IRC log for #asterisk on 20091106

00:05.11Katty:>
00:06.35rdude99Is it possible to have a SIP call in progress but play non-conversation sounds on the console (which are based on the call) via asterisk application commands?
00:10.59Kattyeppigy: once a native maiden and a stranger met, underneath the blue tahitian moon!!!
00:11.15Kattyeppigy: http://www.youtube.com/watch?v=nksJQRXglnc
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00:15.21ardnathey guys how can i excuate a system command with a ' in it
00:15.29ardnatit keeps droping all my (')'s
00:19.12leifmadsen\'
00:23.13jefimenkoare there any docs on how to specify session timers in asterisk?
00:26.38russellbconfigs/sip.conf.sample
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02:35.54nyranger[TK]D-Fender: are you their?
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02:46.53Katty:>
02:47.01KattySTOP
02:47.03KattyHAMMERTIME
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02:59.53jayteecan't touch this
03:02.12_ShrikEgoes back to cali
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03:12.11ChannelZSHOWER TIME
03:12.18ChannelZit's like Hammer Time but with less pants
03:36.04p3nguinthrows his CO detector over the balcony
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04:15.40ZPerteeHi! I'm feeling rather benovelent at the moment and was wondering: Does anyone want a google voice invite? I've got plenty and thought maybe I'd give one away.
04:17.17ChannelZDo I have to show you my boobs?
04:17.52ZPerteeChannelZ, that would work, but no not required. I'm just a nice guy trying to get rid of invites
04:18.36ZPerteeChannelZ, Do you need an invite?
04:18.58ZPerteeChannelZ, or were you just commenting :-)
04:19.44ChannelZjust seeing if you loved me for my mind or my moobs
04:19.58ZPerteeChannelZ, Haha
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06:46.01natlonehathi all
06:48.36natlonehattrying to get func_odbc to compile into asterisk 1.6.1; but little success. end goal: to connect to ms-sql db for realtime storage
06:49.32kaldemarwhat's the problem?
06:50.05natlonehatwell, i installed unixODBC and required packages
06:50.25natlonehatand then when i run ./configure with odbc it ends up reporting success
06:50.44natlonehatbut when i go to menuselect; all the *odbc stuff has XXXX
06:51.22natlonehatnot sure where am going wrong with this.
06:51.41kaldemaryou probably didn't install all the required packages
06:51.53natlonehat<PROTECTED>
06:52.12natlonehatwhich one did i miss?
06:53.30natlonehatI get this: "checking for mandatory modules:  UNIXODBC... ok"
06:53.58ChannelZdo you have libtool?
06:55.34natlonehatChannelZ - thanks. feel like bashing myself on the head with a brick. lol
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06:58.11natlonehatgot it ;)
06:58.22ChannelZyay!
06:58.38natlonehat:D
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06:59.48natlonehathave you tried realtime with ms-sql?
07:00.06ChannelZnot personally no
07:00.42natlonehathmmm...ok
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07:02.22natlonehatI've gotten Freetds working and can query the sql db from the box; lets see how this works out.
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07:26.36bmancan someone point me to a install doc for ubuntu from the debian package?  i would like to use mysql and sip
07:27.08*** join/#asterisk Hazuki (n=Hazuki@cpe-72-229-137-30.nyc.res.rr.com)
07:27.42HazukiIs it possible to buy a phone number from an SIP provider and multiplex it through * among 12 IP phones with extensions?
07:28.21HazukiLike, if the number is called, * intercepts it and says "Thank you for calling Acme Widgets. If you know the extention of your party, please dial it now, or press 0 to be added to the queue for the next available rep?"
07:29.11kaldemarHazuki: yes
07:29.49bmanhaving a hard time finding a install doc that isnt from source
07:30.09Hazukikaldemar: thank you! :) This will make my boss really happy. So, I don't need to buy 12 SIP trunks, one per phone?
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07:30.57kaldemarHazuki: no, one will do. you just have to decide what you do with the incoming calls when they come to your asterisk box.
07:31.43HazukiThis is wonderful news. Okay, so business cable (I believe 10 Mbps down, 3 up) is enough for this? Assuming, let's say, a 60kbps codec, max 6 concurrent calls?
07:32.08Hazukiwe have a WRT54GL with Tomato, so I can QoS SIP packets to the highest level
07:32.11kaldemarit's plenty
07:32.30HazukiThis sounds almost too good to be true, wow
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07:34.52bmandoes anyone know where a good howto for debian from the repository?  i have it installed and i have my odbc source up for mysql but i dont really see anything that tells me how to create the tables and use mysql table for the user authentication with sip calls
07:34.55HazukiSo, I'd need to buy only a single SIP trunk, migrate the phone number to it, then set up an * dialplan that would intercept calls to that #, play the message, and relay the call to the appropriate phone in-house when the extension is added?
07:35.33HazukiSomeone calls 555-555-5555, gets the message, then presses 1 and the phone on desk 1 rings
07:38.24kaldemarbman: the name of the functionality is Realtime, google for that
07:39.23kaldemarHazuki: yes, for example like that
07:39.23HazukiYou guys rock @_@
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07:39.49HazukiMy boss pays almost 400 a month for 4 verizon POTS lines. We have $450 of parts, a $60 router, and 12 x $65 Grandstream GXP 280 phones on order. Within 5 months, this will have paid for itself
07:39.53HazukiAND everyone will have a phone
07:40.09kaldemar~gs
07:40.10infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
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07:40.26kaldemarGrandSuck aka Grandstream :)
07:40.29HazukiYou sure? The GXP280 is getting pretty good reviews from what I hear
07:40.35Hazukiwe only need very basic functionality
07:40.50Hazuki~snom
07:40.51infobotextra, extra, read all about it, snom is like all German products. High quality, but wacky engineering. :)
07:41.13Hazuki~cisco
07:41.14infobotcisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice
07:41.57kaldemarwell i haven't tried out that particular model myself, but everything i've tried from GS has been crap. it's been a while though, but haven't heard much positive feedback from people who have.
07:42.16HazukiOkay...so if GS is crap, Snom isn't so wunderbar after all, and Cisco is also crap...what do people here recommend? Preferably <$100 with Ethernet passthrough
07:42.18ChannelZhumps his Linksys SPAs
07:42.33Hazukier, Linksys is just rebranded Cisco isn't it?
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07:43.29ChannelZThey have different models but yes they are the same company
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07:43.43kaldemarsome were Sipura's until the company got bought and rebranded to linksys/cisco
07:44.25kaldemarpolycoms are recommended by many, but you won't probably find those for <$100.
07:44.28bmancan i use res_odbc.conf with mysql through the system level odbc driver?
07:44.32ChannelZYes Linksys bought Sipura, Cisco bought Linksys
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07:45.56ChannelZKumbang?  Is that what happens after you fingerbang?
07:46.39Hazuki>>
07:46.49Hazukiit's Filipino or Malay or something
07:49.22ChannelZoh :)
07:50.12Hazukiis just taking her best guess at that, anyway
07:50.23ChannelZIndionesian, A beetle.  Says wikitionary
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07:54.41Hazukiwants to visit Indonesia sometime. Not many people know this, but it's the 4th most populous nation (behind China, India, and USA) and the largest Muslim one IIRC
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08:02.12mchouChannelZ: you have a softphone handy atm?
08:02.44mchouChannelZ: sip softphone*
08:03.18ChannelZhmm not really, I rebuilt my system under Win 7 last week and haven't put one on yet
08:03.30mchouwin 7??
08:03.49ChannelZYes
08:04.15mchouhow you run asterisk on win7? :)
08:05.24gushiIn a VM.
