IRC log for #asterisk on 20081022

00:00.09drmessanoI'm using 3.1.16
00:02.38antI'm really not able to formulate a clear qestion at all today. :0(
00:04.16*** join/#asterisk Iron_Forge (i=d8088a9a@gateway/web/ajax/mibbit.com/x-630dcc290f254492)
00:05.21Iron_ForgeI have my asterisk box setup with two trunks, one to an outside PBX for external calls, and a second to an internal sip pbx. The internal sip pbx routes 9|
00:05.37Iron_ForgeI have my asterisk box setup with two trunks, one to an outside PBX for external calls, and a second to an internal sip pbx. The internal sip pbx routes 9|. calls to the asterisk box, how do I get it to re-route these calls out to the external pbx?
00:06.56*** join/#asterisk Corydon76-dig (i=two@pdpc/supporter/bronze/Corydon76-home)
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00:08.18gramulhaozindrmessano: found the problem
00:08.22gramulhaozindrmessano: wrt54gx4 somehow interfering,
00:08.36gramulhaozinthat WRT54GX4 is the buggiest router I have ever seen
00:09.02sbc383Iron_Forge: Have you setup your sip.conf file?
00:09.08lunaphytewhy would a sip endpoint would be able to send media through asterisk to an iax provider even if the ports i've specified in rtp.conf are blocked by iptables?
00:13.20*** join/#asterisk techman97 (n=myweiner@97-91-103-181.dhcp.roch.mn.charter.com)
00:13.47Iron_Forgesbc383: I'm using the GUI. New to this.
00:14.33techman97hello all, I have a Sangoma A102d card in my * box and it *appears* that I'm not getting any callerID from my provider.  What settings can I check to make sure or test if I just have something wonky with my config?
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00:20.38limaunionhi, newbie here, i'm trying asterisknow 1.5 (beta) and for some reason it seems that asterisk is not listening on port 5060 udp (sip), if i type netstat -tunap | grep 5060 i don't get any answer, any ideas ?
00:23.13jayteetechman97, have you tried putting callerid=asreceived in your zapata.conf
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00:23.58techman97jaytee:  Yes, it is there
00:24.23jayteeis this setup as PRI?
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00:25.31techman97jaytee:  PRI, correct.
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00:26.48techman97hmmm
00:27.14techman97either my provider isn't sending it, or my box can't recognize it correctly....I don't see anything in the CLI when a call comes in
00:28.09jayteehave you tried putting something like exten => s,1,NoOp(${CALLERID(all)} in the incoming context in extensions.conf that you defined in zapata.conf and making an inbound test call?
00:28.32jayteeand see what it outputs on the console?
00:31.34drmessanoYeah, so I had a number from stanaphone
00:31.45drmessanoand I kept getting VM e-mails from it..
00:31.50drmessanoBut lost the info long ago
00:32.13drmessanoand are not taking new residential customers
00:32.18drmessanoThey finally added some password retrieval thing to their site
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00:32.21techman97sorry jaytee - on the phone with the provider.  In the CLI, I'm not getting anything on my debug.
00:32.24drmessanoSo I got my pass reset
00:32.27drmessanoand got my number back
00:36.15jayteetechman, sounds like it's your provider not passing the info, you might also have the wrong switchtype= defined.
00:37.36techman97I'm in the US, and setup as switchtype=national.  Calls are going in and out fine, I think it's the provider too, but I'm tackling it from all directions.
00:38.11jayteetechman97, ok that sounds good, national is the most common configuration setting
00:38.21jayteefor PRI in the US
00:38.46techman97I put a NoOp literally at the first line of my dialplan to write out the CallerID, I got none.
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00:46.26jeevjaytee, stop messaging me your pics
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00:47.25jayteejeev, you're hallucinating again. put the crack pipe down and go see a doctor.
00:47.40*** join/#asterisk radiofree (n=radiofre@75.150.198.1)
00:49.11jeevyou're wack
00:49.43radiofree\join #asterisk-dev
00:49.55radiofreeI win
00:50.13radiofree?/join #asterisk-dev
00:50.18jeevwhat do you win
00:50.18jeev?
00:50.27radiofreeEverything
00:50.28radiofreeAnyway
00:51.09radiofreeI'm trying to write this thing with the .NET asterisk api that tracks incoming calls as they get transferred to different agents
00:51.42radiofreeAnd I started trying to use the uniqueId thing, but it seems useless
00:52.08radiofreeSo I guess I'm wondering, what can the uniqueId be used for if anything?
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00:52.59jayteeradiofree, it's used as a unique index field in the CDR table
00:54.11radiofreeThanks.  How frequently does it change?
00:54.49WimpManradiofree: What version?
00:54.58radiofree1.4
00:55.34WimpManhas used AMI to track calls as well, and did not have any trouble identifying calls.
00:55.38jayteeevery call hs a different uniqueid and I believe that includes transfers and each leg of a multi-leg call like with queues
00:56.22WimpManNothing you won't be informed of, as far as I remember.
00:56.44WimpManOtherwise it couldn't possibly have worked.
00:57.08WimpManBut I did have a hard time trying to find out where the call came from and wehere it went.
00:57.22riddleboxcan someone help me out here I have an extension built 701, and I am trying to connect to my asterisk server over the internet, http://pastebin.ca/1233302
00:57.34radiofreeSo I probably couldn't use that to track a call that gets transferred?
00:59.22WimpManYou can.
00:59.50radiofreeHow?
01:00.03WimpManWatch the messages.
01:00.10WimpManIf will inform you about all changes.
01:01.15radiofreeOkay, I'm really new to Asterisk and I really don't know squat beyond the Asterisk.NET API
01:01.52lunaphytei'm messing around with asterisk and x-lite at home.  what's the difference between "user name" and "authorization user name"?
01:01.55radiofreeAnd the events I was looking at kept ch
01:02.01radiofreeWait, nevermind
01:02.34WimpManJust connect to AMI, make a test call and look at what it sent to you.
01:02.35radiofreeSo if I was looking at the NewState event, what is it giving me the uniqueId of?
01:05.43Iron_ForgeWhen trying to call an outside # from sip1.server, which has a trunk to my asterisk box (sip2.server), I get error Channel 'SIP/sip1.server' sent into invalid extension '9outside#' in context 'from-trunk', but no invalid handler. sip2.server has an outbound route to sip3.server, which works if calling from a sip device on sip2.server, but not on sip1.server. What am I missing?
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01:09.41hardwireanybody know of a way to trigger a voicemail email?
01:10.00carrarleave a voicemail?
01:10.11hardwiresay if I move msg0009.* from 0010/INBOX to 0012/INBOX then want asterisk to handle the email notification through app_voicemail
01:11.31jeevis forcing girl cousin to learn PHP so she can program for me.
01:12.48lunaphytei suppose there's worse things you could be forcing your girl cousin to do...
01:16.22CrashSysYou in Kentucky, Carolina's, or Tennessee?
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01:51.41Deeewayneouch!
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02:13.38gene2Can anyone give me a clue why my initial voicemail prompt sounds like it is too loud on asterisk-1.6.0.1? Like garbled
02:13.57[TK]D-Fendergene2: your own recording?
02:14.01gene2no sir
02:14.29gene2Fender: the standard one that says immediate mail (sp?)
02:14.51*** join/#asterisk boch (n=fran@200.68.119.237)
02:14.57bochhi all
02:15.12[TK]D-Fendergene2: I thought they'd have fixed this, but in older versions there was a GCC compiler otimization issue that would cause GSM playback to suck.
02:15.15[TK]D-Fender~gccbug
02:15.21[TK]D-Fender~gsmbug
02:15.21jbot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily.  Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243
02:15.25[TK]D-FenderThere it is
02:15.55[TK]D-Fendergene2: What are you listening to it on?
02:16.00bochanyone know if AgentComplete manager event includes calls completed by the caller ?
02:16.07gene2yep I got 4.2 gcc
02:16.22[TK]D-Fenderboch: Shouldn't.  Thats what "completecaller" is for
02:16.23gene2I'm listening this one a Cisco 7960
02:17.16[TK]D-Fendergene2: Quick way to prove it : download the sound pack in the codec useds on your phone and drop them in.  The native codec version should get chosen.  If it sounds fine (which is all but assured) then we've found the culptrit
02:17.22boch[TK]D-Fender, there is no completecaller in the voip-info wiki
02:17.39[TK]D-Fenderboch: And the wiki is never ever wrong or incomplete.
02:18.22bochis there an official doc for asterisk?
02:18.31bochand updated
02:18.31gene2thanks I will try
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02:31.13[TK]D-Fenderboch: changelogs, the /doc folder, the book is a good place for reference as well
02:31.23[TK]D-Fenderboch: basically the WIKi is like #5 in line :)
02:31.43[TK]D-Fenderboch: here is #6 when all the common written places have failed :)
02:32.05gene2Fender: I have a feeling that patch had been applied already to 1.6.0.1
02:34.39gene2Fender: yep this patch is in: http://bugs.digium.com/file_download.php?file_id=18245&type=bug
02:36.03[TK]D-Fendergene2: its about the only suspect right now.  try the native codec recording... its just a question of isolating if its the transcoding or not
02:37.39gene2Fender: I got to look into this, I'm not all that good with Asterisks so I will need to read up, I've just recently upgraded from a version that was at least 5 years old
02:38.21WimpManWow!
02:38.45WimpMan5 years?
02:39.10gene2It would be hard for me to put a date on it exactly but I think 5 years is a right now, at that table debian sarge was still unstable
02:39.38gene2Maybe 4, who knows. But it was antique..
02:39.39[TK]D-FenderDebian developers have always been unstable ;)
02:39.53gene2=) That is why I'm running ubuntu
02:39.58gene2Everything else works very good.
02:40.19gene2I have about 10 extensions all over, most are in the USA but I got one in London and sound quality is amazing
02:40.43gene2I'm actually running this in a VPS which is based on xen.
02:42.04*** join/#asterisk comfrey (n=comfrey@dsl-66-78-76-161.ipns.com)
02:42.07comfreyhey all...
02:42.21comfreyi am trying to compile sangoma drivers and i get the following error
02:42.24comfreyUnsuppored architecture type x86!
02:42.43comfreyi am using the command:./Setup install --64bit_4GB
02:42.50comfreyanyone have any ideas?
02:43.32orkidis there a way for asterisk to send a hookflash to a linksys spa-3102 for it to send out the fxo port?
02:43.33WimpManI haven't got a clue about Sangoma drivers (yet), but I'd try something like ARCH=i386 ./Setup...
02:43.47[TK]D-Fenderorkid: I answered this 3 times yesterday
02:44.13orkidu did?
02:44.32[TK]D-Fenderorkid: Yes
02:44.52[TK]D-Fenderorkid: There is a special SIP packet you have to send to it
02:44.57[TK]D-Fenderorkid: Google it up
02:46.24orkidyes, but ive also read on forums that its not possible: http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3102-hook-flash-18570.html
02:46.28orkidhogwash?
02:47.07orkidalso reading that 3102 is a major headache, as opposed to say tdm400p
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02:58.13jayteehahaha, Pug Bowling http://www.youtube.com/watch?v=dmUaL3Fp85Y
03:04.46drmessanorofl
03:05.26drmessanoSome people just hear what they want to hear
03:05.46[TK]D-Fenderorkid: You have 1 issue.  Feel free when you think you've found enough for it to deserve being called that.
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03:13.08comfreyso... i just hardcoded $ARCH
03:13.11comfreylooks ok
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03:22.00comfreyok, so... i am getting a horrible buzzing on a sangoma a101 on inbound
03:22.09comfreyany ideas about why?
03:23.37hardwirehai
03:26.07orkidfeel free to what....
03:26.59lmadsenyou know
03:45.15*** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com)
03:45.44rue_mohrhow come I cant get rid of the comedian mail message headers ?
03:45.56[TK]D-Fenderrue_mohr: "man rm"
03:46.16rue_mohrhu?
03:46.34rue_mohrfor voicemail?
03:46.51comfreyso ... i am getting Line Code Violations...
03:46.54[TK]D-Fenderrue_mohr: Perhaps you should be more specific.  A lot more.
03:47.05comfreyon my sangoma card.
03:47.09rue_mohrI just want it to get on with the message instead of spending 2 mins telling me about it first
03:47.29comfreyanyone have idea about what that is caused by?
03:48.02[TK]D-Fenderrue_mohr: envelope=no
03:48.09[TK]D-Fenderrue_mohr: Go read the sample configs again
03:48.14rue_mohrk
03:48.28WimpMancomfrey: More input please. Constantly or transient? What signalling? What errors?
03:48.34rue_mohrI recall trying to edit them to trim out most of the date, but didn't work
03:48.38rue_mohrI go look again
03:48.52comfreyWimpMan: so there is loud buzzing on incoming calls
03:49.15comfreyand the only thing i have seen in errors is Line Code Violations
03:49.26comfreyi think they incriment on incoming calls
03:49.44rue_mohrT1?
03:50.04WimpManWrong configuration or broken cable.
03:50.10comfreyT1 yes
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03:50.33rue_mohrare you sure you have the card configured for the right kind of signaling?
03:50.36comfreyok, i should rerun wancfg or whatever
03:50.41comfreyi will reconfig
03:50.44comfreyand see
03:50.47rue_mohrwell
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03:51.08rue_mohrno, does the config of your T1 service match your config file
03:51.21rue_mohrmine is zapata.conf
03:51.32comfreysame here... i will look into it
03:51.43rue_mohrsignalling = fxs_ks
03:51.46rue_mohrthose things
03:52.10comfreyfxs_ls
03:52.18comfreyyes i believe it to be correct
03:52.22rue_mohrI would dump the relivent lines of my config, but as much as [TK]D-Fender lines me I'm sure he would ban me :)
03:52.28rue_mohrloopstart
03:52.37comfreyyup
03:52.54rue_mohrloopstart is for telephones, but central office
03:53.08rue_mohrunless that backwards cause of my setup...
03:53.14rue_mohrbut I dont think so
03:53.23rue_mohr:)
03:53.40comfreyshould i try other configs... could that damage the card?
03:53.59rue_mohrI dont think so, properly built hardware cant be damaged by software
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03:54.19comfreyright... ok well i will poke at it with other settings
03:54.40rue_mohrboiler, home automation, drat, where DID the terminal to my phone system go?
03:55.08rue_mohrcomfrey, does your incommng service from co come in on the T1?
03:55.17rue_mohror is the T1 to a channelbank of telephones?
03:56.50comfreyservice comes in on t1
03:56.55comfrey22 channels
03:57.09rue_mohrI'm no eignstien, but I think you want fxo
03:57.35comfreyok... i'll try that
03:57.58rue_mohr[TK]D-Fender, hmm I shall switch it off then :)
03:59.46rue_mohrwhat do i have to restart for voicemail.conf changes to take effect?
03:59.52rue_mohrrestart now?
04:00.02rue_mohrer reload now?
04:00.06[TK]D-Fenderrue_mohr: "module reload app_voicemail.so"
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04:02.24rue_mohrthanks
04:03.48rue_mohrNo such command 'module' (type 'help' for help)
04:04.20rue_mohr<PROTECTED>
04:04.36rue_mohryou getting good sleep latley?
04:04.45rue_mohror just keeing me on my toes?
04:05.59WimpManrue_mohr: You must be using a pretty dated version then.
04:06.07rue_mohr:\
04:08.06jameswf-homeping
04:08.19comfreyso... it does nothing if i don't use fxs_ls
04:08.24comfreyand just buzzing
04:08.43comfreyi think it may be a codec or timing thing
04:08.57comfreyi am going to ge some caffeine
04:09.08[TK]D-Fenderrue_mohr: No, you're just not keeping up
04:09.17comfreyand food
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04:10.24rue_mohr[TK]D-Fender, I'm afraid to play with things that wor
04:10.43rue_mohrand I was told to be weary of zapata drivers
04:10.57rue_mohrmumbles kb1kanobe
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04:12.53[TK]D-Fenderrue_mohr: procrastination : the art of keeping up with yesterday
04:13.11coluludoes anyone know how to config chan_ss7 for sangoma card?
04:13.13rue_mohrnote my lack of bleeding edge scars
04:13.21rue_mohroh it worked thanks :)
04:13.40rue_mohrcolulu, I only know zapata cards
04:13.47rue_mohronly the one I have
04:14.06colulurue_mohr: have you tried loopback on one 4-port zapata card?
04:14.20rue_mohronly with help from my guru
04:14.33colulucould you help me?
04:14.45coluluI have problem setting up loopback on one 4 ports zapata cards
04:15.35coluluand I tried to get two sangoma cards working together with chan_ss7
04:15.38colulubut no luck
04:15.49coluluSangoma cards don't work well with chan_ss7, it seems
04:16.42rue_mohrsomeone here can help you, but their not watching right now
04:17.07rue_mohrstick around, reask about every 2 hours
04:17.42jameswf-homesangoma...cause it just must work...... bwahahaha
04:18.12jameswf-homesorry
04:20.13outtoluncunless it doesn't fit <G>
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04:33.39[T]ankwhen a call comes into asterisk and rings the phone, only the name is displayed. then for the number, it shows the extension of the phone that is ringing. here are the pertinent details of the config file. http://pastebin.ca/1233495
04:33.49[T]ankhow can I correct that?
04:34.57[T]anki want it to show name and number of the caller, instead of name of the caller and extension of the phone i am answering on
04:35.02[TK]D-Fender[T]ank: pb your [200] peer
04:35.06[T]ankk
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04:36.45[T]ankhttp://pastebin.ca/1233497
04:39.24[TK]D-Fender[T]ank: don't do "fromuser"
04:39.44[T]ankif I take that out, I get authentication errors
04:39.56[T]ankgetting details....
04:40.05[TK]D-Fender[T]ank: Do it, and remove "insecure" while you're at it
04:40.56[T]ankok... can I pm you this error so that it is not public? too much to delete and still make valuable
04:41.07phpboypastebin
04:41.26drmessanoThe SPA-942 is a router too?
04:41.53[T]ankoh, wait... fromuser does not need to be on the extension side... just the peers connecting to the sip provider, right?
04:42.23[TK]D-Fenderdrmessano: No.
04:43.24*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fc3adf8ae3f7b793)
04:43.28drmessanoIt seemed overwhelmingly silly.  I think someone just took a generic Linksys V5.2 firmware config and used it for the SPA-942.. They have the PPPoE settings in there which you would expect for a SPA-x102
04:43.33jameswf-homecant do pastebin.ca for some reason...
04:43.54hardwiresomebody give me your GPG key ID
04:43.57hardwireis having fun
04:44.06drmessanoWho's?
04:44.19hardwireyou got one?
04:44.33hardwireare you daniel messano?
04:44.34drmessanoYeah
04:44.52WimpManHmm. Why not? That dumb Allnet phone does pppoe.
04:45.03hardwirehttp://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1BED59B1
04:45.05hardwireerr
04:45.09hardwirepub  1024D/1BED59B1 1999/07/22 Daniel R Messano <drmessano@yahoo.com>
04:45.30drmessanoYeah, I need to clean that key up too
04:45.53hardwiredammit
04:45.56hardwireyou're not cool enough
04:46.03drmessanoHow so?
04:46.25drmessanoLow man on the WoT?
04:46.30hardwiretotally
04:46.50hardwireyour wife/sister/mom doesn't count!
04:46.51hardwirehaha
04:46.59drmessanoYeah.. I tried to get friends to use PGP years ago, never could get them to really use it
04:47.00drmessanolol
04:47.15hardwireactually.. you have a ton of sigs
04:47.21hardwirebut I guess your signers don't have lots of sigs
04:47.25hardwireI can't "link" to you
04:47.33Qwellhardwire: how many degrees of separation do I have from Kevin Bacon?
04:47.33phpboy[TK]D-Fender: I wrote a VERY nice chan_agent replacement, where do you think I should put it for other people to make use of it?
04:47.34hardwirehttp://pgp.cs.uu.nl/mk_path.cgi?FROM=1CA9ADE2&TO=46C171A0&PATHS=trust+paths
04:47.41hardwireQwell: does kevin bacon have a key?
04:47.42phpboy(I cannot find a replica anywhere :()
04:47.46Qwellmaybe?
