IRC log for #asterisk on 20090717

00:00.13Cyonleifmadsen:  But RDNIS is set on the forwarded call.  Progress.  :)  (Yes, I have no knowledge of this stuff, I just do what I have to when necessary)
00:00.28jpsharptrial by fire.
00:00.35morko757leifmadsen: the site describes on product:24 port modular analog PCI 3.3/5.0V card with 4 Trunk interfaces and HW Echo Can
00:01.01morko757so this means that it only has 4 actual ports that can attach to the
00:01.04CyonAlways is...years upon years ago I had time to monitor all the mailing lists, and stay active on all the irc channels...but life isn't no pleasant anymore.
00:01.06Cyon*spo
00:01.08Cyon*so
00:01.13morko757device
00:01.30morko757bit it can be upgraded up to 24 ports total
00:01.34morko757is that correct?
00:01.35leifmadsenmorko757: it means there is only 1 module the comes with the card -- you could order more modules later
00:01.50leifmadsenor you could by enough modules to run all 24 ports right away
00:01.54morko757ok, so the 24 was misleading to me.  It really does not have 24 ports
00:02.00leifmadsenit has 24 ports
00:02.07leifmadsenit doesn't necessarily have them all active
00:02.33morko757ok, it physically has 24 ports, but you would need modules to control them
00:02.36morko757gotcha
00:02.46morko757what kind of answering detection do these devices have?
00:02.48leifmadsenhttp://store.digium.com/productview.php?product_code=1TDM2406EF
00:03.02leifmadsenmorko757: you want that one -- 24 ports with 24 trunk interfaces
00:03.09leifmadsentrunk == phone line, station == phones
00:03.31morko757gotcha
00:03.39morko757for FXOs, how does asterisk detect an answer?
00:03.49leifmadsenvoltage change
00:03.51j0does asterisk & zaptel automatically make outbound phone calls from top to bottom? eg, line 4,3,2,1.. i'm getting collisions sometimes where an outbound call collides with an inbound call
00:03.54leifmadsentypically
00:03.56j0if not, how do i change it?
00:04.14morko757voltage change...does that have anything to do with fxo ac impendance?
00:04.17leifmadsenj0: it does if you tell it too --   Zap/g1 vs. Zap/G1
00:04.28leifmadsenmorko757: I have no idea -- I don't do much hardware :)
00:04.39leifmadsenI'd just call the sales department and ask
00:04.46j0leifmadsen: what is that called? (what do I google?)
00:04.50morko757you've been very informative, do you know someone who would know?
00:04.56morko757fxo ac impedance
00:05.03morko757google doesn't come up with that much
00:05.07ManxPowerj0: Are you sure you are using Asterisk and not one of those "Asterisk GUIs"
00:05.27ManxPowermorko757: I believe in the USA phone lines are 600 Ohm, but it should not matter at all to you.
00:05.34morko757found a page describing it, but don't know how it would apply practically to an Asterisk system
00:05.36j0ManxPower: oh common... :) yes, i'm using a "gui" one
00:05.49morko757Max, why does it not matter?
00:05.56ManxPowerCalling an Asterisk GUI "asterisk" is like calling windows 95 "dos"
00:06.13ManxPowermorko757: why should it matter?
00:06.25ManxPowerj0: go to one of the Asterisk GUI channels.
00:06.26morko757well, if I knew what is was, I'd be able to tell you
00:06.30ManxPowerThey should know what you need to know.
00:06.56ManxPowermorko757: It does not matter.  If the device is for USA phone lines then it will have the correct impedence.
00:07.19morko757asterisk can be configured for other countries?
00:07.27leifmadsenpotentially, yes
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00:07.44ManxPowermorko757: Yes, but as long as your tell the hardware driver what country you are in then it will set the correct options for that country.
00:07.45leifmadsendepending on the country -- it does understand stuff in many of the more populace 1st world countries
00:08.11morko757the hardware driver - is that dahdi?
00:08.16leifmadsencorrect
00:08.18morko757ok
00:08.19ManxPowermorko757: Yes.
00:08.34morko757does the impendance effect the answer signal in anyway?
00:08.39leifmadsenhardware is controlled by DAHDI, and Asterisk interacts with the hardware driver via chan_dahdi
00:08.48morko757dahdi, or zaptel
00:09.11morko757I had a gateway that only had battery reversal and voice detection as an answering detection
00:09.19j0ManxPower: what am i asking for at least?
00:09.45ManxPowerj0: I would call it the order of channel usage.  I dunno what GUIists would call it.
00:09.55j0thanks
00:10.00morko757I couldn't get the battery reversal to work properlly and wonder if that had anything to do with the impedance not set up correctly
00:10.01leifmadsenmorko757: what country are you trying to use the hardware in?
00:10.10leifmadsen(or would)
00:10.13j0here's hoping that option is available in a gui
00:10.42leifmadsenj0: potentially hardware group?
00:10.46leifmadsencould be any number of things
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00:14.32morko757what was that?
00:14.32j0ok.. what is it called when i'm using the "real" asterisk?
00:14.32Cyonwb
00:14.32ManxPowerj0: Remember, Asterisk really isn't a PBX.  Asterisk is a TOOLKIT that lets you build a PBX.   Because of this asking things like "what is the default channel order" doesn't really mean much, since you would set that up when you built the PBX.
00:14.33guaxand the world isone again
00:14.33guaxis one
00:14.33ManxPowerj0: It's called "Asterisk".
00:14.33jong2that is deep
00:14.34j0right, and that's how i learned it.. but now i go for quick and easy, which for me, is a standard system that other techs have seen before
00:14.43guaxmorko757, the split? (all people out and in?)
00:14.43ManxPowerTerms for GUI Asterisk are things like "Trixbox", "Switchvox", "Asterisk-GUI", "Asterisk-NOW", or "PoS"
00:14.44j0you're giving me a hard time about capitalization?
00:14.45ManxPowerj0: then you might want to ask your questions on a channel dedicated to your GUI.
00:15.00ManxPowerj0: no, I'm giving you a hard time about calling "asterisk gui" (whatever the caps are) as "asterisk".
00:15.41ManxPowerI suspect the vast majority of people here have never even see an Asterisk gui. 8-|
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00:16.15Cyoncringes if I ever had to
00:16.21j0okok
00:16.29ManxPowerBut I bet the people over on #Asterisk-GUI have seen it quite often.
00:16.29Cyontxt configs and clis for teh win
00:17.17j0what do you use for a standard install then?
00:17.44ManxPowerj0: My own configs that I built based on the customer needs.
00:18.09j0so when another tech needs to work on it, what then?
00:18.17j0or even another company?
00:18.21Qwellthey are a tech and can read configs
00:18.27CyonThey understand how asterisk works and can read a config file?
00:18.33QwellCyon: exactly
00:18.37ManxPowerj0: I hope they understand Asterisk because if they don't then they have no busimess trying to work on a PBX
00:19.04Qwell"But what happens if somebody needs to fix the mail server in the future?"
00:19.10QwellWell, you better have configured it properly..
00:19.32Cyonj0:  You are a microsoft guy originally, aren't you?  ;-)
00:19.45j0Cyon: how'd you guess ;)
00:19.56jong2Safiworkshop can help microsoft guy!
00:20.06jong2shameless plug... cough cough.
00:20.12ManxPowerMakes about as much sense as asking a person that only knows Win95 to fix DOS problems.
00:21.34toughmarketingDoes anyone know of a working app_swift for asterisk 1.6 on a 64bit machine?
00:22.40morko757guax...yes
00:22.43morko757the split
00:23.22guaxmorko757, when two servers lost its connections the network spits into two (or many)
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00:23.44morko757I see
00:23.47guaxso people from one server to another dont see each other
00:23.55Cyonguax:  Correct
00:24.07morko757and when they reconnect, they join up
00:24.11guaxnet split in a nut shell =P
00:24.11morko757again
00:24.17guaxmorko757, exactly
00:24.20Qwellwhen they reconnect...
00:24.22QwellBATTLE TO THE DEATH
00:24.26Qwellwinner gets ops
00:24.29guaxpretty visual attractive
00:24.35guaxQwell, auheuheahue
00:24.52Qwellhow do you think I got ops here? :D
00:25.03Cyonlol
00:25.13CyonI thought it was knowledge and hard work.  :D
00:25.22Qwelloverrated
00:25.44guaxhaha
00:25.48CyonBut yeah, I remember the good old days of chan wars, taking out a server to get ops; deoping people the second they rejoin...