08:05.46ChannelZNeither;  My * run on headless linux boxes
08:06.09ChannelZbut I don't really use linux as a desktop
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08:07.49HazukiLinux is desktop ready. A lot of users aren't desktop ready though
08:08.20ChannelZI have to run a lot of non-linux software
08:08.52Hazuki*nod* Fair enough. You have legit reasons to need Windows
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08:10.00ChannelZI'm a whore, I use a bit of everything
08:11.32Hazukiyou're not a whore, just polymorphously perverse =P
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08:23.45ChannelZARGH
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08:35.05ricko73I know this isn't a 'linux' channel, but I have a Linux file system question that I could use some help with
08:35.44ricko73a friend's system got hacked and we're unable to rm some files or chmod them
08:39.17Faustovtzafrir__laptop: hi, could you be so kind and tell me what changed in dahdi-linux so that it compiled for 2.6.31 kernels?
08:39.18tzafrir__laptopis the filesystem mounted read-only? This can happen after various disk errors
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08:39.27Faustovomg lucky :>
08:39.37ricko73tzafrir__laptop: no it's rw
08:40.07tzafrir__laptopFaustov, try branches/2.2 . I commited some fixes into it (backported from trunk) yesterday,
08:40.12ricko73it's a web server.  We think they got in via ftp when a yum update replaced a config file, allowing anonymous ftp
08:40.22tzafrir__laptopand now it builds with 2.6.32-rc6
08:40.51TJNIIricko73: You probably gor rooted
08:40.57tzafrir__laptopThe default allows annonymous uploads??
08:41.07TJNIIAre you booting off the disk, or off a livecd?
08:41.25tzafrir__laptopcheck the output of dmesg for 'read-only' or something similar
08:41.31ricko73it's in a colo with no way to to access it recently
08:41.40ricko73access it physically
08:41.41tzafrir__laptopalternatively, if you don't trust the binaries there,
08:41.41Faustovtzafrir__laptop: yeah I remember it works now, but I think I'm having the same problem with wanpipe and maybe a similar fix will be in place
08:41.46Faustovtzafrir__laptop: let me show you:
08:41.54TJNIIChances are the chmod and rm binaries arn't what you think they are.
08:41.59Faustovtzafrir__laptop: http://pastebin.com/m6b4c9f07
08:42.03TJNIIRun rkhunter, if you can.
08:42.06tzafrir__laptopgrab a copy of a statically-linked busybox binary
08:42.08ricko73TJNII: we're using a static busybox binary
08:42.22TJNIIOff the disk?
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08:42.26ricko73yes
08:42.35TJNIIIt could have been attacked, too.
08:42.38ricko73I built it on a separate system
08:42.43TJNIImmmkay
08:42.47ricko73it got scp'ed into the system
08:42.59ricko73it's only affecting one directory
08:43.25ricko73strange thing is the directory is not writable and the root user cannot change it to be writable
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08:43.39tzafrir__laptopFaustov, well, that's for Sangoma to fix :-(
08:43.50ricko73Faustov: good luck with that one ;)
08:44.06Faustovtzafrir__laptop: it looks like just something missing declaration...
08:44.19tzafrir__laptopThe netops fix is one that is not exactly search/replace one
08:44.29Faustovhmm
08:44.41ricko73/path/to/busybox chmod 755 file is 'permission denied' even for the root user in single user mode
08:44.56Faustovwell then, I'll go report it
08:45.15TJNIIricko73: I don't know, but I would still recommend trying rkhunter.
08:45.28TJNIIIf nothing else it might give you a clue,.
08:45.57ricko73TJNII: we ran rkhunt already
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08:46.26Faustovhmm
08:46.33Faustovseems sangoma has no bugzilla
08:46.38tzafrir__laptopFaustov, look at http://lwn.net/Articles/2.6-kernel-api/
08:48.24tzafrir__laptopricko73, that's "permission denied" and not "read-only filesystem", so it's not about a read-only filesystem
08:48.45ricko73tzafrir__laptop: correct
08:48.50ricko73it only affects one directory
08:48.54ricko73that's the strange part
08:49.10Faustovtzafrir__laptop:
08:49.10FaustovMost of the callback functions/methods associated with the net_device structure have been moved out of that structure and into the new struct net_device_ops. In-tree drivers have been converted to the new API.
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08:49.22Faustovstill not search&replace?
08:50.27Faustovi'll try...
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08:56.03ChannelZricko73: they use ACLs or something?
08:56.28ricko73ChannelZ: could be, but we can't figure it out
08:56.33ricko73selinux is disabled
08:57.12ricko73we have three of us looking at it right now.  One guy's been doing unix work for about 15 years.  The other girl longer than that
08:58.22ricko73rkhunter and chkrootkit both came up clean
09:01.44ricko73we're giving up for tonight...
09:04.36bmanif i am creating a sip users table for mysql how should the secret be stored? plain text or md5?
09:09.07Hazukiwould guess md5 just for security reasons, but doesn't know
09:12.47bmani dont either
09:12.52bmanin the conf it isnt md5
09:13.01bmanso its a conundrum
09:13.02Hazukihm, maybe that's for a reason then
09:13.14Hazukiit's possible asterisk expects it in plaintext
09:14.56ChannelZit does in sip.conf so I'd imagine it's the same
09:15.09bmanhrmm
09:15.18bmanmysql docs are sparse it seems
09:17.17TJNIISuck it and see.
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09:19.16bmani dont know i keep getting wrong password
09:19.29bmanits definitely reading from the db tho
09:19.46bmanbecause i changed the username and got a error that it couldnt find the user on the console
09:20.01bmanbut its not groking the secret
09:20.09bmantried md5 and plaintext
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09:22.39ChannelZhttp://www.voip-info.org/wiki/view/Asterisk+sip+mysql+peers
09:22.41bmanwould it report bad password if it were another issue?
09:23.07ChannelZ??
09:23.41ChannelZthat seems to say plaintext.. sure it's correct in the table? Or that you don't have duplicate rows in the table or something?
09:24.35bmani only have 1 row in the table
09:24.56bmanit shows ipaddr, port and regseconds, are those required for all users?
09:25.04bmani dont see default values
09:27.34ChannelZno idea, I haven't done realtime on *.  Good luck, it is soo far past my bed time, I gotta scram
09:27.52bmanthanks anyway
09:29.17ChannelZwhat version of *?
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09:36.29Faustovtzafrir__laptop: speaking of dahdi, gentoo patched dahdi in the tree too
09:36.40HazukiGentoo <3
09:36.47Faustovtrue :>
09:36.50Hazuki2.6.31-gentoo-r3 x86_64 Pentium(R) Dual-Core CPU T4200 @ 2.00GHz GNU/Linux
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09:38.03tzafrir__laptopFaustov, looked at their patches yesterday. One fixes soemthing I can't yet reproduce. Fixes for others should be merged now
09:38.05PanicManSpan 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: Linux26) 1" (MASTER)
09:38.21PanicManTiming is not working with IAX trunk :(
09:38.41PanicManany helper care to help ne plz
09:38.56Faustovtzafrir__laptop: awesome.
09:39.37tzafrir__laptopPanicMan, try 'dahdi_test -v -c 3'
09:40.26PanicManOpened pseudo dahdi interface, measuring accuracy...
09:40.27PanicMan8192 samples in 8191.600 system clock sample intervals (99.995%)
09:40.27PanicMan8192 samples in 8199.072 system clock sample intervals (100.086%)
09:40.27PanicMan8192 samples in 8191.544 system clock sample intervals (99.994%)
09:40.27PanicMan--- Results after 3 passes ---
09:40.27PanicManBest: 99.995 -- Worst: 99.914 -- Average: 99.967738, Difference: 100.025294
09:40.29PanicManbasically I'm trying to configure IAX trunk
09:41.12PanicManserver B can compresss the data
09:41.18PanicManbut server A can not :(
09:41.38PanicManserver B has Digium Hardware TE420P
09:41.47PanicManserver A doesn't have any card
09:42.14PanicMan'll be grateful if anyone can help me out
09:44.33PanicManany body work on trunkmtu?
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09:47.42kaldemarPanicMan: you need a timing source on server A also. use dahdi_dummy if there are no cards to use.
09:50.03Faustovtzafrir__laptop: looks like you were right, changing net_device to net_device_ops started other issues...
