IRC log for #asterisk on 20100729

06:59.57*** join/#asterisk infobot (~infobot@rikers.org)
06:59.57*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta1 (2010/07/23), 1.6.2.10 (2010/07/23), 1.4.34 (2010/07/23), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs
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08:26.35Mark22Hello, I have 2 asterisk servers (one in a datacenter with a public IP (no NAT) and one in an office behind a NAT on a DSL connection). The one in the office registers with the one in the datacenter. However we have the problem that one of them says that the other down is and the connection isn't restored automatically. How can I find what the problem is and what the solution is?
08:28.43*** join/#asterisk sulex (~sulex@93-35-11-23.ip52.fastwebnet.it)
08:31.08zoacould you paste how it says that ?
08:32.17zoadoes you office connection have a fixed ip ?
08:34.51Mark22http://yourpaste.net/5855/ << that is on the datacenter side, at the office asterisk server nothing is shown
08:35.02Mark22the connection does have a fixed IP
08:36.04Mark22after shutting down the asterisk server at the office and waiting for 5 minutes and starting that asterisk server again it works (but for how long is the question and I want to solve it for the future if possible)
08:37.35*** join/#asterisk suneeel (~suneeel@115.252.84.170)
08:45.07zoaok so the connection is actually gone
08:45.20zoado you have an idea what type of nat it is ?
08:46.21zoadid you put a qualify in sip.conf ?
08:48.36zoalook up the nat timeout for your firewall and make sure the value in qualify=$value is less
08:48.49zoathat should keep the connection up
08:48.52*** join/#asterisk krion (~seb@unaffiliated/krion)
08:49.59Mark22it is nat on a simple dlink modem, the connection is gone (but only 1 asterisk server "knows" it is gone and the other says the connection is up). at both servers I have set (in sip.conf): qualify=yes
08:51.06zoadecrease this value "defaultexpiry" on both servers to be under 30 seconds
08:51.15zoai think your nat is timing out
08:52.24zoaqualify=30000 should work as well with less load on the server (as long as the ip doesnt change on the office side)
08:52.53zoaoh wait
08:52.57zoai might be saying stupid things there
08:53.48zoaqualifyfreq is the value you want to change
08:53.52zoanot the qualify value
08:54.17suneeelwondering if anyone here has faced call progress detection problems with the tdm400p board
08:54.25*** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net)
08:55.10zoaMark22, without changes the sip connection should go back up after 120s though i think
08:55.16suneeelMy dialplan starts executing way before the other end picks up an outbound call...
08:55.53Mark22it did re-register, however it didn't work :S
08:55.56suneeelAm I asking this in the correct place?
08:56.01zoasuneeel, how does you dialplan look like ?
08:56.44zoaMark22, asterisk will retry the qualify every 60 seconds
08:56.51Mark22zoa: should I place qualifyfreq=30 in sip.conf under the general section?
08:56.52zoaand the reregistration every 120s i think
08:57.05zoashould work there
08:57.11zoabut doesnt need to be there i think
08:57.24suneeelwell, I'm actually issuing an originate over AMI and directing it to an AGI script that plays a prompt to the callee. But the AGI starts executing before the called party picks up
08:57.29zoai'd start with the defaultexpiry and turn of the qualify
08:57.33Mark22i did see that, it did register for 2 minutes and after that was unreachable for 1 minute (but in reality it didn't work at all)
08:57.47zoai think you now have
08:57.53zoanat timeout after 30 seconds
08:57.57zoareregister after 120
08:57.58suneeelI read on one of the discussion boards that this is because the 400p assumes that the call is connected as soon as the exchange picks up..
08:58.01zoaand qualify after 60
08:58.19zoawhich means that even if the register worked, it will take up to 60 seconds before the qualify will approve it
08:58.29zoaand by then the nat might have timed out again
08:58.37suneeelSo I'm guessing i need to set up the correct call processing tones in the config in order to detect the remote end off hook
08:58.58zoasuneeel, i wont be able to help you with that one im affraid
08:59.42suneeeldamn! well thought it was worth a try in any case.. Any idea where I could look for more info?
09:00.11Mark22could it be a solution to use port forwarding and stop using register (but just configure the servers at both ends)?
09:00.22zoai'd say stick around and ask here again in an hour or so
09:00.43zoaMark22, i wouldnt do that, you would need to do too much forwarding for the rtp
09:00.51suneeelbtw.. The same behaviour is exhibited when I put use a dialplan context instead of an AGI.. Which is why i said dial plan i the first place
09:01.04suneeelOh ok
09:01.06suneeelCool
09:01.08zoaaha
09:01.08suneeelWill do that
09:01.23zoaso its not ami related (I'm not familiar with AGI)
09:01.52zoasuneel can you paste the dialplan part somewhere and the cli output ?
09:02.22suneeelk, gimme a sec
09:03.08zoause pastebin please
09:03.15hrhrhr_anything worth shouting about in 1.8?