00:26.09QwellCyon: something like that
00:27.29CyonWell, I'm still totally stuck on this Diversion header thing...but getting closer.  Leaving for CA in the morning, so I'll deal with it another time.
00:27.36CyonLater all
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01:08.17djMaxif I use Dial with A(file), is there a way to get it to ring to the caller while playing that file?
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01:36.56raden_workjpsharp, you still around I cannot find anywhere in the modem control panel to bridge a connection
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01:37.07raden_workclosest thing I can find is share a single IP
01:37.24raden_workSingle static IP configuration to be exact
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01:59.51raden_workanyone know how to get traceroute to work when icmp is blocked
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02:19.19djMaxI doubt you can
02:19.45djMaxunless you can find something "known" upstream that will respond to ping
02:19.54djMaxi.e. if you know the gateway of the target machine
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02:30.54leifmadsenanyone around who wants to test dialing a number for me via DNS?
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02:33.54ariel_sorry folks for the rant... but ABE and the asterisk-gui 2.0x sucks...
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02:35.04leifmadsenIf someone can try dialing sip:1234*460@leifmadsen.com that'd be great!
02:36.05voip_trollIs it possible to use a list or an array or a similar concept in extensions.conf?  i.e. I have a GotoIf statement matching on the inbound CLID, instead of having multiple lines of exten => 1234,n,....... for the match, have a single line and any additional CLIDs to be matched get added to a variable definition
02:36.56leifmadsenvoip_troll: it's called a pattern match
02:37.03leifmadsenor what is it you mean?
02:37.16leifmadsen(its late here)
02:37.26voip_trollheh, I'm a few brews into the night
02:37.49leifmadsenjust do a Goto() once you've matched what you want... ?
02:38.07leifmadsenand after setting any variables you need, like assigning ${EXTEN} to a variable to use in a different extension
02:38.21leifmadsenanyways, going to bed with the g/f :)
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02:38.27voip_trollWell, in PHP I can define an array (something like $tn = array("1112223333","2223334444","5555555555"); foreach ($tn as $num)
02:38.43voip_trollbleh
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02:45.18raden_workwhats the point of traceroute like every place i have internet its blocked 2 or 3 hops in
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03:32.55kb3ienhi. i was trying to implement a park softkey on a polycom. anyone have experience with that?
03:33.29kb3ienmuch has been written, but i'm not getting it to work.
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06:02.42tengulreanybody here using asterisk callback funtion, can give me some tips?
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06:16.47tengulreanybody alive?
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06:56.09xrmx__hi, i have a couple of ip that are trying to register with non existant account, is there a way to find if it's a misconfigured client or someone else? can i see somewhere if the same ip is registered with another account?
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07:02.01eliyahuis there any way through the CLI, AGI or AMI to retreive a channel from a uniqueid?
07:04.46eliyahuanyone?
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07:58.52kaiieliyahu: in which situation where you have access to the uniqueid of an active channel do you not have access to this channels name property?
07:59.19kaiito make the answer short, imho there is no way to do this. i may be proven wrong-
08:00.25eliyahuyeah i don't think so either
08:00.44kaiixrmx__: you could disable "sip debug" for this IP and see what's coming.  or you do a "sip show peers" and look after that IP
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08:01.14kaii*enable.  i just woke up ^^
08:01.22eliyahuthe channel name changes when it switches failover trunks
08:01.33xrmx__kaii, yeah, i've done that, it's a pity that like only match the account and not host btw
08:01.43kaiieliyahu: well then you have no chance with uniqueid ...
08:01.44eliyahuIts a local channel which is bridged into a conference, the other end is dialing out
08:02.14kaiithe uniqueid is associated with the call, the requesting channel (= caller)
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08:02.49eliyahuyou're saying that the uniqueid changes as well? I didn't realize that
08:02.51kaiithere is mostly more than one channel involved in a call
08:02.57kaiino the uniqueid does not change
08:03.38kaiibut you can not look up the name of the outgoing channel with it, because you would get the callers channel if there were such a method
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08:07.18eliyahuok, thanks
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08:21.59dweryI've got this error from chan_iax2.c:10673 build_peer: Unable to support trunking on peer 'reiss-rivalta' without a timing interface . The machine has an Astribank connected, which provides timing. dahdi show channels correctly shows the pseudo interface.  dahd_test shows an accuracy of 99.99%
08:22.22dweryany hint?
08:24.13jmkgreenAnyone here using astmanproxy?
08:24.52dweryouch, found the bug: https://issues.asterisk.org/view.php?id=15191
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08:28.59creativxjmkgreen: jep
08:29.39jmkgreencreativx: it is still used then
08:29.52creativxits still kicking on our box
08:29.57jmkgreenI built it for use with current stable asterisk
08:30.12creativxwe are on the 1.2 branch still . :)
08:30.15jmkgreenWhen I connect to the proxy it immediately segfaults
08:31.03jmkgreenI am assuming it's either buggy under modern OS/s (Ubuntu 9.04) or we're using it wrongly in a dramatic way
08:31.51creativxwell the proxy is rather simple code
08:32.03creativxim not even sure what the latest build is.. 2 years old? or has it had some updates
08:32.17jmkgreenDon't know - I grabbed the latest from github
08:32.35jmkgreenIt fires up fine and says it's received the currect GET request with params
08:32.51jmkgreenthen it issues a buffer overflow warning and segfaults every time
08:33.18jmkgreenhence I'm wondering if it's an old project no longer used with modern editions of asterisk
08:33.26creativxhttp://github.com/davetroy/astmanproxy/commit/f4b952a717b7e982b585bf0daa86398add394a88 ?
08:33.56jmkgreenWhen I send 60 simultaneous HTTPs via multicurl I can get the AMI to trip up on about 6-10 of those connections
08:34.23jmkgreenyeah I checked out sources - we appear to have that patch
08:34.44jmkgreenwhen I say 60 from multicurl I mean directly to the AMI port
08:34.52jmkgreenwell, the AJAM thing
08:35.43jmkgreenI know it's not supposed to be a full web server and spotted astmanproxy, but given it crashes on connection I have to wonder whether it's us using it wrongly or just a redundant piece of software
08:36.18creativxwell yeah
08:36.23creativxthe point of the proxy was that it would be abused
08:37.02jmkgreenthere's very little on Google about this
08:43.07noisewaterphdso I have colocated PBX's, which connect via SIP to VoicePulse. we have multiple offices, 2 with DSL (1.5/.5), 1 with 20mb fiber, and 1 with comcast business cable. We have been having the "underwater" call problems, cutting in and out, and one way connectivity.  Where is my most likely source for this?  I got a notice from even the fiber office today that they were having quality issues.  Colo?  Is this a VoicePulse thing?  
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08:47.19mbrevdaeliyahu: was your question answered?
08:49.57eliyahukind of, I think i have to find another solution
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08:56.49mbrevdadidnt see the question
08:59.23morko757can I install multiple digium cards on one serveer?
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09:00.15mbrevdayes
09:01.16morko757mbrevda: goood...how are they recognized in asterisk?
09:01.28morko757as one big zap device?
09:01.36mbrevdahowever you set them up in dahdi.conf
09:01.39Dovidno
09:01.46mbrevdayou define groups
09:01.50Dovidyou set each one up
09:01.56mbrevdag0=channelsx-channel y
09:02.00Dovidand yes dahdi. no more zaptel
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09:07.54BlinkizHi there. I would like to know if my asterisk 1.4.21 (ubuntu 9.04 repository) supports G.722. How can I check that?
09:09.27Blinkizdoes "core show codecs 722" show this?
09:09.41Blinkizasterisk cli I mean
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09:45.37BlinkizFound it. "core show transaction" shows if g722 is supported. And its not..
09:59.50BlinkizHow can I tune in asterisk so it does not send REGISTER so often to my ISP?
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10:03.12jmkgreenhmm it seems if you HTTP GET against astmanproxy it spits out the buffer overflow. bug noted in github
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10:41.49kiwi_ukhey folks, anyone had any luck with getting mISDN to compile on ubuntu 9.04?
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12:34.23devyllhello. can't seem to convert an mp3 file to a WAV for moh. it's now an uncompressed 16-bit PCM, 8000 Hz, Mono. Am I missing something ?  (err: format_wav.c:201 check_header: Unknown block - not fact or data)
12:45.28jayteedevyll, you need to strip out any ID3 tags in the mp3 and save it using an ID tag editor before converting it.
