:15.50*** join/#asterisk slipkid08 (~actioncom@rrcs-97-77-102-225.sw.biz.rr.com)
14:16.09puzzledhi Katty
14:16.21slipkid08hello everyone, I am having an issue figuring out how to install this thing and get it working, can someone please help me
14:16.51speedlnxthe first link seems to work only with 1.2 ver of asterisk
14:19.24slipkid08it is so hard to get a response from 1 out of 249 people
14:19.25slipkid08wtf
14:19.50n3hxsOnly 3 are watching you slipkid08
14:20.01slipkid08watching me?
14:20.04n3hxsAnd some are working for a living.
14:20.07n3hxsCIA etc
14:20.12slipkid08oh I see
14:20.19KattyFBI, personally.
14:20.22Kattyi'm watching you.
14:20.25slipkid08hehe
14:20.34Kattyno really.
14:20.35slipkid08it's ok I don't do anything illegal anyway
14:20.41kaldemar~ask
14:20.41infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
14:20.46speedlnxok, i will look on google for something like "outlook SIP dialer"
14:20.48speedlnxthank you
14:20.58n3hxsAlso, since you are one of the 249 people you could talk with yourself.
14:21.07slipkid08true that
14:21.09slipkid08:P
14:21.13n3hxsLOL
14:21.15slipkid08I just need some help is all
14:21.22[Jasper][TK]D-Fender did you see my pastebin?
14:21.37Corydon76-digslipkid08: did you run ./configure ?
14:22.03Kattyhi Corydon76-dig
14:22.37slipkid08oh
14:22.41slipkid08so it's terminal
14:22.50slipkid08it doesn't have an auto install
14:23.00Kattyauto install?
14:23.00kaldemarslipkid08: what exactly are you installing?
14:23.01Corydon76-digNot from source, it's not
14:23.17slipkid08Ok I am kind of lost
14:23.22Corydon76-digIf you're installing from a package, there could be a GUI, though
14:23.39slipkid08this is a VOIP system that you install on your server, and use with a SIP client right?
14:23.41kaldemarslipkid08: asterisk from source? asterisk from a package? asterisknow from a cd?
14:23.47Corydon76-digBut you need to consult with your distribution's help channels
14:23.53slipkid08gah
14:24.15Corydon76-digslipkid08: Servers don't generally run a GUI
14:24.27slipkid08whatever lol
14:24.33slipkid08I think I am in way over my head
14:24.36*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
14:24.41slipkid08I know servers don't run a gui
14:24.55slipkid08but I think I have no inclination of what is going on here
14:25.13slipkid08so I'll get out with my butt still attached
14:25.13Corydon76-digSo how were you planning on getting Asterisk installed, if not from the command line?
14:25.20slipkid08well, not really sure
14:25.31slipkid08I was going to try and upload the files to my MAMP
14:25.39Corydon76-digmamp?
14:25.45slipkid08I am not sure how the configuration is supposed to go
14:26.04slipkid08http://www.google.com/url?sa=t&source=web&cd=1&ved=0CBkQFjAA&url=http%3A%2F%2Fwww.mamp.info%2F&rct=j&q=mamp&ei=de6ITOS8AcOAlAeDs_nFDg&usg=AFQjCNE3Eeh78-JAsRygzNjtPVbj-miJkQ&sig2=XavsfQBBOQy_CJ-dLzABMw&cad=rja
14:26.10kaldemar~thebook
14:26.11infobotrumour has it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org
14:27.25Corydon76-digWhile you can run Asterisk on a Mac, I would not recommend it at this point.  We still have a lot to work out with the Mac port
14:28.08slipkid08ohhhh ok
14:28.14slipkid08well let me ask you this
14:28.15[TK]D-Fender[Jasper]: that is NOT * SIP DEBUG. 'help sip" <- go get the syntax for enabling it on your version
14:28.24Corydon76-digLinux and FreeBSD are your best bets at this point in time
14:28.26slipkid08if I VMware it
14:28.30slipkid08with ubuntu
14:28.34slipkid08would that work
14:28.56[TK]D-Fenderslipkid08: * wcan work on jsut about any *NIX
14:29.18slipkid08ok...talk to me in laymans terms
14:29.19Corydon76-digGenerally, yes
14:29.27slipkid08I am way new to this whole thing
14:29.33[TK]D-Fenderslipkid08: YES <-
14:29.35Corydon76-digThose WERE layman's terms
14:29.37slipkid08doesn't mean I won't get it, just have to be patient
14:29.48slipkid08I wasn't talking to you lol
14:30.00slipkid08ok
14:30.11slipkid08so I run ubuntu on vmware
14:30.17[TK]D-FendersliYes, it can work
14:30.24[TK]D-Fenderslipkid08: Yes, it can work
14:30.43slipkid08when I get in there, download the pack, and then run ./configure on the package
14:31.02slipkid08correct so far
14:31.03slipkid08?
14:31.07[TK]D-Fenderslipkid08: Just "sudo apt-get asterisk" or download the tarball and read the instructions
14:31.17slipkid08ok cool
14:31.19[TK]D-Fenderslipkid08: "sudo apt-get install asterisk"
14:31.21[TK]D-Fender(rather)
14:31.23slipkid08that should get me started
14:31.46slipkid08I didn't realize there wasn't a mac port yet
14:32.03*** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com)
14:32.22slipkid08now may I ask some general question?
14:32.25[TK]D-Fenderslipkid08: Mac's are for kiddies, not servers :p
14:32.30p3nguin~ask
14:32.31infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
14:33.11[TK]D-FenderslikAsterisk doesn't have a "Go" button and is therefor not conducive for use on Aplle tech.
14:33.33slipkid08Is this a skype type of thing (i.e. calling only computer-computer) or can it call land-based lines too
14:33.42[TK]D-Fenderslipkid08: No.
14:33.47slipkid08haha fender..
14:34.01[TK]D-Fenderslipkid08: Call likes has nothing to do with the PROTOCOL (which Skype is an example of)
14:34.06slipkid08trust me, I am not really a mac fan
14:34.10[TK]D-FenderCalling lines*
14:34.12[TK]D-Fendergah
14:34.19[TK]D-FenderInsufficiently caffienated this morning
14:34.23slipkid08lol brb
14:34.36*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
14:36.50KattyThe_Boy_Wonder: what is the secret of your power
14:37.15The_Boy_WonderKatty: pancakes
14:37.17tzangerit's the tights he wears
14:37.19n3hxsWhite powder in a little bottle.
14:37.35KattyThe_Boy_Wonder: pancakes++
14:37.36tzangerthere's a lot of power hiding in that tiny spandex
14:37.45Kattyooh la la.
14:38.04slipkid08OK, so is this a Vonage type of system that only calls landlines?
14:38.13n3hxsEwwww
14:38.24slipkid08and/or mobiles
14:38.37Kattyslipkid08: you can use pots lines, t1/pri, or sip providors
14:38.48slipkid08hm
14:38.51Kattyslipkid08: vonage is kinda craptastic, fyi
14:39.04slipkid08I don't like vonage
14:39.22slipkid08what I am asking is the system, once installed, can make phone calls anywhere in the world?
14:39.31slipkid08for the cost of your host?
14:39.40Kattyslipkid08: that probably depends on the terms of your telco
14:39.53ectospasmKatty: or ITSP
14:40.02Katty^- what he said.
14:40.13slipkid08ok
14:40.41Kattyslipkid08: we currently have a pri, with unlimited long distance in the usa, canada, and mexico
14:40.52slipkid08that's pretty cool
14:40.53ectospasmalthough I suppose the two terms are interchangeable, but I always think of telcos as being layer-1 providers, for traditional POTS/PSTN connections
14:40.58Kattyslipkid08: places like norway, cost more
14:40.59slipkid08but pretty far over my head, too
14:41.09slipkid08wow, yeah
14:41.16Kattyslipkid08: think of asterisk like an over glorified answering machine.
14:41.26slipkid08I could just use onebox?
14:41.29ectospasmKatty: that's dumbing it down a lot!
14:41.30slipkid08or google voice
14:41.43Kattyectospasm: i'm on the level, yo
14:41.49*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
14:41.51ectospasmGoogle Voice is essentially a hosted PBX
14:41.53Kattyhi tony
14:41.55slipkid08right
14:41.55ectospasm...or cloud PBX
14:42.03slipkid08I like that
14:42.04slipkid08lol
14:42.17ectospasmslipkid08: no, it's simple, and free, and relatively no hassle
14:42.30Kattygoogle vchat is nice.
14:42.37slipkid08so, basically I would be getting the same thing with google voice as I would going with asterisk?