09:04.08suneeelOk, someone's gone and rebuilt my asterisk box.. Should have checked that before comming on here I guess
09:04.25suneeelWill get back to you guys tommorow
09:04.28suneeelThanks
09:04.49Mark22and it isn't working anymore :( I need to fix it today or we will loose this client :S
09:06.44zoaMark22, just try the defaultexpiry, i think it will work
09:07.01Mark22I did already change that
09:07.06zoano luck ?
09:07.11Mark22no luck :(
09:07.21zoathat doesnt sound good
09:07.29zoawhat dlink is it
09:07.30zoa?
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09:07.42zoacheck if is has some option for statefull packet inspection
09:07.46zoaor packet rewriting
09:08.02zoasome of the dlinks are known to fuck up sip
09:08.17zoatake a tcpdump on the server when you try dialing and see what it returns
09:08.30Mark22I did just recheck it, it is a THOMSON ST546 if I may believe the web interface
09:08.41*** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk)
09:08.52zoaaha, thomson is better
09:09.22zoaif you get lucky you will see am icmp unreachable when trying to connect to the office pbx
09:10.10zoaif you want to rule out sip rewriting issues on the thomson (not sure if it does that, i dont know that model), use iax2
09:11.18*** join/#asterisk soman (~somnath@118.102.130.6)
09:14.15Mark22ICMP echo reply, id 53481, seq 27076, length 44 << looks normal to me, now I did get a "Auto fallthrough, channel 'SIP/1010-0000005e' status is 'CONGESTION'" in the logs
09:14.27Mark22is iax2 more stable compared to sip?
09:15.52zoanot really, helps a bit for situations where you have no audio because of nat, but you get stuck before that
09:17.00zoai'd remove the qualify completely for now, and just use the defaultexpiry
09:17.03zoaon both servers
09:17.30zoaand try iax2 if that doesnt work as you are in a hurry
09:22.19*** join/#asterisk nilclass (~niklas@pD4B9E2F7.dip.t-dialin.net)
09:29.47*** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com)
09:43.01mpeHi do any of you know to set the timeout that is used with asterisk attende transfer *2
09:43.01mpeat the moment the call timeout after 15 sec, where I get this message
09:43.02mpefeatures.c:1957 ast_feature_request_and_dial: We exceeded our AT-timeout
10:00.40*** join/#asterisk Yoda_1204 (~chatzilla@static-96-225-251-220.ptldor.dsl-w.verizon.net)
10:11.59garymcis making my phone show 01412341234 instead of 1412341234 a really hard task to do? I can live without the 0 but it would be nice if I could get the phone to show the 0
10:12.17garymchence the full number
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10:38.18zoawhere do you want to show that ?
10:38.26zoayou mean the caller id for incoming calls ?
10:39.27zoahttp://www.voip-info.org/wiki/view/Setting+Callerid
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11:03.28zoahierse puzzled
11:03.32zoalang geleden
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11:04.59puzzledhey zoa. how are things?
11:05.12zoagoooood
11:05.14zoaand you ?
11:05.57puzzledbesides a massive headache I'm good
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11:19.17*** join/#asterisk sputnick (~sputnick@unaffiliated/sputnick)
11:19.23sputnickhi there
11:19.25puzzledzoa: is zoiper communicator biz 64bits available for Fedora 13? I only see 32bits F10 releases on the website and F10 is EOL for quite some time
11:20.23zoai will check just a sec
11:21.38zoanopez, only for ubuntu apparently
11:21.52zoawill check to make a new one
11:22.01zoawe are a bit understaffed because of the holidays
11:22.04zoai will put it on the todo
11:22.09puzzledno problem. thanks
11:22.47sputnickis there a particular setting to call a mobile number directly to a voicemail ? Because I have had a bug on a asterisk/nagios/dedicated server calling my mobile phone every 5 minutes but never ringing, my French provider Free ask me 89€ this month just for this.
11:24.11sputnick$116
11:24.44puzzledzoa: the Windows version of Communicator Biz is 32bit too?
11:24.48zoayes
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11:24.51sputnickuse SIP
11:24.57zoadoesnt really make a difference to run them in 64 bit
11:25.06zoaonly for DSP Stuff
11:25.25zoathey should run fine in 32bit on 64 bit machines
11:25.41puzzledzoa: ok, ta
11:25.59zoaits a little bit faster, but not a lot and we usually dont use over 4gb of ram :)
11:26.23zoasputnick
11:26.30zoasip cannot do that
11:26.38zoabut, depending on your cellphone provider
11:26.42zoathere might be an option to do that
11:26.55zoawith some dtmf voodoo
11:26.59zoaor special numbers to dial
11:27.01zoai have seen it before
11:27.42zoayou might have to call your cellphone from another cellphone from the same provider though to make it work
11:27.58sputnickzoa: I have no option from Free provider, nor special number, but I have a call every 5 minutes since june 26 that cost me a bit each
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11:29.15*** mode/#asterisk [+o leifmadsen] by ChanServ
11:30.33zoayou will need to check with your mobile operator
11:30.49zoai doubt you can do anything on Asterisk to do that
11:31.02zoaah look
11:31.05zoaits mister leif
11:31.15puzzledindeed
11:31.52zoagood morning sunshine! :P
11:35.13leifmadsenzoa: :)
11:35.17leifmadsenis now writing documentation
11:39.31prgmrchrissputnick: there has to be a way, http://slydial.com/ does exactly what you are describing and im sure they use asterisk to do it
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11:44.51zoasputnik is from france
11:47.05zoathis is how to do it for one us provider: http://www.ehow.com/how_4787651_send-message-directly-voicemail.html
11:47.19zoai imagine that slydial does such a trick for every provider
11:49.05zoabut why would you send it to voicemail ? You will still pay for it ?