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12:47.49devylljaytee: makese sense. thanks .
12:48.07[TK]D-Fenderdevyll: http://www.voip-info.org/wiki/view/Using+ffmpeg+to+convert+Music+On+Hold+files
12:48.38[TK]D-Fenderdevyll: And ID3's aren't the problem.  You are converting to WAV, just an incompatible format, and WAV no longer has ID3's....
12:48.39jayteedevyll, that's an excellent guide [TK]D-Fender just posted the link to
12:50.14jayteeah, yeah. it's more for playing mp3's as MOH files that the ID3 tags were an issue. thought maybe something might not have gotten stripped out in the change of format
12:50.39devyllok. got it.
12:51.07[TK]D-Fenderjaytee: Yeah, but he's not TRANSCODING ;)
12:51.17[TK]D-Fender(in *)
12:51.36jayteegoes to get more caffiene, brain is not working ......
12:52.06[TK]D-Fenderdevyll: And a good way for a few files is to use * itself to convert them : "help convert"
12:52.17[TK]D-Fenderdevyll: And install format_mp3.so from addons
12:57.43devyllexit
12:57.49devyllsorry
12:57.56devyll:) thought I was in the terminal :D
12:58.16devyllin asterisk cli to be specific
12:59.27devyllgreat. it's working. thank you [TK]D-Fender and jaytee
12:59.52[TK]D-Fenderdevyll: You're welcome
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13:06.43OrNixcomrades, help me please. I've installed 1.6.1.1, it doesn't show DTMF in console, when i connect to asterisk kernel, using "safe_asterisk" and then "asterisk -r". But if i start asterisk using "asterisk -cvv", it shows DTMF-signals in the console. (sorry for my poor english)\
13:07.13[TK]D-FenderOrNix: "set debug 10"
13:09.21OrNix[TK]D-Fender, on 1.6.0.10 it worked without debug. There is too much useless information in debug to use it permanently
13:09.42[TK]D-FenderOrNix: Perhaps only debug 1 would do...
13:10.12[TK]D-FenderOrNix: DTMF alone isn't a standard option.. certainly nothing tied to 1.6.0
13:11.10OrNix[TK]D-Fender, may be is it bug? And why it works when i use "asterisk -cvv" ?
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13:11.26[TK]D-FenderOrNix: Verbosity has no impact on DTMFdisplay
13:11.40[TK]D-FenderOrNix: Thats a core debug perk
13:12.38OrNixok, with "asterisk -c" it works too. but doesn't work with safe_asterisk, (console = yes in asterisk.conf)
13:21.20leifmadsenOrNix: and you have it enabled in logger.conf ?
13:21.28leifmadsenconsole => warning,notice,error,dtmf
13:21.57OrNixleifmadsen, yes, certainly
13:22.07OrNixso it looks like a bug
13:22.11leifmadsencould be
13:22.15leifmadsentry the latest 1.6.1 branch
13:22.23leifmadsenjust to verify it hasn't already been fixed
13:22.26OrNixsvn ?
13:22.29leifmadsenif it still persists, then file a but
13:22.30leifmadsenbug*
13:22.33leifmadsenOrNix: correct
13:22.45leifmadsensvn co http://svn.asterisk.org/svn/asterisk/branches/1.6.1 asterisk-1.6.1-vanilla
13:23.29mvanbaaksvn co http://svn.asterisk.org/svn/asterisk/branches/1.6.1 asterisk-1.6.1-vanilla-ice_ice-ice-baby
13:23.36OrNixhow do you think, is it rather stable or not ?
13:24.47OrNixi'v installed 1.6.1.1 because of this patch: https://issues.asterisk.org/view.php?id=15285
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13:26.42leifmadsenOrNix: stability shouldn't matter since you're just installing on your development/test system to verify if the bug still exists, right?
13:26.48leifmadsen~asteriskversioning
13:26.49infobotsomebody said asteriskversioning was Information about the new Asterisk versioning method with the 1.6.x series is available here:  http://www.asterisk.org/node/48602
13:27.41anonymouz666- /* XXX This could block for a long time, and block the main thread! XXX */ - the person who wrote this comment in chan_sip.c didn't know murphy's law
13:27.58OrNixleifmadsen, right. thank you
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13:35.53MichofrHi all...I would like to ask please if there is a specific version of asterisk that works on Solaris...
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13:39.06skirmishahi guys
13:39.19skirmishai have a little problem with asterisk sip
13:39.30skirmishawhy asterisk sends 407 on invites?
13:39.41skirmishaand how can i disable it?
13:40.03jmkgreenis there a limit to the number of simultaneous HTTP connections the AMI can deal with?
13:40.07skirmishathe other question is if user agent is not sending digest username , call is failing as can not be auth
13:40.18jmkgreencause I just broke it with 60 connections
13:40.27skirmishaany siggestions how to disable 407 on invites
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13:42.13[TK]D-Fenderskirmisha: "insecure=port,invite"
13:42.23TSMjust wandering, is there any way when im making an outgoing call, to show the name of the phone im dialing if its in the directlry?
13:42.44skirmishathis is in global or under peer?
13:42.44[TK]D-FenderTSM: Show where?
13:42.52[TK]D-Fenderskirmisha: peer
13:43.06skirmishais it same as insecure=very
13:43.13skirmishaas i already tried that
13:43.21TSM[TK]D-Fender: on my phone? i get incomming caller id, but when i call a extention it only shows the number i dialed and not the name associated with it
13:43.23skirmishaand did not work
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13:44.08[TK]D-FenderTSM: ...
13:44.10[TK]D-Fender~cpid
13:44.11infobot[~cpid] Called-Party ID is possible with * using patches on Mantis.  See : http://bugs.digium.com/view.php?id=8824
13:44.12[TK]D-Fender^^^^^
13:44.44TSMahh
13:45.28*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
13:46.02TSMhas this been put in to later versions of *
13:46.10TSMim on 1.4.24
13:48.27[TK]D-FenderTSM: TSM external patch at best.  1.4 does not get new features
13:48.48[TK]D-Fender1.4.23.1.patch [^] (97,249 bytes) 2009-02-10 20:35 [wget patch] [License OK (v3.0)] wget 'https://issues.asterisk.org/file_download.php?file_id=21601&type=bug' -O - | patch -p0
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13:49.44[TK]D-Fender"All right, everyone, this issue should hopefully be coming to a close very soon. Asterisk 1.6.2 has been branched, meaning that Asterisk trunk is wide open for new feature consideration. This would be a great candidate for 1.6.3." - mmichelson (administrator)
13:50.07putnopvutHey, that's me!
13:54.57[TK]D-Fenderputnopvut: But enough about me... what do YOU think about ME? ;)
13:55.16putnopvutWhat movie is that from?
13:55.32[TK]D-FenderWARNING! chan_narcissism.so is already loaded!
13:55.33anonymouz666any heavy IAX2 user in here? I just wonder if IAX2 users have the DNS sync problem...
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14:12.06FabiOnehi all
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14:12.44FabiOnei've a strange problem with a sangoma A101D
14:13.58*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
14:14.10FabiOnesomeone wanto to hear the strange hangup?
14:16.51*** join/#asterisk bullium (n=RAIN\wbr@216.68.250.30)
14:17.09bulliumdoes anyone use aastra 57i's?
14:17.32*** join/#asterisk andres833 (n=andres83@190.144.75.22)
14:17.49Zeeekanyone have any VoIP or Asterisk news?
14:18.07ZeeekI need some stories like Jajah for dating
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14:21.24bulliumhow would I go about displaying the user/extension information when dialing the extension if it's in the asterisk system on an aastra 57i SIP phone?
14:24.23*** join/#asterisk anthm][ (n=anthm@CPE-72-128-94-253.wi.res.rr.com)
14:25.04[TK]D-Fenderbullium: read :
14:25.06[TK]D-Fender~cpid
14:25.07infobot[~cpid] Called-Party ID is possible with * using patches on Mantis.  See : http://bugs.digium.com/view.php?id=8824
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14:26.27bullium[TK]D-Fender: thanks for the info, I will read :)
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14:36.15kn0x[TK]D-Fender: what do you know about this internal timing?
14:40.05[TK]D-Fenderkn0x: Virtually nothing
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14:48.56pifhi, what is the best reporting tool for asterisk?