14:42.38ectospasmand the voicemail transcription is awesome
14:42.49ectospasmslipkid08: yes, but you'd have more control over Asterisk
14:43.00slipkid08I think it's cool that you guys are building this, but why reinvent the wheel?
14:43.08ectospasmslipkid08: if you don't like something about Google Voice, and it doesn't provide you a way to workaround it, tough
14:43.16slipkid08true
14:43.24Kattyslipkid08: why?
14:43.32[Jasper][TK]D-Fender http://www.pastebin.org/834911
14:43.32Kattyslipkid08: because when someone calls me, i want to rickroll them.
14:43.34[Jasper]thats sip debug
14:43.39Kattyslipkid08: and eject the cdrom drive when i dial extension 42
14:43.40slipkid08haha
14:43.41ectospasmslipkid08: Asterisk was developed concurrently with Grand Central (which Google bought and renamed Google Voice)
14:43.58ectospasm...or maybe even beforehand
14:43.58slipkid08oh I see
14:44.09chuckfslipkid08: asterisk is not geared for the home user/single user audience (though there are many installs for that range)
14:44.25Kattychuckf: i use it at home.
14:44.25slipkid08so, when will there be like a GUI, so that way someone such as myself doesn't have to go to such great lengths for this
14:44.32ectospasmslipkid08: and don't think that one solution is appropriate for all
14:44.36Kattychuckf: but i agree, generally speaking i find it better for small-med size business
14:44.48slipkid08I know what a pbx is
14:44.58slipkid08press one to go here press two to go there
14:45.01chuckfKatty: as do I, but most people don't have/want/need a pbx at home
14:45.12slipkid08it would be geared towards business
14:45.13Kattychuckf: i enjoy rickrolling the telemarketers
14:45.19slipkid08I was wanting it for my band
14:45.20slipkid08lol;
14:45.25ectospasmslipkid08: the beauty of Asterisk is that it connects any kind of telephony technology to any other kind of technology
14:45.31slipkid08so that way people could call one number
14:45.40chuckfKatty: that is a good use for it
14:45.41slipkid08and get all 5 of us
14:45.44slipkid08or our manager
14:45.46Kattychuckf: mhmm
14:45.52slipkid08that's okay
14:46.01slipkid08it was a thought but I don't have time to work around it
14:46.08slipkid08I'll just buy one, I think
14:46.14slipkid08vpbx
14:46.22Kattyslipkid08: asterisk will take awhile to learn, that's for sure.
14:46.37Kattyslipkid08: if you're not willing to spend the time, it would probably be best to find something a bit more suited to your needs
14:46.39slipkid08yeah it seems to have a hefty learning curve
14:46.44ectospasmyeah, it took me two weeks to figure out how to get a call routed in the dialplan
14:46.57slipkid08it's not that I WOULDN'T spend the time
14:47.03slipkid08the time is not there to spend, you know?
14:47.07Kattyslipkid08: depending on what country your in
14:47.08slipkid083 jobs and a kid and a band
14:47.21Kattyslipkid08: you might be just as well off going to Sam's Club and buying a small out of the box phone system
14:47.23ectospasmslipkid08: yeah, Asterisk isn't for the faint of heart
14:47.31p3nguinIf you read The Book, you should be able to make calls within a few minutes after you finish reading.
14:47.36slipkid08Well, I need a VPBX
14:47.43ectospasmp3nguin: heh, the book is over 400 pages!
14:47.46*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
14:47.50Kattyslipkid08: you might consider talkswitch.
14:47.54slipkid08pretty much is what it boils down to
14:47.58slipkid08what is talkswitch
14:48.11chuckfI got it at home to play with at first, then my cousin started doing shows in eurpoe. I got DIDs from there so she could call local there to the *, then dial family and friends here in the US
14:48.15Kattyslipkid08: i would steer clear of samsung boxes.
14:48.20slipkid08huh
14:48.23ectospasmbut remember, Asterisk _isn't_ a PBX.  It's a PBX _Toolkit_
14:48.23slipkid08that's nice
14:48.34slipkid08see I can't make out that grey line
14:48.39KattySome Assembly Required
14:48.44slipkid08oh haha
14:48.48slipkid08I see
14:48.54slipkid08kinda like my son's trike
14:49.01slipkid08that was in 45 pieces lol
14:49.10Kattyholy snickerdoodles, 45 pieces?
14:49.11tzangerheh
14:49.14ectospasmgah, I'd hate Asterisk if I had to know assembly to configure it! (-;
14:49.24slipkid08but now it has rockets on the back and cards in the spokes
14:49.24slipkid08lol
14:49.26tzangersomeone should come up with a set of IKEA-like instruction pdfs for asterisk
14:49.33slipkid08someone should
14:49.42Kattyi'll get right on it.
14:49.53ectospasmtzanger: http://astbook.asteriskdocs.org
14:49.55slipkid08I don't mind figuring stuff out, but I just don't have the time or the energy
14:50.01ectospasmyeah, it's outdated, but still good info
14:50.04tzangerectospasm: no, I am talking pictograms :-)
14:50.12ectospasmtzanger: um.
14:50.15Kattyslipkid08: if you really want asterisk, i'm sure you could find a consultant
14:50.25ectospasmtzanger: configure dialplan using pictograms.  I dare ya
14:50.42slipkid08Yeah but I am not paying $150/hr to have someone tell me something I could find on my own
14:50.46slipkid08if given time
14:50.53ectospasmslipkid08: or use a turnkey solution like Switchvox
14:50.56Kattyk
14:51.06Kattyokay what does TURNKEY mean
14:51.08Kattyseriously.
14:51.11slipkid08haha
14:51.18fileyou turn a key and a cookie appears
14:51.23ectospasmKatty: basically a fully functioning system, COTS/out of the box/etc.
14:51.27[TK]D-FenderKatty: Put the Key in.  Turn.  Product jsut starts working.
14:51.32Kattyohah, kthen
14:51.37slipkid08lol
14:51.42slipkid08I like turnkey solutions
14:51.46Kattyi like file's description better
14:51.48*** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net)
14:51.49Kattyfile: did you go to pax?
14:51.53fileKatty, I did
14:52.08Kattyfile: my sister was there
14:52.24fileKatty, a lot of people were there >_>
14:52.28Kattyfile: yesh.
14:52.42filewhich produced the PAX cellular deadzone
14:52.49Katty:<
14:53.10p3nguin$150/hour?  I'd work on your Asterisk system for less than half that!
14:53.22b0gatyrmorning everyone, can someone please help me see what's going on here..I'm trying to call an international number but I get "Got SIP response 486 "Busy here" back from ..." but if I try calling some other number it goes through just fine.. what can be wrong?
14:54.18slipkid08I'll brb
14:54.38slipkid08I like you guys I think I'm going to lurk here for a while and just watch your conversations
14:55.00p3nguinIf you want someone to set up a system for you, let me know.
14:55.16jamkogaytr sounds like your provider does not have you set up for int.
14:55.48slipkid08how about setup for a trixbox Pro
14:55.55b0gatyrjamko: but it works when I dial another int number
14:56.05Kattyp3nguin: take beer.
14:56.09b0gatyrit's just that number
14:56.16b0gatyrand it's random
14:56.24jamkogayr: ask your provider why they won't place the call.
14:56.28p3nguinkatty: I like beer, but the doctor told me I have to stop drinking beer.
14:56.55n3hxsAsk the doc about Tequila.
14:56.56Kattyp3nguin: whyfor?
14:57.17p3nguinHe's worried about my liver.
14:57.23Kattyoh :<
14:57.57[TK]D-Fenderslipkid08: That is a closed platform and isn't supported here
14:58.03p3nguinI had an enzyme that was elevated 30% over a normal level, and it worried him so he had me stop drinking beer and stop taking Advil.  We'll retest in 90 days.
14:58.15n3hxsp3nguin just add onions... ;)
14:58.16slipkid08[TK]D-Fender: really?;
14:58.19[TK]D-Fenderslipkid08: The kind of thing retard kiddies use :p
14:58.22Kattyp3nguin: i hope everything turns out okay.
14:58.26p3nguinme too!
14:58.53p3nguinIf it stays elevated, he'll probably test me to find out if I have NAFLD.
14:59.03p3nguin(non-alcoholic fatty liver disease)
14:59.03Kattywhat's that?
14:59.08slipkid08what about virtual pbx
14:59.22Kattyp3nguin: and what exactly does that mean?
14:59.33[TK]D-Fenderslipkid08: Trixbox pro runs a closed source proprietary GUI that configures * to its cookie cutter design and isn't made for you to get "creative".