11:51.54*** part/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net)
11:53.29prgmrchriszoa: only thing is how does slydial know which provider a mobile number is on? with lnp i dont think thats easy unless you have some kind of inside info
11:53.37prgmrchrisi doubt they are just bruteforcing it
11:54.38zoanumber portability db or so :)
11:54.48zoai investigated such a functionality for belgium before
11:54.56zoaand did not find a common api
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12:01.52jkroonhi guys, I've got three BRI (B410P) cards in ptmp mode, now they were up just now, except for two lines, then swapped the two dead ports with two others now all four is dead.
12:01.54jkroonany ideas?
12:02.00jkroon(dead => RED alarm)
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12:23.16jkroonok, the only way to get them pack up is to reboot.
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12:27.05*** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com)
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12:39.01DNDguys im getting segfault from asterisk: segfault at 000000000ee04000 rip 00002aaaad526d66 rsp 0000000040bc35a0 error 4
12:39.12DNDim not sure where it came from. i just typed "dmesg"
12:39.22jkrooncan be anywhere.
12:43.05[TK]D-FenderDND: Going to tell us anything about your install?
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12:54.51garymc[TK]D-Fender : have you any idea how long it would take for a polycom software guy to decide if they would add RFC3326 or dismiss it? I know you dont work for polycom but would you have an educated guess?
12:54.54*** join/#asterisk cusco (~trilili@33.83.136.95.rev.vodafone.pt)
12:54.57cuscohi all
12:55.22garymcas the feature request is open and active. just no notes on it
12:55.42cuscoasterisk 1.6.2 on a debian stable machine is keeping a big list of opened file descriptors
12:56.36[TK]D-Fendergarymc: No, and guessing is pointless
12:57.02[TK]D-Fendergarymc: link me to the request page
12:57.26[TK]D-Fendercusco: And that isn't a version #
12:58.56garymcits passworded
12:59.15garymchttps://jira.polycom.com:8443/browse/EXT-3029
13:00.08garymc[TK]D-Fender : Have you got an account ?
13:02.18[TK]D-Fendergarymc: No
13:02.53garymc[TK]D-Fender : does the link ask for a password?
13:03.10[TK]D-Fendergarymc: indeed
13:03.34garymc[TK]D-Fender : Thats the end of that then
13:05.34garymcJust wondering if i would have to add much or how difficult would it be for me to get the 0 to show on my phones. When someone calls me from say 01412342345 my phone show 1412342345
13:06.00garymci know its no biggie, but just wondered how I would do it and what file I need to be in
13:06.12leifmadsengarymc: modify the CALLERID() ?
13:06.15cuscohttp://paste.debian.net/81729/
13:06.34cusco[TK]D-Fender: Connected to Asterisk 1.6.2.9
13:06.49cuscothis was hapenning with 1.6.2.8
13:06.58leifmadsenlots of file descriptors are normal especially if the system is under load
13:07.01garymcleifmadsen i dont know how to do it
13:07.06[TK]D-Fendergarymc: It shows no 0 because there is no 0 coming in
13:07.09leifmadsengarymc: core show function CALLERID
13:07.12[TK]D-Fendergarymc: Add one yourself
13:07.29garymcok how do i add it?
13:07.31[TK]D-Fenderleifmadsen: FreePBX <-
13:07.39leifmadsen[TK]D-Fender: oh then I'm done here
13:07.45garymcyeah but its an asterisk thing is it now
13:07.48garymc*not
13:07.49[TK]D-Fendergarymc: No
13:08.07leifmadsenIn Asterisk, I'd just do:  Set(CALLERID(number)=0${CALLERID(number)})
13:08.08garymcoh n3glv said it was an asterisk thing
13:08.14cuscoleifmadsen: lots of udp connections, it seems that i doesn't terminate them after the calls
13:08.14[TK]D-Fendergarymc: it is dialplan, not device config.
13:08.26leifmadsencusco: check the bug tracker to see if that is an open issue
13:08.28[TK]D-Fendergarymc: n3glv said a LOT of things.