09:51.31bmani went from getting bad password to nothing on the server, the client is showing the error Could not register(Unauthorized)
09:51.49bmani did not change anything in the db or the config files
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09:52.48bmaneither way it wont auth
09:53.02bmanive been on this now for 8 hours straigh
09:53.29bmantired and feeling that asterisk isnt the choice for mysql logins
09:55.44Guggebman: tried setting sip debug on and pasting a login attempt on pastebin?
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09:58.07PanicManback
09:58.39PanicMankaldemar > i did, but trunkMtu is not working from Server A to Server B :(
09:59.04PanicMandid u work on trunkmtu
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10:11.27AdvoWorkHi there, has anyone ever had a problem with say trixbox causing wav recordings to be corrupted, ie played back fast? think its something todo with wav49 but cant seem to fix the problem
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10:37.18jafdethi
10:37.56jafdetdoes anybody try to register in skype with chan_skype?
10:58.45tuxx-hey guys, how many concurrent calls can 1 skype account hooked up to an asterisk box take?
10:59.01tuxx-i think its only 1 concurrent call per skype account, but i cant find it on the web :)
10:59.55tuxx-hm, maybe found something
11:00.14tuxx-one skype account can take multiple concurrent calls, every concurrent call needs its own license...
11:00.18tuxx-is that true? :)
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11:08.07kaldemartuxx-: https://www.digium.com/en/supportcenter/documentation/viewdocs/SFA
11:08.46tuxx-tnx
11:08.47tuxx-:)
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11:24.13wendell_hi...
11:24.55wendell_have problem DND in asterisk
11:26.20wendell_please help me
11:26.57kaldemar~ask
11:26.58infobotQuestions 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.
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11:31.44p4d4w4nMy asterisk does not work without internet
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11:39.13Faustovkaldemar: it's futile
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11:54.37jafdetplease tell me, why I can't login with skype login user .... command? it just say nothing and user stays logged out according to 'skype show user' command
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12:28.21waKKuhey folks.. could someone give a look at this post? http://pastebin.com/m5a81d9cd ... i've described my scenario and a debug call for a "mute on endpoint" problem ...
12:31.34kaldemarwaKKu: you seem to be behind a NAT and have nat=no in your config
12:32.56tuxcrafterhi all, how can i make two logs with asterisk to dump the sip debug to a log file and verbose 10 to a log file
12:33.09tuxcrafteri am having call forwarding issues
12:33.32tuxcrafterand i want to make a log to show you people to see if something is wrong at my end
12:33.39tuxcraftersometimes it works and sometimes it does not work
12:34.38waKKukaldemar yeap.. if i put nat=yes i can't call...
12:35.24waKKumy router/firewall has ip 201.202.203.204 and another interface with 192.168.100.1, my asterisk is 192.168.100.2 and all clients is in range 192.168.100.20-80
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12:37.42AdvoWorkHi there, has anyone ever had a problem with say trixbox causing wav recordings to be corrupted, ie played back fast? think its something todo with wav49 but cant seem to fix the problem, or it could be the playback speeds :S
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12:38.43kaldemarwaKKu: can't call where? why?
12:39.15waKKukaldemar lemme set it to yes... moment
12:41.04waKKu; The nat= setting is used when Asterisk is on a public IP, communicating with
12:41.04waKKu; devices hidden behind a NAT device (broadband router).
12:43.16kaldemaryes, when a DEVICE IS behind a NAT, then you use the setting for a peer.
12:43.57kaldemar~sipnat
12:43.58infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
12:49.21waKKukaldemar damn.. works like a charm :(
12:49.28waKKusorry.. i'd read it wrong
12:50.22kaldemaryou're not supposed to be unhappy when you get it to work. :)
12:53.40waKKui'm unhappy to be silly ;)
12:54.21mchoukaldemar: the [general] section in sip.conf, does it have global scope?
12:55.06kaldemarit has a general scope. :P
12:55.41kaldemarif you mean that do all device definitions get the exact same settings, the answer is no.
12:55.58mchoukaldemar: in other words, if under [general] nat=yes, do [clients] need nat=no or yes (assuming all clients beand asterisk behind same nat)
12:56.56mchoukaldemar: so I'm confused....
12:57.26kaldemarthe default is nat=no. so you don't have to explicitly set nat=no for clients.
12:57.51mchouwhat does nat=yes in [general] logically apply to?
12:57.58mchouasterisk itself?
12:58.49kaldemaryes, asterisk itself. it tells asterisk to put externip setting as address into outgoing SIP messages. otherwise asterisk would put a private address in the messages.
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12:59.18tuxcrafter~book
12:59.19infobot[~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
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13:21.54ariel_Morning
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13:29.24phixwb [TK]D-Fender
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13:31.11c0rnoTaHi all
13:31.57c0rnoTais anybody here?
13:33.41ChainsawYes, the mic is on. Please stop tapping it :)
13:34.47TSMis it possable to
13:34.50TSMwwoops
13:35.11c0rnoTawow, sorry :) i have one question.
13:35.28TSMis it possable to parse pri messages in * to give the correct error message to the user
13:35.45c0rnoTaand could not find any answer in google
13:36.06c0rnoTaso what's mean chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel
13:36.11c0rnoTa?
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13:38.02c0rnoTai'm getting calldrops, and this debug message appears before hangup
13:38.55kaldemarTSM: check cause codes in dialplan
13:40.07kaldemarTSM: variable HANGUPCAUSE, to be exact
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13:48.25TSMkaldemar: 28
13:48.59TSMall cool, ive just looked up the list of numvers for hangupcause
13:49.14TSMso now i can mod the script
13:51.19TSMare variables global for the life of the call session even when you bouce around diffrent contexts
13:53.21kaldemarchannel variables are associated to a channel. it doesn't matter what context you're in, as long as the channel is the same.
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14:09.17eppigyTRABAJO
14:09.37jayteesi! trabajo!
14:14.23Katty)=<
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14:20.21mort_gibHi, Voicemail attendent is stuttering, which is a timing issue, but I have dahdi_dummy loaded??
14:23.44jayteemorning Katty *hugs*
14:24.28Kattyhi jaytee
14:24.30Kattyhugs jaytee
14:24.55jayteemort_gib, might want to try load speech_therapist.so  :-)
14:25.10[TK]D-Fendermort_gib: Onlyt he prompts?
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14:28.33mort_gibI can't even hear the prompts
14:29.03mort_gibmoh sounds fine
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14:32.08Kattystruggles with the grumpies.
14:33.31creativxpms?
14:34.19Kattyno, just cranky
14:34.37creativxbg
14:34.39creativxits friday !
14:38.12Kattytrying :<
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14:41.59jayteehas anyone been to voip-info.org today? it's got a rather odd message at the top of the main page in red text.
14:42.29Naikrovekhah
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14:44.29mort_gibdahdi_test gives me -200.XX% +/-
14:44.33mort_gib???
14:44.50eppigyD:
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14:47.21[TK]D-FenderjayLink harvesting fuck-off was spamming the news with links to every article on a "new site"
14:47.25[TK]D-Fenderjaytee: Link harvesting fuck-off was spamming the news with links to every article on a "new site"
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15:02.08jayteewhat's the silver "joystick" type control with the checkmark button in the middle called on a Polycom phone?
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15:06.59[TK]D-Fenderjaytee: "Navigation keys" most likely
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15:10.35ManxPower-work~answers
15:10.36infobotfrom memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
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15:13.46McLazarusIs there an asterisk channel variable that will tell you the auth username?
15:14.35mort_gibhow come my dahdi_test gives me -200. % ???
15:14.58mort_gibI'm not running an old kernel 2.6.26
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15:15.17leifmadsenMcLazarus: probably in one of the SIP_* functions
15:15.25leifmadsenMcLazarus: I doubt it's a channel variable
15:15.36leifmadsenSIP_PEER() function, or maybe CHANNEL()
15:15.50McLazarusleifmadsen: thanks!  I'll check it out.
15:19.29jaytee[TK]D-Fender, thanks
15:20.07mort_gibWhat would make dahdi_test report -200% ?? it's supposed to be around 99.99%
15:20.09mort_gib??