14:59.50[TK]D-Fender[10:58]<slipkid08>what about virtual pbx <- that is a "concept", not a "thing"
15:00.07*** part/#asterisk skyion (~bradc@siza.thusa.net)
15:00.30p3nguinI guess having a fatty liver isn't acceptable.  I've heard of other people being diagnosed with it, and they were kinda bummed over it.
15:00.38KattyNugget: <3
15:00.51Kattyp3nguin: hmm.
15:01.01Kattyp3nguin: you basically have to change your diet, i'm guessing?
15:01.09p3nguinI would think so.
15:01.13Kattynods
15:01.26*** join/#asterisk grayhame (~grayhame@74-94-250-169-Nashville.hfc.comcastbusiness.net)
15:01.50p3nguinBut my glucose, cholesterol, thyroid, kidneys... all good.
15:04.18Kattyexcellent.
15:05.56n3hxsp3nguin hope you feel better soon.
15:06.44p3nguinI hope the re-test shows that it was my overdosage of Advil that elevated that liver enzyme.
15:06.54Kattynods
15:06.57Kattyi hope so too.
15:07.07p3nguinI had no thought at all when I took 600mg the night before my test panel.
15:07.19slipkid08I am not really concerned with getting "creative" I just want to make and receive calls from anywhere and to anywhere without paying more server fees
15:07.21Kattyi'm guessing that's high?
15:07.43p3nguinThe normal adult dosage is two 200mg pills.
15:07.57Kattyso 1 pill is 100mg?
15:08.04p3nguinI often take three because I figured more was better.
15:08.07Kattyor 1 200mg?
15:08.23Kattyah.
15:08.27p3nguinone is 200mg, and I often take three.
15:08.37Kattydoes 2 not kick it?
15:09.04slipkid08brb
15:09.30Kattyoh, advil is ibuprofen
15:09.32p3nguinI felt like the regular 400mg dosage wasn't enough.  He wasn't very happy with me for that.  He said prescription strength 800mg is NOT the same as taking four regulars, so I was unncessarily overdosing.
15:09.41p3nguinyeah it is.
15:09.49Kattyibuprofen doesn't work for me, for some reason
15:09.55*** join/#asterisk jmacz (~jmacz@190.144.75.22)
15:10.03n3hxsslipkid08  google the ITSPs and look at their rates.  I used les.net for a year and their rates were fairly good.  Then they raised the base rates for just having an account and I dropped it because I was only "playing" with the service.
15:10.19Kattythey had me on that stuff back when the wisdom teeth came out, but it didnt' really phase the pain. have you considered a different type of pain killer?
15:10.27p3nguinAnd he said the enzyme being high could be because of my overdosage of fast-acting ibuprofen.
15:10.34Kattynods
15:10.55Kattyyou might consider asking him what type of pain killer to try if the average dose of ibuprofen isn't doin it for ya
15:11.01p3nguinI forgot that I had a headache the night before the testing.
15:11.07Kattythey switched me over to tylenol.
15:11.13p3nguinHe said I can take that.
15:11.16*** join/#asterisk KavanS (~KavanS@unaffiliated/kavans)
15:11.17Kattywould not recommend asprin....
15:11.38p3nguinI guess aspirin is almost the same as ibuprofen.
15:12.26Kattyi've no idea
15:12.35Kattybut i know it is autotoxic
15:12.38p3nguinAcetaminophen is okay, but he said stop taking ibuprofen.
15:12.44Kattydamages your hearing over time
15:12.55p3nguinMy hearing is bad enough already.
15:13.03Kattyand can cause tinnitus if taken excessively
15:13.17p3nguinHmm, I sometimes have some ringing, too.
15:14.10Kattyi've had tinnitus for nearly a year. thankfull it's not bad.
15:14.19p3nguinfaint ringing?
15:14.27Kattybut i can tell you that caffeine, being tired, and high bp/stress really make it a lot worse
15:14.41Kattyyes. faint now. used to be worse. i hear it a lot more right before i go to sleep.
15:14.46p3nguinIt must not be too loud... I couldn't hear it on the conference last week.  :D
15:14.50Kattyduring the day i can't really hear it anymore unless i listen for it
15:15.01*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)
15:16.03Kattyp3nguin: you might mention the tinnitus next time you're in for testing
15:16.12KavanSacetaminophen damages your liver...
15:16.19KavanSmeh, I try to stay away from all that shit...unless you really need it
15:16.25KattyKavanS: i agree.
15:17.34KattyKavanS: pain killers are a beautiful thing, tho.
15:18.26*** join/#asterisk felipe_ (~felipe@my.nada.kth.se)
15:21.54*** part/#asterisk slipkid08 (~actioncom@rrcs-97-77-102-225.sw.biz.rr.com)
15:25.16*** join/#asterisk Trox (~Adium@85-126-174-18.work.xdsl-line.inode.at)
15:29.57*** join/#asterisk mrbnet (~ryanbantz@74-95-100-233-Minnesota.hfc.comcastbusiness.net)
15:30.00n3hxslikes his TENS for back pain.
15:30.16Kattytens?
15:31.25n3hxsTransdermal eletronic nerve stimulator.
15:31.30n3hxsor something like that.
15:31.37*** join/#asterisk dodavoice (~d@41-134-22-10.dsl.mweb.co.za)
15:32.15dodavoiceKobaz you around?
15:32.38p3nguinlooks around the channel
15:32.50dodavoicehi p3nguin
15:32.54p3nguinhello
15:33.07dodavoicehave you seen him around today?
15:33.17p3nguin~seen kobaz
15:33.25infobotkobaz is currently on #asterisk-dev #asterisk. Has said a total of 161 messages. Is idling for 18h 20m 27s, last said: 'perl.. not pearl'.
15:33.25dodavoiceyip
15:33.44dodavoicethx
15:34.43p3nguinAnything the other 250 people here can help with?
15:35.14dadavoiphe said he would hava a look ata project some customizing of vicidial
15:35.50dadavoipp3nguin how did you get the info regarding Kobaz?
15:35.59p3nguinYou saw me.
15:36.42dadavoipyip
15:36.51dadavoipyou hacker
15:36.54dadavoiplol
15:39.02p3nguinhttp://whatthefuckshouldimakefordinner.com/
15:40.06*** join/#asterisk CunningPike (~CunningPi@204.239.12.183)
15:40.39dadavoip<PROTECTED>
15:41.45p3nguinnod
15:42.02dadavoipyou can put some goto if times
15:42.47dadavoipdifferent menus for diverent days, even have some turkey on for the holidays
15:45.17*** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn250.78-98-184.t-com.sk)
15:51.40*** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net)
15:51.40*** mode/#asterisk [+o Deeewayne] by ChanServ
15:53.02*** join/#asterisk BANSAL (~bansal@117.199.113.144)
15:54.55*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:54.55*** mode/#asterisk [+o leifmadsen] by ChanServ
15:55.10*** join/#asterisk _zoom_ (~user@41.218.41.70)
15:56.05_zoom_hi, for those who has played with termination, have experienced any problem with NICs ?
15:56.37asilvaLittle help.. this message is caused by what action - [Sep  9 12:53:35] WARNING[23860]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1780002, seqno=17)
15:56.37asilva[Sep  9 12:53:40] WARNING[23855]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 2, ts=1785007, seqno=18)
15:56.37asilva[Sep  9 12:53:45] WARNING[23854]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1790002, seqno=19)
15:56.37asilva[Sep  9 12:53:55] WARNING[23858]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1800002, seqno=20)
15:56.40asilva[Sep  9 12:54:01] WARNING[23857]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 2, ts=1806007, seqno=21)
15:56.43asilva<PROTECTED>
15:56.45asilva<PROTECTED>
15:56.47asilva[Sep  9 12:54:05] WARNING[23853]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1810002, seqno=22)
15:56.52asilvapabx*CLI>
15:56.54asilva[Sep  9 12:54:15] WARNING[23859]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1820002, seqno=23)
15:56.57asilva[Sep  9 12:54:22] WARNING[23857]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 2, ts=1827007, seqno=24)
15:57.00asilva[Sep  9 12:54:25] WARNING[23851]: chan_iax2.c:3339 __attempt_transmit: Max retries exceeded to host 200.145.112.7 on IAX2/contadundi-7879 (type = 6, subclass = 11, ts=1830002, seqno=25)
15:57.01*** mode/#asterisk [+q asilva!*@*] by russellb
15:57.03russellb~pb
15:57.03infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
15:57.07p3nguinJesus... use a fucking pastebin.
15:57.47p3nguinI wonder if he'd like it if we all went into his living room and dump a big bucket of shit on his floor.