14:49.25piflike a log analyser for cdr data
14:57.11Joelpif I like awk myself ;)
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15:03.10jayteesed and awk rule!!!
15:03.15jaytee:-)
15:12.35beekjaytee: And vi for editing!
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15:31.26Zeeekhey beek
15:31.30Zeeekand everyone
15:31.38beekHi Zeeek
15:31.42Zeeekwhat's da newz?
15:31.46ZeeekI need news
15:31.58deadpigeonjust missed it
15:32.03deadpigeonnow its the weather report.
15:32.11Zeeeksports first
15:32.28beekI have a local sports page right next to me... care to hear little league scores?
15:33.00Zeeekis so old he can remember arguing with someone abput mousemacs on System 5
15:34.11Zeeekanyone into Freenum.org ITADs?
15:34.53Zeeekhow about the death penalty for using faxes after the year 2000 ?
15:37.24Zeeekanyway, you are welcome to come to #voip-users-conference anytime, especially right now as we'll be starting a live conference ina  few minutes
15:37.47ZeeekTo call you can go to http://VoipUsersConference.org (or http://VUC.me)
15:37.55ZeeekTweet it from the rooftops
15:38.01deadpigeonhm
15:38.07deadpigeonwhat are you gonna discuss?
15:38.41deadpigeoni want to discuss why every affordable ata is a piece of junk.
15:39.17Zeeekthat's NOT NEWS!!!!
15:39.37Zeeekuse asterisk as an ATA
15:39.59TSMis there a list of rev numbers to release numbers?
15:40.00deadpigeonyeah that works great for ppl i want to charge for service.
15:40.32coppicedeadpigeon: the reason ATAs are junk is obvious
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15:40.46Zeeekdeadpigeon: maybe someone will have a suggestion 4u
15:40.49deadpigeonhow are the iax atas
15:41.13Zeeeklooks at his IAXy in the drawer
15:41.19deadpigeoni cant call into the conference via phone?
15:41.23*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
15:41.32Zeeekyiou can call in any way you like
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15:42.02Zeeekhttp://www.voipusersconference.org/how-to-participate/
15:42.14ZeeekHey I missed Alan
15:43.37Joellol
15:43.39deadpigeonwhen does your conference start.
15:43.45JoelI'm all for the death penalty for using faxes past 2000
15:44.07leifmadsenin 15 mins
15:44.19deadpigeoni like this little face machine we got. does everything but fax.
15:44.24deadpigeonffaxx*
15:44.32coppicepeople love complaining about FAX, yet offer no effective alternative
15:44.35deadpigeonsanta ana keeps calling.
15:44.59deadpigeonwe still fax quite a bit.
15:45.27coppicesarbanes oxley has give fax a new lease of life
15:45.31deadpigeonwhich is why we have a pstn fax line, never worked quite right on asterisk
15:46.40deadpigeonheh. white collar crime $$
15:47.36Joelcoppice email.
15:47.38Joelcoppice problem solved.
15:47.53coppicehaha
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15:51.38elementzhi all. are there any users from germany around?
16:01.45Qwellelementz: You'd be better off asking a specific question
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16:02.44coppiceyou mean something like "hi all. are there any users from south east munich around"?
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16:02.47ZeeekQwell: we're live come on by;.. see you all later at #voip-users-conference
16:02.53Zeeekbye coppice
16:02.55Zeeekby DOnner
16:02.59ZeeekBy Blitzen
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16:06.17IBC_jkenneyhello i am having this problem with asterisk 1.4.21.2 does anyone know how to fix it
16:06.18IBC_jkenneyile.c:1204 waitstream_core: Unexpected control subclass '-1'
16:08.44QwellIBC_jkenney: first step - upgrade
16:08.59QwellNo point trying to hunt down a bug that may have already been fixed.
16:09.19*** part/#asterisk tfrew (n=tfrew@75-145-244-108-Richmond.hfc.comcastbusiness.net)
16:10.15IBC_jkenneyI have vicidial running on that box and would perfer to fix it not upgrade it
16:10.29IBC_jkenneyprefer
16:11.30kn0xdoes having a hardware timing source help voip quality?
16:11.44kn0xwould RTP use this then for timing?
16:11.47QwellIBC_jkenney: "fix it" and "upgrade it" are often the same thing
16:11.55kn0xI don't understand this internal_timing issues
16:12.17kn0xwe're having all sorts of weird problems, and I see a lot of INTERNAL_TIMING = no warnings in my logs
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16:15.04alcazarnetHi all, I am seeing occasional dropped calls via SIP wan calls.  Is there anything you can tune to help this?
16:15.21Superbarttsip debug and see why it's dropped?
16:15.24Qwellalcazarnet: not without knowing why it's dropped
16:15.37alcazarnetIt looks like restransmit issues due to a critical packet
16:16.13Qwellif the other side isn't getting it...even after retransmissions...  no.  fix your network
16:16.29SargunWhat's that asterisk conference call today
16:16.35Sargunevery friday
16:16.48*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
16:16.51QwellSargun: /j #voipusersconferense
16:16.56alcazarnetQwell, Its not a network issue, its random.  There aren't any timeouts or the like I can set higher?
16:17.13QwellI assure you, it isn't random. :)
16:17.37alcazarnetWell, I would say 1 out of every 100 or so calls
16:17.49*** join/#asterisk Talirk81 (n=David@rrcs-67-78-39-22.sw.biz.rr.com)
16:18.52Joelalcazarnet time to start capturing data :)
16:19.00Talirk81can someone  look at  http://pastebin.ca/1498398 im having some wierdness with an if when using execif
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16:24.47kaldemarTalirk81: drop the quotes around LEN
16:26.05alcazarnetQwell, So messages like this are generally caused by a network issue? http://pastebin.ca/1498407
16:26.14Talirk81ok ill try that
16:27.27SargunQwell, what's their actual IRC channel?
16:28.35kaldemaralcazarnet: did you see doc/sip-retransmit.txt ? if not, it's a good read.
16:28.59QwellSargun: that is it
16:29.13Qwellor maybe there are dashes...  dunno
16:29.13SargunQwell, there are dashes between the words
16:30.37alcazarnetkaldemar, Yes I have.  I was just wondering if there was anything I could do to make it a little more robust then simply hanging the channel up.  IE Allow more retransmits, or increase a timer?
16:31.12alcazarnetkaldemar, My thinking this is happening due to a few packets being lost.
16:34.18kaldemaralcazarnet: that equals to many packets being lost, not a few.
16:35.35Talirk81kaldemar,  thanks that seems to have fixed it
16:35.58Talirk81didnt realize quotes were requited for  if's
16:38.05kaldemarcomparing numbers is what makes the difference, not the if.
16:39.06*** join/#asterisk oglynn (n=me@173-9-102-81-NewEngland.hfc.comcastbusiness.net)
16:41.01oglynnI have a question on Monitor Vs MixMonitor on 1.6.0.9  using Monitor and the m flag i am getting the 2 call legs in one long file but one after the other. i like how it auto names my files can i get the 2 files truely Mixed or is that the purpose of MixMonitor? And if I use MixMonitor how can I get similar nice file naming
16:42.16errris it possible to listen in on an active sip call?
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16:44.14beekerrr: Via Asterisk, as in ChanSpy()?
16:44.39errrbeek: maybe so. I think I will go read up on chanspy() to see if that is what I am looking for thanks
16:45.35*** join/#asterisk javb (n=javb@190.166.114.116)
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16:52.43Talirk81oglynn,  i use  Monitor() with the m option this  saves as   $FileName-in.wav $FileName-out.wav then merges them using  sox
16:53.28Talirk81for example i have  it merge into Call111.wav  then i have a cron that     uses lame on it and  moves it to a /recordings/mp3/ folder ( to save space)
16:54.08javbIs there any intercom system for apartments, like when you want to call an specific apartment, to ask for permision to enter and so the open the door.. i want to have this integrated with an Asterisk that will manage the whole telephone system in the building.
16:54.13javbany idea?
16:54.31errrbeek: that was totally it. Thanks
16:55.12oglynnTalirk81 With the m option set it takes the in and out files and makes one long file 2 times length of the actual call.
16:55.36jplankI remember hearing at Astricon last year that in 1.6, with chanspy, you'd be able to switch between whisper, spy, ect, while in chanspy (and not having to hang up and redial in), has that been implemented?
16:55.36Talirk81not for me it mixes them  and keeps the same length
16:55.58*** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net)
16:55.59Talirk81maybe yoiur sox  settings are off
16:56.32kn0xTalirk81: why not just use MixMonitor()?