15:22.00McLazarusI was able to get what I wanted using:
15:22.13McLazarusSIPCHANINFO(peername)
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15:24.27Kattyshould write today off as a bad idea.
15:24.49ManxPower-workKatty: I'm installing SugerCRM, and I think that makes today a bad idea.
15:25.01KattyManxPower-work: we already have it running here.
15:25.08KattyManxPower-work: it's generally not the source of my irritation
15:27.33keypI have asterisk hooked up to an NEC PBX via PRI. There's a T1 card in the NEC PBX that's providing a PRI line to Asterisk. I have several extensions on the NEC system routed out the PRI to Asterisk, where matching extensions exist. This works fine, except that when a call is transferred on the NEC system, the transferrer's CallerID is sent until the transferrer hangs up, at which time it...
15:27.35keyp...changes to the transferree's CallerID. Asterisk seems to ignore this change. Is there a way I can make Asterisk handle an update to the CallerID info after the phone has started ringing?
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15:29.59ManxPower-workKatty: I don't enjoy working with, installing, or managing "web applications"
15:30.39ManxPower-workkeyp: "core show application dial"  Pay special attention to the "o" option
15:30.40KattyManxPower-work: there are a great many things in life we don't enjoy.
15:30.44jayteegoosh.org FTW!!!
15:31.17FinboySlickThis might sound a bit dumb, but I'm trying to connect a type=friend to a non-standard port.  I tried host=ip:port but it still hits the remote box with 4569.
15:31.36ManxPower-workFinboySlick: I don't think IAX2 support non-standard ports.
15:31.42ManxPower-workI'm not SURE however.
15:32.21FinboySlickManxPower-work: Heh, it suports listening on them (bindport) :P
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15:33.13ManxPower-workFinboySlick: it's not the same thin
15:33.16ManxPower-workg
15:33.40mort_gibWhy would dahdi_test -v report 8192 samples in 32768.109 system clock sample intervals (400.001%)
15:33.55FinboySlickManxPower-work: No, obviously.  But it's a bit odd that you'd support listening on a different port with no means of connectiong to that different port.
15:33.57ManxPower-workmort_gib: that sounds like a question for Digium Support.
15:34.15ManxPower-workFinboySlick: I agree with that.
15:34.19mort_gibSo a later version of dahdi??
15:34.36ManxPower-workmort_gib: I assume you have the latest version before complaining about problems.
15:35.57mort_gibNo, sometime upgrading creates more problems, but i use 2.2.0.2
15:36.12Kattyand sometimes upgrading is what fixes a bug.
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15:36.29mort_gibKatty: true... I hav tried both
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15:40.42keypManxPower-work: That doesn't seem to help, It's not Asterisk that's sending the CallerID, it's the NEC system. The problem is that on the NEC system, the extension of the person transferring the call is sent first. Then, once the person transferring the call hangs up, the CallerID is changed to the transferred number. I'm only seeing the first set of callerID info sent.
15:41.28ManxPower-workkeyp: put a Wait(.5) in your dialplan before sending the call to the NEC
15:41.56ManxPower-workBTW, if anyone is considering going with XO -- just don't.
15:42.18keypOther way around. The NEC is sending the call to Asterisk.
15:42.47ManxPower-workkeyp: then put a Wait(1) as at the beginning of the part of the dialplan that accepts the call
15:43.09FinboySlickManxPower-work: For the record, it's merely a matter of adding port= to the account.
15:43.30ManxPower-workCallerID NAME is always sent as a message separate from the call setup.
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15:47.24CGMChrisAnyone familiar with AMD()?  I'm trying to use it over SIP channels but it always returns "TOOLONG", even if I set totalAnalysisTime to 30000
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15:51.03Kattyoh where oh where has my perky gone, oh where oh where could it be.... :<
15:52.00Faustovporky?
15:52.23Kattythat is not what i typed.
15:53.36jayteelikes perky
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15:57.38Kattyanyone familiar with the Quince fruit?
15:58.29eppigynegative
15:58.44Tim_Toadymy mother used to make marmelade using it and other sweets
15:59.27UQlevKatty: what do you want to do with it?
15:59.32Kattyi'm not sure.
15:59.38Kattyin fact, i'm not even sure how to prepare it.
15:59.47Kattytho, i've found a recipe for honey baked quince.
15:59.51Kattymuch like cored, baked apples.
15:59.55UQlevKatty: you can boil it
16:00.02UQlevroast with meat
16:00.07UQlevmake jam
16:00.09KattyUQlev: how would you suggest preparing it?
16:00.55UQlevKatty: I like to put it inside duck and roast
16:01.07Anth8708Hey guys, I'm using queues and passing a macro to the queue.  I would like to pass an argument to the macro, but I can't find any documentation on syntax.  Does anyone know?  Here's what I have now: exten => s,1(onhold),Queue(helpdesk,hHnt,,,,,say-call-length(${CDR(duration)}),,)
16:01.25Kattyhmm. duck.
16:01.31Kattydon't think i've ever had duck.
16:01.36Kattybut, this is a good a time as any to try it.
16:01.42UQlevKatty: turkey
16:01.54Kattyno, no..i want to try duck now.
16:02.04Kattyidly wonders where to buy Duck.
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16:02.27Faustovi know how to kill one!
16:02.41ChainsawI know a pond where they swim!
16:02.49Katty>.<
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16:03.54KattyUQlev: how many pounds do you figure a Duck generally is? or do you simply buy cuts of duck.
16:04.35Kattyoooh! a local place carries duck.
16:04.50Katty6 to 7lbs
16:05.00UQlevKatty: whole
16:05.53Kattyhow does one prepare, Whole Duck
16:06.00Kattyand what do you do with the fruit
16:06.02Naikrovekomfg i hate linux sometimes
16:06.15KattyNaikrovek: me too. let's talk about duck instead.
16:06.16Naikroveksamba in particular
16:06.20Naikrovekduck
16:06.24Naikrovekduck is yum
16:06.26Kattyroast duck.
16:06.27Naikrovekand good pet
16:06.31Kattywith quince...and probably other things.
16:06.35Kattyduck makes wonderful pet.
16:06.57Naikrovekuntil they wake you up by quacking in your ear at 4:30am
16:07.15Katty^_-
16:07.17Naikrovekheh
16:07.19Kattyducks are kept outside.
16:07.22Kattywith a pond.
16:07.26Kattyand chickens.
16:07.36Kattychickens do not make good pets.
16:07.37Naikroveki'm so fed up with samba i'm going to overturn a table or two
16:07.44Naikroveki'm borderline nervous breakdown here
16:07.55*** join/#asterisk momelod (n=smelo@CPE00a065c98ce6-CM0012c91df0bc.cpe.net.cable.rogers.com)
16:07.58Katty:<
16:08.00momelodgreetings channel
16:08.02Naikroveksamba is such a pile of stool
16:08.08Kattyi am sorry.
16:08.10Kattyhugs Naikrovek
16:08.13CGMChrisDo any of the duck lovers know anything about AMD() ?
16:08.14ChainsawHi momelod.
16:08.15*** join/#asterisk scalex000 (n=chatzill@80puntacana02.codetel.net.do)
16:08.32KattyNaikrovek: any chance you can get away from the problem for an hour or so? go have a drink.
16:08.33*** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM]
16:08.35momelodcan anyone tell me if its possible to give callers an option so that when they hit a voicemail box they can hit a button to get back to the main menu, or operator?
16:08.35*** join/#asterisk clyons (n=clyons@unaffiliated/clyons) [NETSPLIT VICTIM]
16:08.39KattyNaikrovek: clear your head a bit.