13:08.35garymcits dial plan so it displays the incoming number with the 0?
13:08.47garymcyou wernt there I dont think
13:08.50[TK]D-Fendergarymc: You modify CallerID in the dialplan.
13:09.01garymcok ill go over to freepbx
13:09.10[TK]D-Fendergarymc: Probelm is you have to integrate this right to work with your GUI
13:09.16[TK]D-Fendergarymc: And we don't want to deal with that./
13:09.28garymcok, so its alot of hassle for me, then for you too :P
13:09.44[TK]D-Fendergarymc: Yes, you are a lot of hassle for me...
13:09.49garymc:P
13:10.03garymcyou will grow to love me
13:12.44*** join/#asterisk diegomad (~mad@190.147.221.78)
13:12.51garymcor maybe not
13:12.54zoagarymc, i asked you before and i sent you in the right direction
13:12.57zoascroll up please
13:13.05garymcyou did ?
13:13.25zoa<zoa> http://www.voip-info.org/wiki/view/Setting+Callerid
13:13.26garymcI got disconnected and had to reboot
13:13.34zoaaha
13:13.42garymcmy chatzilla doesnt cache
13:13.56garymcnot sure if i can get it too either
13:14.22zoaso its possible and you can either go read on the set command or you can cheat and listen to leif :)
13:14.50garymcthat link says thats for outgoing only
13:16.28*** join/#asterisk FlashDeluxe (~FlashDelu@static-87-79-94-28.netcologne.de)
13:16.53FlashDeluxeHi, Ive got a problem, if i want to dial i get an error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
13:17.06FlashDeluxeany suggestions?
13:18.20[TK]D-FenderFlashDeluxe: Tell us useful details about what you've got, configs, and the actual failed call to look at
13:20.50FlashDeluxeOk, thx :) i`ve asterisk 1.6.2.10 installed with dahdi-linux-2.3.0.1 and dahdi-tools-2.3.0
13:21.03FlashDeluxethe configs are default
13:21.21[TK]D-FenderFlashDeluxe: PASTEBIN <-
13:21.23[TK]D-Fender~pb
13:21.24infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
13:21.32[TK]D-FenderFlashDeluxe: "Default" is meaningless
13:21.58garymcleifmadsen : could I just add Set(CALLERID(number)=0${CALLERID(number)}) to sip_custom.conf file?
13:22.01*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
13:22.09[TK]D-Fendergarymc: No
13:22.11leifmadsengarymc: probably not
13:22.19[TK]D-FenderDEFINITELY not
13:22.22leifmadsengarymc: I don't use FreePBX so I will be of no further assistance
13:22.24garymcok
13:22.26FlashDeluxeokay. extensions.conf: [default]  exten => 015771360963,1,Dial(DAHDI/g1/${EXTEN})
13:22.27[TK]D-Fendergarymc: This has nothing to do with SIP
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13:22.45garymcok
13:22.48[TK]D-FenderFlashDeluxe: PASTEBIN.  You have multiple things to be showing us and you haven't told us what you're using./
13:22.54leifmadsenFlashDeluxe: first of all, that's dangerous -- you should not allow outgoing calls via [default]
13:22.54garymcIs there I file I would add Set(CALLERID(number)=0${CALLERID(number)}) too?
13:23.05leifmadsengarymc: please use #freepbx -- we can't support it here
13:23.15[TK]D-Fendergarymc: extensions(something).conf
13:23.28[TK]D-Fendergarymc: And all sorts of other GUI changes to point things there
13:23.32FlashDeluxeits just for testing, i will change it later if it works
13:23.37[TK]D-Fendergarymc:  ...
13:23.39[TK]D-Fender~wglwat
13:23.39infoboti guess wglwat is well, good luck with all that
13:23.51[TK]D-Fendergarymc: 2nd door to your left.
13:23.55FlashDeluxewait a minute i will show you my configs
13:23.56prgmrchrishaha
13:24.04garymcim in there :S
13:24.09prgmrchrisgarymc is at it again
13:24.24[TK]D-Fenderprgmrchris: Never really stops.. he just slows down a little on occasion
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13:25.41prgmrchris[TK]D-Fender :)
13:26.59drmessanoIf I went to the pound and they had a dog that was missing its left legs, I would buy it, name it garymc, bring it home, set in the middle of the living room, and watch it run around in circles all day.
13:27.03leifmadsen#asterisk is not 2nd level #freepbx support though
13:27.39garymcok
13:28.01drmessanoleifmadsen: #trixbox Is Tier 2 Windows Vista support, however.
13:28.17garymcA dog missing its left legs would find it hard to stand up, why not try a boat with one paddles
13:29.20drmessanogarymc: "try a boat with one paddles" <-- PLURAL, and a good indicator of your underlying lack of understanding
13:29.30Carp1I have an old install of asterisk and when I choose 1 to update it says its timing out trying to reach the server.