04:48.07hardwireheh
04:48.27hardwiredamn.. kevin bacon has no signers
04:48.32Qwellhttp://pgp.mit.edu:11371/pks/lookup?op=get&search=0xB80EA480
04:48.39drmessanoI had to make PRZ take his sig off my key, it was drawing too much attention to me
04:49.02hardwirehttp://pgp.cs.uu.nl/mk_path.cgi?FROM=1CA9ADE2&TO=D730A262&PATHS=trust+paths
04:49.06hardwirewoot
04:49.16Qwellbah!  file!
04:49.47drmessano2006?
04:49.51drmessanon00b
04:49.58hardwireQwell: wots yers?
04:50.11Qwelleh?
04:50.14drmessanolol
04:51.12QwellKarsten has WAY too many signages
04:51.16comfreyso... still stuck with this buzzing on inboud over the t1
04:51.40hardwirecomfrey: what kind of handset?
04:51.53comfreythe logs say that it is playing messages
04:52.01drmessanoIm not even sure if I have GPG on this machine
04:52.03comfreyi am calling in from a cell to a t1
04:52.10hardwirecomfrey: who's your PRI provider?
04:52.19comfreysangoma fxs_ls
04:52.23comfreyit is integra telecom
04:52.33hardwirehave you had issues with them before?
04:52.42comfreyi had it working briefly before....
04:52.52drmessanoI started off with 2.6.3i, then 5.0, 6.5.x.. I even own a box copy of McAfee PGP 6.5.2 lol
04:52.55hardwirenotes, just because your calls hit PRI to a telco.. doesn't mean they aren't using shortwave analog trunks to istanbul
04:53.00comfreyi am migrating to asterisk from some nec box
04:53.05hardwirethe nec was fine?
04:53.12comfreyyeah.
04:53.22comfreystill is if i plug it back in
04:53.24hardwireusing sangoma fxs for standard telephones?
04:53.31hardwireor for old NEC telephones?
04:53.40comfreyuh, no digital old things
04:53.45comfreyyes
04:53.53hardwiretried a regular phone?
04:53.56comfreyso... this loud buzzing
04:54.09comfreyno.. as i have no access now
04:54.20hardwirecomfrey: the buzzing is probably due to the handset you are using
04:54.33comfreyi am onsite... but... as i said it was working on pbxina flash
04:54.37comfreyfrom my cell
04:54.45hardwirewhat is pbx in a flash?
04:54.47comfreyi am just trying to get the setup on debian
04:54.56comfreysome just add water solution
04:55.02comfreyi used to test
04:55.04hardwirelinux/asterisk ?
04:55.14comfreycentos and asterisk
04:55.16comfreyyes
04:55.19hardwireweird
04:55.29hardwirehow is the T1 provisioned?
04:55.30comfreyi could try the 32 bit system i have here...
04:55.40comfreycoding:B8ZS, framing:ESF.
04:55.48comfrey<PROTECTED>
04:55.55hardwire23B and 1D channel?
04:55.56comfrey2. Nortel DMS100 *
04:56.07comfrey22 channels
04:56.09comfreyb
04:56.14comfreynot pri
04:56.16comfreyjust t1
04:56.22hardwirewhat signalling?
04:56.25hardwirerbs/e&m?
04:56.40hardwireif it's fxs I'll freak out
04:56.51comfreyfxs_ls
04:56.56comfrey:)
04:56.57hardwirefreaks out
04:57.05hardwireso the NEC PBX is still installed eh?
04:57.06comfreywhat do you know?
04:57.11drmessanoI would like to set up Asterisk to do a GPG lookup on every SIP packet I send between 2 endpoints
04:57.13comfreyyes
04:57.14hardwireyou just want to extend it with an asterisk box using a tie?
04:57.24drmessanoNEC Electra Elite?
04:57.42comfreyyeah, electra something
04:57.43hardwireI'm sure you heard this already.. but tieing to an NEC is a huge pain in the but.
04:57.48drmessanoOMG you poor bastard
04:57.48hardwirewith an extra t
04:57.52comfreya tie?
04:58.00comfreyi don't understand
04:58.09comfreymy sangoma can't talk?
04:58.13hardwireit can
04:58.14drmessanoI quit a job over an NEC
04:58.17hardwireyou already proved it.
04:58.24comfreyheh
04:58.26hardwireI'm letting drmessano take it from here.
04:58.40comfreywhas the deal drmessano ?
04:58.42hardwireI have to go to bed now.. everybody is feeling sick now that you said NEC
04:58.48comfreyheh...
04:58.53hardwirepoor bastard..
04:58.55comfreyuh...
04:58.58comfreywtf?
04:59.03hardwirenight
04:59.13comfreydrmessano: can you explain?
04:59.40comfreyi think the T1 can talk... should i order another t1?
04:59.49comfreydo it all sip?
05:00.02drmessanoNEC PBX's are horrid, horrib pieces of overpriced, overly reliable shit.  NEC has a flair for making PBXs that are low in feature, with highly priced addons, that seem to last WAY TOO LONG, so stupid ass bald headed managers can't justify replacing them
05:00.24comfreywell we are replacing it...
05:00.25drmessanoThe best thing you can do to interface Asterisk to an NEC is go into work after hours and set the NEC on fucking fire
05:00.40comfreyi am not interfacing the nec.
05:00.53comfreyi am plugging the t1 into my asterisk box
05:01.02comfreyis the signaling a problem?
05:01.11comfreyi am not getting something.
05:01.14drmessanoThe T1 from the NEC?
05:01.25comfreyno... the t1 from the adtran
05:01.29comfreyi am sorry
05:01.35comfreyi did not explain that part
05:01.43comfreythe t1 is going bye bye
05:01.50comfreyi mean the nec
05:01.52drmessanoI'm confused.. I just want to get to the part where we set the NEC on fire
05:01.52comfreysorry
05:02.00comfreyyes ... me too :)
05:02.12comfreybut i need a working phone system first
05:02.23comfreyso adtran -> asterisk
05:02.35comfreywas adtran -> nec
05:02.59comfreysome thing with adtran -> asterisk  =  noise on line
05:03.05comfreyloud buzzing
05:03.18drmessanoI've not done any work with an adtran or T1s, but I can assure you that if I had, I would stay up all night helping you do anything possible to make the BBQ of the NEC go much faster..
05:03.19comfreyis it zap, sangoma, asterisk?
05:03.26drmessanoMaybe you have a ground loop?
05:03.38comfreyi can try that.
05:03.57drmessanobuzzing sounds like ground loop
05:04.12comfreyok, that sounds interesting
05:05.19comfreyreconfiguring wancfg_zaptel
05:05.42drmessanoA good asset for a tech's toolbag, and I say this cautiously, is a standard power cable with the 3rd prong cut off
05:05.56drmessanoHelps to isolate group problems
05:06.00drmessanoground
05:06.43drmessanoSome dickheads will leave the equipment plugged in that way, rather than fixing the problem..
05:07.42comfreywhy is wancfg_zaptel not asking about the switchtype?
05:08.36drmessanoDunno
05:09.20comfreythat is important no?
05:09.49comfreystill buzzing with ground start
05:10.02comfreyi'll try fxo
05:10.03*** join/#asterisk ming_zym (n=ming_zym@202.165.107.100)
05:10.04comfreyi guess
05:10.09comfreyls and gs
05:11.19comfreyhere goes fxo ls
05:11.30comfreyand...  nothing...
05:11.42comfreygoes to upstream autoattendant
05:12.11comfreyi wonder about the switchtype thing
05:12.15comfreyis this important?
05:16.46*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d58e183ace186925)
05:20.57hardwirecomfrey: analog over digital circuits makes me insane.
05:21.42hardwirebut if that's how the nec is set up.. then I guess you need it to stay the same for now
05:21.53comfreyi am heading that direction
05:21.56hardwireI thought you were hooking the asterisk box up to the t1 at first.. then to the nec.. now we are setting things on fire.
05:22.19hardwirecomfrey: how did the pbx in a flash system set up the zaptel?
05:22.22hardwireapparently that worked.
05:22.37comfreyyeah, i might try asterisk now
05:22.42comfrey1.5b
05:22.42hardwirescribble it down after backing up zaptel.conf and /etc/asterisk/zapata.conf
05:22.43drmessanoI wasnt talking about Ground Start
05:22.54comfreyoh... i see
05:22.58comfreya ground issue
05:23.15drmessanohttp://en.wikipedia.org/wiki/Ground_loop_(electricity)
05:23.24comfreyyeah
05:23.26hardwirecomfrey: is the system you are using the same system you set up pbx in a flash on?
05:23.30comfreyyes
05:23.50hardwirewhat changed between when pbx in a flash was installed.. and debian was installed. ?
05:23.50comfreyi was thinking codecs or something
05:24.02comfreyuh... dependancies? or...
05:24.03hardwirethere are no codecs on PRI like you would find in a voip world
05:24.28comfreythen... dunno.
05:24.28hardwirecomfrey: but hardware wise.. everything is okey dokey?
05:24.31WimpManThat's not exactely true.
05:24.33drmessanoWait, you had this all working with PBX In a Flash?
05:24.38hardwiredrmessano: yeh.
05:24.39comfreyyep
05:24.51drmessanoand you switched, why?
05:24.52hardwireWimpMan: the codecs part?
05:25.02WimpManyes
05:25.03comfreyit was a bit too busy and i am a debian loyalist
05:25.11drmessanoOh fuck me
05:25.13hardwire99.99% of the world uses PCM over PRI when it's full digital :)
05:25.17drmessanoI mean
05:25.21drmessano"thats nice :)"
05:25.24comfreysure...
05:25.32comfreyi want to know how this shit works too
05:25.42comfreyi may go back
05:25.50hardwireit probably set up your zap correctly
05:26.02WimpManhardwire: There is at least the question of G.711 a-law or G.711 µ-law. And then there's also the broadband option.
05:26.10drmessanoFiguring out how it works on a working system is a lot easier than the pile of monkey poo you have now
05:26.16hardwireWimpMan: I'm sorry
05:26.22hardwireforget I tried to confuse somebody
05:26.22drmessanoActually, let me take that back
05:26.25comfreywell i had to run wancfg_zaptel on pdx in a flash too
05:26.40comfreyand i told it the same thing as the deb box
05:26.44orkidcool, i have Line1 on the Sipura connecting to *, and also a softphone :)
05:26.56hardwireWimpMan: thankfully.. both ends of the PRI don't have to "negotiate" what to use.
05:27.08comfreywell this is just t1
05:27.10comfreyno pri
05:27.14drmessanoO.o
05:27.15hardwirecomfrey: so no codec.
05:27.53hardwireorkid: woot.
05:28.15WimpManhardwire: Actually they might have to. That's one ose of LLCs.
05:28.20hardwirecomfrey: are you sure its fxs?
05:28.40hardwireand not rbs? e&m?
05:28.43WimpMans/ose/use/
05:28.46hardwireNEC loves it some E&M
05:29.00hardwirejbot: uh.. thanks tiger.. I can grok RE's
05:29.39comfreypretty sure fxs_ls
05:29.50comfreythat is what worked on pbx ia f
05:30.00hardwirecomfrey: south america?
05:30.06comfreynorth america
05:30.10hardwirewhereaboot?
05:30.14comfreyoregon
05:30.17comfreypr0tland
05:30.19drmessanoOhhhhhhhhhhhhhhh
05:30.24drmessanoThat explains it
05:30.25*** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d58e183ace186925)
05:30.31hardwiredrmessano: too much honey beer?
05:30.32comfreyoh, it does?
05:30.43comfreydead guy ale tonight
05:30.45hardwirecomfrey: send me a box of spire cider.
05:30.48drmessanoOregon is like a big time machine.. You step over the border and it's like going back to the 1880s
05:30.50comfreyi haven't opened it
05:30.54comfreyye
05:30.55comfreyt
05:30.56hardwiredrmessano: lol
05:30.57drmessanoThey still have Party Lines
05:31.08hardwiregoes to bed for reals.
05:31.14comfreythanks hardwire
05:31.21drmessanoWhich part of Oregon?
05:31.27comfreymy wagon is like stuck in the mud out front
05:31.40comfreya bit south of portland
05:31.47comfreynot too fat
05:31.49comfreyfar
05:31.53comfreya 30 min bike ride
05:32.27drmessanoSo you're not down with the wackos in "Jefferson"
05:32.38comfreyi didn't say that
05:32.49comfreyi believe in local autonomy
05:33.15comfreythis whole federal business has been out of hand for some time
05:33.41comfreybut that might be off topic
05:33.47*** join/#asterisk bmg505 (n=leon@196-209-79-7-tbnb-esr-2.dynamic.isadsl.co.za)
05:33.50drmessanoThe local autonomy movement in Oregon is just a thinly veiled plow to allow them to overthrow the drug laws of the federal govt at a local level so everyone can grow and smoke weed.
05:33.55drmessanoploy
05:34.03comfreyso
05:34.04comfrey?
05:34.06comfreylol
05:34.09drmessanoI am on to you people
05:34.13drmessanoO.o
05:34.26comfreyur just jealous
05:34.40drmessanoAt least be honest about it, Oregonians..
05:34.44drmessanoThats all I ask
05:34.55drmessanoIf you wanna secede to smoke more weed, say so
05:35.12comfreyit is much more complicated eh?
05:35.42drmessanoI think it's less complicated than it's being made out to be.. It's about smoking weed.
05:36.01comfreywell at least i am not trying trixbox anymore
05:36.07comfreythat is some progress
05:36.08drmessanolol
05:36.16drmessanotrixbox is like smoking crack
05:36.19comfreyi read some of the forums
05:36.24comfreygod dang
05:36.34comfreysome crazy people in the world
05:36.43drmessanoTalking about the fanboys?
05:36.43comfreyand some of them run software companies
05:37.07comfreyillusory fanboys
05:37.14comfreyphantom fanboys maybe
05:37.22comfreytalk about weed smoking
05:37.34comfreyi haven't seen anything like that
05:37.37drmessanoFonality has always been about smoke and mirrors
05:37.45comfreysad sad story
05:38.00jameswf-homesmoking weed and lookin in the mirror?
05:38.27drmessanoWhen you can't make a decent product, you BS your way through with the cow chips you are given to pawn off on people
05:39.06drmessanoUnder the hood, trixbox is as unstable as my ex wife.. on the surface, it looks like Angelina Jolie.. with the flu
05:39.11drmessanoPretty, but Green
05:39.17*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-86d47a11e39673a2)
05:39.20jameswf-homedrmessano: your magnet failed... tim was forgotten
05:39.41phpboydrmessano: I've had nothing but good expereiences with trixbox on smaller installations
05:39.42drmessanojameswf-home: Didn't they lock it down?
05:39.49drmessanolol
05:39.56drmessanotrixbox is absolute crap
05:40.20jameswf-homeits all crap you just need the right toilet paper
05:40.30jameswf-homered corn white corn
05:41.22drmessanoThe repos are unstable, the RPMs are crap, they don't understand versioning and naming, they spy on the users with perl scripts, they lie about security holes, strip features that are broken instead of fixing them, and when they fuck something up and it breaks YOUR system, guess who is at fault?  YOU.. for running an update, EVER
05:41.43comfreyyup that's what it looks like to me too
05:43.07drmessanoThey forked FreePBX because they got tired of being browbeat over using it and not giving back.. so they forked and and rebranded.. Then did nothing with it.  They could care less about the open source projects they use to make up their suite, and if you say "Well, thats the beauty of open source, you can use what you want", not only do they use and not give back, but they thumb their nose at
05:43.15drmessanothe implication it's even a good idea
05:43.22comfreywell... here goes asterisk now 1.5b
05:43.56comfreyat least they are obcessed with enforcing the gpl...
05:43.58drmessanoThey are a very evil corporate entity who brought a bowling ball to a T-Ball game, and eventually, as we are now seeing, their time will be/is coming.
05:44.00comfreyor whatever
05:44.50drmessanoROFL
05:44.50drmessanoGPLLaw?
05:44.50comfreysad
05:44.50drmessanoKerrys fake handle he trolls forums with, then blames some buddy that came to his house, got drunk, and used his wifi?
05:44.50drmessanoThat was all too good
05:45.03comfreynot the sharpest knife...
05:45.30comfreyi've seen some trolls
05:45.36comfreybut that shit...
05:45.39comfreydag
05:46.38drmessanoKerry got thrown into too many lockers in high school and still isn't taking it well
05:46.48orkidi'm afraid to ask ... but here it goes ... my spa-3102 Line1 is registered with *, and so is a softphone. I call the softphone extension form the phone attached to the SPA-3102 FXS port (ie. Line1) and answer with the softphone. I can hear both ends/etc, so voice works... When I quickly put the phone on-hook and then off-hook again (ie. flash hook), I get an Unable to parse INFO message from hex_stuff@ip_of_spa-3102. What is the problem? I set CW Settin
05:46.56orkidwhat is the problem?
05:46.59orkidcan anyone help?
05:47.05orkidcovers his head
05:47.30jameswf-homeI use 2048 bit encryption on my WPA32 PRO wifi so the haxors cant use my ip to troll
05:47.58comfreyjameswf-home: or your drunk buddies
05:48.09drmessanojameswf-home: GPLLaw uses 128 bit budweiser
05:48.30comfreyhe needs some of the good weed
05:48.47jameswf-homeDOH forgot about the drunk buddies... f me
05:48.48orkidbefore i disabled the CW and three way stuff, a flash hook gave me a second dialtone (i'm guessing it was intercepted by the spa-3102 and not sent out to *)
05:49.52orkidhmm, seems like i get the hookflash recognized by * now.
05:50.17orkidi had to re-enable cw/three way, but keep hook flash tx method:info
05:50.54*** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com)
05:51.40orkidsorry for the clutter
05:52.48orkidbtw, asterisk is real cool. i'm already pretty amazed at the whole thing.
05:54.07jameswf-home~whatnow
05:54.08jbotsomebody said now was a good time to tell you that I have 6 gigabytes of data
05:54.17jameswf-home~nowwhat
05:54.18jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
05:59.21orkidman this is worse than #openbsd ?
05:59.38orkidrite of initiation? amirite?
06:09.31orkidno fading of moh?
06:09.34orkid:P
06:09.39comfrey<PROTECTED>
06:09.46orkidhow 1980s :P
06:09.55comfreythe other box was overkill
06:11.26orkidmaybe now ill fit
06:11.31orkidlol
06:16.11*** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net)
06:16.18drmessanowanpipe is 80s?
06:16.38comfreyok... well maybe it is related to 64bit
06:16.47comfreysounds great on 32 bit
06:17.00comfreyget the whole the number is not in service
06:17.03comfreythang
06:17.08comfreywell well
06:17.24comfreyso much for quad core intel box
06:17.29comfreyoverkill right
06:17.54comfreya amd duron 1600 mhz
06:18.07comfrey.5GB ram
06:18.14comfreyok for a 11 extension office?
06:18.33comfreyi just need another disk for raid 1
06:20.24drmessanoI have 70 extensions with 5 or 6 concurrent calls on a P4 2GHZ box with 512MB Ram
06:20.43comfreyand what is utilization like?
06:20.55drmessanoNot remarkable
06:21.10drmessano5% unless there's some transcoding
06:21.20comfreyk... well... i hear about folks running embedded asterisk so...
06:21.28comfreyi think it should hold up.
06:21.31comfreywe'll see
06:21.36comfreybeer is open
06:21.44comfreystarts provisioning phones
06:21.58comfreyand happy that i can stick with debian 32bit
06:23.26comfreyi hope to burn the nec tonight
06:23.27*** join/#asterisk synchris (n=synchris@athedsl-220386.home.otenet.gr)
06:26.35*** join/#asterisk cobex4 (n=cobex4@unaffiliated/cobex4)
06:26.43cobex4Hello world :))
06:27.21*** join/#asterisk nuonguy (n=john@c-71-198-1-139.hsd1.ca.comcast.net)
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06:40.13ionixHey guys, I'm looking for help on what hardware to use
06:41.09ionixI'm using a 4 years old machine as a PBX and I want to plug 2 analog lines into it and 2 fax machines as well (So it can bridge if it's a fax, otherwise take the call)
06:41.19ionixWhat hardware do I need?
06:47.09Maliutaionix: if you only need 4 lines then a tmd400p would do it
06:47.16Maliutaionix: there is also the 800p
06:47.45ionixAre the Openvox card any good?
06:47.52ionixOpenvox A400P
06:48.04ionix800p is for 8 modules, right?