16:08.46NaikrovekKatty: i've been doing that all week
16:08.57*** join/#asterisk w-m- (i=w@dengue.zephyr.ws)
16:08.57Katty:<
16:08.58Naikrovekworking on it for an hour, then taking an hour break, then working for an hour
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16:08.59Naikroveketc
16:12.54*** join/#asterisk infobot (i=ibot@rikers.org)
16:12.54*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.9 (2009/11/04), 1.6.0.17 (2009/11/04), 1.4.26.3 (2009/11/04), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs
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16:14.14eppigySHAKE YOUR ASS LIKE YOU KNOW HOW TO DANCE
16:14.25Chainsawmomelod: http://www.pastebin.ca/1659671
16:14.33FinboySlickForgive my noobism again, but can I send a call to another context sort of like this:  exten => _NXX,1,Dial(${EXTEN}@someothercontext)   ?
16:14.42Kattyhm. dancing.
16:14.46Kattyeppigy: let's go dance.
16:14.55Kattyeppigy: perhaps i'd find my lost perky on the dance floor.
16:15.02FinboySlickHmm, that'd be a goto statement, no?
16:15.17KattyFinboySlick: yes.
16:15.28KattyFinboySlick: goto(foo,s,1)
16:15.31FinboySlickKatty: Thanks :)
16:16.39momelodChainsaw, Thanks!
16:16.56FaustovChainsaw: exten => o instead of 0?
16:17.08ChainsawFaustov: This was pulled from a working dial plan.
16:17.19*** join/#asterisk jeff (i=jeff@unaffiliated/jeff) [NETSPLIT VICTIM]
16:17.38FaustovChainsaw: sure, just asking why is it o rather than 0 that you are waiting for
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16:19.29jayteetouched M.C. Hammer
16:20.17eppigyKatty: yes that would be fun
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16:23.45momelodChainsaw, perfect thanks brother
16:24.13Chainsaw(For the record, o is operator)
16:26.05Kattyjaytee: STOP
16:26.09Kattyjaytee: HAMMERTIME
16:26.23Kattyeppigy: I HAVE LOCATED MY LOST PERKY!
16:26.39CGMChrisIf I call AMD() immediately after Answer(), no speach is detected.  However, if I first call Background(Silence/1), then AMD(), IT WORKS.  Anyone have any idea why this could be?
16:27.12Kattyeppigy: it appears i miss placed it in a glass of iced tea.
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16:28.19Katty:>
16:28.25Kattyi knew you were going to show up.
16:28.28Zeeek{{{Katty}}}
16:28.30Kattyi have been waiting.
16:28.36Kattyhugs Zeeek
16:28.39Zeeekso sweet
16:28.44Zeeekthank you so much
16:29.03Zeeekat my age, huggled in my blanket, my day is now complete
16:29.26beekKatty: When are you going to join us for the conference?  We'd love to have you.
16:29.56ZeeekKatty said one thing once in IRC on #voip-users-conference
16:29.57Corydon76-digThat sounds slightly creepy
16:30.04ZeeekI think it was something like "Hi"."
16:30.17[TK]D-FenderCGMChris: Because the audio path isn't established until you play a sound
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16:30.29beekCorydon76-dig: Only slightly?
16:30.45Corydon76-digLike a gaggle of vampires
16:30.53[TK]D-FenderZeeek: And I was like your first VUC guest on Polycom provisioning, presence, etc :)
16:31.15ZeeekVillage Telco project is our topic today it's open source hardware and software to bring telephony to the farthest reaches of the galaxy for little or no cost
16:31.17Kattybeek: well it's rather difficult because my lunch break is when the conference starts
16:31.22[TK]D-FenderCorydon76-dig: Gaggle is for geese which are herbivores.. vampires are certainly not in "gaggles"
16:31.30ZeeekTK you were and I have sorely missed you ever since
16:31.43Corydon76-dig[TK]D-Fender: A murder of vampires?
16:31.46ZeeekKatty: that's easy, skip lunhc!
16:31.48[TK]D-FenderCorydon76-dig: typically they follow the occult style namings like "covens"
16:31.51Katty^_-
16:31.53Kattyi think not.
16:32.01ZeeekYou don't care about us.
16:32.05[TK]D-FenderCorydon76-dig: Murder of crows... close... at least considerably more deserving
16:32.11beekKatty: Bring your lunch to your desk and eat it during the conference.
16:32.16ZeeekKevin Spacey, yeah great movie
16:32.16Corydon76-digMurder of cows, actually
16:32.35[TK]D-FenderCorydon76-dig: O RLY?  don't think I was sure on that one...
16:32.38Zeeekno murder of cows is the Meatrix
16:32.51Kattyutterly homicidal.
16:33.22momelodanyone here using cisco's 7960s?
16:33.23Zeeekutterz
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16:34.02ZeeekOpenSky has driven me nuts
16:34.10Zeeekchalk my madness up to that
16:34.15momelodIve got an issue w/ cisco 7960's directory button which points to a script which generates the list of contacts from the db into an xml format the phone can read... but it only lists the first 32 entries...
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16:34.33momelodif i run the same script on the cli, it returns all the results
16:35.47ZeeekOk, one final spam: http://VUC.me for info or #voip-users-conference to hear more about Village Telco. And once in a great while, [TK]D-Fender may be available for questioning. But almost never
16:36.11Zeeek[TK]LesPaul may be there, though
16:37.47Chainsawmomelod: If you run the 7960 on SIP firmware...
16:38.05Chainsawmomelod: Then the XML support is extremely limited, and 90% of what the documentation says it can do does not apply. Welcome to my world.
16:38.42p3nguinI still don't understand why you have so much trouble with your Cisco phones.
16:39.36Chainsawp3nguin: Wasn't it you that recommended a version 6.3 SIP firmware?
16:39.37momelodChainsaw, do u have the same artifacts? where your xml lists are trunkated ?
16:39.46p3nguinchainsaw: Negative.  I use 8.11.
16:39.48*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
16:39.52momelodall the other xml stuff has worked for me fine.
16:40.05Chainsawp3nguin: Right. Avoid 8.12; it truncates caller ID.
16:40.16Chainsawp3nguin: (And has similar flaws in other areas)
16:40.56momelodoh, im running 8.12
16:41.04momelodshould i revert back to 8.11?
16:41.18p3nguinmomelod: Is 8.12 broken for you?
16:41.34ChainsawEasy to check. Have a caller ID string with a space in it.
16:41.50Chainsaw"Red phone" will become "Redphone"
16:42.02momelodp3nguin, no, its working well except for this one xml thing..
16:42.16p3nguinWouldn't truncating it make it "Red" and that's all?
16:42.21ChainsawIf they screwed up something as elementary as whitespace in a caller ID string, god knows what other things are broken.
16:42.58Chainsawp3nguin: Perhaps corruption would fit you better then.
16:43.16p3nguinI guess I could use 8.12 and see what happens.
16:43.25p3nguinShouldn't be hard to go back.
16:43.47Chainsawp3nguin: Downgrade works correctly, indeed.
16:44.59momelodim trying a downgrade right now
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16:53.36momeloddang it, it doesnt want to downgrade
16:53.56momelodthe tftp server shows it requesting the 8.11 frimware but the phone doesnt install it
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16:54.34p3nguinDid you ensure that you have changed the version number in all the appropriate files?
16:55.55p3nguinI know it needs to be changed in, at least, SIPDefault.cnf and OS79XX.TXT.
16:57.03Chainsawp3nguin: Why yes, I did change it there.
16:57.24Chainsawp3nguin: Please note that a dash has moved in the version string.
16:58.00p3nguinBetween 8.11 and 8.12, it should be the same.  The dash moved in like version 7 or something.
16:58.25ChainsawIt was an extra 0, or a 0 went away. It is different and it bit me the first time.
16:58.39momelodtrying again, i may have missed the "S" in P0S-8-11-00
16:58.57Chainsawmomelod: And check for that 0 please.
16:59.05Kattyhas stuff in the oven
16:59.18beekYou're pregnant?
16:59.19p3nguinimage_version:P0S3-08-11-00
16:59.21Kattyno.
16:59.25Kattyit's a casserole.
16:59.33beekAhhh.... wrong oven.
16:59.34p3nguinP003-08-11-00
16:59.36momelodChainsaw, sorry what "0"?