13:29.33garymctypo
13:29.39drmessanoPLURAL
13:29.50[TK]D-FenderCarp1: There is no such thing as "1 to update"
13:30.05leifmadsensounds like a GUI problem
13:30.07Carp1Whoops, I think it's AsteriskNOW
13:30.16leifmadsen302 Redirect #asterisknow
13:30.18[TK]D-FenderCarp1: 3rd door to your left ...
13:30.23Carp1Sorry.
13:30.23drmessanoIf I had a boat with one paddles, I would be up the creeks
13:31.24[TK]D-Fenderdrmessano: WITH a paddles!
13:32.13drmessanoSorry, I was typing with one hands
13:32.32eppigygood morning
13:32.52eppigyi too have challenges typing with one hand
13:33.34drmessanoeppigy: Can you row a boat with one paddles?
13:33.40drmessanoThen your in luck
13:33.49garymcdrmessano was your other hand touching your vagina's ?
13:34.10[TK]D-Fenderdrmessano: Of course.  I've canoed for almost 20 years :)
13:34.18drmessanogarymc: No, she's at work
13:34.27[TK]D-Fendergarymc: His vagina's what?
13:34.38garymcyou have another half? whats his name?
13:34.57drmessanogarymc: You're going nowhere with this.  Stop while you're ahead
13:35.15garymcwhats up, i thought this was "Lets take the piss time"
13:35.45eppigyi will never understand that term
13:35.48drmessanoI'm not into golden showers
13:35.49eppigytakin the piss
13:35.54drmessanoThat's pretty gross
13:35.55[TK]D-Fendergarymc: so far you're gender, plural, possessive, and grammatically challenged :)
13:35.56garymcits a uk thing
13:36.01eppigyoh i know
13:36.11garymcno need to gang up on me
13:36.30garymc[TK]D-Fender your not the other half thats at work are you? :P
13:36.36eppigyrude
13:36.47[TK]D-Fendergarymc: "that's"
13:36.48drmessano[TK]D-Fender: You left off the part about being GUIetically challenged :)
13:37.02drmessano"You're"
13:37.03garymcZZZZzzzzz
13:37.24[TK]D-Fenderdrmessano: Thought I'd leave you one ;)
13:37.51eppigythis mcdonalds coffee is actually pretty tasty
13:37.56drmessanoI would call garymc on Skype and tell him to his face, but I don't have weeks to help him set it up
13:38.03eppigyloool
13:38.11eppigyburn
13:38.13Carp1you got to goto "Stewarts Shops" for coffee
13:38.20Carp1I think they're only in NY though
13:38.20zoapuzzled
13:38.24zoawas that for communicator ?
13:38.31eppigyYeah I live in hotlanta
13:38.33garymcyou got more than weeks , well you would need more than weeks
13:39.01drmessanogarymc: Thank you for the props, mcgary.  Word.
13:40.01puzzledzoa: yes the comm. biz pro/enterprise uber version
13:40.43garymcdrmessano : word? like WORD UP! ?
13:40.56garymcOWW
13:41.37[TK]D-Fendergarymc: Haven't you heard?
13:41.43[TK]D-Fendergarymc: THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD...
13:41.45[TK]D-Fender...BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE...
13:41.46[TK]D-Fender...WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!
13:41.53garymclol
13:42.07garymcI like Peter Griffins take on this wonderful song
13:42.47garymc[TK]D-Fender : ?pastebin
13:42.54garymc!pastebin
13:43.15kaldemar~pb
13:43.15infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
13:43.24kaldemar[TK]D-Fender: ^^^^^
13:43.40[TK]D-Fenderkaldemar: Pardon?
13:44.06[TK]D-Fenderkaldemar: that was 1 line which IRC's silly limit split in 3.  Thus not OVER 3.  FAIL :p
13:44.39kaldemar[TK]D-Fender: showed up as 11 on my tiny screen. but i was obviously just kidding. :)
13:45.14*** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk)
13:45.40kaldemari'll re-read on a 1800x1600 console.
13:47.15puzzledzoa: if it was not clear I mean the commercial version and very much prefer a 64bit version for F13
13:47.31zoaah we dont have biz versions for it i think
13:47.42zoaneed to modify the shop a bit for that :/
13:47.47zoaalso on the todo :)
13:47.55zoawill see what i can do
13:47.59puzzledzoa: ok. i that case I'll settle for what you have that works on F13 x86_64
13:48.06puzzledthanks
13:48.08zoaam already installing f13 here
13:49.35*** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net)
13:49.57coppicedoes anyone here get a problem when using FAX for Asterisk that the FAX appears to complete OK, but the resulting TIFF contains only the first centimetre of the image?
13:50.52joobiehttp://tinyurl.com/36vp3fv
13:51.21joobiecheck it out
13:51.29*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
13:51.37*** kick/#asterisk [joobie!~chatzilla@216.191.106.163] by [TK]D-Fender (joobie)
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13:51.46joobie:P
13:51.56joobiei take it u clicked it TK
13:51.57[TK]D-Fenderjoobie: Don't ...