16:56.52Talirk81i found it to be slower
16:57.38beekerrr: NP
16:58.15Talirk81also off loading the   mixing to  another  system would let you create a  "recording web server" that scp's the wav files from your * server converts then send out emails or what ever else you need done
16:58.33Talirk81deligation of cpu basically to allow asterisk to only worry about the call itself
16:58.48*** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net)
16:59.02hardwireiax rtp diagnostics can be done somehow.. I just forget how.
16:59.15hardwireI just want to know how many out of place or dropped rtp packets there are in an IAX2 trunk
16:59.26Qwell0 RTP packets would be dropped over IAX2.
16:59.43hardwirethat answers that.
16:59.56hardwirenow on to the real question of how many IAX2 packets are dropped.
17:00.09Qwell'iax2 show netstats' is what you want, I think
17:00.16hardwirewoot~!
17:00.19hardwirekisses Qwell
17:00.27Qwellbuy me dinner first
17:00.38hardwireMcDonalds OK?
17:00.54Qwellcan I order outside the value menu?
17:01.03hardwireno.
17:01.10Qwellthen yes
17:01.13hardwirehttp://hardwire.pastey.net/117640
17:01.35hardwirethat's 12 from asterisk echo() going to a meetme at the remote site.
17:01.45oglynnTalirk81 I dont have any sox settings?
17:01.53hardwirethe RTT fluxuates between 64 and 8000
17:01.53hardwire:(
17:02.02Superbarttdoes anyone know how/where to get support for the free asterisk fax software? I'm planning on building a big fax-server for in the office... but the normal tech support said they could only support it for paying customers... while i'm ofc. am not going to pay before it's tested and working :x
17:02.10oglynnTalirk81 or where should i be putting them.  I just tried MixMonitor and it worked
17:02.23oglynnI would prefer to offload the work as well
17:02.46Talirk81oglynn,  http://www.voip-info.org/wiki/view/Monitor+stereo-example  is a good example
17:03.03hardwireoglynn: what are you up to?
17:03.16Talirk81it should you how to set  MONITOR_EXEC which run when the call closers  or STOPMONITOR() runs
17:04.14Talirk81i have sox combine the files , thenin a dead agi use php and lame to convert to mp3 ( mainly so i can have a php scp it to another host  if future growth warrants is)
17:05.11oglynnok i dont have my MONITOR_EXEC set at all sorry for the stupidity
17:06.24Talirk81its ok it took me about 2 months of  planning to build out my system for  growth and to do everything i wanted
17:06.26hardwireQwell: crapsteaks.
17:06.52Talirk81monitor took a fair amount of time  ( due to volume  level issues among other things)
17:07.18hardwireoglynn: I have a good solution for you
17:07.20*** join/#asterisk a2_ (n=a2@c-71-197-243-66.hsd1.wa.comcast.net)
17:07.27a2_Hello everyone
17:08.03oglynnhardwire
17:08.07hardwireoglynn
17:08.08a2_Im hoping you guys can clarify the startup process for a newbie
17:08.23oglynnhardwire you hve a prefered solution?
17:08.37hardwireindeed.. making a pastebin of it
17:08.44a2_I have bought a couple of DID's. Now all i do is setup asterisk, and setup the peer for my DID's to be my asterisk server ip address, is this the basic laydown(minus configuration) ?
17:08.50*** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca)
17:09.02hardwiremine uses speexenc.. but you can replace that with sox
17:09.13*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
17:10.45a2_Can someone pelase clarify if that is the correct process to follow? Thank you
17:11.26Talirk81a2_ your sip/iax provides should give you settings to put into asterisk
17:11.41Talirk81then all you have to do is dial your did and the provider will route to your * server
17:12.03a2_Talirk81: Yea they do. I guess what im trying to figure out, is if i need any other hardware or anything. All i need is my provider and asterisk with a dedicated server right?
17:12.07hardwireoglynn: http://hardwire.pastey.net/117642
17:12.18Talirk81if your using voip yes
17:12.28hardwireoglynn: that makes everything very "nice"
17:12.43Talirk81if you need pstn lines then is gets more "intresting" lol
17:12.58a2_Yea i dont need PSTN lines
17:13.07a2_Is it possible to detect if a line is PSTN or voip?
17:13.18a2_Without the register database
17:14.41a2_I will worry about that later i guess, be back
17:15.06Talirk81hardware: why not do the stop monitor in the h ext?
17:15.35hardwireTalirk81: whatever works for you.
17:15.46Talirk81doing it in the main  ext level will cause the channel to stay open while converting
17:15.54Talirk81is why i ask
17:16.10hardwireTalirk81: see the & in the first script?
17:16.35hardwirecrap.
17:16.37hardwirewhere'd it go.
17:16.48hardwireoh.. nevermind
17:16.52hardwirepostpone does all that for me
17:16.59Talirk81oh ok
17:17.26Talirk81ill have to look into postpone never seen it used beforwe
17:17.51hardwireit's a very excellent method of locking.
17:18.02hardwirethat way I'm only transcoding one recording at a time.
17:19.23Talirk81unfortually i need to have the system immedatly  "update" an api for  another section of our system ( not used in a callcenter evirmoment but something more particular so i have to do RT coding currently)  :( definatly not prefered but i do what the bosses say hehe
17:19.57Talirk81what provides postpone ?
17:21.55hardwirepostpone provides postpone
17:22.00hardwireit's in ubuntu and debian.
17:22.09Talirk81hmm centos doesnt have it  in its repo's
17:22.15hardwireotherwise you'll have to play with lockfile-progs
17:22.17hardwirewhich.. meh..
17:22.20hardwirenot as clever.
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17:32.15oglynnthanks to all.
17:32.58hardwireword
17:33.12hardwirethe part that makes my solution work is buffer
17:33.33hardwirelimiting how many bytes per second can be sent to the transcoder before archiving the recording.
17:33.42hardwirethat's the part that makes the CPU happy
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17:50.29DefrazI have a grandstream HT502 and a fax machine plugged into that, then it regs to my asterisk box, then that has a PRI into it.
17:50.42DefrazCan I use T.38 to send faxes out the PRI?
17:51.02QwellDefraz: T.38 is a Fax over IP protocol
17:51.15Qwellprotocol?  wrong word maybe.  coppice can correct me
17:52.15QwellDefraz: That would require gatewaying, which I don't think any of the solutions do.  You could save the fax from the fax machine, and resend the file out the PRI as a new fax
17:52.36*** join/#asterisk sfire (n=sfire@businessservers.info)
17:52.36WindowsUser?
17:53.06sfirequick question... I need to make it so that an extension calls an outside number.. is this possible?  could someone point me to some instructions
17:53.10coppicecallweaver does T.38 gateway, and I think someone has patches for * to add it
17:53.28[TK]D-Fendersfire: "core show application dial"
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17:55.37sfire[TK]D-Fender, thanks for the lil snippet of info.. I'm going to go search for how to do it with that
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17:56.56Defrazyea i guess i could
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17:59.03sfirebasically the business has onsite security that is part time.. and whenever that extension isn't available I want the IVR to give an option to call the security company.. basically they press 4 then it connects them to an outside line to their 800 number
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18:15.56Talirk81sfire,  wouldnt you want to have   that happen on outside of biz hours , and go to vm other wise?
18:16.22sfireno.. I want them to be able to call the security company out if needed
18:16.47sfirebut I don't want them to have to directly dial it.. I just want them to have to press a button and have it go through the office system
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18:17.26sfire12 cars stolen in the last month or so
18:17.43sfirean insane number.. now we have armed guards on site and their office is right up the street
18:18.15*** join/#asterisk adam420 (n=a2_@c-71-197-240-228.hsd1.wa.comcast.net)
18:18.15[TK]D-Fendersfire: "core show application read"
18:18.40*** join/#asterisk eighthour (n=IceChat7@24.144.176.62)
18:18.41sfireI'm looking up the data on that.. trying to find an example similar to what I need
18:19.30timeshell_atworkIs there a way to retrieve the XML config from a Polycom phone to see what it's current config is?
18:20.18[TK]D-Fendertimeshell_atwork: No
18:20.23*** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve)
18:20.29timeshell_atworkWell that sucks
18:20.57[TK]D-Fendertimeshell_atwork: You should ahve to guess whats in its config.. you should be setting it
18:21.00Joelyeah just one of the things that makes polys suck imho
18:21.15timeshell_atworkYes, TK, that's quite the obvious
18:21.25timeshell_atworkI'm trying to debug something
18:21.34adam420Should i install 1.4 or 1.6?