15:58.22*** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk)
16:02.06*** mode/#asterisk [-q asilva!*@*] by russellb
16:02.23_zoom_hi, for those who has played with termination, have experienced any problem with NICs ?
16:02.37[TK]D-Fender_zoom_: Huh?
16:02.39p3nguinhuh?
16:02.59p3nguinHaha, glad I wasn't the only one with that thought.
16:03.36*** join/#asterisk DaCoD (~DaCoD@189.4.108.113)
16:03.45[TK]D-Fender_zoom_: http://tinyurl.com/y72t82u
16:04.19frigidzephyr_zoom_: you mean like hardware level issues with flakey NICs?
16:04.36frigidzephyrthats a pretty broad, vague question
16:05.02DaCoDhi all
16:05.06DaCoDCan someone tell me what this error means and how to fix
16:05.15DaCoDExt: 1  Cause: Interworking, unspecified (127), class = Interworking (7) ]
16:05.41DaCoDasterisk 1.4.32, dahdi, isdn
16:07.27frigidzephyr"Cause No. 127 - SW56 disconnect (Internetworking, unspecified) This cause indicates that an interworking  call (usually a call to SW56 service) has ended. May also be seen in  the case of a non specific rejection by your long distance carrier (try  again at a different rate)  "
16:07.53leifmadsenKatty: pictures now exist!
16:08.24Qwellleifmadsen: of the new hoose?
16:08.35leifmadsenQwell: yes! new counter tops specifically
16:08.41Qwellnice.  link?
16:08.44*** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net)
16:08.56Qwelloh.
16:08.59leifmadsenQwell: http://www.facebook.com/album.php?aid=159742&id=512680761&l=20abfce25a
16:09.07Qwellyes, I figured facebook
16:09.13leifmadsenheh, you don't need to be signed in though
16:09.23Juggieleifmadsen, geek
16:09.30Qwellthose chairs are too much
16:09.46leifmadsenQwell: the black ones?
16:09.46Qwellthey remind me of those clear plastic spoons
16:10.03leifmadsenQwell: I thought you liked spoons?!
16:10.09leifmadsenba-dunk-chink
16:10.14*** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142)
16:10.17leifmadsenJuggie: well duh :)
16:10.25leifmadsenJuggie: slate is sealed now!
16:10.27Juggiewheres the furniture :P
16:10.29Juggienice
16:10.38leifmadsenJuggie: still in the process of getting it into the house... couch comes tomorrow
16:10.45leifmadsenwe're still working on furniture a bit, heh
16:10.49leifmadsenno rugs or coffee table yet
16:10.56Juggieno pictures :P
16:11.06p3nguinIs that a single-family home?
16:11.18DaCoD<frigidzephyr>: OK, And how can I change the rate ?
16:11.34leifmadsenp3nguin: townhouse, ya
16:12.00p3nguinOh great...  Another nick copy-and-paste person...
16:12.26frigidzephyrDaCoD: not sure, i would make sure you are formatting your dialed number correctly as the telco wants it
16:12.30beardyDoes anyone know of a "Jp" person?
16:12.58Qwellbeardy: "Jp" could mean lots of things.
16:13.05frigidzephyrDaCoD: compare a local number dialed, vs long distance, etc see if you get same error
16:13.50frigidzephyrDaCoD: I think I saw a guy with that error once, when had some kind of restriction set on his service, couldn't dial  a certain area, or long distance or something
16:14.02beardyQwell: Yeah.. I mean with Jp as nickname. Thought I'd check if he was around here.. made a helpful comment on an * related article I wrote..
16:14.12p3nguin~seen jp
16:14.21infobotjp <n=JP@m195e36d0.tmodns.net> was last seen on IRC in channel #norganna, 608d 12h 37m 8s ago, saying: 'hi'.
16:14.30DaCoD<frigidzephyr>: support  operator, told me I should switch to National subescriber
16:15.01frigidzephyrDaCoD: look into the   pridialplan setting in chan_dahdi.conf then
16:15.15*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
16:15.20frigidzephyrDaCoD: try setting that to national, or unknown,
16:15.31p3nguindacod: Instead of copying and pasting his nick, why not type a couple letters of his nick and then press the TAB KEY?  fr <TAB> will work just fine.
16:15.38DaCoDI change in chan_dahdi.conf but don't work, change  National subescriber to National
16:16.02frigidzephyrDaCoD: I don't understand what you mean there.
16:16.32frigidzephyrDaCoD: did you set pridialplan=national     ?      Are they maybe talking about the switchtype?  you could try national and ni1 for that
16:17.00p3nguinSmells as if someone is making garlic bread... I should investigate.
16:17.03DaCoDhttp://paste.debian.net/88896/
16:17.18DaCoDlook my group config
16:18.05_zoom_frigidzephyr: yeap
16:18.09*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
16:18.55frigidzephyrDaCoD: try changing pridialplan to national ?  that will change the TON, i'm not sure what they mean by National Subscriber ?
16:19.38frigidzephyrmaybe they mean  the subscribers TON should be national
16:20.06frigidzephyryou may want to set pridialplan to dynamic and have it set based on the prefixes you are using
16:20.53*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
16:21.22_zoom_frigidzephyr: could a normal PC handle 10K termination ?
16:21.32_zoom_using asterisk
16:22.35frigidzephyrif you mean an average desktop PC, no, probably not, maybe 10K registrations, as long as the registrations are staggered or quite delayed.  Definitely not 10K active sip calls
16:22.58*** join/#asterisk war9407 (war@liquidswords.org)
16:23.14leifmadsenschmits in #asterisk-dev has been working on over 20k registrations
16:23.17DaCoDp3nguin: ok, I try on the next
16:23.24p3nguindacod: THANKS!
16:23.42leifmadsenbut it's still in development and requires code changes that are not yet merged into asterisk :)
16:23.59p3nguinSo what is the current limit on registrations?
16:24.03DaCoDfrigidzephyr: tks for your help
16:24.17frigidzephyrDaCoD: you're welcome
16:24.43*** join/#asterisk pif (~ldm@zenon.apartia.fr)
16:29.22*** join/#asterisk imox1234 (~imox1234@p4FC5C53A.dip0.t-ipconnect.de)
16:29.41*** join/#asterisk myster (~myster@207.148.172.210)
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16:42.17Deeewayneleifmadsen, nice house
16:45.21*** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-170-232.mia.bellsouth.net)
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16:50.56*** part/#asterisk _zoom_ (~user@41.218.41.70)
16:52.32leifmadsenDeeewayne: heh thanks :)
16:53.30*** join/#asterisk sol (~sol@unaffiliated/sol)
16:55.26asilvaback!
16:55.45asilvaa little help here. - http://pastebin.com/yy3AcEG5 these messages cause my asterisk to FREEZE could it be like a DoS?
16:55.56*** join/#asterisk ahowlader (Adnan@180.234.35.175)
16:58.01Kattydrags in
16:58.22Kattyi think my trainer is trying to kill me.
16:58.26Qwelldrugs Katty
16:58.32dadavoipasilva do a top command and see what processes is taking uf resources(do you have iax trunk or extension?
16:59.19KattyQwell: :<
16:59.26Qwellin a good way!
16:59.33Kattydrugs are never good!
16:59.38Kattyunless you are going into surgery
16:59.38Qwellflintstones chewables!
16:59.42Katty:>>>
16:59.48Kattythose are acceptabuhls.
17:00.10asilvadadavoip, iax account used by dundi.. type=friend
17:00.23KattyQwell: do you care about my fiber?!
17:00.35asilvadadavoip, i'll check the %% of cpu and asterisk process next time it happens
17:00.35QwellThere's no good way to answer that.
17:00.56Kattyinfobot: fiber?
17:01.09Kattyinfobot: fiber is You care about my fiber?! http://www.youtube.com/watch?v=DjyklC02MNM
17:01.10infobotACTION stuffs is You care about my fiber?! http://www.youtube.com/watch?v=DjyklC02MNM with fiber.
17:01.48Qwellthat response is...odd
17:01.59Kattyyes.
17:02.01Kattyinfobot: fiber?
17:02.04Katty:<
17:02.09Qwellinfobot: stuffs
17:02.33Kattyinfobot: you care about my fiber?
17:02.40Kattysuspicious.
17:03.01Letoriccan anybody tell me what variable I can put in a mixmonitor file name to make sure it is truly unique?
17:03.04asilvadadavoip, tkz for the tip!
17:03.21LetoricIE, is there a variable for the uniqueid that is assigned when the call is placed in cdr-csv/Master.csv
17:03.32Qwell${UNIQUEID}
17:03.47Letoricthat variable works in a dial plan?