06:48.39*** join/#asterisk comfrey (n=comfrey@dsl-66-78-76-161.ipns.com)
06:50.47Maliutaionix: yeah. I can only speak for digium stuff
06:50.53Maliutanever used openvox
06:50.56ionixof course
06:51.01MaliutaI am happy with my 400p though
06:51.12Maliuta2 FXS, 2 FXO
06:51.16phpboyionix: I'd guess so
07:03.34*** join/#asterisk mikkel (n=mikkel@130.226.36.170)
07:07.06*** join/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net)
07:08.45sivadnzAnybody know how to turn debug on with dahdi, and where it's output goes?
07:10.27comfreytalk to your mother
07:10.29comfrey;)
07:10.40comfreysorry...
07:10.58comfreyyou should just turn it up with regular debugging no?
07:11.02comfrey-vvvvvvvvvv
07:11.47milouxPossible to send SMS to a cell phone using asterisk connected to a SIP provider?
07:11.58ionixno
07:12.10ionixyou need an SMS provider. and I'm not sure if they integrate to asterisk
07:12.49milouxionix: ok thanks. Just as i thought...
07:13.08sivadnzcomfrey: dahdi docs say add 'options dahdi debug=1' to /etc/modprobe.d/dahdi which I have done. restarted, run asterisk with -rvvvvvvvc but still no additional output.
07:14.48comfreyhmmm... no logfile specification option?
07:15.02comfreyor something showing up in /var/log/asterisk?
07:17.35milouxQueue(queuename[,options][,URL][,announceoverride][,timeout][,AGI][,macro][,gosub][,rule]) <--- gosub i asume its gosub if the timeout is reached...Am i right?
07:17.51milouxso i can gosub QUEUESTATUS
07:18.15coluludoes anyone know how to make outgoing PSTN call with SPA3102
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07:19.39sivadnzcomfrey: nothing new in /var/log/asterisk
07:23.01comfreysivadnz: you try using the tools to get info?
07:27.29sivadnzconfrey: logger.conf needed debug=>debug uncommented
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07:34.12comfreysivadnz: in /etc/modprobe.d/dahdi or what?
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07:35.04comfreythe doc kinda sucks
07:35.14comfreyand the doc dir is empty
07:35.30comfreyjust the README
07:35.41comfreywhere did you find that info?
07:37.37milouxanyplace where all the columns in queue_table actually are documented (dont say http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue as its not documented there)
07:38.24milouxnvm
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07:48.32coluludoes anyone know how to make outgoing PSTN call with SPA3102
07:48.41colului have difficulty setting up SPA 3102
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08:06.23mockergets ready to break asterisk.
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08:07.44IsUphey
08:08.53coluluhi
08:09.38Greek-Boyhow does one see the current value of a global from the CLI?
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08:11.39IsUp'core show globals'
08:12.04drmessanoAnyone know how to remove menu item options on the DISPLAY of a Linksys SPA-94x?
08:12.28drmessanoI can fix the Web interface with the XML, but it has NO effect on the display, apparently
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08:19.34Greek-Boyi dont understand how to read a ODBC function
08:19.45Greek-BoyI have already set it it up in func_odbc.conf
08:19.55Greek-Boyall it does is read an sql statement
08:20.06Greek-BoyI dont know how to refer to that value from the dialplan
08:23.48comfreyc-yas in the mornin
08:23.54comfreyand thankd
08:23.58comfreys
08:24.10comfreywe'll be burning that nec soon
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09:02.10beniwtvHi all, I'm trying to make a webcallback application with asterisk. For that, I originate a call to a local extension (121), which in turn will do the call handling. However, asterisk seems to keep killing the agi, and thus, the originate fails. Any ideas?
09:03.36mockerbeniwtv: Are you using call files?
09:03.52beniwtvmocker: No, the originate command from the management console
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09:19.45beniwtvmocker: The thing is I can't use originate directly making the calls, because often my provider fails and then I have to switch to another provider. So I need a script which handles this on both calls. I also tried using sipsak to send SIP commands (using REFER), however, this did not work out (Asterisk declines the REFER), not sur why (I copied the SIP dialog form a softfone that works) .(
09:19.59sivadnzAnyone using Grandstream GXP2000 and Idle Screen Refresh, the latest firmware 1.1.6.37 say that it's now supported, just need to know how? Checkout http://www.grandstream.com/BETATEST/BT200_GXP/
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09:26.06zamolxeshello. is there a way to flush the current stuff in the cdr csv file?
09:27.27thomasI would like run a script if i recieve a call
09:27.27thomasexten => s,n,agi(anruf_ident.php|${CALLERID(num)})
09:27.29thomasis it correct?
09:29.02tzafrir_laptopzamolxes, no need. They are re-opened every time. Just move the existing one
09:29.52coluluHi, does anyone know about how to setup spa 3102?  I need some help
09:30.11thomastzafrir_laptop: is my line correct? hello :-)
09:32.25tzafrir_laptopthomas, seems correct
09:32.37tzafrir_laptopis this 1.4 or 1.6.x?
09:32.41thomas1.4.22
09:32.49thomasi think the script is not run
09:33.25tzafrir_laptopthe script is? the astagidir is?
09:33.46thomasvar/lib/asterisk/agi-bin/anruf_ident.php
09:33.50thomasthat is my script
09:34.01thomasbut i think why.. wrong extention :)
09:34.10tzafrir_laptopis it executable to the user running asterisk?
09:34.19tzafrir_laptopno need for an extension
09:35.17thomastzafrir_laptop: no no, i mean i have this line a a wrong [bla]
09:35.18thomas:-)
09:37.08thomastzafrir_laptop: now
09:37.08thomas<PROTECTED>
09:37.11thomas<PROTECTED>
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09:37.14thomas.-))
09:37.15Footmanhello
09:37.30FootmanIf a user click on a phone number in my interface, I want to ring his phone and call the phone number when he answers. I use the manager with this command :
09:37.36FootmanAction: Originate\r\nChannel: ".$channel."\r\nExten: ".$number_to."\r\nPriority: 1\r\nCallerid: \"".$caller_id."\" <".$number_from.">\r\n\r\n
09:37.42FootmanThe problem is that the phone of the user is unreacheable when I use the good $number_from. If I use an other number, it works... What's the problem ?
09:37.50FootmanIn fact, John Smith with the phone 300 wants to call Mike Affy with the phone 400. John clicks on 400 on the interface. With the originate command, the call informations are the same on the two phones. So, John should see on his phone "John Smith 300" and Mike will see the same thinks (it's an internal call, of course). The problem is that asterisk says that 300 is unreachable or busy when I want to display 300. If I display 200, it works but I don't
09:37.56Footmando you have a solution ?
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09:40.41jblackhuh. LNUX is a buck a share.
09:47.49tzafrir_laptopLNUX is a company that has some news sites and also sells proprietary software
09:48.06tzafrir_laptopIt is no longer in the Linux business
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09:55.23hescoI thought I could run * applications interactively at a console, not just the console status commands.  But I keep getting No such command when I tell it to Dial().  Any ideas?.
09:55.40hescoyes, I used a fully qualified Dial command.
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10:06.11gramulhaozinae
10:07.51yangI have one question regarding Dialplan...how could i change between these 2 CALLERID's when needed ?
10:07.54yang;exten => _0.,1,Set(CALLERID(num)=018109${CALLERID(num)})
10:07.56yangexten => _0.,1,Set(CALLERID(num)=0158631${CALLERID(num)})
10:08.21yangI can either use one or another? maybe putting things into different contexts would help?
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10:12.16yangbut then again, when the number is being called over a trunk it canniot distinguish between the wanted string, hrm?
10:19.09hesco~paste
10:19.10jbotpaste is, like, http://rafb.net/paste/, or see also pb
10:19.18hesco~pb
10:19.19jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
10:21.37yangwell, i paste those 2
10:22.06yangI would like to know how to route calls sometimes over first and over the second string
10:22.14yangdepending on the number
10:22.29IsUpwhat exactly you mean?
10:22.49yangsometimes the outgoing number is 58631xx and sometimes 8109xxx but the route is on the same trunk
10:24.08yangand this trunk links to the second trunk, so i don't know how to send the correct string from the second trunk
10:27.20hescoI've posted a summary of an issue I'm having include'ing => a trunk in another context, Can anyone please take a look and provide some guidance?
10:27.24hescohttp://pastebin.ca/1233659
10:27.38yangI guess it has to be sorted with different contexts on the primary trunk?
10:37.26hescoI thought I could run * applications interactively at a console, not just the console status commands.  But I keep getting No such command when I tell it to Dial().  Any ideas?.
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10:43.23IsUphesco: you are trying to Dial() over CLI?
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10:47.47joathesco.. the command line applications are different than those available via the dial plan...
10:48.01joatfor console dialing, try originate
10:48.02arekmhello, which context is going to be used if I do atxfer (features.conf) to a number which is currently busy?
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10:52.39hescojoat: thanks will do.  appreciate the lead
10:55.44hescook, but what does the tech/data mean in originate <tech/data> application <appname> [appdata] ???
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11:08.52tzafrir_laptophesco, SIP/trunkname/blabla - tech: 'SIP', data: 'trunkname/blabla'
11:09.21tzafrir_laptopBasically a channel
11:09.38tzafrir_laptopThe same as you use in Dial
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11:12.14hescothanks
11:15.22arekmafaik when atxfer fails then I should get back to the caller right? but instead I get "invalid extension"
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11:26.33hescoI keep getting "Call rejected by 69.71.222.196: No authority found", but host 69.71.222.196 says: not found: 3(NXDOMAIN)
11:27.01hescoThis was working only a day or so ago.  What might have changed to give me these errors?
11:29.33Maliutaif you can't get an host response I'd say something DNS related
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11:37.52hescoHere is a transcript illustrating the issue I'm having: http://pastebin.ca/1233681
11:38.16hescoI'm being rejected by an IP, immediately after registering to it.  What is that about?
11:38.17mort_gibMaliuta: I have had similar issues with several clients lately, use hosts instead...
11:39.24Maliutamort_gib: don't look at me, my stuff is working just fine
11:40.20mort_gib:-) Chill out, I'm not looking at you, but strangely 4 different clients had "DNS" problems in the same week....
11:40.58mort_gibIn the and I changed them to hosts, so now I have to check the DNS~IP every week or so!
11:41.05Maliutaif you have issues with _your_ DNS server completing lookups then it's possible
11:41.16MaliutaDNS is more than just other peoples servers
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11:45.39mort_gibYes, but in this case it was 3 different public DNS servers (ISP's servers)
11:45.45mort_gibStrange though!
11:46.17mort_gibIf I use mode=mp3 in musiconhold.conf it should play louder right??
11:46.39IsUpno, its not.
11:47.43mort_gibIsUp: Then how do I play it louder?
11:48.58hescoWell, I just tried this from a different server (and a different IAX2 account) and got this message: Call accepted by 69.71.222.196 (format gsm), as the call went through.
11:49.22hescoSame IP which my host command cannot find
11:49.45Maliutahas the IP changed?
11:49.54IsUphesco: maybe its about with context? i remember completely but i had same problem, "No authority found". maybe you can check your Dial string?
11:50.05Maliutabecause it will live until the TTL expires, and some ppl set stupidly high TTL's
11:50.54hescomeanwhile back on my problem server I see: chan_iax2.c:8667 socket_process: Call rejected by 69.71.222.196: No authority found
11:51.24Maliutahesco: have you done any call debugging?
11:51.34hescohow would I do that?
11:51.40IsUpmort_gib: i dont think using mp3 making a difference. i can't suggest anything, i am not using MoH stuff so much.
11:51.47hescoI am watching a console invoked with 3 v's
11:51.59IsUp'iax2 set debug'
11:52.04hescook
11:52.52pacakI am trying to activate 'Pickup feature'. I have three SIP accounts (401, 402, 441), two of them (401 and 402) have pickupgroup=1 callgroup=1. I am calling from 441 to 402, then i am trying to pickup call from 402 by dialing *8. As far as i can see pickupgroup=0 and callgroup=0 on channel, which leads to 401 and have status RINGING.
11:53.30pacakBut i can see pickupgroup=0 and callgroup=0 when i type sip show peer 401 in asterisk console
11:53.40pacak(asterisk 1.6.0)
11:53.44pacakAny ideas?
11:53.49mort_gibpacak: What type are your SIP users??
11:54.11pacakmort_gib: type=friend
11:54.38mort_gibThere you have your problem
11:54.38IsUppacak: try without pickupgroup and callgroup
11:54.49mort_gibtype=peer
11:56.22pacakOk, thanks. I'll check...
11:58.16hescoI'm assuming that 'No authority found' means that my server was unable to authenticate with the IAX2 provider.  Is that right?
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11:59.31Maliutahesco: debug a call and we'll see
11:59.32IsUphesco: no, may its about with context.
11:59.42IsUpjust debug a call, and let's see whats wrong.
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12:02.20hescook, one call, debugged: http://pastebin.ca/1233688
12:04.46Maliutahesco: show us your iax.conf
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12:05.56hescocoming you way:
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12:09.40ronrI got some zap errors, anyone knows their meaning: http://pastebin.com/m3b0ca199
12:10.23IsUpwhen you get these errors?
12:10.38hescothis is the meat of my iax.conf:  http://pastebin.ca/1233694
12:11.04ronrthey were there when a connection was disconnected
12:11.27IsUpyou are using PRI?
12:11.31ronryes
12:11.45IsUpconnected to a GSM GW or what?
12:12.00ronrno, I''m guessing it's a telco error but not sure
12:12.11Maliutahesco: and these calls are being done as Dial(IAX/diamondcard)?
12:12.20hescoyes.
12:12.43hescoeither with a .call file or with an originate command at the console.
12:12.54pacakmort_gib: I tried type=peer and it does not works as well. Any other ideas?
12:13.12IsUppacak: try type=friend and remove pickupgroup and callgroup options.
12:13.36IsUpronr: these errors coming up when a "call disconnects", right?
12:13.45Maliutahesco: and the "secrets" match?
12:14.15hescoyes, I've copied them back in to the .conf files from source a couple of times now.
12:14.46ronrIsUp: yes
12:15.23Maliutahesco: because the remote host is definitely rejecting the password
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12:15.45IsUpronr: its probably about with telco. i dont think theres another problem.
12:15.59Maliutahesco: what does an iax2 show registry give us?
12:16.00IsUpmay you can try showing your zapata.conf
12:16.13ronrIsUp: thx
12:16.41Maliutahesco: I am thinking you should attemt a login to the providers sight and check your account status
12:17.02pacakIsUp: No luck. Actually i already saw asterisk's sources and it requires (chan->pickupgroup & cur->callgroup) to actually pickup call from group.
12:17.20hescodoes reloading a dialplan reload the iax.conf file as well?
12:17.25Maliutano
12:17.26IsUppacak: okay, hold on please
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12:17.29pacakAnd i can pickup calls using Pickup application.
12:17.36Maliutahesco: you need to do an iax2 reload
12:17.37IsUp'iax2 reload'
12:18.04hescoI'm on my provider's site, and I have plenty of funds on deposit with this account.
12:18.30Maliutahesco: did you just tell us you've been changing the iax file and not reloading it?
12:20.09hescoI just reloaded it and tested again.  I also just copied the passwords in again to each of the three places in occurs in this file.  It all looked great.
12:20.21hescoI had been reloading the entire server.
12:22.21Maliutahesco: iax2 show registry
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12:29.31IsUppacak: do you have an extension for pickupping?
12:30.07pacakIsUp: Nope, i am using *8 (actually *0) which is defined in features.conf
12:31.30IsUplet's try with "exten => _8XX,1,Pickup2(SIP/402)"
12:31.33IsUpif its working or not
12:31.39pacakIsUp: Pickup works if i set correct pickupgroup in features.c or comment out (chan->pickupgroup & cur->callgroup) check
12:31.43pacakIsUp: it works
12:37.21hescoHost                  dnsmgr  Username    Perceived             Refresh  State
12:37.21hesco69.71.222.196:4569    N       461170      71.236.31.107:4569         60  Registered
12:37.21hesco71.236.31.107:4569    N       biko3       <Unregistered>             60  Timeout
12:37.22hesco2 IAX2 registrations.
12:38.06hesco.
12:38.57hescoI took the IAX2 account I have on my sandbox server and used it working connection parameters on the new production box I'm building now.  They also failed.
12:39.27Maliutais the sandbox connected with those credentials too?
12:41.18hescotwo iax2 accounts.  Acct A works on sandbox, fails on production box.  Acct B fails on production box (and has not been tested on sandbox)
12:43.36hescoI may have found the issue.
12:43.52IsUphesco: it's about with auth. i dont see any other problem.
12:44.29hescoMy production box where I'm having these issues is hosted on a dyndns account.  It seems my IP may have changed.  I'll update and see if my issue persists.
12:44.51IsUp...
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12:46.41Maliutathe IP changes between registration and the calls? repeatedly?
12:46.50Maliutaset the registration timeout lower
12:47.07Maliutaif it's changing every 60 seconds you have issues
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12:50.20pacakIsUp: Any other ideas?
12:51.40IsUppacak: no idea about your problem.
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12:58.19jasonwootwoot
12:58.26Maliutajason
12:58.35Maliutayou missed a bit there
12:58.35jasonwootwerd
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13:06.36milouxanyway you can localize the "Wednesday, October 22, 2008 at 03:04:42 PM" thats printed with ${VM_DATE} in the voicemail functionality?
13:10.33milouxI need to do that in app_voicemail.c don't i?
13:10.52Maliutamiloux: ~book
13:10.58Maliutapage 488
13:11.05milouxMaliuta: Thanks
13:11.16MaliutaRTFM people
13:11.44MaliutaRTFB is READ THE F*CKING BOOK
13:12.26milouxMaliuta: Dude...Information on asterisk is pretty scattered. There isnt a single place with enough information on asterisk. That you need to look at several places... That makes it pretty hard to find information you need and pretty easy to ask in #asterisk...
13:13.05[TK]D-Fendermiloux: IIRC the sample config shows you how to set the zone for each box.  Go ready it
13:14.24Maliutamiloux: google for "asterisk voicemail timezone" and press "feeling lucky"
13:14.41Maliutamiloux: try and find the answer yourself first
13:14.58creativxdo you feel lucky, punk?
13:15.04Maliutamiloux: your googlefoo must _really_ suck
13:15.18milouxon line 657 in app_voicemail.c its hard coded an at (static char emaildateformat[32] = "%A, %B %d, %Y at %r";)  so you mean setting the timezone changes that AT?
13:15.51[TK]D-Fendermiloux: You should not be looking at the source for something normally controlled by the configuration files <-
13:16.28milouxand FYI the timezone is correctly set and is properly displayed everywhere except the mail that the voicemail system sends out
13:18.42IsUpmiloux: voicemail.conf, take a look 'tz='
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13:20.15IsUpcd /usr/share/zoneinfo;ls -al
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13:40.03LeddyHMasterisk 1.4.22: starting music on hold, stopping music on hold (instantly) and no music plays to the caller. Do I need to bump up my -vvv to troubleshoot?
13:41.28[TK]D-FenderLeddyHM: You need to see what class is called, and verify that it has anything to play and if its an external app, that it is compatible.
13:41.55LeddyHMdefault, just points it to /var/lib/asterisk/moh
13:42.01LeddyHMwhich has the default wavs
13:42.09LeddyHMand is readable
13:42.20[TK]D-FenderLeddyHM: pastebin..... configs, folder dumps, etc
13:42.44[TK]D-FenderLeddyHM: CLI....
13:49.12LeddyHMfixed it
13:49.27LeddyHMmode=quietmp3 needed to be mode=files
13:51.35[TK]D-FenderLeddyHM: Yup, good 'old mpg123 not working out.... never would have seen that one coming...
13:52.03LeddyHMyeah I was reading some docs that it was removed and supported in addons
13:52.20LeddyHMjust missed the important line change
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14:05.57tmjbhello does anyone have idea why exten => s,n,WaitExten(15|m) music with this command on asterisk 1.6 ?
14:06.34tmjbi also setup exten => s,n,SetMusicOnHold(default) whic should setup default moh
14:07.48*** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp)
14:08.12*** join/#asterisk errr (n=errr@fedora/errr)
14:08.56[TK]D-Fendertmjb: Whats the problem?
14:10.36Kattywocka!
14:10.55tmjb[TK]D-Fender, i dont here the moh when the dial plan gets to s,n,WaitExten(15|m) is this changed in 1.6
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14:11.16*** mode/#asterisk [+o putnopvut] by ChanServ
14:12.02Kattygoooooood morning all you wonderful asterisk users!!!!