18:21.39Joeladam420 both?
18:21.40[TK]D-FenderJoel: What other phones export configs?
18:21.42timeshell_atworkSomething that works in one environment isn't working in another with the same config
18:21.48Joel[TK]D-Fender aastra.
18:22.10Joel-extremely- useful when troubleshooting.
18:22.15[TK]D-FenderJoel: Congrats for them.  Once again, if you do the job right yourself you don't ask the phone what its doing, you TELL IT
18:22.35eighthourcan someone help a complete noob figure out why i cant get the web gui working....???....
18:22.53[TK]D-Fendereighthour: GUI's are not supported in the channel.  Please read the /topic for related channels
18:23.25eighthour[TK]D-Fender, thanks....
18:23.49*** join/#asterisk Shaun2222 (n=Shaun222@ip68-5-154-128.oc.oc.cox.net)
18:23.54Joeltimeshell_atwork indeed, exactly where downloading the config off of the phone would be extremely helpful :)
18:24.09Shaun2222is there a command that will show a devices status?  Ex: in-use, busy, etc?
18:25.23[TK]D-FenderShaun2222: "core show function DEVICE_STATE"
18:25.54Shaun2222need to be able to see this using the CLI
18:26.36Shaun2222sip show inuse show's nothing, even when a phone is inuse
18:26.38[TK]D-FenderShaun2222: "show channels concise"
18:26.49*** part/#asterisk eighthour (n=IceChat7@24.144.176.62)
18:27.55Joeltimeshell_atwork I've seen all sorts of models of phones say one thing, and the configs have another :\ it sucks
18:28.49[TK]D-FenderJoel: Only reason for that to happen is that the phone isn't pulling configs.  And with Polycom, you'd know
18:29.10[TK]D-FenderJoel: Which typically falls down to PEBKAC
18:30.49timeshell_atworkEveryone who drives slower than TK is an idiot and whoever drives faster than him is a moron.
18:32.03[TK]D-Fendertimeshell_atwork: Faster?  Find me one ;)
18:32.27[TK]D-Fendertimeshell_atwork: I hear a lot of hollow talk about people who can't read a log or an admin guide.
18:33.00[TK]D-Fendertimeshell_atwork: people come up with ever wussy-assed cop-out they can.
18:33.19[TK]D-Fenderevery*
18:33.37Joeltimeshell_atwork he suffers from perminant PMS, pissy male syndrome, I find his help not worth the trouble of his presentation, just ignore him.
18:33.49timeshell_atworklol
18:34.02timeshell_atworkTK an I have a pissing match every once in a while
18:34.13timeshell_atworkHe usually pisses more than me
18:34.25timeshell_atwork:)
18:34.29[TK]D-Fenderis 5% more water than most
18:34.41timeshell_atworkAll that beer.
18:35.02[TK]D-Fendertimeshell_atwork: Not really.... I don't drink.
18:35.07jkroonrofl @ timeshell_atwork + [TK]D-Fender
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18:35.40timeshell_atworkAhh, well neither do I.
18:35.44timeshell_atworkMust be all the coffee then
18:35.50Joeltimeshell_atwork this is what i see when the diarhea falls out of his mouth: http://yfrog.com/13screenshotdogj
18:35.58timeshell_atworkErr
18:36.38Kattysighs
18:37.21psilikonhow do I tell what type of swithtype I need on a PRI?
18:37.33Kattyyou call your pri providor and ask them.
18:37.38[TK]D-Fender[14:22]<[TK]D-Fender>Joel: Congrats for them. Once again, if you do the job right yourself you don't ask the phone what its doing, you TELL IT <--- apparently this was the offending line that had him ignore me.
18:38.15jkroonhahahaha!  that's funny.
18:38.28Joel[TK]D-Fender you've been on my ingore list for years.
18:38.59[TK]D-Fender\o/
18:39.23jkroonyea well, [TK]D-Fender might be a bastard but at least he usually shoves one in the right direction - if he bothers.
18:39.39jkroonand yes, [TK]D-Fender - coming from me that's a compliment.
18:41.01[TK]D-Fenderjkroon: Bastard?  I wouldn't say so.  I can be biting at times, but its also proportionate to whatever I'm reacting to.
18:41.12[TK]D-Fenderjkroon: And always the right direction... even if it happens to be the door.
18:41.28[TK]D-Fenderjkroon: And I seem to bother a lot ;)
18:41.29jkroon... point in case :p
18:41.47timeshell_atworkProportional???
18:41.48timeshell_atworkLMAO
18:41.56[TK]D-Fenderjkroon: Nifty Freudian slip there
18:42.04psilikon5ess isn't the same as national I take it...
18:42.07timeshell_atworkYour definition of proportional means you would have nuked North Korea and Iran 10 years ago.
18:42.16[TK]D-Fenderpsilikon: clearly not
18:42.29[TK]D-Fendertimeshell_atwork: For what?
18:42.39psilikonWhy wouldn't it cause serious issues to have it mismatched?
18:42.44timeshell_atwork... exaggeration TK
18:42.50psilikonthis pri works *most* of the time
18:42.59[TK]D-Fendertimeshell_atwork: Korea never did jack shit to North America, neither did Iran.  Who cares?
18:43.06[TK]D-Fendertimeshell_atwork: RP 2008!!!!
18:44.01*** join/#asterisk micols (n=mio@rlogin.dk)
18:45.37lirakisany one know of a commnd line speech to text (not text to speech) tool for linux ?
18:46.39[TK]D-Fenderlirakis: Lumenvox
18:48.23lirakis[TK]D-Fender:  Thanks - im looking for wav -> plain text.  Ill see if lumenvox does that.
18:49.44[TK]D-Fenderlirakis: I can't see them as only running "live". Got to be a way to feed it
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18:55.41paulgmDoes anyone have an example .cnf.xml for a 7941G-GE phone?
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19:09.55Shaun2222is there a command via the CLI to show number of calls in a queue?
19:11.50Joelshow queue <queue>
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19:16.03[TK]D-FenderShaun2222: "queue show [queue]"
19:16.06*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
19:16.19Shaun2222ah... i was doing core queue....
19:16.27Shaun2222and core show queue
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19:20.28Kattyeppigy: :<
19:20.41[TK]D-FenderKatty: Mew.
19:20.56Katty[TK]D-Fender: mew.
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19:22.46adam420Whats the best text to speech program for asterisk?
19:23.34adam420Is festival good?
19:23.37[TK]D-Fenderadam420: Cepstral is the big one these days
19:23.58adam420cool thanks fender
19:26.00paulgmHere's my SEP<mac>.cnf.xml for the Cisco 7941G-GE phone:  http://pastebin.ca/1498584  Can anyone see anything wrong?  The phone downloads the config every minute or so from the tftp server but still says "Unprovisioned".  Also, the phone status messages have "Error Verifying Config Info" but don't tell me what that error is or where it happened.
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19:32.28toughmarketingCan anyone recommend a reasonable business type sip phone?
19:32.46paulgmanything that isn't made by Cisco apparently
19:33.00toughmarketinglol i read it was a pain to deal with cisco
19:33.09toughmarketingin the asterisk book it mentions it
19:33.21Alfiotoughmarketing polycom
19:33.30Kattyinfobot: mew.
19:33.31infobotA MIME mail reader for Emacs/XEmacs. URL: http://www.mew.org/
19:33.43toughmarketingAlfio: thanks, let me look them up :)
19:34.00Kattyinfobot: :<
19:34.01infoboti guess < is redirection of stdin to a program
19:34.59Katty:<
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19:40.39sumasumadoes anyone offers scalable telephony ?
19:40.50sumasumasuch that i can make calls from 1 to 1 million calls
19:42.54[TK]D-Fendersumasuma: That would take a very large setup, racks of servers
19:44.08[TK]D-FenderCisco is a decent phone.  However the challenge in running it legally (costs), and the second-rate SIP support make them a harder choice.  cisco = pain.
19:44.28*** join/#asterisk Kobaz (n=kobaz@its.kobaz.net)
19:45.01ariel_sumasuma: PortaOne offers a sip scalable system, so does Enswitch.