17:04.03p3nguinWhere else would it work?
17:04.10*** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net)
17:04.11Letoricheh, in the csv files! ;)
17:04.15Letoricthank you
17:04.54leifmadsen~fiber
17:05.01leifmadsen~stuffs
17:05.16Kattyleifmadsen: i think infobot got some wires crossed :<
17:05.33*** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net)
17:06.15ChainsawKatty: It is stressed. It needs a spa day.
17:06.20*** join/#asterisk EndEng (~epierce@mail.endeavoreng.com)
17:06.40*** join/#asterisk Ad-Hoc (~nimbus@193.92.83.151.dsl.dyn.forthnet.gr)
17:07.04Kattyinfobot and me both.
17:07.04infobotmoi?
17:07.11Kattyinfobot: yes, you dear.
17:07.47Kattyand possibly a manicure.
17:11.04*** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net)
17:11.27Letoricif I want to make SURE all calls are monitored regardless of how the call flows, do I have to create a monitor statement in each context? Is there a better way to ensure this?
17:11.49[TK]D-FenderLetoric: No
17:12.03LetoricI'm currently catching it, but it's high maintenance when I want to do something like change the file name, and setting a global variable for the file name doesn't seem to work, it doesn't find the uniqueid or callerid properly
17:12.11[TK]D-FenderLetoric: Recording is a dilaplan option.  It goes where you put it.
17:12.12drmessanoTell your users to record all calls on a tape recorder or you'll punch them in the face?
17:12.17drmessanoThat actually works
17:12.29p3nguinWhy would you try using a global for that?
17:12.40Troxwhat about using a macro for your Dials?
17:12.43drmessano[TK]D-Fender:  So does a punch to the face, usually
17:12.47Letoricso that I wouldn't have to change it for every different spot we initiate monitoring
17:13.02LetoricI'm kind of new, I haven't jumped too much into macros, just the stdexten one
17:13.22[TK]D-FenderLetoric: That name isn't implict
17:14.41*** join/#asterisk Blackvel (~blackvel@dslb-084-057-081-245.pools.arcor-ip.net)
17:15.42Blackvelhi all. how can help me about voip delays (not echos) on snom 370 and pstn/isdn + patton isdn media gateway 4634?
17:15.46Blackvelwho can...
17:17.18Blackvelits just a matter of some millisecs...but still enough delay that a 1h business interview is difficult enough
17:17.58Blackvelnoone of both parties knows how to interrupt the party...question / answer playing gets a mess when noone knows when to start/stop talking
17:19.17Blackveli am running snom370 firmware 7.3.23 which improved that over older versions..but its still quite far away from what we all know from pstn/isdn
17:19.53BlackvelAND voip is only local at the office (ISDN pstn, 1meter to patton + 1meter to asterisk/snom
17:27.12*** join/#asterisk nachoguy (~boster@nacho-tested.hrapproved.com)
17:30.31*** join/#asterisk cusco (~trilili@62.28.152.206)
17:30.32cuscohi
17:30.44Kattyohai2u
17:30.50cuscoasterisk for some reason is not listening om port 5060
17:30.55cuscoon external IP
17:33.00*** join/#asterisk polk330 (4297116e@gateway/web/freenode/ip.66.151.17.110)
17:33.21drmessanoHow do you know?
17:34.06drmessanoYou can't just make a blanket statement like that, bro
17:34.28drmessanoDon't tase your Asterisk box without proof
17:34.37polk330Hey Guys, Useing AsteriskNow 2.8.0.2 Just wondering if there are any Front end typr thing for something like a Customer Interface. For someone that my company would sell VoIP to. Somewhere that thye can chage there Followme/VM etc
17:34.54cuscodrmessano: telnet on port 5060 gives connection refused
17:34.54Nuggettelnet is eeeeeeevil!
17:35.06drmessanocusco:  It's UDP, of course it is
17:35.06cuscoand I had permission to launch nc -l -p 5060
17:35.13cuscoit is UDP?
17:35.18drmessano5060 UDP
17:35.18Troxcough - udp - cough
17:35.36cuscoo.O
17:35.37leifmadsenp3nguin: regarding "how many registrations can asterisk handle right now?" see the [Code Review] Rate limit astdb->sync() calls" thread on asterisk-dev. Stefan Schmidt says 1.6.2.10 can do about 8k registrations, with the patch in that thread it goes to 15k, and with that patch plus another that will be on reviewboard soon, he gets over 20k peers without a problem. To quote him, "20k peers with qualify=yes would cause a l
17:35.37leifmadsenoad of 0,07 on my system, sending around 350 to 400 sip packets per second and around 2 mbit of bandwith."
17:35.48cuscosorry
17:36.01*** join/#asterisk n3hxs (~HAMming@63.68.135.4)
17:36.02Qwellholy crap.  2mbit from qualifies?
17:36.08leifmadsen20k registrations :D
17:36.20cuscowell I can't find out what is wrong then, router has forwarding set OK... why can't I register from outside, and no error comes up on cli?
17:36.23drmessanopolk330:  You don't have AsteriskNOW 2.8.0.2, you have FreePBX 2.8.0.2, which isn't supported in this channel.  In any case, there is no portal for customers, as it's not designed to run a VoIP business with.
17:36.25leifmadsenI'm guessing that is burst traffic
17:36.43polk330My bad thanks, freepbx channel then?
17:36.44polk330lol
17:36.51drmessanoYes, but the same answer applies
17:36.56Troxcusco: portforwarding for udp?
17:37.01cuscoTrox: yes
17:37.08drmessanoYou can ask me in there and I can cut/paste my answer
17:37.20polk330drmessano any suggestions?
17:37.50[TK]D-Fenderpolk330: No.  You are using FREEPBX, and that is not meant for you to run a telco out of.
17:38.01[TK]D-Fenderpolk330: That is made for SMB PBX use.
17:38.02cuscohow do I test udp connection?
17:38.07polk330Ok, well i mean it has been heard of to be done
17:38.15[TK]D-Fenderpolk330: Not with FreePBX
17:38.21polk330<PROTECTED>
17:38.23[TK]D-Fenderpolk330: * as a core sure.
17:38.43drmessanopolk330:  FreePBX is designed to setup/admin a PBX, not an ITSP.. You need another GOOEY, which you probably have to roll yourself
17:38.55polk330Yeah, via PHP.
17:39.01drmessanoor whatever
17:39.15drmessano.NET, something that rhymes with PAILS, whatever
17:39.42polk330Would you have any idea where to start?
17:39.47drmessanoGenerally your Asterisk box wouldn't even host this GUI.. it would be done on some backend and perhaps a MYSQL connection to Asterisk
17:40.02polk330Yeah for security reason
17:40.27drmessanopolk330:  No, because you dont run your customer frontend or billing on your call handler
17:40.48drmessanoWhy not just pass all the calls through the secretarys laptop?
17:40.54polk330Lmfao,
17:41.49drmessanoDo you realize what you're going to need to start an ITSP?
17:42.26drmessanoIt isn't just about having an Asterisk box and some ATA's or phones connected.  It's the PSTN end of the cable where the money starts
17:42.28leifmadsenno one realizes what they need to start an ITSP :)
17:42.32Qwelldrmessano: lies.
17:42.39Qwell~next vonage
17:42.50Qwell~vonage
17:42.50infobotit has been said that vonage is a bunch of monkeys
17:42.54Qwellshrugs
17:42.59leifmadsen~nextvonage
17:43.00polk330<PROTECTED>
17:43.36drmessanoOk, well you need to get your Asterisk knowledge well above "I just installed AsteriskNOW" before beginning to attempt it
17:43.51drmessano~book
17:43.51infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
17:43.55drmessanoRead that, 3 times
17:44.10Troxand terrorize jared a bit ;)
17:44.22Qwelladds an option 7 to the AsteriskNOW kickstart list.
17:44.22polk330Ive installed AstersikNOW b4
17:44.31Qwell"7) The Next Vonage"
17:44.42drmessanopolk330:  We are WAY beyond AsteriskNOW here, period
17:44.43polk330Ive even set it up with a customer side,
17:44.49polk330What are you at now?
17:44.51polk330lmfao
17:45.07b14ckpro troll
17:45.08drmessanopolk330:  You're talking about IIS on an XP box, we're talking about Apache clusters in a datacenter
17:45.25polk330Yeah? well you dont have to be a huge dick?
17:45.32drmessanoI am?
17:45.33*** join/#asterisk tzanger_ (~tzanger@gromit.mixdown.ca)
17:45.43tzanger_stupid freenode.
17:45.44b14ckpolk330, what's your goal?