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16:59.41Chainsawmomelod: My OS79XX.TXT has P003-08-11-00
16:59.47p3nguinJust use what I pasted.
17:00.09momelodbut the files themselves have P0S3 ..right?
17:00.24c0rnoTayes
17:00.35c0rnoTaOS79XX.TXT  should have P00
17:00.35p3nguinP003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2
17:00.40c0rnoTabut firmware POS
17:00.43Chainsawmomelod: the files themselves, yes. P0S3-08-11-00.loads
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17:01.14Chainsawmomelod, p3nguin: Ah yes, it was a 0 gone away, in comparison: P0S3-8-12-00.loads
17:01.19momelodso in SIPDefault and xmldefault is there an S in there?
17:01.42Chainsawmomelod: In SIPDefault I have image_version: P0S3-08-11-00
17:01.56p3nguinmomelod: I've already pasted the exact info.
17:02.19Chainsawp3nguin: Second opinion is a normal part of the technical support process.
17:02.21Qwelleasy way to upgrade Cisco phones: Sell them on eBay and buy Polycoms
17:02.32p3nguinThat's not an upgrade.
17:02.48ChainsawQwell: Yes, the 7960 does not do IPv6. That'll be my reason.
17:03.32*** join/#asterisk lucasb (n=bussey@office.telifon.com)
17:03.43p3nguinI haven't seen any Polycom that I like as much as Cisco.
17:04.56momelodDoesnt wanna downgrade for me. "OS79XX.TXT : P003-08-11-00", "SIPDefault.cnf: image_version: "P0S3-8-11-00"", "xmlDefault.cnf: <loadInformation7 model="IP Phone 7960">P003-8-11-00</loadInformation7>", ""
17:05.04Qwellwhat does a Cisco have that Polycom doesn't do - better?
17:05.19Kattygeneration of profit.
17:05.27Kattyduh, Qwell.
17:05.41Chainsawmomelod: SIPDefault.cnf: P0S3-08-11-00
17:06.11momelodohhhh
17:06.15momelodthats the extra 0 u mean
17:06.16momelodlol
17:06.18ChainsawExactly.
17:06.22ChainsawIt bit me too :)
17:06.34p3nguinsigh
17:06.59eppigy8[]
17:07.09Chainsawp3nguin: Hey, turn that frown upside down.
17:07.27momeloddangit called into a meeting
17:07.35momelodback in a bit
17:08.20*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
17:11.35Anth8708I'm having an issue with GotoIf not working right.  I'm sure it's a syntax issue, but I can't find it:(.  exten => s,n,GotoIf($["${CDR(duration)}" >= "99"]?vmlong) ;  A NoOp clearly shows the CDR duration over 99, but I don't skip to vmlong:(.  Any ideas?
17:12.00eppigyQwell: about 66.8 billion more in assets
17:12.05eppigyYOU KNOW WHAT I MEAN
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17:23.49jayrod422anybody know of any examples for the asterisk manager that use async and originateresponse with the originate command?
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17:40.38Kattyeppigy: i don't think Qwell appreciated our comments.
17:42.08kaldemarAnth8708: don't use "" when you're comparing numbers
17:42.42leifmadsenprefix the variable names with a 0 if you are unsure whether the value in the variable will be NULL
17:42.53Kattyhi mister madsen.
17:42.57leifmadsen$[0${FIRST_VALUE} > 0${SECOND_VALUE}]
17:42.59leifmadsenfor example
17:43.24leifmadsenthen if ${FIRST_VALUE} contains nothing, you don't get a WARNING and a failed comparison.
17:43.46leifmadsen$[0 > 025] vs. $[  > 25]
17:43.51leifmadsenKatty: ms. katty
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17:55.41eppigyKatty: :<
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18:01.42vallyi had a look at the rtmp branch and compiled the source. when starting asterisk i get an error while loading the module: undefined symbol: av_audio_resample_init. thats a ffmpeg method he obviously can't find. do i have to link it somewhere else? i am confused because i linked the ffmpeg libs in the configure script and it built successfully. anyone? :)
18:03.08TSMin my dialplan im trying to test to see if a variable has been set, when it has then all good, but when it has not i cant evaluate it to true
18:07.47Kattyhmm. my lunch is entirely vegan.
18:08.21QwellKatty: broccoli has souls too
18:08.26vallyKatty: you should appreciate that
18:08.40Kattyi used to be vegan.
18:09.19russellband then you remembered how tasty meat was
18:09.39Kattythen i went to the doctor with extreme deficiencies
18:10.04ryduhKatty: what kind of deficiencies?
18:10.16russellbnot enough meat
18:10.17Kattycalcium, iodine, and b12
18:10.46vallyyou must eat more legumes :)
18:11.09Kattylentils for lunch today.
18:11.29FinboySlicklikes lentils, but also tartar steak.
18:11.46Kattyi enjoy bison tacos.
18:11.58ryduhKatty: did you know of the deficiencies before going to your doc?
18:12.11Kattyno, just generally feeling ill and lethargic.
18:12.47ryduhKatty: what kind of test(s) did your doc run to figure out what you were lacking?
18:13.01Kattythey did blood tests.
18:13.07Kattybut i'm not sure of what Exact Tests were ran.
18:16.09Kattyi still have to take b complex.
18:16.13Kattyand i'm also taking d now too
18:17.52*** part/#asterisk FinboySlick (n=shark@207.134.8.4)
18:18.12ariel_~nat
18:18.13infobotmethinks nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
18:19.26leifmadsenvally: you might want to ask that question in #asterisk-dev
18:19.35vallyok
18:19.54vallyi guess there are not so many c developers in this channel
18:20.02leifmadsennope, that's what -dev is for :)
18:20.15vallyk
18:22.06Kattyeppigy: STOP
18:22.08Kattyeppigy: CHOCOLATETIME
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18:42.30ariel_heads to drink a hot chocolate since Katty posted the word...
18:44.35CGMChris[TK]D-Fender: Any idea what minimum length a Background() file must be in order to properly establish the audio channel for use with AMD() ?  In my tests, a 100 milisecond file containing silence works, but a 10ms file does not.
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18:58.11Kattyariel_: :>>>>>>>>>>>>>>>>>>>
18:58.14Kattyhugs on ariel_
18:59.16p3nguincgmchris: AMD is used for making automated outbound calls, right?
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19:01.23Naikrovekhas 12 google wave invites left, if anyone wants one
19:01.54FinboySlickI have multi-line sip phones.  In some cases, I would like to drop straight to voicemail (not ring another line on the phone) if it the phone is already busy.  Is there some way to see if a sip client is already busy before ringing it?
19:02.29[TK]D-FenderFinboySlick: "core show application chanisavail"
19:02.59FinboySlick[TK]D-Fender: I meant more as part of the dialplan...
19:04.10FinboySlick[TK]D-Fender: Okay, sorry ;)
19:04.16ChainsawFinboySlick: You can indeed use ChanIsAvail in the dial plan; note that you'll have to limit the maximum amount of calls for that SIP peer in order to use it for busy detection.
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19:05.50[TK]D-FenderChainsaw: Yeah, because every SIP phone should require supporting 100's of calls :)
19:07.05ChainsawAnd the train wifi just misplaced my session cookie. Great.
19:09.11p3nguinI would still like to know if there is any way to make my phone ring when I'm already on the phone, as opposed to the call waiting tone.
19:10.09[TK]D-Fenderp3nguin: depends on the phone
19:10.16QwellNaikrovek: I'll take one
19:10.21p3nguinCisco 7940 or 7960
19:10.32NaikrovekQwell: email address?  pm it to me if you like
19:12.04*** part/#asterisk scottsmith7 (n=hammer@64.201.141.80)
19:12.45p3nguinI have tested it under several conditions:  using the same SIP account for all lines, using different SIP accounts for each line, enabling call waiting in the menu, disabling call waiting in the menu
19:13.57p3nguinUsing the same SIP account for all lines, or using different SIP accounts on every line, I get the call waiting tone no matter which line I am using and no matter which number is being called.