13:52.02joobieheeh ok
13:52.21joobieyou inspired me
13:52.29joobiewith your THE BIRD BIRD BIRD THE BIRD IS THE WORD
13:53.07FlashDeluxehi, can anybody show me his chan_dahdi.conf? :)
13:53.26Carp1what ever happened to NuFone?  I havn't been on the internet in a long time.
13:53.29drmessanoNice, NSFWOH
13:54.18seanbrightwelcome to 1998
13:54.28seanbrighthe's been holding that pose a long time
13:55.26drmessanoseanbright:  Waiting..
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14:02.05FlashDeluxeHi@all: Ive got a problem: When i edit the chan_dahdi.conf by adding a 'channel => x-x' all of the dahdi commands are not longer available, they do not exist :( but as i delete the line 'channels => x-x' all of the dahdi commands do exist :S?? Any suggestions? My system is:asterisk 1.6.2.10 installed with dahdi-linux-2.3.0.1 and dahdi-tools-2.3.0
14:02.24seanbrightpastebin your entire chan_dahdi.conf file
14:02.29seanbright~p
14:02.30infobot[p] q and not q
14:02.30seanbright~pb
14:02.31infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
14:03.58FlashDeluxehere it is http://paste.debian.net/81737/
14:13.46chazzamif you run 'dahdi_cfg -vvv' what gets turned up?
14:14.25*** join/#asterisk patrick^ (~patrick_@dhcp-0-24-14-f5-ac-e2.cpe.mountaincable.net)
14:16.17chazzamFlashDeluxe: ^^
14:16.25FlashDeluxenothing
14:16.27FlashDeluxeDAHDI Tools Version - 2.3.0
14:16.36FlashDeluxethats all :(
14:16.41chazzamdoes it say "No channels to configure" or anything?
14:16.56FlashDeluxeno it says nothing^^
14:16.56chazzamyou have to configure the driver before asterisk can configure the channels
14:17.29FlashDeluxedamn...the maschine freezed
14:17.41FlashDeluxei have configured it
14:18.41chazzamit sounds like you may have lower level problems than your asterisk config
14:19.39FlashDeluxei guess sp -.-
14:20.20[TK]D-Fender[10:17]<FlashDeluxe>i have configured it <-- doesn't look like
14:21.47chazzamwhat card do you have?
14:22.52FlashDeluxetwo no name hfcs [TK]D-Fender: But i configured it, but after that i installed a new asterisk version and since that it freezes
14:23.18[TK]D-FenderFlashDeluxe: show use "dahdi_cfg -vvvv"
14:24.15FlashDeluxeon moment plz, the machine is still booting
14:24.22hrhrhr_when did they rename it to daddy
14:25.31[TK]D-Fender~dahdi
14:25.32infobot[~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav
14:26.19FlashDeluxeOK, after i executed the command i get this: http://paste.debian.net/81739/
14:27.03FlashDeluxeand then the machine freezed
14:27.11tzafrir_laptopFlashDeluxe, segmentation fault by daudu_cfg? That's bad
14:27.34chazzamheh daudu
14:27.36*** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net)
14:27.49FlashDeluxei know^^ but what can i do?
14:27.59*** join/#asterisk yabadabado (~root@static-213-115-44-229.sme.bredbandsbolaget.se)
14:28.08yabadabadohmm
14:28.09tzafrir_laptopThere's a patch or two I got for bug reports
14:29.52*** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br)
14:31.15FlashDeluxetzafrir_laptop what for patches?
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14:38.20Kattyguten morgan
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14:38.42FlashDeluxemoin
14:38.42seanjohnchan_sip.c:13649 handle_response: Remote host can't match request CANCEL to call '176314020af796be26ef805e45dd962d@173.50.101.11
14:38.49seanjohnmorning
14:38.58seanjohnterrible yesterday and today
14:39.15seanjohnthe audio quality went from perfect to choppy as hell
14:39.23seanjohnI see that in the damn manager
14:40.05seanjohnanyone know what's going on?
14:40.31zoausually it means too much harddisk activity
14:41.16seanjohnok zoa. that would help my diagnosis
14:41.21*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
14:41.34seanjohnone of the hard drives in raid 5 is failing
14:41.51seanjohnI can't tell which one as noone can tell me any tool for linux to do a smart
14:42.13seanjohnotherwise, I would just remove it
14:42.19zoais it hardware raid ?
14:42.25seanjohnnope
14:42.28zoaif so, ask the raid controller
14:42.29zoaah damn
14:42.50seanjohnits only one bad sector
14:42.56seanjohnand linux is having a fit
14:43.11zoahmm now that i think of it the raid 5 in software might already not be a very good idea
14:43.23zoawhat are you doing that requires raid 5 ? do you monitor a lot ?