17:46.00*** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net)
17:46.09b14ckpolk330, I'll give you some practical advice.
17:46.19polk330k
17:46.20drmessanoHe can sell you a big Trixbox
17:46.21nachoguybeing a huge dick would be to tell you to take your tinkertoys and play elsewhere.  He's (fairly kindly) telling you that you haven't thought the problem through
17:46.27b14ckdoesn't use trixbox.
17:46.43polk330Trixbox is shitty to.
17:46.45[TK]D-FenderUsed to WORK for them if I recall..
17:46.55b14ckI used to work for them.
17:47.00b14ckI'm not stupid though.
17:47.01Qwell[TK]D-Fender: he had the sense to QUIT
17:47.01b14ck=p
17:47.34*** join/#asterisk cusco (~trilili@213.63.137.210)
17:47.44[TK]D-Fender[13:46]<b14ck>I used to work for them. [13:46]<b14ck>I'm not stupid though. <--- auto-contradiction FAIL :p
17:47.59b14ckTo be fair, [TK]D-Fender, it was my first telephony-related job.
17:48.11b14ckMy experience is in programming, and before working there I didn't know anything about telephony stuff.
17:48.25*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
17:48.28tzangertelephony's fun
17:48.38tzangerthe realtime constraints of tdm networks doubly so
17:48.52b14ckYah, telephony is ok =)
17:49.00Qwellphones suck
17:49.09upbb14ck: how do you rate the dialplan implementation/format ?:P
17:49.14b14ckupb, it sucks
17:49.18b14ckbut so does AEL.
17:49.25drmessanopolk330:  You can call me names if you want, but you really need to understand the scale of what you're trying to do here.. Setting up a box designed for an office PBX, that controls the Asterisk dialplan to accomplish JUST that, is not the way to go here.. you need to be thinking at a much higher level.. Dedicated vanilla Asterisk install, likely a custom backend to glue it all together
17:49.39b14ckThe best telephony libraries are that which come with twilio/tropo =p
17:49.42upbheh yeah, just wanted to know someones opinion who has design experience
17:49.54b14ckThere's almost no room for proper design patterns with asterisk stuff :(
17:50.17EndEngshould you asterisk box live in your DMZ or internal network?
17:50.19*** join/#asterisk Martinblr (~Miranda@61.12.17.170)
17:50.27b14ckEndEng, internal network.
17:50.39drmessano...or just stick with Asterisknow and send me your customer list when it hits around 200 and you can't bill or scale.  Whateve.
17:50.51MartinblrI am trying to connect linksys SPA3102 box to my Asterisk and trying to send FAX but not able to send..
17:50.56drmessanoHe's gone anyway :(
17:51.11*** part/#asterisk ShaunR (~shaun@freenode/sponsor/NDChost.com)
17:51.25Martinblrthe extension created in the Asterisk is configured in Linksys SPA3102 and fax machine is connected..
17:55.34*** join/#asterisk RypPn (~TuMbL@rosscom.co.uk)
18:11.27*** part/#asterisk Trox (~Adium@85-126-174-18.work.xdsl-line.inode.at)
18:12.20voxterHey, any of you guys know of/use a tool that is preferably open source, that will take in a pcap stream (or just listen on an interface) and analyze sip/rtp to identify when there are periods of poor audio quality?
18:12.41voxterIm having an intermittent issue with audio quality degrading then going back to normal and I'd like to be able to pinpoint it happening
18:16.14cuscosipp (sip-tester package in debian)
18:16.36cuscoits hard to handle tho
18:17.28voxtersipp can take an active stream from a live box and analyze RTP as well for low MOS?
18:17.38voxterI thought sipp only told you about SIP
18:17.48cuscoow sorry it is
18:17.58*** join/#asterisk Ad-Hoc (~nimbus@62.1.130.40.dsl.dyn.forthnet.gr)
18:19.19coppicea rolling SIPP gathers no MOS
18:19.52*** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com)
18:20.04asilva<PROTECTED>
18:23.42[TK]D-Fenderasilva: It means that no function with that name exists
18:24.19[TK]D-Fenderasilva: Perhaps you should show us your failed call that generated that error.
18:26.29asilvahere it goes
18:26.30asilvahttp://pastebin.com/CEkENJr0
18:29.43[TK]D-FenderaliAnd the source line that coased it from your dialplan?
18:31.10asilvais under configuration
18:31.19asilvaon the pastebin
18:31.22asilvaline Set...
18:31.35LetoricOk, I found this yesterday, but today it is evading my googlefoo! Can anybody point me to a good method for setting up a feature code that allows our team to activate a different greeting on our phone system when we're in a meeting? IE, an 'all our people are busy, if it's an emergency, dial 911, otherwise sit and wait!!' ;)
18:32.15asilvabefore the Set verbose message asterisk prints the error message
18:32.15Letoricsorry asilva, I was scrolled up, didn't mean to disrupt your conversation ;)
18:32.29asilvaohhh np!!!
18:33.24cuscoasilva: means what it says. You are using IfTime but asterisk doesn't have that function registered
18:33.45asilvacore show funcion command shows IFTIME there
18:33.54[TK]D-Fenderasbeacuse that is in UPPERCASE
18:34.06[TK]D-Fenderasilva: Which was your mistake.  Functions are CASE SENSITIVE
18:35.18asilvalol that's right
18:35.25asilvasolveddd
18:35.39asilvahead full of things can't see a little thing..
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18:36.59Letoricok, now for mine ;)
18:37.18LetoricOk, I found this yesterday, but today it is evading my googlefoo! Can anybody point me to a good method for setting up a feature code that allows our team to activate a different greeting on our phone system when we're in a meeting? IE, an 'all our people are busy, if it's an emergency, dial 911, otherwise sit and wait!!' ;)
18:37.22Letoricperty please!
18:37.53Letoricneed to be able to activate and deactivate it by feature code
18:40.31[TK]D-FenderLetoric: "core show application gotoif", "core show function DB" <-----------
18:41.05[TK]D-FenderLetoric: Set a db value to indicate what state you should be in.  Checkk that state where your call comes in and go somewhere else accordingly
18:42.16Letoricok, I understand the basic theory, but may need a bit more help
18:42.39Letoricwill try out my googlefoo and come back in a bit if I'm stuck. Thank you
18:45.02Blackvelwhat settings of a jitter buffer do you use on LAN (patton smartnode)? default seems to be 60ms
18:46.35Blackvelwhat is the default for snom phones?
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18:53.56Blackvelhm. both patton GW 4634 and snom370 have alaw/ulaw with 20ms
18:54.25Blackveli have no idea where i should "tune" for low latency / LAN
18:56.30Blackvelcan reducing jitter buffer in patton gw from 60ms to 20ms improve delay/latency?
18:58.02Chainsawlooks for a jitter buffer setting in his 4634 config file and can't find any
18:58.08ChainsawSo I must be on whatever the default is...
19:01.02Blackvel60ms is default
19:01.21Blackvel+ adaptive
19:01.38Blackvelthis delay/latency makes me crazy and any business talk useless
19:01.47QwellWhy do you have a jitterbuffer enabled, if it's a meter connection?
19:01.54Qwellyou said it's just LAN, right?
19:02.09Blackvelsnom370 - asterisk - patton gw - ISDN
19:02.40Blackvelthere is no turn off...if i see it right...but i could set to a static one (from adaptive)
19:02.55Blackveli doubt that will changes anything anyway
19:03.39Blackvelwhat could i be doing that packets get transmitted "faster"?
19:03.42Blackvelor shorter?
19:04.04Blackvelpatton gw would allow 10ms setting
19:04.11Blackvelbut snom only let me choose 20ms
19:04.34Blackvelagain...dunno if that would change anything...
19:08.27Blackvelcan it be that snom FW messes voice packets and doesn't get voice to the point?
19:08.54Blackveldo you all have the same problems within offices and smartnode/pstn/isdn gateways?
19:09.53*** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk)
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19:51.33KnewHey
19:51.42Naikrovekhey
19:51.58KnewAnyone know of a PBX/IVR system that presents the customer (the caller) with a web interface?
19:52.32KnewIVR systems suck. Even the speech ones.
19:53.00Naikroveki had an idea once where the phone you were calling from would present you the menu on its local interface, like softbuttons or something
19:53.18Naikrovekthe communication is already digital <--> digital, why rely on DTMF
19:53.22KnewHas anyone seen a solution where the user makes their menu selections/enters extension or account info on the web site securely, then gets called by the menu system and patched through?
19:53.22Naikrovekbut no one thought it was a good idea
19:53.37NaikrovekKnew: that is possible, but i don't know of a system that already does this
19:53.48KnewNaikrovek, do you think it's a good diea?