19:45.20ariel_Enswitch is based on asterisk with a SER setup
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19:47.18ariel_anyone here from Digium, I can't call into you guys due to phone issues.  I am on a ship and need some support.  I have ABE 250 user  Lic and also using TE420b, but main issue is with ABE C 2.0
19:47.39paulgmWhat I'm leaning towards is probably just getting updated hardware and running the latest Cisco CallManager.
19:49.48*** join/#asterisk lcuevas (n=lcuevas@fw1-ext-core.net-uno.net)
19:51.17sumasuma[TK]D-Fender: I was looking for an option like Amazon web services, where we can make outgoing calls
19:52.18sumasuma[TK]D-Fender: I would think amazon would have invested more into servers for even the  EC3, SQS etc.,
19:52.41[TK]D-Fendersumasuma: Have you calculated the bandwidth costs on that?  AWS itself scales at a funny rate...
19:57.50*** join/#asterisk wadaw (n=wadaw@c-71-197-240-228.hsd1.wa.comcast.net)
19:58.27wadawI have festival setup, in my festival_log.conf it shows " : accepted from localhost
19:58.27wadaw", but in asterisk there is no info about festival at all..i believe everything is setup correctly, i dont know how to debug this..
19:58.33toughmarketingAlfio: Thanks for that I just ordered 8 of them, they are very reasonable.
19:58.55toughmarketingI have a new problem from asterisk 1.6 on my development server.
19:58.57toughmarketing[Jul 17 12:57:51] WARNING[17318]: chan_sip.c:3324 __sip_xmit: sip_xmit of 0x10522c0 (len 1150) to 64.2.142.214:5060 returned -1: Address family not supported by protocol
19:59.04toughmarketingDoes anyone know what this means?
19:59.21toughmarketingit happens on a Dial() outbound
20:03.21sumasuma[TK]D-Fender: 104448 simultaneous calls at $0.27 per second, You think it is costlier ?
20:03.38sumasumait is at G711 codec
20:04.25sumasumaincluding IP overhead
20:04.37*** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk)
20:04.45rob0"Address family" is IPv4, x.x.x.x where x is 0-255, 64.2.142.214 in your error. "Protocol" in this case is UDP. Guess: broken kernel.
20:05.51toughmarketingOh this is wierd, I did a sip reload and a dialplan reload and now it works.
20:06.00toughmarketingI didn't change anything though
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20:09.46[TK]D-Fendersumasuma: AWS != you getting a bank of servers...
20:10.00lirakissumasuma: what are you looking for ? a platform, or termination
20:11.18wadawI have asterisk16_gui installed and configured, however. When i browse to http://localhost:8088 i get
20:11.19wadaw"The requested URL was not found on this server."
20:11.34wadawI really need help this is frustrating..
20:12.13[TK]D-Fenderwadaw: GUI's are not supported here, they have their own channels.  See the /topic
20:13.23djMaxif I use Dial with A(file), is there a way to get it to ring to the caller while playing that file?
20:13.42lirakisi guess sumasuma is off to attemt 1 million simultaneous calls on opensips, lol
20:13.51lirakis*attempt
20:15.01lirakisdjMax:  i assume you tried passing the r flag in the dial command too?
20:15.21djMaxyeah, didn't seem like it worked, but perhaps I screwed it up
20:15.34djMaxit rings, but as soon as the called party answers there seems to be a couple second pause
20:15.57lirakisdjMax: ... so you want ringing while the audio is playing to the called party
20:17.34[TK]D-FenderdjMax: "vi app_dial.so"
20:18.08*** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com)
20:18.29djMaxright, yes.
20:18.48djMaxhuh, you mean look at the compiled code?
20:18.55sumasuma[TK]D-Fender: I have not used the EC3, I have used the SQS
20:19.36[TK]D-FenderdjMax: Look, no.  CHANGE, yes
20:19.39mltlnxWhy is it that I loose my subscriptions after a reload?
20:19.43sumasumait works cool with more than 100K queries awesome per second
20:19.46djMax:)
20:20.12mltlnxin 1.6.0
20:20.13sumasumaSimilarly if someone provides telephony services it will be awesom
20:20.14psilikonhttp://www.savethetaint.org/stretching.php
20:20.20sumasuma*awesome
20:21.12lirakissumasuma: dude .. what are you looking for ?
20:28.29*** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com)
20:29.15[TK]D-Fendercheckout time, BBIAB
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20:36.11lcuevaslcuevas: hi, why when im installing freepbx i am getting "does /var/run/asterisk.ctl exist?"
20:36.58*** join/#asterisk Know1edge (n=Know1edg@66.18.99.218)
20:38.21lirakislcuevas: go to #freepbx
20:38.44lcuevastxs
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20:48.19mltlnxDoes anyone know how to adjust  the rate at which a Linksys SPA 942  subscribes?
20:51.31xpot-mobileQuestion: anyone know where I can find information for setting up multiple periodic-announcements in a queue?  I have 5 sound files I would like to play every 30 secs.  So far it looks like the queue will only play a single periodic-announcement.
20:51.50howiehow do i get a soft phone to configure to static ip? im following the asterisk book but it only shows dhcp?
20:53.46xpot-mobilehowie: that sounds like a softphone issue and not an asterisk issue?
20:54.47eppigyKatty: <3
20:54.50eppigywin 14
20:55.10howieprobly, but its saying to change my dhcp server config file but im using static
20:55.41xpot-mobile<PROTECTED>
20:56.05howieIm following the asterisk book to install and config
20:56.37howiedo i need to make my server dhcp?
20:57.15xpot-mobilehowie: I would say no, in fact, I would skip that part.  is it referring to a dhcp.conf in /etc/asterisk?
20:57.47xpot-mobilehowie: I don't even think that exists
20:58.01howieno its refering to my networking config file on my box
20:58.34*** join/#asterisk thansen (n=thansen@7.247.sfcn.org)
20:58.49xpot-mobilehowie: oh.  Perhaps you should find out why the book would have you configure a dhcp server.
20:58.51howiethe book says the xlite softphone should auto sync but i dont have dhcp setup.. soo i dont see how thats possabe
20:59.26*** join/#asterisk GuillermoHurtado (n=Guillerm@fw1-ext-core.net-uno.net)
20:59.27howieim really thinking this book is out of date...
20:59.27xpot-mobilehowie: does the computer you have xlite setup on have an IP?
20:59.33howieyes
21:00.00xpot-mobilehowie: then your done.  Now open up xlite and create a server connection to your asterisk box.
21:00.33xpot-mobilehowie: once you setup xlite, then it will auto sync to the information you tell xlite to connect to.
21:00.51xpot-mobilehowie: if xlite doesn't know what to do, then it will not do anything. ;)
21:01.14*** join/#asterisk GuillermoHurtado (n=Guillerm@fw1-ext-core.net-uno.net)
21:01.29howieic, so i just tell it to go to 1.2.3.4 i dont remember setting any passs user
21:02.42xpot-mobileuser is usually the extension or whatever your context is in sip/iax .conf  ie: [user1] or [100]
21:02.48*** join/#asterisk GuillermoHurtado (n=Guillerm@fw1-ext-core.net-uno.net)
21:02.56howieok
21:02.58xpot-mobilehowie: and pass will be whatever secret is set to in those files
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21:04.13howieok ty
21:04.24xpot-mobilehowie: np ;) enjoy
21:04.51howieI will cant wait to get everything working :P
21:05.08GuillermoHurtadois it possible to know when an extension is being rung directly from Asterisk CLI?
21:06.07xpot-mobileGuillermoHurtado: you should see a Dial() command in the cli execute something like Dial(SIP/100)
21:08.47GuillermoHurtadothanks
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21:12.50howieok so i got the soft phone to connect on 2 comps but when i use it, it says :chan_sip.c:14035 handle_request_invite: Call from '1000' to extension '100' rejected because extension not found.
21:14.12*** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
21:16.42xpot-mobilehowie: is the ext 1000 or 100?
21:17.12howieboth? 2 serperate extensions
21:18.10howieok i found it in the book
21:18.13howiety
21:19.29*** join/#asterisk Airozi (n=user@200-171-41-207.dsl.telesp.net.br)
21:19.40leifmadsenyw :)
21:19.56*** join/#asterisk voxter (n=voxter@190.241.15.56)
21:28.58*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
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21:37.51*** mode/#asterisk [+o putnopvut] by ChanServ
21:43.54howie<PROTECTED>
21:45.50[TK]D-Fenderhowie: Go look at the peer status
21:46.44howie100/100                    (Unspecified)    D          0        Unmonitored
21:46.44howie1000/1000                  192.168.1.107    D          35472    Unmonitored
21:46.54howieis it because my 100 is off? lol..