14:12.02[TK]D-Fendertmjb: Instructions say no.  pastebin the CLI of your attempt, your musiconhold.conf, and the folder dump of where your MoH files are
14:12.04[TK]D-Fender~pb
14:12.05jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
14:12.06[TK]D-FenderKatty: Mew.
14:12.12Kattyit's going to be a Super Duper day!
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14:20.22lmadsenKatty: lies!
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14:27.16tmjb[TK]D-Fender, http://pastebin.com/m5f9f7924 here it is tnx
14:27.59Kattylmadsen: not lies!!!!
14:28.08*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-867a9fdc54cca32c)
14:28.08lmadsenKatty: that sounds like a lie
14:28.08*** mode/#asterisk [+o Deeewayne] by ChanServ
14:28.40Kattyhugs lmadsen
14:28.45[TK]D-Fendertmjb: the rest please...
14:28.49lmadsenday is going to be crazy busy... lots of consulting to do, a presentation to update, dinner with a speaker from the FSOSS conference I'll be speaking at tomorrow, etc....
14:29.19Kattylmadsen: well at least try to be happy (=
14:29.34lmadsenKatty: bah humbug!
14:30.32tmjbtmjb, and the files in /var/lib/asterisk/moh here it is http://pastebin.com/m3c8c55d6
14:30.49[TK]D-Fenderlmadsen: FSOSS?  Free and Semi-Open Sours Software?
14:30.56[TK]D-FenderSource*
14:31.14lmadsen[TK]D-Fender: heh... Free Software and Open Source Symposium :)
14:31.27lmadsenor as I like to call it... Free Sauce
14:31.47*** join/#asterisk mog (n=mog@nat/digium/x-5188dbd0d1e782e3)
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14:32.11[TK]D-Fenderlmadsen: Sounds.... dirty...
14:32.22[TK]D-Fendertmjb: Go test your MoH directly.
14:32.34tmjbI did exten => 123,3,MusicOnHold(default)
14:32.37tmjband in works
14:33.49[TK]D-Fendertmjb: Please set the class in your call to waitexten, and I'd like to see more past your call to it as well.  Also include your direct test attempt
14:34.17[TK]D-Fendertmjb: And include your actual dialplan
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14:40.20Kattylmadsen: now now, mister party pooper.
14:40.38UnixDawg_so dod asterisk 1.4.22 move to dahdi or can it also still use zaptel
14:40.38lmadsenKatty: I'm the grinch donchaknow
14:40.46UnixDawg_dod/did
14:40.52lmadsenUnixDawg_: I believe it can still use zaptel
14:40.57Kattylmadsen: you keep that mood up and im going to start tickling!
14:41.03lmadsenbut zaptel won't be getting any more releases
14:41.12lmadsenKatty: I'm not ticklish!
14:41.15Katty:M
14:41.17Katty:<
14:41.23lmadsennice stache
14:41.27Kattythx.
14:41.30Kattyi bought it myself.
14:41.34UnixDawg_everyone is ticklish you just have to find the right spot
14:41.41lmadsenI have a couple of fake mustaches myself
14:41.41tzafrir_laptopyou can use either. Though it's a build-time decision
14:41.53UnixDawg_ok
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14:48.05logicwrathim getting the following error: No channel type registered for 'Local'
14:48.20[TK]D-Fenderlogicwrath: Look at your modules.conf
14:48.23lmadsenyou don't have chan_local.so loaded
14:48.53[TK]D-Fenderlogicwrath: this is a cre module that really rare to see someone not get compiled & loaded unless they are running a phsyco slimmed down install
14:48.59[TK]D-Fendercore*
14:49.23logicwrathreviewing
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14:57.20logicwrathmy modules.conf is showing autoload=yes
14:57.28logicwrathis it possible a module got corrupted?
14:58.01logicwrathhttp://pastebin.com/m4ad19e2c
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15:04.20Katty[TK]D-Fender: http://www.flickr.com/photos/izaah/2963781759/sizes/m/in/set-72157608195837483/ <- Mmm!!! Delicious rawhides!
15:04.51ajohnsonAnyone know if it is possible to add CDR variables when using cdr_odbc and write them to a database
15:04.57[TK]D-FenderKatty: Yums... webcam pic?
15:04.59ajohnsonother than using the hangup extension
15:05.19Katty[TK]D-Fender: no, that was taken last night
15:05.37ajohnsonrawhide nom nom nom
15:05.42Katty[TK]D-Fender: my camera just has a time stamp on it
15:05.57[TK]D-FenderKatty: What'd you take it with?
15:05.59Kattyajohnson: we have a lil butcher shop out here that smokes bones for doggies.
15:06.07Kattyajohnson: i can't wait to get him some (=
15:06.20ajohnsonooohhh
15:06.23ajohnsonThat's pretty cool
15:06.25Katty[TK]D-Fender: http://www.flickr.com/photos/izaah/2963781759/meta/in/set-72157608195837483
15:06.31Kattyajohnson: oh yes.
15:06.38coppiceI passed a place serving smoked dog last week
15:06.46Kattyajohnson: they smelled pretty good last time i was in there to get some bacon, but i didn't think about getting him any at that point...
15:06.54Kattyajohnson: and they were out yesterday when i dropped by :/
15:07.23Katty[TK]D-Fender: it's a mostly el cheapo camera from walmart
15:08.04[TK]D-FenderKatty: Its got more grain than Quaker ;)
15:08.30Katty[TK]D-Fender: yeah the zoomy bit doesn't work too well
15:08.42[TK]D-FenderKatty: Mind you I went there last night for other stuff and saw their film & development costs were pretty good...
15:08.46Katty[TK]D-Fender: good enough for me tho (=
15:09.41Katty[TK]D-Fender: yeah the camera was about 70
15:09.49Katty[TK]D-Fender: the 2gig card was around 15, i think
15:10.19Katty[TK]D-Fender: still wanna get a camcorder so i can do proper videos with audio
15:10.47[TK]D-FenderKatty: I've become a camera enthusiast in this past quarter, so I start to notice these things
15:11.10Kattylemme get the camera i've had my eye on for awhile
15:12.01Katty[TK]D-Fender: http://www.newegg.com/Product/Product.aspx?Item=N82E16830120256
15:12.46[TK]D-FenderKatty: What do you expect out of a camera?
15:17.55knightfal[TK]D-Fender: You around man?
15:18.02[TK]D-Fenderknightfal: Yup
15:19.15knightfal[TK]D-Fender: Im kind of at a loss here I got a new set up with a tyan board and dual AMD processors and for some reason I cant get my cards to fire up.
15:19.38knightfalI keep getting  DAHDI_SPANCONFIG failed on span 1: Invalid argument (22)
15:20.03knightfalBut when I install on an Intel based system everything goes fine.
15:20.50knightfalIs there something different needed during install/compile time when running with an AMD based system?
15:21.37*** join/#asterisk XnOSX (n=XnOSX@212.145.55.118)
15:21.39[TK]D-Fenderknightfal: Nothing I've seen.
15:21.47[TK]D-Fenderknightfal: And less that you're showing
15:23.22knightfalhttp://pastebin.com/m2888d27b
15:24.06knightfalthis is my out put of /etc/dahdi $ cat /proc/dahdi/*; cat /etc/dahdi/system.conf
15:24.23knightfalIt looks like on boot up its loading ztdummy
15:24.37knightfalor dahdi_dummy
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15:25.36Footmanbye bye
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15:26.11jasonwootAnyone here using Cacti with asterisk?
15:26.28[TK]D-Fenderknightfal: What card?  Where is ztcfg?  Where is the verification that chan_dahdi is even loaded?  Did you try manually?
15:27.21*** join/#asterisk hfb (n=hfb@96.247.65.63)
15:27.23knightfalI did try to bring them up manually.  The cards are 2 te410p
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15:29.45tzafrir_laptopknightfal, the driver for your card has failed to load for whatever reason
15:29.54tzafrir_laptopWhat is the output of:  dahdi_hardware
15:30.01knightfalone sec
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15:32.09knightfalok My guy put elastix on it lastnight to see if he could bring up the cards like we had before
15:32.55knightfallemme look again
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15:35.07jasonwootCacti, or maybe Nagios anyone?
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15:37.59mockerjasonwoot: What's your question?
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15:38.35tzafrir_laptopknightfal, you realise that if you use dahdi you'll have to rebuild Elastix's asterisk as well
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15:39.25knightfalIts an install from thier ISO so its using zaptel
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15:39.32jasonwoota voip-info user posted a great cacti template for monitoring asterisk processes, but the zap-channel template only seems to show total zap channels, not zap channels in use
15:39.34knightfalBut its the same issue on this machine
15:39.43shrivenhey guys I can't find an answer to this... Does anyone know what this cryptic error actually means?
15:39.44shriven[Oct 22 11:37:12] WARNING[28811]: app_fax.c:174 phase_e_handler: Error transmitting fax. result=17: Received a DCN while waiting for a DIS.
15:39.48jasonwootwondering if that's tweakable
15:40.14knightfalthe card will not come up at all
15:41.17mockerjasonwoot: Should be, but tweaking cacti sucks. :P
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15:42.48jasonwootmocker: a more accurate description would be "bewilderingly undecipherable"
15:43.01jasonwootaka, chinese algebra
15:43.07jasonwootbut it sure is pretty
15:44.08*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
15:44.23mockerjasonwoot: Indeed.
15:44.33mockerjasonwoot: Zabbix is a little easier, but still a pain.
15:44.39Kattyindeederoonies!!!
15:46.21coppiceChinese algebra? you mean things like 三X + 四Y = 四十二?
15:46.39jameswf-homecoppice: the answer is 42
15:46.58coppiceso, you can read Chinese
15:46.59jasonwootsays "some people call it a sling blade but I call it a Kaiser blade"
15:47.55*** join/#asterisk dvyjones (i=dvyjones@wikimedia/Dvyjones)
15:48.29dvyjonesIs it possible to wire up asterisk as an advanced "voicemail" for my phone?
15:48.36tzafrir_laptopknightfal, I asked you if you see the card in the output of dahdi_hardware .
15:48.54dvyjonesSo it rings on the phone, but if no one answers, or */# is pressed, it goes into voicemail/menu.
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15:49.21tompawblah
15:49.38Kattynearly almost lunch time (=
15:49.39tompawI quickly need to call a H323 trunk on 1.6 - any quick howto url please?
15:49.41Katty[TK]D-Fender: what are you having for lunch?
15:51.12knightfalTzafrir: It doesnt have dahdi now we reinstalled it with zaptel
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15:51.43*** mode/#asterisk [+o russellb] by ChanServ
15:56.03knightfalTzafrir Whats the equivelant for zaptel
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16:04.21*** join/#asterisk stintel (i=stijn@madwifi/support/stintel)
16:05.38stintelgood evening / timezone all. is there any place where I can find a description of a red alarm (ZT_ALARM_RED) with a Zaptel PRI card ?
16:06.07*** join/#asterisk IsUp (n=nocturne@88.255.77.200)
16:07.15coppicedescription? you mean something like "its red"?
16:07.30n3hxsIf it has the same meaning with other PRI circuits.... it is down hard
16:07.54holosn3hxs: Yah, it's down hard...
16:08.05holosthats soft of a standard alarm
16:08.08n3hxsLike if you can't make a call... red alarm.
16:08.34jeevisUp
16:08.40IsUphello
16:08.41jeevi just woke up, it's been doing it all night.
16:08.43holosn3hxs: like someone unplugged the cable or the far side is down because a backhoe operator was not paying attention..
16:09.00coppicea red alarm on an E1 or T1 means the card can't sync to the received signal, typically because there is no received signal
16:09.28stintelcoppice: thanks, so that would be a L1 problem ?
16:09.52tzafrir_laptopknightfal, what do you mean by "equivalent"?
16:09.57IsUpwell, i've just travelled 3 hours to my telco for plugging an E1. it's was in RED alarm. i plugged a new rj45.
16:10.11stintelIsUp: sucks :/
16:10.17IsUpwho is the lucky guy having red alarm?
16:10.22stintelIsUp: that would be me :)
16:10.29IsUpwhats your card?
16:10.30tzafrir_laptopknightfal, BTW: to "ping" someone you should normally mention their complete nick name. Use the tab key to complete them
16:10.45stintelIsUp: description=Digium Wildcard TE110P T1/E1 Card 0
16:11.12IsUpwell, no idea about digium cards. i can help you about sangoma but i can tell you some tips
16:11.47IsUpa plugged E1 should have 2.5 - 3.5db or -2.5 RX level
16:12.06[TK]D-FenderKatty: Campbell's soup "Italian Wedding"
16:12.11stintelIsUp: np. I'll go to the datacenter first, maybe something with the cable or the card just fried - we have a spare one
16:12.27*** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com)
16:12.33IsUpit's probably about wiring. replace your wiring.
16:12.44*** part/#asterisk dvyjones (i=dvyjones@wikimedia/Dvyjones)
16:12.48IsUpby the way, how you control Digium cards? any diag. tools?
16:13.09stintelif only I knew that :P
16:13.11n3hxsAllways call the phone company and blame them first.  :)
16:13.31stinteln3hxs: don't have a contact yet :P
16:13.36stintelbut I will have tomorrow
16:13.49n3hxsdid it work before and just now failed?
16:13.55IsUpyou are going to use SS7 or what?
16:15.15stinteln3hxs: it worked before. but I ran update-source today on my PIAF system. however, if I normally do that, I get a yellow alarm, because genzaptelconf adds crc4 which is not supported by the telco
16:15.30stintelso removing that from zaptel.conf and service zaptel restart should do the trick
16:15.33stintelnow it doesnt
16:15.36*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:15.51tmjb[TK]D-Fender, sorry I was busy is here my dial plan for http://pastebin.com/m60c3f0a5 and belive music on hold does work but not with waitexten only with not with waitexten ? any ideas
16:16.20IsUpstintel: can you pastebin your zaptel.conf?
16:16.25IsUpand other outputs
16:16.38stintelIsUp: sure
16:16.38[TK]D-Fendertmjb: exten => 444,2,WaitExten(10|m) <-- "|" is not a valid separator in 1.6
16:16.39[TK]D-Fendertmjb: "," only
16:17.15tmjbtnx
16:18.10IsUpi cannot use ChanSpy with chan_ss7, how can i listen channels? any ideas?
16:18.15stintelIsUp: http://stijn.tintel.eu/zap
16:18.18*** part/#asterisk bbryant (n=Brett_Br@adsl-153-43-19.chs.bellsouth.net)
16:18.36stintelIsUp: donno if you want anything else
16:18.55IsUpstintel, did you try setting your span to "slave" clock?
16:19.21IsUptelco is always master clocking
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16:20.41stintelIsUp: if I understand http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax correctly, it is configured like that
16:21.07IsUpstintel: just try with slave clocking
16:21.15IsUpspan=1,1,0 -> span=1,0,0
16:21.42stintelIsUp: If '0' is used asterisk will try to provide timing to the span
16:21.55stintelso that doesn't seem right to me
16:21.56stintelbut ill try ;)
16:22.20coppiceit isn't right. you need something like span=1,1,0
16:22.30stintelIsUp: same problem
16:23.01stinteloh hmmm
16:23.08IsUpokay, 1 min.
16:24.21tompawhttp://pastebin.com/m410435b5
16:24.28tompawcould you please have a look guys?
16:24.48stintelI remember I had to use span=1,1,1 because we had line drops, but changing that now still gives red alert
16:24.48hardwiredrmessano: how dod comfrey turn out last night?
16:24.55tompawI'm simply calling it with exten => _00XXX.!,1,Macro(trunkdial,OOH323/trunk_wiz/${EXTEN:2})
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16:32.51murdock_utDoes anyone know why in /dev I'm seeing dahdichannel, dahidctl,dahdipseudo, dahditimer instead of each of dahdi/channel and so on.
16:33.10jameswfmurdock_ut: do channells loaded
16:33.59Qwellmurdock_ut: I think because you don't have the udev rules installed
16:34.18Qwelldid you install the release version?
16:34.41murdock_utQwell: I looked in /etc/udev/rules.d and the dahdi rules are there.
16:34.42Qwellthere was a problem that was fixed before the release..  I don't recall exactly what it was though
16:35.01murdock_utI just downloaded the dahdi files from the website this morning.
16:35.06Qwellwhat version?
16:35.32murdock_ut2.0.0
16:35.40Qwellof linux and tools?
16:35.48murdock_utSame version
16:35.51murdock_utof tools.
16:36.06QwellI wonder if tzafrir_laptop remembers..
16:36.29murdock_utWhen I try to start dahdi I get "Error: missing /dev/dahdi
16:36.47jameswfmurdock_ut: what kenrel
16:36.58tzafrir_laptopmurdock_ut, it means that there are no udev rules
16:37.16Qwelltzafrir_laptop: yeah, that was what I thought..  he says he has them though O.o
16:37.20murdock_uttzafrir_laptop: That's what I thought.  Kernel version 2.6.27 rc8
16:37.21jameswfhow do I turn off the drunl local.... argh
16:37.28tzafrir_laptop(and that the patch to use dahdi!ctl and such was not yet applied)
16:37.46hardwirehas anybody compiled a "burn in test" of hardware to channel ratios for different hardware specs?
16:37.57tzafrir_laptopmurdock_ut, grep dahdi /etc/udev/rules.d/*.rules
16:38.17jameswfhardwire: I am always told thats a stupid question
16:38.38murdock_uttzafrir_laptop: Do you want me to pastebin the results?
16:38.41Qwelljameswf: you at 1 million calls yet?
16:38.47hardwirejameswf: I wait for them to tell me that.. then I wait longer for people to tell me it exists.
16:38.53Qwellmurdock_ut: show us the dahdi/ctl line
16:39.08murdock_utQwell: /etc/udev/rules.d/dahdi.rules:KERNEL="dahdictl", NAME="dahdi/ctl"
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16:39.25Qwelllooks right
16:39.41Qwelltried reloading the dahdi modules?
16:39.46tzafrir_laptopmurdock_ut, any chance that the rule was not there when the module dahdi was loaded?
16:40.16hardwirejameswf: I've always wondered if a test could be standardized.. like loopbacks per t1 and some dialplan logic.. and a sip tester
16:40.57murdock_uttzafrir_laptop: excuse my ignorance, but I have restarted the system, would that of cleared that up?
16:41.43tzafrir_laptopsure
16:41.47Qwellmurdock_ut: if you've restarted since the udev rules were there
16:42.00murdock_utQwell: I have, twice.
16:43.57murdock_utQwell: when I run lsmod I do see dahdi in there if that means anything.
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16:44.41tzafrir_laptopmurdock_ut, what distribution is it?
16:45.41murdock_uttzafrir_laptop: Mandriva 2009.0 (don't shoot me, I'm just more familiar with this distro.)
16:45.58yareckonhello all, let me ask a question which you all probably get a lot, yet I can't find a simple answer to
16:46.03tzafrir_laptopok. So it's not an ancient udev
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16:46.16murdock_uttzafrir_laptop: It's a new Distro, so new challenges.
16:46.18Qwelltzafrir_laptop: well, it is Mandriva...  who knows what they've got
16:46.20yareckonwhy can't I use a modem to make outgoing calls controlled by asterisk
16:46.28Qwellyareckon: You can.
16:46.32QwellIF you write drivers for it.
16:46.45Qwellso effectively - no, you can't.
16:46.47yareckonbut noone has drivers
16:46.53Qwellexactly
16:47.11yareckonbecause quality is poor and anyone who is anyone buys a digium card?\
16:47.13QwellIt's not cost-effective to do so
16:47.36coppiceso many people want winmodems to work with zaptel, yet nobody can be bothered to do the work :-)
16:47.36Qwellno, because for the price that you would pay for a driver to be written (or the time you'd spend writing one), you could EASILY justify buying "real hardware"
16:47.56*** join/#asterisk mchou (n=mchou@unaffiliated/mchou)
16:47.56yareckonit would certainly be cost effective to use a modem for such a purpose, as there are hundreds of thousands lying around
16:48.00murdock_utQwell: Well, this is the first time I've had this problem.  :)
16:48.22tzafrir_laptopyareckon, if you have some extra time this would be an interesting project
16:48.26Qwellyareckon: Each different modem needs a different driver.
16:48.39QwellIf you'd like to write hundreds of drivers, you are more than welcome to.