19:53.49Knew*idea
19:53.57Naikrovekyeah why not
19:54.14Naikrovekwould make a lot of sense in customer service
19:54.16Kattyhi Naikrovek
19:54.21Naikrovekhi katty :)
19:54.28Deeewayneknew, is your user an agent?
19:54.35KnewRight now with Chase, they have me entering my account information and PIN number over unencrypted, analog phone lines using DTMF. How secure
19:54.37Kattyhugs Deeewayne
19:54.44Kobazit's teh katty\
19:54.55Kattymhmm
19:54.55Deeewaynebear hugs Katty
19:55.09KnewDeeewayne, what user? :p
19:55.39Deeewayneknew, re: 'user' in your 14:53:21 comment
19:55.47KnewI'm going to have about four developers working under me for a week and I'm thinking about what we want to do for a project.
19:56.03KnewDeeewayne, oh. The user is the caller.
19:56.15*** join/#asterisk moa_ (~moa@65-19-228-168.vnet-inc.com)
19:57.01KnewI was thinking of developing a PBX/IVR solution that actually handles all extensions/menu navigation/account info entry over the web site, and then calls them.
19:57.17KnewVirtual queueing would of course exist automatically, too.
19:57.48KnewIt would tell them on the web site how much longer to wait
19:58.28KnewWhat do you guys think?
19:58.38nucceMay I have any ideas what too do with this problem? Ive got a wav file that I want into my asterisk.. So I did simple convert it into a alaw pcm64a file.. and when I call with a static phone it sounds great.. but when I call with a mobile phone it sounds crap.. It seems it try too play it too loud and makes the noise very bad.. Anyone have some tip howto solve this?
19:59.05NaikrovekKnew: what you are proposing is a great idea imho
19:59.07nucceI did try to lower the volume with sox but the result still aint any good.
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19:59.47Naikroveknucce: turn down the volume on the mobile phone, and try it on more than one mobile phone
19:59.52Naikrovekit could be a GSM codec thing as well
20:00.05nucceI did try it on 3 different phones with 3 different operators.
20:00.12Naikrovekthen it's GSM or something else
20:00.23Naikrovekthe GSM audio codec is optimized for voice and nothing else
20:00.47nucceWell, other helplines got thoose audietunes and it sounds good from mobile phone
20:01.22Naikroveknucce: yes but they pre-process the file.  removing almost all low and high frequency sounds so that GSM doesn't chop them out and make them sound awful
20:01.37Naikrovekput it through audacity and turn down the bass and treble, and try again
20:01.44Naikroveki think you'll find a much better outcome
20:01.56nuccethank you! that is what I call a good tip!
20:02.02Naikrovekwell
20:02.07Naikrovekit's only a good tip if it's useful
20:02.21Naikrovektime will tell
20:02.48nucceI will try now.. :)
20:03.26nucceIm not so good with this audicity due
20:03.28*** join/#asterisk adyn (~adyn@unaffiliated/adyn)
20:03.31nuccefirst time I download it now
20:03.49hardwirevoxter: poke
20:04.17voxtersup
20:04.41hardwiredo you guys operate as a hosted wholesale platform provider?
20:05.03voxterhardwire: we do now yep
20:05.11hardwireInteresting
20:05.12*** join/#asterisk uqlev (~yuriy@91.184.221.31)
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20:05.19voxterhardwire: north america coverage only for now, though
20:05.54hardwireI was thinking we'd use a hosted wholesale platform that somebody else maintains and manage our trunks through that platform as well as rates and LCR
20:06.09hardwirenot specifically use a wholesale provider, but have somebody host the wholesale platform
20:06.19voxterhardwire: interesting, yeah we are just about finished rolling out a new API that allows us to give direct access to our provisioning platform for just that reason
20:06.38voxterhardwire: and we also have it set up to work for resellers as well, with multiple clients below them
20:06.46hardwirewhat about least cost routing and the like?
20:07.01*** join/#asterisk guilept (~guile@a213-22-53-84.cpe.netcabo.pt)
20:07.22guilepthei
20:07.31hardwirehush
20:07.45guileptwhats up?
20:07.48hardwireshhhhhhh
20:08.39Qwellwe're mourning.
20:13.20frigidzephyri read that as mounting
20:13.32frigidzephyrconfused me for a minute
20:13.51guileptmy condolences
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20:28.44ahowladerhi tzafrir_laptop
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20:39.58EndEngis there a asterisk PPA for lucid?
20:40.41QwellEndEng: #ubuntu
20:40.53EndEngyes ubuntu
20:41.59Qwell/join #ubuntu
20:42.01Naikrovekheh
20:42.13tzafrir_laptopahowlader, hi
20:42.41ahowladerhello tzafrir_laptop
20:42.50*** join/#asterisk xuser (~xuser@unaffiliated/xuser)
20:43.01QwellLOOP DETECTED
20:44.25*** join/#asterisk cjk (~cjk@vodsl-8936.vo.lu)
20:44.30tzangerQwell: does the loop give a funky groove?
20:44.44Letoricok Fender: I did some brief research. The samples I found on voip-info.org seem to reference out of date commands
20:44.47LetoricDBPut and DBGet
20:45.07Letorichow do I set and read the database with 1.6.2.11?
20:46.03*** join/#asterisk unspin (~unspin@S010600031d02196a.vc.shawcable.net)
20:46.24Naikrovekcontinue your research and you will find that info, surely
20:46.42Naikrovek([TK]D-Fender has stepped out)
20:46.48Letoricthanks
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20:55.46KattyATTENTION
20:55.51KattyIT IS NOW 1 HR UNTIL 5PM, CST
20:55.53Kattythat is all.
20:56.08Kobazheh
20:56.12Kobazit's 4:56 here
20:56.19Kattyi hate you.
20:56.26Kattypouts.
20:57.37Kobazheh
20:57.56LetoricNaikrovek - I don't quite follow what the wiki is saying about the database stuff
20:58.19LetoricI did find the updated commands
20:58.30Letoricbut it's not quite clicking on how to mold that into what I need heh
21:00.04Letoricin the example, is family/key literal, or is that something like table and row?
21:00.11Letoricwhat is $foo supposed to be?
21:00.25Letoricerr, ${foo}
21:01.07KattyKobaz: i am so bored.
21:02.06Kobazyou can come visit
21:03.10Kattybrt
21:03.18Kobazbrt?
21:03.36Kobazbears remember tennis?
21:03.45Qwellbacon's really tasty
21:04.33Kobazbeavers racing through?
21:06.12Kattyi like Qwell's better.
21:06.29Qwellthat's wha...  nevermind.
21:06.42Kattymhmm
21:06.59hayasking again since many joins and parts happened.. I would like to have asterisk running on FreeBSD and on an ISDN line (1 BRI)... what PCI ISDN cards are tested and working in this combination... of course, as cheap as possible :-) thx
21:07.16Kattyhay: are you looking for a consultant?
21:07.22Kobaz~cheap
21:07.22infobotcheap is probably a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
21:08.07hayKatty: I suppose freenode IRC network isn't a right place of looking for a consultant...
21:08.22Kattyhay: well it can be...i'm just not really sure waht you're asking for.
21:08.48KattyQwell: and no comments about previous statement. kthx.
21:09.02Kobazhay: your question could either mean "i want someone to set this up for me". or "i want a point in the right direction for getting this going"
21:09.02Qwellhuh what?
21:09.07hayKatty: I am looking for some experiences of experienced users using FreeBSD and ISDN cards... as I mentioned before
21:09.28Kattyhay: that's still not specific enough.
21:09.28hayKobaz: second meaning was almost exactly what I wanted...
21:09.32Kattyhay: what, precisely, do you want
21:09.47KobazKatty: someone to tell him which isdn cards work in freebsd
21:10.05Kattyah.
21:10.20hayKobaz: and in Asterisk ...  I hope that isn't too tough question...
21:10.43beardyWell, if you couldn't find out yet it can't be that easy.
21:10.47Kobazhay: i'm not sure if we can handle that
21:10.53Kattyhi beardy
21:11.04Kobazit sounds hard
21:11.07beardyHi Katty
21:11.27Kattyyou know what sounds good.
21:11.32Qwellbacon.
21:11.34hayif there are some docs anywhere I will be glad to read them... however, I wasn't able to find anything useful...
21:11.36Kattytacos.
21:11.40Qwellbacon tacos.
21:11.42beardyhay: There might be hardware compatability lists for FreeBSD, as there is for Linux, looked for that?
21:11.47Kattymmmm, no.
21:11.49Kattybeef/turkey tacos
21:11.56QwellI think I might have to go get tacos this weekend now
21:11.58Kattybison would also be acceptable.