21:47.33[TK]D-Fenderhowie: Well it hasn't registered and * has nowhere to call
21:48.03howieok got it its working!
21:48.19apeironSet up voicemail! :D
21:51.05toughmarketingHey guys, I just setup my first IVR which is working perfect.  Only problem is if I type an invalid extension I am not sure how to make it say that extension is invalid and replay the menu.
21:51.34Qwelltoughmarketing: i exten
21:51.43toughmarketinggotcha
21:51.54toughmarketingQwell: thank you!
21:55.31*** join/#asterisk serph (n=serph@CPE001f5b006e5e-CM001371144daa.cpe.net.cable.rogers.com)
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22:16.21Know1edgeAny of you guys know of a receptionist app that will integrate with asterisk?
22:17.02QwellKnow1edge: Asking in multiple channels is considered to be rude.
22:18.12*** join/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com)
22:21.15Slade-hey i recoded a custom record (which seems to be a wav file)  but when i attempt to play it. it seems to not play anything..  do i need to convert it to gsm or something like the rest of the files?
22:23.40*** join/#asterisk DrAk0- (n=luisjose@nelug/coreteam/luisjose)
22:23.42wwalkerSlade-: you have to convert it to a format for which you are loading the format_<type> module
22:23.43*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
22:23.52wwalkeror load more modules
22:24.15Slade-thaks ;)
22:28.15WindowsUserSlade-: does your .wav have a 8khz somethingrate or is it 44.1khz?
22:29.00Slade-WindowsUser: it was recorded by asterisk the *77 stuff..
22:29.17Slade-i converted it to sln. and its working now
22:29.23WindowsUserodd
22:30.03WindowsUser*77 stuff? is that something you put in the dialplan or is that part of features.conf?
22:30.14QwellFreePBX no doubt..
22:30.22Slade-asterisknow
22:30.29WindowsUserah
22:30.53Slade-guess its not working entirely as intended. :)
22:31.18Slade-i also discovered my voice is a complete failure for ivr work
22:31.47WindowsUseryou sound like a redneck?
22:32.38Slade-no. i just sound completely inferior to alison smith
22:32.49WindowsUserah
22:33.02*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
22:33.45WindowsUseryou could practice?
22:33.54Slade-and buy a new mic
22:34.05KavanSheh
22:34.07KavanSI got rid of alison
22:34.19WindowsUserwhatcha using?
22:34.21KavanSI didn't like everyone knowing that I was running asterisk...
22:34.28KavanSI had a friend record for us
22:34.42KavanSI love asterisk, I just don't like being easily ID'd :P
22:34.48WindowsUserhaha
22:35.09KavanSyeah I had a chick who used to do phone sex...doesn't sound bad :)
22:35.15WindowsUserniiiiice
22:35.36WindowsUserI'm using the Alex TTS thing that OSX has
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22:36.08WindowsUserit would be nice to tweak it to sound more like doctor manhatten
22:36.12securevoipis there a good way to send an email from the dialplan?
22:36.39WindowsUserwhat kind of email? is there anything in it?
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22:37.41securevoipbasically, i just want to be notified at a certain step in the dialplan...
22:38.10*** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com)
22:38.22Slade-cant just call an agi script?
22:38.52WindowsUseryea either agi or system, probably agi tho
22:38.57securevoipsure...  just being lazy...
22:39.07Slade-system? soudns clever
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22:39.16securevoipdidn't know if there was a non-AGI way...
22:39.29WindowsUserVerbose()?
22:39.30KavanSsecurevoip, just use a script...
22:39.34KavanSsecurevoip, I use something similar...
22:39.38juju2143hi
22:39.42KavanSI use jabber for callerID
22:39.51juju2143can someone help me?
22:39.52KavanSso I get an IM with callerID info for incoming calls
22:39.54securevoipok, so, one more friday question...
22:40.28securevoipwhat is the best way to call an external party and play a sound file?
22:40.57KavanShrm, not sure...I'll leave that answer to a pro :P
22:41.00WindowsUserread the callfile thing
22:41.00KavanSI've never done that...
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22:41.34WindowsUserhttp://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt
22:41.39juju2143i'm a total n00b in asterisk
22:41.55juju2143how I use it?
22:43.02WindowsUsersecurevoip: make a callfile taht runs the playback application
22:43.04juju2143someone?
22:43.10WindowsUser~ask
22:43.10infobotQuestions 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.
22:43.14ariel_juju2143: you need to ask a direct question of issues your having or items like that.  If your just starting you should do some reading
22:43.19ariel_~doc
22:43.20infobotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
22:43.22juju2143ok.
22:43.55KavanS"it's evolution baby..."
22:44.10juju2143I have 3 phone numbers, how I bind to one of them?
22:44.52ariel_bind? don't understand this bind
22:45.32juju2143i mean how I can make asterisk listen to one of them?
22:45.38juju2143for calls
22:46.25KavanSjuju2143, SIP or Zap?
22:46.32juju2143is this works with my winmodem on /dev/ttyAGS3?
22:46.57ariel_ohhh one of those,  good luck
22:47.16WindowsUsermost people skip that
22:47.22ariel_I am out of here going to do some walking in Juneau, Alaska before we have to set sail again....see you all.
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22:48.28WindowsUserjuju2143: for using a winmodem you'd have to muck around a lot just to get a crappy pieve of hardware to work with asterisk
22:48.37Slade-is there a asteriskforge or clever place for publishing asterisk/agi libs?
22:49.11juju2143like what I want to do is something like press 1 for doing something
22:49.23juju2143is this possible?
22:49.30WindowsUserwell, yes
22:49.34WindowsUserbut winmodems are shit
22:49.50juju2143and how this works with my winmodem?
22:50.09juju2143mine works well
22:50.28juju2143I have a good driver for it
22:50.57Qwelljuju2143: You can't use that hardware with Asterisk.
22:51.11toughmarketingAnother IVR question, you know when you call big companies they have the press 9 to return to the previous menu... I can't quite figure this one out.
22:51.42juju2143really? even when I make Asterisk listen ti /dev/ttyAGR3?
22:51.45WindowsUsertoughmarketing: you keep track of where they came from
22:52.02Qwelljuju2143: how are you going to get Asterisk to listen to /dev/ttyAGR3?
22:52.34toughmarketingWindowsUser: I think the idea of it working in my current ivr is in my head I just havent made it work yet.  I was just curious as to how others are doing it.
22:52.52juju2143Qwell: some .conf in /etc/asterisk?
22:53.00Qwelljuju2143: no..
22:53.16Qwellit doesn't work that way.  You need real telephony hardware.
22:53.26Alfiotoughmarketing exten => 9,1,Goto(ivr,s,1)
22:53.26juju2143a voice modem?
22:53.56WindowsUsertoughmarketing: how is your menu built? is it a tree and can you go in circles?
22:54.01Qwelljuju2143: no..
22:54.21juju2143anyway, someone knows a good tutorial to asterisk?
22:54.25Qwell~book
22:54.26infobot[~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
22:54.31WindowsUserwhoops
22:54.36toughmarketingWindowsUser: Basically, yes.
22:54.45juju2143thanks Qwell
22:55.21WindowsUserif its a tree you just keep track of where in the tree you are, otherwise implement a stack?
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22:57.07Slade-juju: you can use those x100p modems for testing and stuff
22:57.11Slade-oh, hes gone
22:58.30WindowsUseraww asteriskforge.org is registered :(
23:00.00WindowsUserwhen i have something worth sharing I'll probably use google code
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23:15.41ruben23hi
23:15.59ruben23where do i cehck log erros of asterisk on my server..?
23:16.42jblackTry /var/log/asterisk
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23:27.45a|3xhi, i am looking into building a automated phone directory using asterisk, with names being recognized automatically, is there any solution out there which i can use for voice recognition?
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23:37.28Qwella|3x: non-free, but Lumenvox is quite good
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23:43.17a|3xhow about in the domain of russian language?
23:43.27Qwellto that, I say good luck sir
23:45.31Qwella|3x: the amount of data you need to build a good acoustic model is staggeringly high
23:46.49a|3xi see
23:47.16a|3xso there is no good solution already prepared?
23:47.25Qwellnothing I'm aware of
23:53.12a|3xok, thanks
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