16:48.48yareckonwhy is that... don't we have hardware abstraction for the audio generation?
16:49.11coppiceits not hundreds. a few chips covers 90%+ of the computers out there
16:49.27Qwellcoppice: overstatement, but I think (thought?) I made my point
16:49.41*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
16:50.04*** part/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
16:50.12coppiceit used to be hard to get the info to develop drivers for them, but its easy now
16:50.22yareckonseems like an interesting project... if I was a c programmer
16:51.20yareckonas a sysadmin... I have 20 old modems lying around... and it would be nice to play around with a super answering machine without buying any new hardware
16:51.29murdock_utQwell: I've got Dahdi running on 2008.1 and it seems to load fine..  So something has changed I guess.
16:52.19yareckonor have my home computer call me up to deliver mesages without having to pay a gateway
16:52.44yareckonof course all of this works once you buy a card... but more people would play around if you just needed a modem
16:52.48yareckonthanks for the explanation
16:52.52[TK]D-Fenderyareckon: And how much dev time is it worth to try to make some random garbage modem workable as a TDM voice interface?
16:53.31yareckonyes... it is not worth it for one person
16:53.47*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
16:54.06murdock_utQwell: So is the prefered asterisk distro, CentOS?
16:54.12coppice[TK]D-Fender: how come all these perfectly servicable modem cards become random garbage when asterisk enters the picture?
16:54.19Qwellmurdock_ut: depends on who you ask
16:54.25yareckonthe only thing that makes it attractive is the hundreds of millions sunk into hardware that is now useless for IP communication, but still might work as a pstn bridge
16:54.36*** join/#asterisk Steak__ (n=Alex@81-61.3-85.cust.bluewin.ch)
16:54.41Steak__hello
16:54.46yareckonwell, it's still useful for ip commincation if you can't get DSL
16:55.13*** join/#asterisk nx5 (n=pocholo@unaffiliated/nx5off/x-000000001)
16:55.27Steak__anybody with a working SPA3102 and asterisk setup here?
16:56.10hardwireom nom nom
16:56.44[TK]D-Fendercoppice: As each had their own driver for whatever OS, what's it worth to make an * driver for each one you'll find?
16:57.05coppicehow many are there? about 5 or 6?
16:57.42[TK]D-Fendermurdock_ut: Any one you're capable of administering and providing *'s requirements on will do.  The "majors" will have more documentation and support resources (and people) to guide you should you run into hangups
16:58.08[TK]D-Fendercoppice: 5 or 6 drivers required you mean?
16:58.18[TK]D-Fendercoppice: to cover the majority of modems out there?
16:58.28[TK]D-Fender(sufficiently of course)
16:58.51coppicejust support Agere, conexant and ST and most notebooks can be used
16:59.26Steak__... how do I check which number an extension is dialing? from the CLI, I mean...
16:59.27jeevshit! my friend at the office wants all the outbound calls to show caller id! but one problem, i make the calls private and send them out my home viatalk connection and save a lotttttttt of $
17:00.09jblackSo set your callerid on the calls leaving your system to the number of calls entering it.
17:00.17[TK]D-Fendercoppice: how much work do you think it would take from an X100 base to support these other chipsets?
17:00.31jeevhuh, what you mean entering it
17:00.44jeevviatalk residential service i have doesn't allow me to set caller id, it's my home phone.
17:00.46coppicebased on the Linux source code they already provide, not that long
17:01.00jblackIf 123-4567 is the number calling in, set your callerid to 123-4567 before you redirect the call.
17:01.21jeevno, their outbound call, like if they call their house, i append *67
17:01.23jblackWell then, you're screwed because of an overly limited provider.
17:01.25[TK]D-FenderSteak__: an "extension" is a number in your dialplan.  To see what a device dials, look at your channel debug as the call is placed (SIP debug for a SIP device for instance).  To see where things are at a point in time you would show the list of active channels "core show channels"
17:01.40*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
17:01.50jeevjblack, it's a residential service, $200/2 years for unlimited local and long distance.. and i've had it for 30 months now. decided to plug asterisk into it and use it from the offices too hah
17:01.58*** join/#asterisk zuez (n=sf@catalyst.httpd.org)
17:02.22Steak__[TK]D-Fender thanks
17:02.59Steak__so to define a SIP trunk it is just an exten line in the extensions.conf that handles a pattern and dials the SIP trunk with the number passed through, right?
17:03.07coppiceI've actually had the ST USB modem chip working for telephony, but not plugged into zaptel. it took less than a pageful page changes to make that function
17:03.10zuezDo any of you folks use some sort of say Nagios plugin to automate a SIP conversation?
17:03.16jblackjeeve: Ok.
17:03.37jeevnagios plugin to automate sip
17:03.38jeevwhat
17:03.38[TK]D-Fendercoppice: And to plug to Zaptel?
17:03.40jeevto test?
17:04.39zuezjeev: Yeah just to make sure the conversation takes place, that say a 503 isn't coming back, etc.  Crazy idea, not sure if someone was crazy enough to do it yet though.
17:04.41coppicemaybe another couple of pages of changes.
17:05.30jeevcool
17:06.08[TK]D-Fendercoppice: Now for the downsides that people ride the X100P for, how many of these other modems would be as bad or worse?
17:06.52*** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net)
17:07.07tzafrir_laptop[TK]D-Fender, for startes, they work great as introduction hardware
17:07.10coppicea modem is *far* more demanding of quality than voice. these x100p cards actually work pretty well with a competant echo canceller, like OSLEC
17:08.02yareckonfree teh modems.sourceforge.net
17:08.03tzafrir_laptopIf you could just use Asterisk with the modem you already have on your motherboard, it would be nice
17:08.26jasonwootjeev: we're using nagios to alert if number of sip channels in use falls below certain threshold
17:08.31coppicenotebooks are perhaps more interesting than motherboards
17:08.45yareckontzafrir_laptop, that's really a good point
17:08.47jeevthat's cool jasonwoot, you should tell zuez, it hink he's lookin for sometin like that
17:08.54tzafrir_laptopyareckon, err... could you give a working address?
17:08.55*** join/#asterisk ddunavant (n=David@75.145.240.14)
17:09.09jasonwootdoh, damn these eyes!
17:09.12yareckonjokes don't resolve in dns very offten
17:09.24Qwellgood jokes do
17:09.32jeevQwell.
17:10.06yareckonyeah... not many of my jokes rise to the level of "good"
17:12.28yareckonok... thanks for the discussion folks.... I'll keeep lurking... and perhaps do some archaeology to see if previous projects tackled this case
17:12.37Steak__I have this dialplan... as far as I can understand it should dial the passed number after pressing the 9 for the line: '_9XXXXXXXXXX' => 1. Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
17:12.41Steak__is this right?
17:12.55Steak__OUTBOUNDTRUNK is the PSTN gw on the SPA3102
17:15.04tzafrir_laptopand BTW: *.sourceforge.net resolves DNS . But this is really pointless
17:15.22Steak__because when I dial 9+a number I only get a busy tone
17:17.50Steak__a total n00b when it comes to this dialplan coding :P
17:18.33tzafrir_laptopSteak__, OUTBOUNDTRUNK is 'SIP/something'?
17:18.57Steak__yes, in my case SIP/pstn, which is the PSTN interface on the SPA3102
17:19.18Steak__I have defined it only in my sip.conf, do you think I have to define this somewhere else as well?
17:19.40*** join/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net)
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17:24.36murdock_utQwell: So where should I go from here?  Is this considred a Mandriva issues or a Asterisk Issue?
17:24.48Qwellmurdock_ut: got me
17:24.57Qwellis it an upgrade from an old Mandriva?
17:25.04Qwellmaybe the udev rules stuff moved?
17:25.39*** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net)
17:26.01zuezjasonwoot: Do you folks open source that plugin or nah?
17:26.13murdock_utQwell: No, it's a new install.
17:26.45murdock_utQwell: There are other rules in there.
17:26.49Qwellsounds like their bug then
17:27.05tzafrir_laptopif it's "their bug": is it a udev bug?
17:27.34Qwelltzafrir_laptop: I would doubt it's upstreams bug
17:27.46*** join/#asterisk Waimanu (n=Waimanu@p5488BC04.dip.t-dialin.net)
17:27.48Qwelltzafrir_laptop: you might find this silly...
17:27.51tzafrir_laptopI have no idea what the bug is, yet
17:27.53*** part/#asterisk Waimanu (n=Waimanu@p5488BC04.dip.t-dialin.net)
17:28.00QwellYesterday, I referred to ourselves as upstream.
17:28.06*** join/#asterisk bitshuffler (n=bitshuff@unaffiliated/bitshuffler)
17:28.08Steak__tzafrir_laptop: when I try to dial 9 +  a number from my softphone it says "no such user found"
17:28.14Qwelldue to context, it was the only choice O.o
17:28.16murdock_utQwell: Let me check the release notes for 2009.
17:28.35*** part/#asterisk bitshuffler (n=bitshuff@unaffiliated/bitshuffler)
17:28.39Steak__and I do not see anything particular going on on the asterisk cli
17:29.06Steak__like the rule is not parsed or so...
17:29.43[TK]D-FenderSteak__: "sip set debug peer [yourphonespeerwithoutbraces]" <- go look at the actual SIP oncversation to see what happens.
17:29.49[TK]D-FenderSteak__: pastebin is your friend...
17:29.51[TK]D-Fender~pb
17:29.52jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
17:29.54[TK]D-Fender^^^^^^
17:29.57[TK]D-Fenderconversation*
17:30.20Steak__thanks :)
17:30.36*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:32.05LeddyHMI've created a 000000*-directory.xml for my polycom phones which is downloaded to the phone when I boot it, however I don't see anything in "Directories"
17:32.30*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)
17:32.32[TK]D-FenderLeddyHM: because that file is only downloaded ONCE the very first time the phone provisionins itself.
17:32.47[TK]D-FenderLeddyHM: its a "primer" fil, not a corporate override
17:32.56LeddyHMahh
17:33.04[TK]D-FenderLeddyHM: Polycom directories were not means to be "corporate", but rather "personal"
17:33.16LeddyHMguess I'll go back to the webpage format
17:33.19j0lol.. i think i'm having the same issue as LeddyHM
17:33.32j0if i reset the phone *456 will it reload the contacts list then?
17:33.40[TK]D-FenderLeddyHM: Web is one option, MicroBrowser is another
17:34.06[TK]D-Fenderj0: Factory + reprovision? Should
17:34.06LeddyHMYeah we had microbrowser configured for "services"
17:34.21[TK]D-FenderLeddyHM: Yeah, that would be a decent way
17:34.48LeddyHMHad a nice php script that queried our sharepoint site
17:34.52LeddyHMdamn I'm gonna miss that script
17:35.05j0i just switched to aastra's.. :)
17:35.24[TK]D-FenderLeddyHM: You use SPS?  My company is looking at it and I haven't gotten pricing in yet.  Can you give me an est to run with by any chance?
17:35.44j0my buddies list isn't updating either.. even if i go into contacts and enable buddy watch.
17:36.00[TK]D-FenderOnly think on Aastra's side is thier soft-keys, which absolutely rock.  Everything else is squarely inferior to Polycom.
17:36.15j0agreed
17:36.47*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
17:36.55[TK]D-Fenderj0: Most older model polycom's ship with firmware that is not preconfigured to support presence.  They need to be provisioned for it
17:37.25j0[TK]D-Fender: i have the buddies list and it works for those buddies in it (all 8
17:37.36j0), but i'm unable to add any more
17:37.49[TK]D-Fenderj0: What model?
17:38.02[TK]D-Fenderj0: For mass-presence I use the MicroBrowser anyway...
17:38.11*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
17:38.19j0[TK]D-Fender: 501
17:38.39[TK]D-Fenderj0: Better off on the MB with them...
17:38.56[TK]D-Fenderj0: More flexible as well
17:39.00j0[TK]D-Fender: can i re-map the Buddies key to use the microbrowser? i have a 601 and a pile of 301's here as well.. will it work with all of them?
17:39.17Steak__[TK]D-Fender: http://pastebin.com/d598919ab
17:39.22Steak__here is the output
17:39.23LeddyHMtk: it's free with our partnership. however most people only need WSS which is free
17:39.24[TK]D-Fenderj0: No, use "Services" and the 601's would work as well
17:39.51[TK]D-FenderSteak__: That does not include a call attempt...
17:40.07Steak__that are the only things displayed when I place my 9XXXXXXXXXX call
17:40.19Steak__do I have to enable other kind of logging?
17:40.32[TK]D-FenderSteak__: then your phone doesn't like what you are dialing and its not even passing it to *
17:40.41Steak__oh
17:40.42Steak__ok
17:40.50Steak__but it is strange
17:40.52[TK]D-FenderSteak__: What are you using?
17:41.14Steak__this is a normal analogue phone on the FXS port of the SPA3102, if I try to call my softphone (Xmeeting on a Mac) then it is ok
17:41.29Steak__as soon as I dial the 9 then the number that is what I get
17:41.43[TK]D-FenderSteak__: You need to configure the dialplan on your SPA.
17:41.50Steak__ok...
17:42.03Steak__is there a way to have the SPA always refer to *?
17:42.13[TK]D-FenderSteak__: login as admin on it and it should be pretty visible at the bottom of one of those pages.
17:42.16Steak__so that I can get crazy only with the * dialplans? :D
17:42.26[TK]D-FenderSteak__: Yes... set your dialplan to accept "whatever".
17:42.49*** part/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
17:42.53Steak__ok... currently the VOIP-to-PSTN is using a dialplan that is : (xx.)
17:43.00[TK]D-FenderSteak__: I would use "(x.T|#x.T|*x.T)" personally
17:43.06Steak__while the PSTN to VOIP has S0<:0417602486@192.168.25.30>
17:43.13[TK]D-FenderSteak__: No, on the FXS side, not the PSTN
17:43.13Steak__trying your settins
17:43.36Steak__dial plan on the FXS is (xx.<:@192.168.25.30)
17:43.49Steak__which I understand is "refer to 192.168.25.30 for everything" , right?
17:44.23hardwiregrr.. I hate itsp's that only gro g729
17:44.24hardwirek
17:45.51murdock_utQwell: Does Dahdi require any ptys?
17:45.51hardwiremurdock_ut: using virt?
17:46.07murdock_uthardwire: ??
17:46.12hardwirenm
17:46.35murdock_utQwell: Mandriva doesn't load any by default...
17:46.53*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
17:47.20jaytee[TK]D-Fender, the full version of Sharepoint Portal Server 2007 runs approx 4K for the server license.
17:47.31jayteethat's in US dollars
17:47.37Qwellsharepoint is freaking terrible
17:47.46KattyQwell: YOU"RE freaking terrible!!!
17:47.51QwellKatty: :(
17:47.52KattyQwell: <3
17:47.55Kattyhugs Qwell
17:47.58*** part/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell)
17:48.03Kattyhaha
17:48.16*** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell)
17:48.16*** mode/#asterisk [+o Qwell] by ChanServ
17:48.18Steak__this SPA box is driving me crazy
17:48.19Katty:>
17:48.20jayteeQwell, well I was waiting for you to write something better but the deadline for the project was almost a full year before hell freezes over.
17:48.22Kattyhugs Qwell
17:48.24Qwell:p
17:48.29KattyQwell: i loves you.
17:48.34*** join/#asterisk hi365_m (n=hi365@213.151.44.2)
17:48.39Kattyjbot: Habanera?
17:48.56[TK]D-FenderSteak__: Steak__>dial plan on the FXS is (xx.<:@192.168.25.30) <- this is not normal for the USER port,
17:49.04[TK]D-Fenderjaytee: And per seat?
17:49.17jaytee[TK]D-Fender, about 40 bucks a CAL
17:49.24jayteethat's by user or device
17:49.39Steak__[TK]D-Fender: that is on the PSTN to VOIP
17:49.39Kattyjbot: Habanera is bork Bork BORK Bork! http://www.youtube.com/watch?v=EDFgtFXfnv0
17:49.40jbotKatty: okay
17:49.46Kattyjbot: Habanera?
17:49.47jbothmm... habanera is bork Bork BORK Bork! http://www.youtube.com/watch?v=EDFgtFXfnv0
17:49.47*** part/#asterisk ant (n=ant@67-207-113-226.static.wiline.com)
17:49.56Kattyjbot: <3
17:49.57jbotsomebody said <3 was not >4, or not 3, or the emulation of a love symbol
17:49.57[TK]D-FenderSteak__: NOT PSTN!  Please follow. PST is NOT FXS!
17:50.10orkidSteak__: ru trying to set up a 3102?
17:50.14Steak__orkid yes
17:50.17*** join/#asterisk ant (n=ant@67-207-113-226.static.wiline.com)
17:50.43[TK]D-Fenderjaytee: Joy... My company is looking at SPS following my trying to just get a WIKI in use here...
17:50.58*** join/#asterisk XnOSX (n=XnOSX@212.145.55.118)
17:51.14Steak__[TK]D-Fender, sorry got confused
17:51.19XnOSXhello friends i need help for one question
17:51.21KattySteak__: don't feel bad. i stay confused.
17:51.25Steak__I know that FXS is not PSTN :D
17:51.32jaytee[TK]D-Fender, the SPS is free, you only need to spend the big bucks to have a web site interface for internet/intranet users
17:51.35[TK]D-Fenderjaytee: I wanted something free-form everyone could contribute to as a generalized information resource.  One user in marketing (a Mac non-tech person) uses SPS at her school and this shit is infecting my intent
17:51.45*** join/#asterisk glaz (n=strke@ta.blonde.et.moi.sommes.enbonstermes.com)
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17:52.01orkidi'm going to have to revisit this.. i'm doing the same. so far i have the fxs port configured as a sip extension, and can call between it and a softphone. maybe this will help you http://www.voipspeak.net/2007/guide-to-the-spa-3102/
17:52.21XnOSXif i want to playback of a audio file but i need to select a audio file with DTMF how can i make it?
17:52.25j0sharepoint is hard to beat since it's "free" most of the time, integrates well with ms office apps, and stupid techs can install it
17:52.30theharis anything that is needed to be added on zapata.conf for the echo cancel mod on digium hardware?
17:52.35glazHi, I use a queue for incoming calls, and I want to limit the calls to 1 per sip, otherwise if a SIP is already on the phone he gets an other call on the second line if someone joins the queue
17:52.38jaytee[TK]D-Fender, there's a bunch of OSS based WIKI stuff out there that can easily do that. Sharepoint is overkill for most solutions.
17:52.51Steak__orkid: that I am already able to do, thanks... what I want to do now is to be able to place calls to any number using my landline and to receive calls from the landline in the * box
17:53.06glazI use, call-limit=1 , but that makes sip user impossible to transfer calls or call someone else on the second line
17:53.14glazhow can I limit only inbound calls
17:53.20j0jaytee: do you know of any good oss sharepoint'ish app that has easy to use document versioning and filing features?
17:53.39[TK]D-Fenderjaytee: I know, I already INSTALLED Tikiwiki & MediaWiki here and was trying to sell its USE in the office.  This got transformed into a scheduling and collaboration suite for file management more than a "documentation" resource like a WIKI.
17:53.41jayteej0, not off the top of my head, no.
17:53.48[TK]D-Fenderjaytee: dangnammit
17:53.49glazcall-limit=1 limits inbound and outbound at the same time..
17:54.51jaytee[TK]D-Fender, we're using SPS for document management, reporting, group calendaring and electronic forms plus live statistics pages on zoo attendance etc.
17:54.52orkidSteak__: that link talks about it too..
17:54.53*** join/#asterisk nx5_off (n=pocholo@unaffiliated/nx5off/x-000000001)
17:55.33Steak__I'll have a look, but I think I already went through it... and furthermore, it uses the FreePBX stuff which I am not using (going .conf files here)
17:56.08glazanyone?
17:56.51*** join/#asterisk horzuh (n=horza@216.186.152.22)
17:56.52Steak__orkid: I'll try to follow that tutorial and see what happens
17:56.55Steak__thanks guys for the help
17:56.57Steak__bye
17:57.10j0ok.. i formatted the file system on my 501 and it STILL doesn't update the buddies list!
17:57.44jaytee[TK]D-Fender, what's your office environment like? Mostly Windows servers and clients?
17:59.35*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:59.51glazHow can I limit inbound calls per sip user?