21:12.06beardyKatty: Mm tacos
21:12.12*** join/#asterisk jamicque (~jam@80.50.125.74)
21:12.18Kattythose volcano tacos at taco bell are win.
21:12.22NuggetQwell: any chance you've got a sample cisco phone config file for use with sccp/skinny?  I want to take another pass at using that instead of sip and the internets are coming up empty
21:12.25Qwellwait, they still make those?
21:12.27KattyNugget: ohai
21:12.32jamicquehi, can anyone recomend me a solution to make a CTI for Asterisk?
21:12.33QwellNugget: nope
21:12.34KattyNugget: i was looking for you earlier. something about facebook.
21:12.35Nugget79x1/79x5 series, not that old 79x0 shit
21:12.40KattyNugget: but it seems to have slipped my memory
21:12.42Nuggetok, thanks
21:12.46Nuggethuggles Katty
21:12.54Kattyhugs Nugget, goes to refresh memory
21:12.56haybeardy: yes, our course... but combining that with asterisk and overall experience using it and getting some information from people that use it looked like something useful...
21:13.27haybut I suppose nobody is using freebsd with asterisk and ISDN at all? or are you all hiding it? :-)
21:13.41Qwellhay: Here's a tip.  If you're having issues finding information about using FreeBSD..  Don't.
21:13.56beardyhay: Best of luck with it. Asking meta questions usually isn't the best approach though.
21:14.01KattyNugget: oh yes, the tea party thing.
21:14.22KobazQwell: hey... freebsd's not that bad
21:14.23KattyNugget: did you find that off of reddit?
21:14.27KobazQwell: it's got zfs
21:14.32Kobaz(which i use at home)
21:14.34QwellKobaz: you get to support him then ;)
21:14.36Nuggetno
21:14.41Kattyoh ah
21:14.43Kattydigs up link
21:14.45hayQwell: actually I am pretty satisfied with FreeBSD docs (a long time sysadmin of it), but asterisk's hardware lists seem to be hard to find for me :-)
21:14.58KobazQwell: well i never set up asterisk or isdn cards in freebsd though... i couldn't even get asterisk to compile
21:15.13NuggetFreeBSD is doubleplusawesome, but if you want to run Asterisk just use Linux.  I hate Linux just as much as the next guy, but it really is the path of least pain.
21:15.14hayoverall... I am really surprised by the reactions of all of you here... I didn't want to hurt anybody's feelings at all... :-)
21:15.27KobazNugget: why all the hate?
21:15.32KattyNugget: http://i.imgur.com/1pWX5.jpg
21:15.33beardyKatty: So you're hungry again? :)
21:15.41NuggetI love Unix and Linux is a pretty shitty Unix.
21:15.44hayNugget: excellent... that was an excellent answer... thanks :-)
21:15.59Kattybeardy: are you kidding me? i have the metabolism of a ferret.
21:16.13KobazKatty: is that high or low?
21:16.21KattyKobaz: that means i'm starving every 2 hours
21:16.22beardywonders too
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21:16.53NuggetKatty: yeah, that's the pic that led me to the site.
21:17.01Nuggetsomeone posted it somewhere I saw it, so I googled.
21:17.50Kobazlinux is supposed to get a native port of zfs soon
21:17.57Kobazbut not in the mainline kernel because of the stupid cddl
21:18.05Nuggets/cddl/gpl/
21:18.08Nuggetfixed that for you
21:18.27Kobazbut btrfs looks like a better fs anyway... it just needs to be finished
21:18.36Nuggetthe gpl is the license that was designed to be intentionally difficult to cooperate with other licenses.  Not fair to blame the cddl for that.
21:18.47KobazNugget: no it wasn't
21:18.57Nuggetyes it was.  stallman is quite clear on that.
21:18.58KattyNugget: excellent.
21:19.02KobazNugget: the cddl was designed like that... you have it backwards
21:19.21Kobazstallman is clear on that he wanted a very specific definition of free
21:19.27Kobazplenty of other licenses are compatible with the gpl
21:19.32Nuggetno, the gpl did it first.  stallman expressly said he viewed the gpl's intentional incompabatibility to be a key mechanism he wanted to use to drive developers to use the gpl.
21:19.53Nuggetthat was a design *goal* of the gpl -- to be hard to cooperate with other licenses
21:20.00Nuggetthere's plenty of writing on fsf.org that explains that
21:21.07Kobazthere's rumors that sun specifically picked cddl to keep zfs out of linux
21:22.37Nuggetthere's rumors that nasa faked the moon landing.  I'm not sure that tells us muc.
21:23.17Nuggetand, even if it's true, that doesn't change the fact that the gpl's deliberate incompatibility is what made the decision possible.
21:23.39Nuggetit's trivial to make an open source license incompatible with the gpl -- witness the old form bsd "advertising clause" license
21:24.22Nugget(which pre-dates the gpl, obviously.  I'm not saying it was crafted to be incompatible, that would be impossible.  just that even the most innocuous constraints can exclude gpl cooperation)
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21:40.03Kobazhmm
21:40.37*** join/#asterisk DruRoland (~dru@173.200.93.214)
21:41.47DruRolandanyone available to help with some, probably, basic questions regarding extensions?
21:43.41ChainsawQuite likely. Please proceed with your question.
21:44.18DruRolandI've set up an outbound context that handles outbound NANPA calls, but I'm unable to set up another extension to forward calls elsewhere if they're in a 88XX format
21:44.59DruRolandbasically, I want the pbx to outbound dial via a ITSP, but route over a different SIP provider for internal extensions between branch offices
21:46.15voxteranyone here use VQManage?
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22:13.24LemensTSi changed the ip of pbx.myserver.com   ... is there anyway to make the devices reconnect to the new server? i could shut the old asterisk off or something
22:16.52[TK]D-FenderLemensTS: Perhaps you should check your device's respective manuals
22:17.12[TK]D-FenderLemensTS: Because no sever tells some random client to connect to it.  That is backwards
22:17.28pabelangerIf you are using DNS, it should resolve itself
22:17.56Qwellpabelanger: I see what you did there.
22:19.46pabelangerhttp://captionsearch.com/pix/2rub7eoeem.jpg
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22:32.49DruRolandHmm, anyone gotten asterisk and televantage to play together? I'm trying to get the asterisk server to register as a SIP peer on the tv server
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23:03.05[TK]D-FenderDruRoland: Perhaps you could show us your actual problem
23:03.08[TK]D-Fender~pb
23:03.08infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
23:03.09[TK]D-Fender^^^^^^^^^^^
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23:22.36DruRolandhttp://pastebin.com/y9jCGsgG
23:23.06[TK]D-FenderDruRoland: Where is the FAILURE?
23:23.27DruRolandI haven't gotten far enough to have a failure yet
23:23.37DruRolandI'm still working on where to start
23:23.50[TK]D-FenderDruRoland: Then what have you shown me?
23:24.02[TK]D-Fenderthat isn't a "Start"?
23:24.33DruRolandTrue
23:24.46[TK]D-FenderDruRoland: I see no dialplan that would use your [televantage] peer.
23:25.13DruRolandyes, I'm not sure how to set that portion up
23:25.14[TK]D-FenderDruRoland: And that same peer points to [inbound] which you seem to show as being empty.
23:25.37DruRolandit is empty.. that's what flowroute has for their sample astrisk config
23:25.47[TK]D-FenderDruRoland: And you haven't told us which DIRECTION you are looking to work on
23:25.59[TK]D-FenderDruRoland: What does flowroute have to do with this?
23:26.51DruRolandI'm trying to send 88xx to the [televantage] peer, and all other numbers to the [flowroute] peer
23:27.41DruRolandso, to answer your previous question, this is all dealing with outbound calling
23:28.06[TK]D-FenderDruRoland: You have no dial() at all using that peer.
23:28.12[TK]D-FenderDruRoland: So go MAKE one.
23:28.25[TK]D-Fenderdru you have failed to claify WHO is dialing the 88xx
23:29.33DruRolandThe caller is a web app integrated with the asterisk server
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23:36.54[TK]D-FenderDruRoland: Which does not tell us how it communicates to * at all.  Or what context it's calls fall into.
23:38.58[TK]D-FenderDruRoland: And you have no exten anywhere to match that patter, and none that use the [televantage] peer
23:41.53LemensTSwhat file has the cli output info for verbose, debug, warning, etc?
23:41.58LemensTSi thought it was console.conf but dont see it
23:42.42LemensTSwait nevermind its CONSOLE in logger.conf
23:48.51[TK]D-FenderBAI BAI
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