19:14.35p3nguinWith the call waiting disabled, no second call is accepted, regardless of which exten is being dialed.
19:14.57p3nguinI feel that I have exhausted all combinations of possibilities.
19:16.11[TK]D-Fenderp3nguin: Easy possibility : replace them with Polycoms :)
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19:16.53p3nguinI suppose Polycom provides a user option to ring or beep?
19:16.59[TK]D-Fenderp3nguin: yup
19:17.08p3nguinfigures
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19:18.45hardwirehi
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19:30.17donnibhi
19:35.13Kattyeppigy: ONCE A NATIVE MAIDEN AND A STRANGER MET UNDER THE BLUE TAHITIAN MOON
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19:35.52Kattymister madsen
19:36.43leifmadsenyes!
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19:37.55Kattymy brain always wants to type madison.
19:39.59[TK]D-Fenderleifmadsen: NA NA NA NA NA!
19:39.59[TK]D-FenderLeif is Leif!
19:42.51FinboySlickA laibach fan...
19:44.18*** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl)
19:45.32donnibanyone have experience with Fritz!Box and asterisk server ?
19:46.18QwellFinboySlick: not quite
19:47.01*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
19:47.18donnibi am having troubles registering the box to asterisk
19:47.30donnibit registers but then looses registration after a wile
19:47.32donnibwhle
19:47.54FinboySlickQwell: They should make a set of asterisk sounds :)
19:49.37donnibif anyone is curios what Fritz!Box is here is a link: http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_WLAN_7270/index.php
19:49.49donnibi would appreciate if someone can help me with the registration issues
19:50.04QwellFinboySlick: Laibach's is a cover, is what I was hinting at
19:50.38Kattyi'm thinking of a vin diesel movie.
19:50.46Kattyand he's smuggling some girl from russia to somewhere else.
19:51.14Kattysomething about tigers.
19:51.24Kattyhelp :<
19:51.31FinboySlickQwell: Well, yes, that goes without saying.  But the 'leif' bit sounded more like the cover than the original.
19:51.47leifmadsenI'm the original Leif Madsen
19:53.05FinboySlickHeh, that makes more sense now.
19:55.35coppiceleifmadsen: I'm sure there were some before you
19:56.53leifmadsencoppice: they don't count though
19:56.56*** join/#asterisk donnib_ (n=donnib@0x555281d0.adsl.cybercity.dk)
19:57.05donnib_anyone ?
19:57.06eppigyKatty: yes they did
19:57.12Kattyi can testify that leifmadsen is the original Leif Madsen
19:57.24Kattyeppigy: did?
19:57.25Kattyeppigy: what?
19:57.27Kattyeppigy: who?
19:57.57eppigy< Katty> eppigy: ONCE A NATIVE MAIDEN AND A STRANGER MET UNDER THE BLUE TAHITIAN MOON
19:58.10Kattyeppigy: eppigy that was forever ago.
19:58.35Kattyeppigy: http://www.youtube.com/watch?v=nksJQRXglnc <- Ref.
19:59.24Kattyeppigy: do you know the vin diesel movie where he's comissioned to smuggle someone from russia somewhere else. there were tigers in the movie.
19:59.40Kattyeppigy: can't recall much else :<
20:00.41eppigyI have not seen it
20:00.50Katty:<
20:01.21Kattykeeps searching
20:01.50eppigyKeira Knightley 8[]
20:02.12Kattyeppigy: she's one of my favorite actresses.
20:03.23Kattyeppigy: who's yours.
20:07.12lost_soulBabylon A.D.
20:07.46lost_soulI believe is the Vin Diesel movie your referring to
20:07.52Kattyimdbs.
20:08.05Katty:>>>>>>>>>
20:08.07Kattyhugs lost_soul
20:08.11Kattythank you, sir.
20:08.15lost_soulanytime
20:14.24Kattynetflix doesn't have it on insta-play :<
20:16.27eppigyKatty: lily thai
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20:19.46Kattylily chai?
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20:24.58AeroCloudServer was running fine for about 1 month.. now getting this error.
20:24.59AeroCloudERROR[31777]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe
20:25.12AeroCloudanyone have an idea of what went wrong? or what is broken?
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20:55.22Kattysways.
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21:14.34wcselbyo/
21:15.38*** join/#asterisk twanny796 (n=twanny@85.232.220.197)
21:16.02twanny796I need a  zaptel module for kernel 2.4.36...
21:16.31*** join/#asterisk gilevy (n=gil@c-24-10-28-163.hsd1.ca.comcast.net)
21:16.43tzafrir_laptophands twanny796 http://svn.asterisk.org/svn/dahdi/branches/1.4
21:16.56gilevy<PROTECTED>
21:17.07gilevyeverything was working fine earlier
21:17.14gilevyi don't know what i changed
21:17.16gilevyany help?
21:17.35tzafrir_laptoptwanny796, There are no bug reports saying it fails to build. So it must be ok
21:18.05tzafrir_laptopgilevy, what specific file?
21:18.31gilevyany of the gsm
21:18.35twanny796tzafrir_laptop: it's failing to build for me
21:18.38gilevythat come packaged
21:18.42*** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com)
21:18.44gilevylike "is"
21:19.05tzafrir_laptoptwanny796, can you provide a trace of the build?
21:19.47Anth8708Guys, I am attempting to use the o option with the Dial command, which should allow a forwarded call to retain the original caller id, correct?  It doesn't seem to be working.  Is there anything else that is needed?
21:20.00twanny796tzafrir_laptop: will try, but it might take me some time
21:21.04tzafrir_laptoptwanny796, if you don't get a response, please file a bug. ATM I don't have any 2.4 build system around
21:21.28gilevyhttp://forums.digium.com/viewtopic.php?p=77554&sid=1487bf8143cef831c8efde47dd532a42
21:21.34gilevyi have the same problem as he does
21:21.57gilevywhat does he mean by copying back in the codec
21:22.08gilevycopying the codec to where?
21:23.22leifmadsengilevy: he means he didn't copy in the codec_g729a.so module into /usr/lib/asterisk/modules/
21:23.49leifmadsencodec_g729a.so is a pre-compiled module you have to get from Digium and register against, and copy the module into the modules directory to use
21:24.05tzafrir_laptopgilevy, please provide a slightly more complete trace
21:25.57gilevy<PROTECTED>
21:26.00gilevy[Nov  6 13:42:02] WARNING[9526]: channel.c:2948 set_format: Unable to find a codec translation path from 0x1000 (g722) to 0x2 (gsm)
21:26.03gilevy[Nov  6 13:42:02] WARNING[9526]: file.c:954 ast_streamfile: Unable to open freeze (format 0x1000 (g722)): No such file or directory
21:26.06gilevy[Nov  6 13:42:02] WARNING[9526]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/101-0886c5d8 for freeze
21:26.16wcselby~pb
21:26.17infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
21:26.21wcselbygilevy ^^^^
21:26.24gilevyah
21:26.29gilevyone sec
21:27.08Kattyit's wcselby! :>
21:27.20gilevyhttp://pastebin.com/m1541a1ae
21:27.33wcselbyhey Katty! :)
21:27.52[TK]D-Fendergilevy: What ver of *?
21:28.07gilevy1.4
21:28.19gilevyit was working earlier
21:28.25[TK]D-Fendergilevy: * 1.4 does not support transcoding to G.722
21:28.32Kattyherroes :>
21:28.36gilevyo
21:28.50[TK]D-Fendercheckout time, home in a bit
21:29.02KavanScan anyone point me to some reading for database (mysql) authentication with asterisk....i.e. user dials up, enters their clientID (verified by mysql), then is forwarded to the correct person
21:29.58*** join/#asterisk PanicMan (i=PanicMan@122.102.33.68)
21:30.12gilevysweet it works thanks guys!