18:00.05glazwithout limiting outbound calls at the same time
18:01.24[TK]D-Fenderjaytee: Much worse.  Est 40 PC's, WinXP Pro.  File server is Novell 5, due for eventual replacement by Windows Server.  Lotus Domino+Notes 7
18:01.36kate|workWow
18:02.03[TK]D-Fenderjaytee: Worst of breed :)
18:02.22jaytee[TK]D-Fender, trust me when I say that the last thing you'd want is to try and integrate SPS in that environment.
18:02.49[TK]D-Fenderjaytee: I'm not sure the words "I want to axe-murder my head-office IT dept" really do my spirit justice...
18:03.03*** join/#asterisk ddunavant (n=David@75.145.240.14)
18:03.58jayteemy boss shot down the first request for an Asterisk Bootcamp class and sent our trainer/admin/tier1 help desk person to to a $3K Sharepoint training. She did alot to help refine the system after the class but then she left when her husband got a new job in Florida.
18:04.39[TK]D-Fenderjaytee: See?  return on investment!
18:05.39*** join/#asterisk pikachu2000 (n=pikachu2@196-209-197-70-rrdg-esr-2.dynamic.isadsl.co.za)
18:06.00jayteeso now we have to limp along and no one else on our team has any real "expertise" in Sharepoint now. It's an ugly animal to configure and is kinda finicky when it comes to using LDAP authentication from MS Active Directory (should be called "i'm either out-to-lunch or took early retirement directory)
18:06.36jayteewell, at least I get to go to Advanced Asterisk.
18:06.53*** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk)
18:07.45mort_gibjaytee: Sounds a bit like a SAP implementation I has the misfortune to be sidetracked from
18:08.12jayteeOh dear God! SAP
18:08.33mort_gibAs in, I was happy to not be involved, spend £1.000.000 on an accountancy package
18:08.36KattySap.
18:08.38KattyRogue Sap
18:08.42mort_gibYes!
18:08.43KattyQwell: consider yourself SAPPED
18:08.52jayteeHi, We're going to completely change the way your business is structured to conform to OUR software that we're charging YOU a shitton of money for.
18:08.58KattyQwell: does blink get you out of SAP?
18:09.47[TK]D-FenderSAP : A product, and how it makes you feel when they rip your heart out through your wallet.
18:09.51mort_gib-Yes! Our "Financial Controller" who by the way knew EVERYTHING about IT, designed the installation along with a SAP sales rep
18:10.26mort_gibNeedless to say, he got ALL the reporting whistles and bells he wanted
18:11.05QwellKatty: not anymore!
18:11.16mort_gib-And the staff that was actually dealing with clients has an average of 11 grey-in-grey screens to go through before being able to answer simple questions
18:11.25Qwellat least, I don't think so
18:11.38mort_gibBend (yourself) and your business to SAP
18:12.07j0my polycom will not change any buddy settings.. formatted the file system, can't add to the list using the contact manager on the phone or using <bw>1</bw> in the contacts file (which it reads on boot).. any ideas?
18:12.33[TK]D-Fendermort_gib: If it makes you feel any better, we run J.D.Edwards 8.x in Green-Screen mode on an AS/400 :)
18:13.18jayteewe run J.D. Edwards but I can't remember which version. It uses a Windows client though
18:13.25[TK]D-Fenderthinks some more about axes
18:13.25mort_gibNot really, I feel with you, especially since JD. Edwards is being bent towards Oracle....
18:13.36[TK]D-Fenderjaytee: We turned that down....
18:13.46mort_gibTK I though you were into Katanas??
18:14.04[TK]D-Fendermort_gib: I am, but I'm in a Bruce Campbell kinda mood...
18:14.10murdock_utQwell: I fixed it.
18:14.16Qwellhow?
18:14.48mort_gibWell, clubs, baby seal style should be more appealing then :-)
18:14.57[TK]D-Fendermort_gib: Katana is more refined and what I would use in a more controlled assassination kind of attempt.  Axes are for when you WANT to be really messy leaving only giblets.
18:15.05*** part/#asterisk glaz (n=strke@ta.blonde.et.moi.sommes.enbonstermes.com)
18:15.15mort_gibHe, of course
18:15.16murdock_utOriginal Dahdi rule: KERNEL=="dahdictl", NAME="dahdi/ctl"
18:15.25murdock_utNew Dahdi rule: KERNEL=="dahdictl", NAME="dahdi/ctl"
18:15.43QwellO.o
18:15.54Qwellwhat...changed?
18:16.02murdock_utKERNEL="dahdictl", NAME="dahdi/ctl"
18:16.04murdock_utThere we go.
18:16.12Qwell=?  seriously?
18:16.17murdock_utYes
18:16.27j0why doesn't format file system in the polycom remove all the settings? i even set the volatile memory options for 2,4,8mb to yes in sip.cfg!
18:16.29murdock_utI was looking at the other rules in the rules dir.
18:16.29jayteeInterviewer: So what makes you think you'd be a good candidate for our RPG developer position? Interviewee: Well, I designed a MMORG for my fraternity house while I was at the university.
18:16.32Qwellumm
18:16.38Qwellwhat version of udev do they have?
18:17.18murdock_utQwell: How would I find out?
18:17.27Qwellmurdock_ut: = in the new or in the old?
18:17.35murdock_utQwell: Double = is new
18:17.37Qwellrpm -q udev probably
18:17.49Qwelludevinfo -V | cut -f3 -d" "
18:17.51Qwellwhat does that say?
18:18.21murdock_utI don't have that command
18:18.24Qwell...
18:18.33Qwelltzafrir_laptop:
18:18.41Qwellthere's your problem.
18:19.00murdock_uthttp://rpmfind.net//linux/RPM/mandriva/2009.0/i586/media/main/release/udev-128-2mnb2.i586.html
18:19.17murdock_utI think that would be the rpm for it.
18:19.22QwellDo you have any similar command?
18:20.10*** part/#asterisk pikachu2000 (n=pikachu2@196-209-197-70-rrdg-esr-2.dynamic.isadsl.co.za)
18:20.47*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
18:20.47*** mode/#asterisk [+o lmadsen] by ChanServ
18:21.38murdock_utQwell: looking
18:22.15Qwellhow about `udevadm info -V` ?
18:22.47KattyQwell: WHAT?!
18:22.53KattyQwell: no blink desapping? :<
18:23.06KattyQwell: i hope bestial wrath still gets you out of sap :<
18:23.34murdock_utQwell: udevinfo, version 128
18:23.42Qwell*groan*
18:23.47murdock_ut??
18:23.54jeevis that the noise you make when russellb slaps your ass
18:24.05Qwellmurdock_ut: I'm annoyed
18:24.10russellbblinks
18:24.14murdock_utQwell: Why?
18:24.16*** mode/#asterisk [+b jeev!*@*] by Qwell
18:24.16*** kick/#asterisk [jeev!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell)
18:24.21russellblol
18:24.23Qwellmurdock_ut: because Mandriva are stupid.
18:24.32jayteeQwell rocks!!!!!!!
18:24.35lmadsenQwell: +1
18:24.37Qwellwhy the crap would you remove a symlink?
18:24.40Kattygosh.
18:24.44jayteeQwell +10
18:24.50Qwell~paypal
18:24.51jbotpaypal is, like, at http://www.paypalsucks.com and http://www.paypal.com
18:24.58murdock_utQwell: Ok, could you please enlighten me.  I could use all I can get.
18:25.00j0ok, seriously, how can i get my polycom to delete everything and re-download the contacts.. formatting the file system will not remove the settings.. even if i delete the contacts list on the server
18:25.09Qwellmurdock_ut: we have to fix it...
18:25.26murdock_utQwell: ??  Did udev change something?
18:25.31Qwellmandriva did
18:25.35Kattyj0: polycoms will look for a directory xml
18:25.37Qwellthey removed udevinfo...
18:25.48j0Katty: i removed that
18:25.48Kattyj0: take the mac-directory.xml and put it in the boot directory
18:26.01*** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk)
18:26.03Kattyj0: you need to make it empty
18:26.05murdock_utQwell: Let me see if that command is on a old version.
18:26.06j0katty: with that removed, and formatting the fs, it still keeps the old directory
18:26.06Kattyj0: not delete it
18:26.10Qwellmurdock_ut: it is
18:26.14QwellI read their changelog
18:26.20lmadsenj0: have it there, but empty
18:26.22j0i've tried it empty.. or with different contacts too.. with and without the mac-directory
18:26.25lmadsenis what Katty is saying
18:26.33murdock_utQwell: Ok, where is this changelog?
18:26.37Kattyj0: is your polycom using dhcp?
18:26.38QwellI googled it
18:26.46Kattyj0: i can see it be quirky like that on dhcp settings
18:26.58j0Katty: yes, and i can see it trying to download all those files..
18:27.18Kattyj0: take it's settings, and stick it on static just to test
18:27.23murdock_utQwell: Link?
18:27.25jayteej0, is it writing a file in the LOG directory of the provisioning server?
18:27.27Kattyj0: i've had an issue like that before.
18:27.31Qwellhttp://google.com/
18:27.41KattyQwell: google? what is dis?
18:27.53j0jaytee: it's never written a log
18:28.09jayteej0, mine do
18:28.12Kattyj0: yeah that's not normal then
18:28.17jayteesounds like yours aren't setup right
18:28.20Kattyj0: put it on static and see what happens
18:28.25j0ok
18:28.30murdock_utQwell: Ha Ha
18:28.32jayteej0, is this a new provisioning setup?
18:29.05j0jaytee: nope
18:29.49jayteedo  you have the LOG directory setup under the /home/PlcmSpIP folder and the permissions set so PlcmSpIP can write to it?
18:31.17j0jaytee: under polycom/log, and it's a+rw
18:31.20tzafrir_laptopmurdock_ut, what I don't understand is how you can have a working general-purpose (recent) distribution without udev
18:31.25Qwellif [ "`which udevinfo`" ]; then udevinfo -V | cut -f3 -d" "; elif [ "`which udevadm`" ]; then udevadm info -V | cut -f3 -d" "; fi
18:31.31Qwellmurdock_ut: run that?
18:31.36tzafrir_laptopit must be required by a number of other packages
18:31.44Qwelltzafrir_laptop: he has udev...
18:31.56Qwellbut the script is detecting v54 because udevinfo doesn't exist.
18:31.56Kattyj0: so what happened on static?
18:31.58Qwellthey removed the symlink
18:32.18tzafrir_laptopudevinfo is in /sbin ?
18:32.20j0Katty: it's still rebooting... the worst part about owning a polycom
18:32.23Qwelltzafrir_laptop: no, they removed it
18:32.31murdock_utQwell: CLI right?
18:32.37Qwelltzafrir_laptop: run ls -l `which udevinfo` on a box
18:32.42jayteej0, and you've been able to setup phones from this server prior to this ok? it's just a problem with the "MAC"-directory.xml?
18:32.47murdock_utwhich: no udevinfo in (/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin:/usr/local/bin:/usr/local/sbin)
18:32.49murdock_ut128
18:32.49Qwellit's just a symlink to udevadm some places
18:32.58murdock_utQwell: That is what I get
18:33.02*** part/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net)
18:33.03j0jaytee: i provisioned all the phones this way.. but my buddies list and contact list wont update
18:33.19j0just finished booting.. same problem
18:34.02tzafrir_laptopQwell, the fallback should be to udev > 0.54 . Only Centos4 and such has that old udev
18:34.24jayteethe filename should be macaddress-directory.xml    you have to delete the tilde at the end of the filename.
18:34.41murdock_utQwell: So should I submit that as a bug to Mandriva?
18:35.04Kattyj0: hmmm yes, a bit slow :<
18:35.12murdock_utQwell: Could you send me a link to that chang log you were reading?
18:35.13Kattyj0: this is all internal lan, yes?
18:35.15*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:35.26Qwellhttp://pastebin.ca/1234011
18:35.38Qwelltzafrir_laptop: Does that look alright?
18:36.31*** join/#asterisk svenx (n=sven@cD12C76D9.static.bluecom.no)
18:36.36j0katty: yes
18:36.54j0jaytee: i've tried with and without that file.. it reads it fine
18:37.34svenxi'm trying to figure out how call diversion/forwarding works. the goal is that when my screensaver starts, it triggers call forward activation to my mobile..
18:38.02svenxwhen using SIP, is it asterisk that maintains the call diversion/forwarding info, or is it contained in the client?
18:38.17Qwellmurdock_ut: does running the script with those mods give the expected result?
18:38.24Qwellsee the pastebin
18:38.33*** part/#asterisk nx5_off (n=pocholo@unaffiliated/nx5off/x-000000001)
18:38.51tzafrir_laptopQwell, I wonder if it's possible to actively detect the ancient versions, and otherwise assume a decent one
18:39.12jayteej0, then why isn't it logging when you force the phone to reconfigure?
18:39.13*** join/#asterisk noni (n=aurangze@119.154.31.90)
18:39.54j0jaytee: wish i knew.. could it be because i use tftp?
18:40.00j0actually nvm.. i'm using ftp
18:40.01nonianyone can tell best site and resources for asterisk
18:40.12murdock_utQwell: Give me a sec
18:40.16nonibest sites
18:40.37j0i had write disabled.. let's try this again
18:41.50jayteej0, if you were using TFTP that would just mean it would keep redownloading the files. FTP supports date/time stamps so it won't download the same file more than once.
18:42.08Kattyj0: oh noes! another reboot!
18:42.22Kattyj0: what type of server are you using? tftp? ftp? vsftp?
18:42.22j0:)
18:42.30j0Katty: vsftp
18:42.33Kattyj0: :>
18:42.36Kattyj0: i likes vsftp
18:42.37Qwelloh, sorry...  "the script" == build_tools/genudevrules
18:42.40Qwellmurdock_ut: ^^
18:42.59jayteej0, try setting up Solarwinds TFTP (it's free) on another system, copy your configs there, set the phone to use TFTP and point it to that address and reboot it.
18:43.18murdock_utQwell: I just patched the file.  Now compiling.
18:43.33Qwellyou can just run the script directly
18:44.03j0woo log files :)
18:44.31jayteewhat'd ya change?
18:44.37Kattyj0: woohoo!!!
18:44.44Kattyj0: so what's it say?
18:44.47j0jaytee: enabled write on the ftp server
18:44.57murdock_utQwell: So far so good.
18:44.58jayteeoh? thought I'd asked that
18:45.00Kattyj0: and what's the phone doing now?
18:45.15jayteewell, yeeeehaw!
18:45.51*** join/#asterisk hi365_m (n=hi365@213.151.44.2)
18:46.06j0lots of nothing.. the log file doesn't tell me much at all.. it downloads the contacts, and that's it
18:46.11murdock_utQwell: Yup, works.
18:46.15murdock_utQwell: Thanks.
18:47.49Kattyj0: i have an idea
18:47.55Qwellgood, time to commit it and flame them a bit :p
18:48.17Kattyj0: take the ftp settings out of the phone, then format the file system
18:48.42Kattyj0: then put in a mostly blank mac-directory.xml file - all the usual bits, but not <item> enteries
18:48.45j0Katty: and then?
18:49.29Kattyj0: so it'd basically be <directory> </directory>
18:50.01murdock_utQwell: Did the change log say why they made the change?
18:50.07Qwellno
18:50.07j0Katty: what will that tell us?
18:50.32Kattyj0: well you want an empty directory, yes?
18:50.34*** join/#asterisk gr0mit (n=tim@81.187.32.146)
18:50.50j0Katty: and to update the directory in the future... as well as have a working buddy list
18:50.57*** join/#asterisk citywok (n=andrewp@65.249.42.130)
18:51.06Kattyj0: well i don't know a way to do a 'mass update' with those directories
18:51.19murdock_utQwell: Would it be worth my time to put in a bug report for it?
18:51.24Qwellalready fixed
18:51.27Kattyj0: i'm pretty sure that as soon as a phone sees the 00000000000whatever-directory file, it copies it and makes MAC-directory
18:51.49murdock_utQwell: Alright, I won't worry about it.  Thanks for your help.
18:51.51Kattyj0: then what i usually do is edited the MAC-directory files, then reboot all the phones
18:52.17j0yeah.. mine isn't using settings from either 0000* or mac-dir
18:52.25j0it downloads them though!
18:52.35j0here goes with a blank one like u said.
18:52.37j0formatting...
18:54.41Kattyare other phones having this problem?
18:54.42Kattyor just that one?
18:55.16*** join/#asterisk Greek-Boy (n=email@41.222.89.114)
18:55.20j0good question.. i'm pretty sure the 601 is working.. 1 min
18:56.51citywoki'm trying to provide a streaming-radio version of a conference bridge, and i've found a tutorial on how to do it using icecast.  is this the best solution?
18:57.20j0Katty: all the 501's seem to be having the same problem
18:57.31*** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
18:57.32*** join/#asterisk zippytech (n=ron@75.149.24.162)
18:57.57j0the 601 updates fine.. it even re-downloads its *-mac file every time and updates the contact list
18:58.04zippytechhow do you setup multi line phones?
18:58.25Greek-Boyhowcome when use the Answer() app in an extension ODBC functions don't work
18:58.32zippytechdo you create an extention on each line then use ring groups for each line?
18:58.47Kattyhmmmmmmm
18:59.14zippytechthats how i have it setup but i would like line 1 to match pot line 1 and line 2 to match pot 2
18:59.16Kattyj0: does your sip software work with the 501s and the 601s?
18:59.31Kattyj0: the software from polycom's website?
18:59.40j0Katty: i hope so! they all upgraded properly.. i stole the firmware from trixbox
18:59.51Kattymmm, that might be the problem
19:00.16Kattyj0: make a new directory, and temporarily put everything in there.
19:00.19[TK]D-Fenderzippytech: multi-line SIP phones are not key-system phones...
19:00.30Kattyj0: download, and extract, these two files into your temp ftp directory
19:00.41Kattygets links
19:01.02j0ty :) i got lost on polycoms site..
19:01.11Kattyyeah it's a smidgen messy
19:01.26jayteegets out his phone book to call Kerry Garrison and NARC on j0
19:01.47Kattyj0: http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_0_0_release_sig.zip <- sipsoftware
19:01.50Qwelljaytee: 310-861-4300...x...
19:01.51Qwell:P
19:01.59Kattyj0: http://downloads.polycom.com/voice/voip/sp_ss_bootrom/BootRom_4_0_0_release_sig.zip <- bootrom
19:02.20jayteethen remembers the hatchet job Fonality did on poor Engineer Tim and decides "screw it" and goes looking to loot their site too
19:02.27j0my phone says bootblock 2.5.0 and bootrom 4.1....
19:02.44Kattyj0: copy the one $MAC-directory.xml file into the main ftp folder and see what happens
19:03.28j0Katty: well the empty <directory> worked.. and now it's downloaded another contact list that doesn't seem to work.. maybe my directory format isn't compatible with 501s? (except it used to be? wtf)
19:03.43Kattyj0: i'm not sure about that one.
19:03.51Kattyj0: i can paste you a functional directory xml if you'd like it tho
19:03.57Kattyj0: i know it works with the 501s
19:04.04*** join/#asterisk CrazyTux (n=brandon@user-vcaun1e.dsl.mindspring.com)
19:04.09j0katty: sure.. i'll match that format and see what happens.
19:04.14Kattyj0: i'd try the new sip software and bootrom tho
19:04.18Kattyksec
19:04.53Kattyj0: http://pastebin.ca/1234032
19:05.32jayteeI don't have 501's but I remember there were some caveats about them and 301's for versions of SIP 2.2.0 and above and I'm not sure about the BOOTROM
19:05.52Kattyugah, 301s :<
19:06.24*** join/#asterisk glaz (n=strke@ta.blonde.et.moi.sommes.enbonstermes.com)
19:06.35Kattyj0: as far as i know, polycom only has one sip/bootrom for all the models.
19:06.59glazWhy if a call is ended, in my dialplan, it doesnt go to the next extension step
19:07.15glazLet say I have exten => XXX,1,Dial(bleh)
19:07.25glazexten =>,2,Bleh will never happen
19:07.41Kattyglaz: XXX,1,Dial(Bleh,timeout,g)
19:07.46Kattyglaz: also
19:07.52j0katty: this will be a downgrade from what i have.. from 3.0.1 to 3.0.0
19:07.53Kattyglaz: h,1,Dosomething
19:07.59jayteeKatty, yeah they did but now they've come out with phone specific versions of sip.ld for each phone model
19:08.09Kattyjaytee: hmm. neat (=
19:08.16jayteestill same basic one bootrom for multiple models
19:08.19Kattyj0: kay then
19:08.43Kattyglaz: g is if the destination hangs up
19:08.51Kattyglaz: h is if the caller hangs up
19:09.43*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:12.28Kattydoes 'make samples' in asterisk-1.foo overwrite stuff in /etc/asterisk/?