14:43.35seanjohnno, just for things like this
14:43.57seanjohnbackups; its 4 drives in raid 5 and two can fail
14:44.24zoak
14:44.39seanjohnbut the damn thing won't go ahead and fail so that I get an error on post
14:44.40zoawhich means you should be able to pull 1 more out and it should still work
14:44.46*** join/#asterisk af_ (~getsmart@78.134.21.122)
14:44.48zoaso you could find the bad one with trial and error :)
14:45.31seanjohni need to take the whole array out and use the backup drive but I think I need to ask others how to copy WHOLE partitions from one to another. I'm a windows geek
14:45.59seanjohnwith the array out, I can test one at a time
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14:47.04seanjohnthis isn't the proper channel for this
14:47.11seanjohnbut I would appreciate some commands
14:47.48seanjohntar up the whole drive? that's not going to restore the compiled programs and yum installs (centos 64) is it?
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14:50.33kl4mI have a question concerning ztmonitor
14:50.45seanjohnzaptel who uses zaptel anymore
14:51.51anonymouz666those stuck on 1.2
14:51.56*** part/#asterisk sekil (~sekil@80.93.247.26)
14:53.07seanjohn1.4 is just a dialplan change away
14:53.10seanjohnlol
14:54.41[TK]D-FenderLOL...... NO
14:55.19KavanSsersly?
14:56.03kl4mso anyway, I have a local line and asterisk is sending milliwatt() over it. Shouldn't ztmonitor show ~14000 tx? it shows only 4600
14:56.03seanjohn[TK]D-Fender: just joking
14:56.25seanjohntrying to keep myself from bashing my machine
14:56.46seanjohnNEW hard drives and one is failing??? western digital raptors
14:57.10seanjohnI thought the days of "1 in 10 fail within the first month" are gone
14:58.12zoaseanjohn i cant help you with that raid stuff
14:58.15zoai only do hw raid
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15:13.47Ken85hi. can you suggest any good voip cordless phones? I'm thinking of buying one
15:13.48nnypssh thank god
15:14.19nnytcpdump + wireshark + Telephony Tools + Graph Data = foolproof test to show that DTMF is being sent in RTP. Hooray!
15:14.48WIMPyKen85: Don't go for wifi, go for dect.
15:14.57nnyfun part is you can reconstruct calls and play them back too
15:15.16nnySnom M3 is ok, had some glaring support issues when one I had died though
15:15.18Ken85WIMPy: yeah i dont prefer wifi although i dont know what dect is.. whats dect?
15:16.21nnyhttp://en.wikipedia.org/wiki/Digital_Enhanced_Cordless_Telecommunications
15:16.49Ken85nny Nice. do you have any recommendations?
15:16.50*** join/#asterisk Tim_Toady (~moi@178.128.17.211.dsl.dyn.forthnet.gr)
15:17.28[TK]D-Fender~wifivoip
15:17.29infobot[~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended.  Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc
15:17.33[TK]D-FenderKen85: ^^^
15:17.38nnyKen85: I have used the snom m3 extensively. It's a bit light and simple interface. (Almost like a cheap cell phone) only issue I had was it took me 3 weeks to return one, their support process is a bit convoluted
15:17.57nnythey have a new one out too I think, but haven't used it.
15:18.02[TK]D-FenderM3 = range and battery issues (from reports)
15:18.11Ken85nny: how about the siemens gigaset?
15:18.19nnyKen85: haven't tried it
15:18.20zoaeverythin on wifi is miserable on battery
15:18.28nny[TK]D-Fender: 150 feet is about right
15:18.44nny[TK]D-Fender: after my last support nightmare I am looking for a new one anyways
15:18.47WIMPyKen85: If you're concerned aboput privacy, you could take a look at encryption used by various models on www.dedected.org.
15:19.14nnyKen85: if you get the siemens let me know, my business partner suggested it recently
15:19.31Ken85nny: i want something for home use
15:20.02Ken85just a phone which i ll be able to do my landline calls with normal phone and also be able to do sip phone calls
15:21.26WIMPySiemens have combo things for both POTS and SIP, but the one I have doesn't work with multiplae phones.
15:21.55WIMPy(althouh it supports five or was it six)
15:22.30FlashDeluxeNow I`ve reinstalled dahdi, current release and when i execute dahdi_genconf i get 'Empty configuration -- no spans Empty configuration -- no spans' i guess that zaphfc misses? What can i do?
15:24.29Naikrovekanyone know a good replacement for a wrtp54g
15:24.39KattyNaikrovek: a cookie.
15:24.47KattyNaikrovek: snickerdoodles.
15:24.56Naikrovekwhile tasty, these do not meet my requirements
15:25.30*** join/#asterisk intralanman (~lanman@va-67-76-163-226.sta.embarqhsd.net)
15:25.32Kattywell you didn't list any requirements :P
15:25.36Kattyjust a good replacement ;P
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15:28.00Ken85WIMPy: yeah i want that combo hybrid thing. i'm mostly searching. i dont want to waste money. i want to be able to use lowratevoip with it. it uses G.711 so i suppose its fine. i dont believe that there would be any lags
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15:28.52crowb4rHey so google is comming up a little short on a good example og asyncagi. Anyone have some links handy with good examples of using it?