21:30.15PanicManlooking for some help regarding IAX
21:32.21PanicMan:(
21:32.30PanicManeveryone is busy or sleeping
21:32.59QwellPanicMan: or on the edge of their seats waiting for you to ask a question
21:33.11wcselbyQwell - shhhh, we're sleeping
21:34.45PanicMan:(
21:34.58PanicMansorry if i hurt anyone :(
21:35.16PanicManI'm really in trouble with IAX trunkMTU issue :(
21:35.41*** join/#asterisk errotan (n=errotan@a1612.adsl.pool.eol.hu)
21:35.41PanicMantried in the google , but found not that much info :(, so, this channel is the last hope :(
21:36.13PanicMantrunkmtu=1240 ; may not be supported on your Asterisk depending on the
21:36.13PanicManversion
21:36.38PanicManso, which version supports it :( which not, how can i get that info sir ?
21:36.59donnibhow can one get more debug info from chan_?
21:37.03*** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net)
21:37.11donnibthan using sip set debug
21:38.01gilevyis there a function for entering numbers after answering call
21:38.06gilevysuch as an extensions number?
21:38.38QwellPanicMan: Don't msg people without asking, please.
21:38.44PanicMansorry :(
21:39.01*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
21:43.16gilevyis there a function for entering numbers after answering call
21:47.30Kattygilevy: read()?
21:48.56gilevythat reads what the caller is dialing
21:49.15gilevycan you dial back is the question
21:49.22gilevysend dtmf back
21:49.23*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
21:49.27Kattyyou simply dial the variable.
21:49.34Kattyread(foo)
21:49.36Kattydial($foo)
21:49.51gilevyoh easy as that
21:49.57gilevyaight thank you
21:51.59*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:52.17hardwireanybody got an spa-942 for sale in Oahu?
21:52.17Kattyhi fender.
21:52.29Kattywhere is Oahu
21:52.35hardwireHawaii
21:52.50hardwireNevermind.. they found it.
21:52.52hardwire:P
21:53.31Kattyhawaiians are snobs.
21:54.31Kattythey're very mean :<
21:55.51hardwireuh.. ok :|
21:56.18Kattyhardwire: i hope you don't scowl at people who aren't from hawaii when they talk to you.
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21:58.10ruben23hi anyone knows this error log on my asterisk CLI..--->http://pastebin.com/m124230
21:58.49Skeeter-hi, when i do a chanspy it active a channel, is there anyway to remove that channel because it stays active after even if i hangup
22:00.08beekGuys -- I'm having a heckuva time trying to get FUNC_ODBC to be kind to me and do a write.
22:00.17beekPertinent data:  http://www.pastebin.ca/1660101
22:01.14beekThe read works fine.  The write, not so much.   I can use isql from the * box and confirm that I am able to execute the write successfully, so it's something I'm doing vis-a-vi Asterisk.   Can anyone spot the error (and let me know)?
22:01.37hardwireKatty: I'm not from there.. but I work with a bunch of nice people from Oahu.
22:01.49hardwireNow.. if we were talking about Paris.. I'd be on yer boat.
22:01.57hardwirebbl.. gotta fix thingz
22:02.29*** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk)
22:07.02[TK]D-Fenderbeek: Set(${ODBC_ACCESSCONTROL(${CALLERID(num)})}=test)); <-- bad
22:07.11*** join/#asterisk twanny796 (n=twanny@85.232.220.197)
22:07.16[TK]D-Fenderbeek: You are referncing the function as a CALL, not a WRITE
22:07.24[TK]D-Fenderbeek: remove its outer ${}
22:08.01[TK]D-Fenderbeek: Set(${CALLERID(num)}=12345) <- bad
22:08.03[TK]D-Fenderbeek: Set(CALLERID(num)=12345) <- good
22:08.56beekDAMMIT DAMMIT DAMMIT -- [TK]D-Fender thanks!  You have no idea how much time I've wasted staring at that.
22:09.16TJNIIThat looks like one of those mistakes you can look right at and not see...
22:09.21Corydon76-digSo you're a time-waster?
22:09.32beekA huge one.
22:09.35beekFMTT
22:10.35Corydon76-digbut enough about your weight
22:10.55beekCorydon76-dig: we met at Astricon... it's accurate in any event.
22:10.57ruben23http://pastebin.com/m124230
22:11.28Corydon76-digAre you sure we met?
22:11.33twanny796tzafrir_laptop: http://www.pastebin.org/51517 <--- problem regarding compiling zaptel on 2.4.36 kernel
22:11.45beekThe func_odbc session.
22:11.55Corydon76-digWe might have walked past one another with no recognition
22:12.16twanny796tzafrir_laptop: this is when I did make install
22:12.46Corydon76-digYes, talking to the author of a module about his most famous module is going to distinguish yourself...
22:13.23Corydon76-digI talk to a LOT of people about func_odbc.
22:14.07beekI didn't expect that you'd recognize or remember me, and I wouldn't have mentioned it except for the weight comment.
22:14.37*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)
22:15.44beekNext time I'll either punch you or moon you -- either should more readily "distinguish" me...   ;-)
22:15.52beek(the latter will no doubt scar you for life)
22:16.53Corydon76-digNah, I like big butts
22:17.03*** join/#asterisk riddlebox (n=user@159.251.13.3)
22:17.27Corydon76-digMy bf has a big bubble butt
22:17.47*** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net)
22:18.13*** join/#asterisk autojack (n=owen@nerdnetworks.org)
22:19.06Godfather_hi
22:20.32ruben23hi anyone knows this error log on my asterisk CLI..--->http://pastebin.com/m124230
22:20.39autojackI'm running Asterisk 1.4.19. I have a DID that routes into it over SIP, my PBX takes input from the caller of a phone number to dial, and then calls out using a VOIP termination provider, also over SIP. 90% of the time this works fine, but sometimes Asterisk shows a call as answered, but no audio comes through. there's no firewall or NAT on my system, and attempts to troubleshoot with my external provider have led nowhere. I have a snapshot of logs with
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22:24.21wcselbyruben23 - looks as though you can't negotiate a codec with whomever you're calling
22:33.46TSM2ruben23: is this external trunk SIP or internal Phones
22:35.47donnibhow does one debug chan ?
22:36.12donnibi tried turning on the debug and setting the core set debug 128 but still i don't get log from chan
22:36.29donnibanyone has an idea how to accomplish this ?
22:40.15*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
22:40.33ruben23TSM2: internal phones calling outbound...
22:41.50*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:42.08TSM2are they outbound SIP trunks, or PSTN
22:45.50ruben23TSM2: sip trunk outbound..
22:45.56ruben23voip
22:46.18Kobazdoes toneduration have any affect on pri channels?
22:48.57Kobazcoughts on [TK]D-Fender
22:49.25Kobaztyping skills have dwindled from being sick
22:54.27donnibso no one knows how to debug chan ?
22:55.14jayteeJackie Chan?
22:58.23*** part/#asterisk autojack (n=owen@nerdnetworks.org)
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23:10.01*** join/#asterisk rps2 (n=rick@adsl-99-74-144-118.dsl.lsan03.sbcglobal.net)
23:10.25rps2Hi, gang.
23:11.17*** join/#asterisk airtonarantes (n=airton@intra.aloo.com.br)
23:11.37rps2May I pose a quick question?  The answer may be painfully easy, but I don't know it.
23:12.42airtonarantesFolks, does anybody knows how to configure SIP-T trunk???
23:13.33rps2Some outgoing calls from Asterisk via a TDM400P seem to not pick up the first bit of the audio when the other end answers.  Any ideas why?
23:14.14rps2The "r" is not on any of the Dial() commands I see in the dialplan.
23:14.17*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
23:14.57airtonarantesrps2, seems to be silence supression setup.
23:17.22*** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
23:21.49*** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com)
23:22.49[8none1]Carlos_PHX: I think we fixed our call quality problems we talked about a week or two ago
23:23.05[8none1]We finally found that we were having overruns on the PRI
23:23.21[8none1]Sangoma helped us fix it
23:28.49*** join/#asterisk Caplain (i=shayne@2001:470:5:fb:b433:246f:9ea:a023)
23:29.25Kattyello.
23:29.47ryduhallo
23:29.54*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
23:30.48Kattyhad a jog
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