19:12.35glazKatty: ok
19:12.49[TK]D-FenderKatty: Yes
19:12.53glazis there a variable for both?
19:12.54Katty[TK]D-Fender: oh goody.
19:13.04Kattyglaz: no.
19:13.12glazSo I need to set both
19:13.12Kattyglaz: they do that so you can do two seperate things
19:13.15Kattyglaz: yes.
19:13.22Kattyglaz: maybe goto statements
19:13.27Kattyglaz: pointing to the sample new context
19:13.30glazyeah
19:13.34Kattyglaz: s/sample/same/
19:13.42glazWell, I just want to unpause a queue member
19:15.12*** part/#asterisk mog (n=mog@nat/digium/x-5188dbd0d1e782e3)
19:15.16glazso it would be XXX,g,4,Dosomething ?
19:15.25glazI don't get the g/h part I think.
19:15.40glazI'm trying to find an example on google
19:17.01Kattyglaz: i'll pastebin something
19:17.17glazok
19:17.49[TK]D-Fenderglaz: these are PARAMETERS for Dial.  Go read its instructions
19:17.59[TK]D-Fenderglaz: And read up on your Asterisk Standard Extensions.
19:19.35*** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200)
19:20.18Kattyglaz: http://pastebin.ca/1234042
19:20.49Kattyglaz: if you put in a timeout, it will go to the next thing automatically
19:20.57Kattyglaz: g is for destination hangs up
19:21.03Kattyglaz: h is if caller hangs up
19:21.32Kattyglaz: and core show application dial is quite handy
19:21.50glazexten => 15732004005,2,Goto this will never happen if you hangup
19:22.15glazbut thanks, I found what I needed!
19:22.17Kattyglaz: that's what the g is for.
19:24.22Kattyglaz: i use the h, bit for sending recordings of audio files to peoples email
19:24.47Kattyglaz: start mix monitor, dial the number, h to mutt it out and delete the original
19:25.16glazok, I do everything in a sh script called by MixMonitor() for this
19:25.42Katty[TK]D-Fender: make samples didn't overwrite stuff :<
19:25.45Katty[TK]D-Fender: makes me sad.
19:26.44jameswfmy next asterisk server http://www.hardcorecomputer.com/ProductCategoryDetail.aspx
19:26.44*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
19:27.13j0wow
19:27.57Kattyhttp://www.tweaknews.net/reviews/inwin_allure_case_review/img/1.jpg <- my next asterisk server.
19:28.17QwellKatty: My First IP PBX?
19:28.42KattyQwell: it's a pretty case.
19:28.52KattyQwell: i like pretty :<
19:29.11iCEBrkrhrrm.
19:29.18jameswfI wonder if they will send me a demo unit to qualify :)
19:29.18iCEBrkrI swear there were Spanish prompts
19:29.25iCEBrkrMy 'es' directory is empty :(
19:29.30Kattyoh noes!
19:29.36iCEBrkrZOMG!
19:29.58jameswfiCEBrkr: if you were in arizona i would have a funny joke for that
19:30.01Qwelljameswf: I'd prefer it over the green...
19:30.05Maliutajameswf: http://www.bb-shopping.com/prod_images/image_cat36/thumbs/300/thumb_161_image1_brick-lg.jpg  my next * server ....
19:30.23Maliutajameswf: http://windsweptsoftware.com/MyHouse/brick1.jpg and in it's dual core mode
19:30.23iCEBrkrawww shitballs
19:30.26KattyMaliuta: hehehehe
19:31.09MaliutaKatty: trust you to go for the girly option ;)
19:31.41MaliutaKatty: you don't think your case is a little, well, light on the pink? ;P
19:31.41Kattyyou say that like it's a negative thing.
19:31.54KattyMaliuta: i actually prefer the purples to the pinks.
19:32.06KattyMaliuta: thankyouVERYmuch
19:32.53Kattyi just realized i'm wearing pink today :/
19:33.00Kattyi feel like such a hypocrite now. heh
19:33.29iCEBrkrTODAY IS ALL CAPS DAY BTW:
19:33.34KattyOH YEAH IT IS
19:33.35KattyI FORGOT
19:33.37Maliutais it frilly? ;)
19:33.48KattyMaliuta: not particularly.
19:33.56j0anyone have a script to create or edit polycom contacts.xml files?
19:33.59[TK]D-Fender"Double standards : when you don't have the budget for quadruple standards"
19:34.00KattyMaliuta: it's a pink shirt with little pink hearts, and a ping kangaroo shirt over it.
19:34.12Kattys/ping/pink
19:34.52Maliutammm roo
19:35.01Maliutahas roo steaks in the fridge
19:35.08Maliutagoing to make some curries
19:35.09[TK]D-Fenderj0: Maybe somebody out there did something for this, but I've never seen anything published.  Its too small & custom for most people to care about.
19:35.20j0ty
19:35.38[TK]D-Fenderj0: Keeping in mind that directories are supposed to be personal, not corporate
19:35.44j0i'm pretty sure my problem is the polycom 501's dont understand the contacts file.. whereas everything else did
19:36.05j0[TK]D-Fender: yeah.. i realized that too late.. now i just need it to work :)
19:36.12jameswfi lied I would rather have http://www.reghardware.co.uk/2008/09/11/wacky_toaster/
19:36.14Kattyj0: if the xml directories are corporate, shouldn't there be a fancy master file instead of $MAC-directory. xml for all of them?
19:36.50*** join/#asterisk Maxous (n=Maxous@74.7.13.242)
19:37.30j0katty: i used symlinks and just edit 0000
19:37.40Greek-Boyis it possible to sms over PRI?
19:37.51KattyGreek-Boy: yes.
19:37.56Kattyj0: really? that's a neat idea.
19:38.00Kattyj0: why didn't i think of that.
19:38.02Greek-Boyhow nice :)
19:38.03Kattyj0: i've got copies of everything
19:38.23Kattyj0: oh wait. i know why.
19:38.28Kattyj0: my windows box is hosting it
19:38.45Kattyj0: didn't know how to do linux based ftp back in the day
19:38.59Kattyj0: then i got lazy and didn't want to change all the phones, which weren't on DHCP
19:39.35KattyGreek-Boy: i use the sms bit in voicemail, and in the dialplan with System( | mutt) functions
19:39.39j0lol
19:41.06Kattyclearly needs more caffeine >.<
19:41.59j0Katty: msg me ur paypal email.. i'll buy you a few cups :)
19:42.04kate|workhands Katty a glass of mt dew
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19:42.27Greek-Boykatty: nice
19:42.45*** part/#asterisk zydoon (n=zydoon@41.225.158.64)
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19:45.33*** join/#asterisk Winkie (n=urmom@ur.fa.gs)
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19:52.01cesar_CRhello guys, I have a Cisco 7911 phone where can I find a howto of this one for asterisk
19:52.02*** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150)
19:53.11Greek-Boycesar_CR: do you know the wiki? search on there...
19:53.45cesar_CRvoip-info :) Greek-Boy  ?
19:54.04Maliutacesar_CR: google?
19:54.15Maliutacesar_CR: have you _tried_ a search?
19:54.23seanbrightcesar_CR: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
19:54.29Greek-Boycesar_CR: yeah voip-info
19:54.46*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk)
19:54.52Greek-Boycesar_CR: I guess seanbright, being the kind man that he is, did it for you :)
19:55.00cesar_CRok ok I will :) thanks
19:55.18seanbrighti've never been accused of being kind
19:55.21seanbrightjust impatient.
19:56.49Maliutaseanbright: encouraging them again
19:57.12Maliutaseanbright: you need to be slapped ... I just think you might enjoy that too much
19:57.16Maliuta:)
19:58.17seanbrightyeahhhhhhhhhhhhhhhhh.... moving on
19:58.33*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
19:58.59*** join/#asterisk mark_csi (n=csiadmin@host86-131-112-45.range86-131.btcentralplus.com)
19:59.18mark_csihello
20:00.08*** join/#asterisk StephenF[W] (n=none@198.144.197.28)
20:01.00Greek-BoyI noticed that smsq doesn't get installed by default. but it doesn't appear in menuselect either
20:01.30seanbrightit gets built
20:02.02seanbrightin util/ i think...?
20:02.44seanbrighthmmm, it's installed here
20:02.58Greek-Boyhmmm, maybe I gotta enable the module
20:02.59seanbrightsame directory the asterisk binary gets dropped in
20:03.18*** join/#asterisk troubled_ (n=troubled@pdpc/supporter/sustaining/troubled)
20:03.19seanbrightis it in utils?
20:03.23seanbrightlike, built?
20:04.05murdock_utQwell: On a new * install there is a bunch of sample configs.  Most will never be used.  Is there a list somewhere that shows the required confs?
20:04.17j0well i think i found my problem.. my polycom 501 won't monitor more than 7 or 8 people in the buddy list! RAAH!
20:04.18seanbrightno
20:04.26Greek-Boyseanbright yeah its in utils
20:04.29Qwellj0: upgrade
20:04.34Greek-Boybut from the CLI i can't do smsq
20:04.36j0Qwell: to?
20:04.43*** part/#asterisk glaz (n=strke@ta.blonde.et.moi.sommes.enbonstermes.com)
20:04.47Qwellsomething recent?
20:04.49[TK]D-Fenderj0: It's not meant to, and its documented in the admin guide
20:04.56j0Qwell: firmware or phone? lol
20:04.59Qwell[TK]D-Fender: didn't new firmware remove that restriction?
20:05.05seanbrightGreek-Boy: from the asterisk CLI?
20:05.18seanbrightmurdock_ut: there is no list of what is required.
20:05.19[TK]D-FenderQwell: Not that I'm aware of... a while back it raised it, but not that much.
20:05.35j0even 20 will do
20:05.42[TK]D-FenderQwell: For a phone to track more people than it has live indication for is a rare thing.
20:05.44seanbrightmurdock_ut: there was talk at one point of a "minimal sample conf install" but no one has gotten around to it
20:06.08[TK]D-FenderQwell: 6XX's for example can support a tad more than the max consoles worth of line-keys that represents
20:06.11murdock_utseanbright: gotcha.
20:06.11jayteelmadsen, when the book was written what was the current version of Asterisk 1.4.x at the time?
20:06.28lmadsen1.4......5?
20:06.36Greek-Boyseanbright: yeah smsq is supposed to work from asterisk CLI if i'm not mistaken
20:06.43seanbrightGreek-Boy: you are mistaken
20:07.10seanbrightGreek-Boy: there is a command line utility built and installed called smsq
20:07.19seanbrightGreek-Boy: and there is a dialplan app called SMS()
20:07.28Greek-Boyok
20:07.33jayteeI'm just trying to figure out in which major and minor version release that AgentCallbackLogin was deprecated. It states that in the book but I wanted to read the corresponding UPGRADE.txt file.
20:07.35Greek-BoyI want to use the command line for starters
20:07.36Greek-Boysmsq
20:07.50seanbrightGreek-Boy: right, check in the same directory as the asterisk binary.  should be there.
20:07.57seanbrightGreek-Boy: if you did a 'make install' and such.
20:08.17Greek-Boyits in /usr/src/asterisk-1.4.22/utils/smsq.c
20:08.22Greek-Boyits not in binary form
20:08.24seanbrightGreek-Boy: ...
20:08.29seanbrightGreek-Boy: oh
20:08.39seanbrightmaybe 1.4 is different, let me look
20:09.23*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:09.32[TK]D-Fenderjaytee: Deprecation happens at the release of a new major
20:09.39seanbrightGreek-Boy: there is a dependency on libpopt, but other than that, it should build and install.  have you actually run 'make' and all that already?
20:10.11jaytee[TK]D-Fender, thanks
20:10.21*** part/#asterisk yareckon (n=yareckon@h-68-164-254-138.mclnva23.covad.net)
20:11.06jayteeI really wish they hadn't deprecated that app because I like the way it functions versus a standard agent login
20:11.20[TK]D-Fenderjaytee: cycles goes : Introduced > Deprecated > Removed.
20:11.35[TK]D-Fenderjaytee: new apps trump it...
20:11.40seanbrightdeprecation is a no-no from now on i think
20:11.46seanbrightunless there is a really really really good reason
20:12.01j0has anyone developed and released a buddy list for the polycom microbrowser?
20:12.01lmadsenagreed
20:12.05jaytee[TK]D-Fender, trump it how so? are we talking AddQueueMember?
20:12.20[TK]D-Fenderjaytee: yup, etc
20:12.47lmadsennew stance is to not deprecate and remove things; you may see them deprecated in favour of a new method, but the methods are not supposed to be removed, unless absolutely necessary
20:12.54seanbrightGreek-Boy: ?
20:14.29*** join/#asterisk Maxous (n=Maxous@74.7.13.242)
20:14.47*** join/#asterisk srf21c (n=srf@ip98-165-60-42.ph.ph.cox.net)
20:15.09Greek-Boyseanbright: yeah I have run it
20:15.26seanbrightGreek-Boy: then it should be in utils/
20:15.35seanbrightGreek-Boy: if it is not, something else is screwed up.
20:15.55Greek-Boyok
20:16.01Greek-Boyinstalling libopt now
20:16.08seanbrightlibpopt
20:16.12Greek-Boyand then rerunning mke
20:16.24[TK]D-Fenderlmadsen: so "deprecation" now is more of an implementation thing as to "ppreferred apps for a given functionality"?
20:16.25seanbrightyou'll need to re-run ./configure first
20:16.27Greek-Boyoops
20:16.30Greek-Boyk
20:16.30*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
20:16.38seanbrightinstall libpopt, ./configure, make
20:16.46*** join/#asterisk CrazyTux (n=brandon@user-vcaun1e.dsl.mindspring.com)
20:17.50Greek-Boythanks seanbright
20:17.51seanbrightdid it work?
20:18.17lmadsen[TK]D-Fender: correct -- we are going to attempt to not remove applications and functionality if at all possible, and just try to direct people to the "better way"
20:19.03[TK]D-Fenderlmadsen: Or until some core change "break" the less desirable way enough that it gets left to the wayside?
20:19.29seanbrightgod willing
20:19.30seanbrightheh
20:19.55seanbrightif there is a bug in *any* functionality, be it deprecated or not, it should be resolved.
20:20.03seanbrights/should/would/
20:21.28seanbrightbut deprecated functionality is not going to get enhancements afaik
20:22.06[TK]D-FenderThis ought to have an interesting effect on the release & dev cycle as we move from the X.Y.bugfix format to X.Y.Z.bugfix format.
20:22.41[TK]D-FenderMore to maintain, more regularly.
20:22.55[TK]D-FenderWithout necessarily any gains in so doing.
20:23.02seanbrightyeah, i'll need a raise
20:23.07seanbrightfrom $0 to something else.
20:23.08seanbrightheh
20:23.19[TK]D-FenderThis is of course the basis of Windows dev... write 8-bit code, and support it for 30 years :)
20:23.23Qwellseanbright: back to work, slacker
20:23.36seanbrightQwell: i *am* working... just not on asterisk
20:23.42Qwell*real* work
20:23.45seanbrightwriting some super nifty perl code, however.
20:23.47Qwellie; Asterisk :p
20:23.59*** part/#asterisk srf21c (n=srf@ip98-165-60-42.ph.ph.cox.net)
20:24.01PeterFAHow much would someone charge to do an Asterisk installation with a blank computer all the way up to supporting 11 lines?
20:24.20seanbrightPeterFA: $1000101
20:24.37Qwelldepends - why is the computer blank?
20:24.50PeterFAQwell, because it's a fresh new computer just ordered for this purpse.
20:24.56*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:25.04seanbrightPeterFA: what kind of lines?  analog?
20:25.27Kattyseanbright: clothes lines.
20:25.36PeterFAinternet carrier.
20:26.33lesouvageIf there is heavy recording and conversion to .mp3 files does it make sense to add a dedicated .mp3 conversion server to the setup between Asterisk and the database server?
20:26.44seanbrightlesouvage: yes
20:26.48Qwelllesouvage: don't use mp3
20:26.51Qwellmp3 is terrible for voice
20:26.55seanbrightlesouvage: that too ^^^
20:27.21j0lesouvage: mine is low volume, but i have it in cron during off hours and use 'nice'
20:28.07lesouvageQwell: What do you advice, the customer wants mp3 because it is a format that everybody  can play and is compressed to a reasonable level.
20:28.15lmadsenwav49?
20:28.37lesouvagej0: do you use sox with the .mp3 additional packages
20:29.20mark_csiCan anyone give me any pointers on writing a dialplan that limits voip to 2 channels and then falls back to pstn zap channels?
20:29.34lmadsenmark_csi: look into using GROUP() and GROUP_COUNT()
20:30.01lmadsenexten => s,n,Set(GROUP(voip)=limit)
20:30.07seanbrightlmadsen: you could dedicate a whole chapter to this two functions
20:30.30lmadsenexten => s,n,GotoIf($[${GROUP_COUNT(voip,limit)} > 2]?use_zap,1)
20:30.40lmadsenseanbright: I could also smack you with a large trout! :)
20:30.46lmadsenchooses not to
20:31.01lmadsencan't remember the GROUP_COUNT() syntax off the top of his head since he hasn't used it in a while)
20:31.03lmadsenoh!
20:31.16lmadsen${GROUP_COUNT(limit@voip)} I believe
20:31.37lmadsenexten => s,n,GotoIf($[${GROUP_COUNT(limit@voip)} > 2]?use_zap,1)
20:31.38lesouvagej0: is libmad the choice to go?
20:31.51mark_csilmadsn: thanks a lot!
20:32.02lmadsenexten => s,n,Dial(SIP/my_provider/${NUM_TO_DIAL})
20:32.04lmadsenetc...
20:32.24lmadsenexten => use_zap,1,Dial(Zap/g1/${NUM_TO_DIAL})
20:33.12mark_csiAny recommendations on how many IAX channels you could have on a DSL line?
20:33.15lmadsen30 slides for 60 mins of presentation I hope is enough... :S
20:33.18*** join/#asterisk bipolar (i=bflong@216-164-162-138.pa.subnet.cable.rcn.com)
20:34.50lmadsenmark_csi: http://www.asteriskguru.com/tools/bandwidth_calculator.php
20:35.57mark_csilmadsen: thanks again I've been crawling the net for hours and these two were beating me.  It's late here
20:36.14lmadsenis what they call.... teh awezome
20:37.22PeterFAWhat kinds of multi line VoIP desktop phones are recommended?
20:37.34PeterFANot soft phones, but the old fashioned physical kind.
20:38.03lmadsenPeterFA: pretty much anything polycom
20:38.12PeterFAlmadsen, thanks.
20:38.30lmadsensome people like the other stuff, but really... polycoms are almost the de-facto standard phone to use with asterisk
20:39.08lmadsenwonders how much better polycom's sales have been since asterisk has gotten popular
20:39.24mark_csilmadsen: thanks again and good night!
20:39.34lmadsenmark_csi: good night!
20:39.38*** part/#asterisk mark_csi (n=csiadmin@host86-131-112-45.range86-131.btcentralplus.com)
20:39.44lmadsengoes off to shower and get ready for dinner :)
20:40.59*** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec)
20:43.01*** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar)
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20:50.31hescoSince last night I've been experiencing errors when attempting to terminate a connection through diamondcard.  The details are documented here: http://pastebin.ca/1234112  My errors read: "Call rejected by 69.71.222.196: No authority found".  Is there any advice here for how to resolve this issue, please?
20:50.52jblackYou're probably not registering.
20:51.05*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk)
20:51.31hescoAs shown in the pastebin, I start with a reload which says to me: "Registered IAX2 to '69.71.222.196', who sees us as 71.236.31.107:4569 with no messages waiting"
20:51.48lesouvageI'm looking for a pointer where to look/start reading for if I want to setup a kind of  spool directory, like the /var/spool/asterisk/outgoing. The idea is to scp a soundfile to that dir on a server, have it processed to .mp3 and then with scp moved to the database server.
20:51.53jblackAs shown by the channel, I didn't look at your pastebin.
20:52.17jblackexten => s,n,Dial(IAX2/diamondcard/${ARG1},60,WK)
20:52.25jblackThat's what I use with that particular provider.
20:52.42hescowhat is the 60,WK about?
20:53.02jblackYou can find that in the help for dial on the asterisk console.
20:53.13hescothanks, looking now
20:53.47lesouvageat this moment I have no idea where to look/google/search for
20:54.11jblacklseouvage: so that you transcoede one at a time. I do that.
20:55.24*** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net)
20:55.43jblackGoogle for 2wav2mp3 , written by dietmar zlabinger
21:00.29lesouvagejblack: thanks!
21:00.38*** join/#asterisk lanning (n=lanning@66.151.128.195)
21:01.44*** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola)
21:03.09*** part/#asterisk RachedTN (n=rached@41.224.175.166)
21:04.06*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-211-196.phlapa.east.verizon.net)
21:05.40*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
21:08.55cvnetwhere can you get a toll free number with good pricing ?
21:09.59*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
21:11.12hescook, I just copied these non-working configuration from the server which is giving me these issues onto a server which is working.  These connection parameters work.  I was able to initiate a call across this trunk w/o issue.
21:11.24hescoI'm more baffled than ever.
21:12.16hescoI have two accounts with diamondcard.  They both work from  server-sandbox, neither works from  server-soon-to-be-production.  Why might that be?
21:17.48*** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk)
21:22.00ajohnsonSo I've got a weird realtime issue
21:22.18ajohnsonPeers registered to server A can call peers registered to server B directly
21:22.29ajohnsonBut peers registered to server B cannot call peers on server A
21:22.52ajohnson<PROTECTED>
21:22.52ajohnson[Oct 22 14:09:55] WARNING[23132]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
21:25.20*** join/#asterisk Juxt (n=denis@c-76-110-186-156.hsd1.fl.comcast.net)
21:25.26Juxthello party people
21:26.04Juxtquick question - how can i carry a particular channel variable from one extension to another?
21:31.44lesouvageI just found the app ControlPlayback() . Is it stable and rubust enough to use it for serious production? It looks great.
21:32.53fileit is a wrapper around the same API call that is used by Background, Playback, etc
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21:38.33Greek-Boyvery weird. Time stamp in my dcr Master.csv are appearing to be GMT
21:38.40Greek-Boybut my system time is GMT+3
21:39.50lesouvagefile: so it most be working fine?
21:40.08theharany zaptel + xen experience in here?
21:40.34filelesouvage: I have never heard of an issue with it.
21:41.02lesouvagefile: thanks
21:42.08jblackthehar: Yes.
21:42.16theharhaving a hell of a time =)
21:43.17jblackWhy are you messaging me?
21:44.01Kattybecause it's fun?
21:44.01Juxt@thehar what is the issue? i have it running just fine
21:44.13Kattyfile: ATTENTION FILE
21:44.16Kattyfile: I"M GOING TO MESSAGE YOU
21:44.25Kattyfile: not really. that was funnier in my head actually.
21:44.31fileKatty: :(
21:44.38theharlets see i don't have it running in dom0..
21:44.51Kattyfile: i haz your number.
21:44.51theharzaptel is loaded into the kernel, etc.. can't seem to get asterisk to run for the life of me.
21:45.00fileKatty: yes :(
21:45.01Kattyfile: bee vewwwwyy vewwwwy skeered
21:45.07Juxt@thehar what errors r u getting
21:45.17theharERROR[31365] asterisk.c: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection.
21:45.17filetickles Katty
21:45.20theharon stargup
21:45.30thehars/stargup/startup/
21:46.41Kattyhugs file
21:49.39*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-b7ce0cf963dade95)
21:49.39*** mode/#asterisk [+o Deeewayne] by ChanServ
21:54.13Juxtwhat's a good softphone for osx?
21:56.20ajohnsonSo I'm having a bizare realtime issue
21:56.34ajohnsonUsers registered to server a can call users registered to server b
21:56.37ajohnsonbut not vice versa
21:57.19*** join/#asterisk seanmh (n=seanmh@216.31.101.11)
21:59.01*** join/#asterisk CunningPike_ (n=arodgers@vpn.dnv.org)
22:01.34*** join/#asterisk gandhijee (n=user@mail.win-ent.com)
22:01.46gandhijeehey guys, is there something up with the asterisk now 1.5 beta?
22:01.51gandhijeei can't get to the web interface on it....
22:08.13*** join/#asterisk newmember (n=chatzill@static-66-11-81-77.ptr.terago.net)
22:09.19lesouvageWhen skiping beyond the end of the file with ControlPlayback() this message shows up :     [Oct 23 00:10:40] WARNING[4815]: format_pcm.c:150 pcm_seek: offset too large 1305440, truncating to 1291076   and the connection ends. How complicated is it to add somethng to the app like "you reach the end of the soundfile".
22:10.44lesouvageI might have a customer who is willing to pay for this patch.
22:14.09sp00k3ycan anyone tell me some common causes for garbled sound other than high overhead on the system?
22:14.51hescoI have two accounts with diamondcard.  They both work from  server-sandbox, neither works from  server-soon-to-be-production.  Why might that be?
22:15.55hescoI consistently see: [Oct 22 17:27:38] WARNING[29826]: chan_iax2.c:8667 socket_process: Call rejected by 69.71.222.196: No authority found
22:15.55hesco<PROTECTED>
22:17.17*** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp)
22:19.08hardwireI wish I could change the dialplan on my cell phone
22:22.34*** join/#asterisk MalMen (n=asdadasd@bl10-52-107.dsl.telepac.pt)
22:23.28MalMenhello, i am using trixbox... i am trying to create a sip trunk to make outbound, that sip trunk make register, but is allways unvaliable
22:23.34Qwell~trixbox
22:23.35jbot[trixbox] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
22:23.49MalMenhmm
22:23.57creativx~siptrunk
22:23.58jbotNo such thing, my friend.. Like too much salty plum soda.
22:24.01creativx;)
22:24.13MalMenwow
22:24.21MalMendefine:salty ?
22:24.24MalMenpleaseeee
22:25.07MalMeni have a virtual machine with asterisk + asterisk gui
22:25.09legishardwire: Hi
22:25.16MalMenbut dont know create extensions yet
22:25.23MalMeni am starting with voip
22:25.39hardwirelegis: howdy ho
22:26.37hescoWell, I finally got a successful outbound call through my IVR.  In my Dial() I had to give it: IAX2/userID:password@host/17707551543
22:27.53hescoI thought the idea was that I could create a context in iax.conf which would store my password and not require that I put it in extensions.conf.  But that does not seem to work.  Any ideas why it would not?
22:29.32hescothis method is putting my password directly in the logs on console.  I'm not sure to what extent that represents a security issue, but it doesn't seem right.
22:29.33*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
22:32.22hescoThis is: Asterisk 1.6.0.1 built by root @ biko3 on a i686 running Linux on 2008-10-17 06:28:09 UTC
22:32.39hescoMy Debian 1.2.13 version does not seem to require this.
22:33.20*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
22:36.43PeterFAHow much does the first time trying to configure an Asterisk server takes?
22:36.51PeterFAtime... how much time?
22:37.56ajohnsonAnyone here able to help with SIP realtime?
22:38.07*** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net)
22:39.36hescoPeterFA: that would all depend on what 'configure' means?
22:39.49hescoWhat features do you need to implement?
22:39.55creativxPeterFA: you mean how long does it take before you understand the syntax and logic?
22:40.42hescoSo am I reading this right?  Is this the correct syntax?  exten => 51,n,Dial(${TRUNK}/${PERSON_A_HM}&${TRUNK}/${PERSON_A_CL})
22:40.49PeterFATo make a functional system your first time including errors mistakes, corrections.
22:40.58PeterFAhesco, 11 line that connect to an Internet carrier.
22:41.07PeterFAhesco, VoIP all the way.
22:41.10hescoDoes that dial both his home and cell simultaneously and accepts which ever is answered first, dropping the other one?
22:41.18PeterFAhesco, no.
22:41.22PeterFAhesco, good idea though.
22:41.58creativxhesco.. [Description]
22:41.59creativx<PROTECTED>
22:42.02creativxshow application dial in CLI
22:42.08creativxa natural well of information
22:43.56X-Robhesco, you might want to look at the code I wrote to do ring groups with confirmation.
22:43.59PeterFAI'm going to read the book on the site for certain, but I don't want to include that time.
22:44.15PeterFAHow much time did everyone else's first time take?
22:44.18creativxPeterFA: depending on how complicated you want it.. from 3 days to 3 weeks
22:44.44creativxi spent a good time trying to figure out extensions/queues etc.. then moved onto how to integrate it with the AMI and our own CMS app
22:44.51hescoPeterFA: I find that the work proceeds in fits and starts.  I work out a concept through studying the docs and experimentation to implement a single feature.  That takes either twenty minutes, or if I run into snags and wind up trolling through google results a day or so for each little feature.  Once I figure that piece out, replicating what I learned (say across 11 extensions) is usually a straight-forward process.
22:45.24PeterFAhesco, ok.
22:45.29PeterFAIs there a special fax module?
22:45.37creativxPeterFA: and step 2.. learning to macrofy so you dont repeat code
22:45.53PeterFASo a guy could have a fax line that doesn't do something wasteful like packaging it in VoIP sound data?
22:46.05PeterFAcreativx, ok.
22:46.26X-Robhesco - you need to look at http://freepbx.org/trac/browser/modules/branches/2.5/core/etc/extensions.conf lines 944 onwards
22:46.27creativxour fax is rerouted at the itsp to an analouge line here
22:46.40hescoX-Rob: thanks, will do.
22:49.29riddleboxis there a list of area codes that you should not dial, I think I have heard like 809 or something like that?
22:50.02*** join/#asterisk IsUp (n=nocturne@88.242.205.12)
22:50.28*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:53.56*** join/#asterisk robba (n=robert@mail.ampwest.com.au)
22:53.59*** join/#asterisk LiNeTuX_Home (n=LiNeTuX@253.238.95.24.cfl.res.rr.com)
22:54.39robbaHi guys
22:54.44hescoWell that test seems to have worked.
22:54.57robbahaving a wierd issue with g729
22:55.09*** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150)
22:55.43hescoNext on my agenda is to resolve some cdr issues I've had since this past weekend.  I'm getting errors liks: cdr_pgsql: Failed to insert call detail record into database!\n  cdr_pgsql: Reason: ERROR:  column "calldate" specified more than once
22:56.09robbawe have our * box set to use ## as blind transfer from within asterisk. when using ulaw this seems to work fine, when using g729 it just drops the call
22:56.10robba~pb
22:56.11jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
22:56.27codefreeze-laphesco: yeah, sounds like... wait a min, let me peek at the source
22:57.23hescoWhy would it attempt to insert that same field more than once in the same query?  Where is that configured?  I've checked all the cdr_*.conf files and haven't found the issue.
22:57.25*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
22:57.43robbathis is what shows on the CLI http://rafb.net/p/2MEygI38.html
22:59.57codefreeze-laphesco: nope, it's not a dumb typo in the source. calldate is only mentioned once in 1.4 in the INSERT command. But trunk has some logic in it concerning calldate, so that must be where the prob is.
23:00.32hescoI installed just last week from upstream, version 1.6.01
23:03.10*** join/#asterisk lunaphyte (n=lunaphyt@unaffiliated/lunaphyte)
23:03.50*** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net)
23:04.59*** part/#asterisk ftp3 (n=none@pool-96-225-238-78.ptldor.fios.verizon.net)
23:05.30*** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net)
23:05.52kerxHowdy everyone, what is a good free PBX solution?  I see there is AsteriskNOW and Trixbox, out of those two which is the more better and if there are others?
23:06.05Qwelltrixbox is too green.
23:06.18IsUphaha
23:06.20IsUpright
23:06.37kerxWhat about AsteriskNOW?
23:06.41kerxToo blue?
23:06.42kerx:P
23:06.43Qwellmuch less green
23:07.14creativxgreen is the new black
23:07.17Qwellkerx: lighter weight and packaged by somebody who knows what he's doing
23:07.33jeff_smokerANYONE:: I just installed AsteriskNOW 64-bit on a server. When I am prompted for the "localhost login" I type "admin" and when I am prompted for my password I type in the same that I entered during install. I receive "login incorrect". Also, I cannot see the password as I enter it and the cursor does not move. Can anyone explain this?
23:07.33kerxQwell, well said
23:07.35Qwellall the cool kids are doing it
23:07.46Qwelljeff_smoker: root
23:07.56Qwelljeff_smoker: 1.5, right?
23:08.01jeff_smokercorrect
23:08.04Qwellroot, not admin
23:08.06kerxQwell, when you say lighter weight, does that mean less of the advanced features?
23:08.12Qwellkerx: it means less crap.
23:08.21kerxRoger
23:08.23jeff_smokerWhen I type root, how do I get to the main menu once logged in?
23:08.26kerxOk, well I'm 45% downloading it
23:08.29Qwelljeff_smoker: what menu? O.o
23:08.40kerxI realized from the Trixbox site, that there were too many hoola-hoops to download the damn .iso :P
23:08.45kerxI appreciate the insight.
23:09.15Qwellkerx: it was only a little biased.
23:09.20Qwelljust like ||.
23:10.01*** join/#asterisk glaz (n=strke@ta.blonde.et.moi.sommes.enbonstermes.com)
23:10.18jameswfQwell any word in the rumor mill about an endpoint manager for freepbx
23:10.31Qwellendpoint manager?
23:10.33jameswfthats the only feature people complain about
23:10.47kerxWhat about FreePBX vs. AsteriskNOW?
23:10.49creativxwhat is an endpoint?
23:10.52creativxin pbx context
23:11.01Qwellkerx: AsteriskNOW 1.5 (which I was referring to) uses FreePBX now
23:11.08kerxok
23:11.11jameswfa utility to do all the tftp magic and provision phones
23:11.22Qwelljameswf: got me.  I don't actually use the stuff
23:11.34glazHi, I know that if I use Dial(number,,g) the extension plan will continue if *destination* hangs up, How do I continue extension plan if *source* hangs up?
23:11.34jameswfheh
23:12.47StephenF[W]Qwell why has digum decided to use FreePBX instead of their own Asterisk GUI?
23:13.09Qwellhttp://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/
23:13.13QwellStephenF[W]: ^^
23:13.15StephenF[W]thx
23:13.16IsUph,1,NoOp(see you dude)
23:13.53kerxDo you know if there is another function that does a call transfer that doesn't establish a second SIP initiation?
23:14.03StephenF[W]hmm, I wonder if they will continue to maintain Asterisk GUI then...
23:14.04kerxI am paying for 2 x the minutes when it connect's to another phone #
23:14.07kerxIt seems
23:14.12kerxI use Dial() right now
23:15.07sah-worki have an issue. asterisk -r does not work. tells me that asterisk.ctl is not there. it is running and i can use the phones. how do i reconnect then
23:17.47hescoIs it safe to assume that I can ignore: /etc/asterisk/res_pgsql.conf if I only want to use postgreSQL for CDR?
23:19.08IsUpsah-work
23:19.13sah-workyes
23:19.23IsUpwhich user running asterisk
23:19.32IsUpand which user trying to do asterisk -r?
23:19.37*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-211-196.phlapa.east.verizon.net)
23:19.41sah-workroot
23:19.49sah-workand it has worked for months and nothing has changed
23:20.09sah-workasterisk is running as nobody
23:20.15IsUpcan you try 'rasterisk'
23:20.46drmessanorasterisk?
23:20.56sah-workuh, do not seem to have that command
23:21.48drmessanoThere is no such command
23:22.23sah-workokay am i screwed here?
23:23.15drmessanoare you using safe_asterisk?
23:23.24IsUpchmod 777 /var/run/asterisk.ctl and then 'asterisk -r'
23:24.37sah-workfuckit. rebooted
23:25.04sah-workthe thing is a mess and i need to replace it anyway
23:26.45gene2Can anyone give me a clue why my initial voicemail prompt sounds like it is too loud on asterisk-1.6.0.1? Like garbled? I've looked into that gsmbug (Fender pointed me to it, 1.6.0.1 already patched for gcc4.2) I have cisco 7960 and the prefered codec on it is 7.11 ulaw
23:27.36glazI don't really get how dial(ext,,M()) works, the M() part, I want to use CONTINUE, someone done this before?
23:28.31*** part/#asterisk robba (n=robert@mail.ampwest.com.au)
23:28.35glazIsUp: h,1,DoSomething will affect all the extention plans from the same context
23:28.40gene2~gsmbug
23:28.41jbot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily.  Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243
23:29.42LiNeTuX_HomeHas anyone ever seen different PRI cards not alarm yellow when there's a problem on the line?  Had a Rhino card showing green - changed to another device, it showed yellow... sure enough, AT&T had to change a card out....
23:29.43glazany idea?
23:29.46hescowhat all does the make samples target do?  It was after re-running make samples that I started having these cdr_pgsql issues about duplicated calldate columns
23:32.24jeff_smokerANYONE: How do you get to the GUI from the command line?
23:32.37drmessanowhat?
23:32.58jeff_smokerThe AsteriskNOW GUI that I see screenshots of on the website.
23:33.01drmessanocmd /c start iexplore.exe
23:35.50*** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net)
23:35.56*** join/#asterisk robba (n=robert@mail.ampwest.com.au)
23:36.47robbaok i discovered my problem was due to g729 sounds not being installed now i'm having another issue with call parking
23:39.23creativxwell golly gee.. copying 500k jpegs takes its time
23:40.04robbausing ## to BXFER to park, it says Unable to find a codec translation path from g729 to slin, although the call seems to park fine. although when trying to pick it up from being parked it drops the call.
23:40.12robbaCLI -> http://rafb.net/p/njeBzi27.html
23:42.56drmessanoDo you have G729 licenses?
23:44.44gene2Can anyone give me a clue why my initial voicemail prompt sounds like it is too loud on asterisk-1.6.0.1? Like garbled? I've looked into that gsmbug (Fender pointed me to it, 1.6.0.1 already patched for gcc4.2) I have cisco 7960 and the prefered codec on it is 7.11 ulaw
23:50.51robbaone license
23:51.57jameswfverry importannt everyone on Nov 4th
23:52.06jameswfgoes out and buuys a blackberry bold
23:52.29robbaonly testing on one hand set on one call
23:52.38*** join/#asterisk gnarf37 (n=gnarf@west-rock.rockriver.net)
23:52.43gnarf37hey guys
23:52.54*** join/#asterisk nikko (n=nikko@adsl-074-182-164-013.sip.hsv.bellsouth.net)
23:52.56gnarf37so, I'm getting ready to redo our asterisk servers
23:53.01robbanot much of a fan of the blackberry
23:53.13gnarf37and I'd really like to work with 1.6, but I'd also like to look/play with asterisk-gui
23:53.35gene2and the problem
23:53.51gnarf37supposedly, the asterisk-gui doesn't work with 1.6
23:54.22gene2ic.
23:54.41gnarf37I don't mind having to help fix it to work with 1.6  :)
23:54.55gnarf37just curious if anyone can point me in the right direction
23:54.58gene2if gsm is a codec and 711ulaw is a codec too, can a file be encoded in 711ulaw?
23:55.40jayteegnarf37, why not check out the beta of AsteriskNOW 1.5? it's CentOS based and supports either 1.4 or 1.6 and has the FreePBX gui?
23:56.02jayteegene2, yep.
23:56.21gene2jaytee: what would be the default extension for ulaw then?
23:56.26jayteeyou can download all the system prompts in ulaw format or record your own
23:56.47jayteegene, the default extension is (drumroll) .ulaw
23:57.16gene2Comedian mail voicemail prompts are gsm by default and they sound garbled someone suggested changing them to something else to see if it will fix it, so i've attempted to convert to wav and they still sound like shit
23:57.17gene2thanks
23:57.20gene2let me give it a try
23:57.51kerxDo you know if there is another function that does a call transfer that doesn't establish a second SIP initiation?
23:57.57gnarf37jaytee: did they decide to drop asterisk-gui in favor of freepbx?
23:58.07kerxI am paying for 2 x the minutes when it connect's to another phone # it seems
23:58.09kerxI use Dial() right now
23:58.36jayteegnarf37, for AsteriskNOW 1.5 yes I believe they decided that, at least that's the only gui in the beta.
23:59.26jayteekerx, 2 x? so you're transferring an inbound call to another number outbound?
23:59.50kerxNo, if somebody makes an outbound call and transfer's them to another number

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