15:30.40*** join/#asterisk CrimsonX (ccc1490a@gateway/web/freenode/ip.204.193.73.10)
15:31.56CrimsonXwhy doesn't the name show up correct on inbound h323 callerid?  it shows the far end IP instead
15:33.14[TK]D-FenderFlashDeluxe: Genconf is for chan_dahdi, not the core configs.  /etc/dahdi/ <-
15:33.20[TK]D-FenderFlashDeluxe: system.con, etc
15:35.03tzafrir_laptop[TK]D-Fender, actually, aso /etc/dahdi/system.conf
15:35.14tzafrir_laptopBut it does not run dahdi_cfg on its of
15:35.16tzafrir_laptopown
15:36.02FlashDeluxei made reinstalled everything and i executed the patch you gave me this morning tzafrir_laptop
15:36.25FlashDeluxebut now it doesnt work
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15:42.22carrarsnickerdoodles?
15:42.29carrargimmie
15:45.00traxxhi. i have the following in my dialplan: exten => s,n,Dial(DAHDI/1/1234567)
15:45.21traxxi'm calling in from an external number and would like my callerid to be transferred to the new channel
15:45.26traxxany ideas how to do that ?
15:45.51[TK]D-Fendertraxx: it is unless your provider or tech prevents you
15:46.38zoayou might need to tweak it a little
15:46.52[TK]D-Fendertraxx: Which given you are dialing out a single channel... I'm wonding what youa re actually using...
15:46.53zoaeg. if it comes in with international prefix, but your provider will only allow national prefix etc
15:47.09zoaah yes, you need pri or bri
15:47.13traxxgoing into that context from an internal number, i can manipulate the CALLERID(num), but not when calling from extern.
15:47.16zoaand bri usually doesnt allow a lot of numbers
15:47.29zoatraxx, that should not matter
15:47.37zoaunless you hardcoded the callerid somewhere
15:47.58[TK]D-Fendertraxx: and I'm not seeing a call to debug or description of what you're using
15:48.16carrarYou're not reading between the lines!
15:48.21traxxok, maybe i should paste that context
15:48.34[TK]D-Fender~pb
15:48.35infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
15:48.36[TK]D-Fender^^^
15:48.56timeshell_atworkI have an issue with 1.6 asterisk built in 64 bit that doesn't appear in 1.6 asterisk built in 32 bit.  The issue occurs when a SIP trunk on the 64 bit build is unable to connect to it's remote host.  The result is that the whole chan_sip appears to experience severe lags or hang ups.  However, this doesn't occur on the server with 32  bit build.  Both servers are using CentOS 5.5, 64 bit...
15:48.57timeshell_atwork...and 32 bit respectively.
15:50.10zoatimeshell_atwork, your text is a bit confusing
15:50.18timeshell_atworkWhen I say server lags or hang ups, this is manifest in that all the SIP phones experience delays or inability to receive or make calls.
15:50.27*** join/#asterisk Ad-Hoc (~nimbus@62.1.166.114.dsl.dyn.forthnet.gr)
15:50.30zoaso you have 64b and 32b each trying to connect to the same 3rd server ?
15:50.41timeshell_atworkNo
15:50.53timeshell_atworkI have 2 servers.  a 64 bit and a 32 bit.
15:51.21zoaand the 64b calls the 32b ?
15:51.32timeshell_atworkWhen the internet routing is not available for any SIP trunk, chan_sip appears to become unresponsive on the 64 bit server.
15:51.39zoaaha
15:51.43timeshell_atworkHowever in the same situation on the 32 bit server, this doesn't happen.
15:52.11zoai cant think for any reason for this
15:52.31zoaare you sure the rest of the config is the same ?
15:52.32timeshell_atworkThey do have sip trunks to a common server, but that doesn't appear to be relevant as the issue also occurs on a trunk that loses routing that's not related to the 32 bit server.
15:52.34zoadns etc ?
15:53.05zoacant help im affraid
15:53.09timeshell_atworkThey are both using very similar configurations, same version of asterisk.
15:53.23zoawhy dont you go to 32 bit as a workaround ?
15:53.27zoado you need >4gb memory ?
15:53.41timeshell_atworkI've thought about it.  I just don't want to rebuild the server.
15:53.42QwellI highly doubt it's an issue with bittedness
15:53.49zoame too
15:53.51timeshell_atworkI'm open to suggestions.
15:53.57zoai think i was the very first person to run asterisk in 64 bit
15:54.07zoathe only issue i had was md5 in iax2
15:54.41zoaits probably still somewhere in the mantis :)
15:54.54timeshell_atworkI've tried several things to get around this issue, even creating a local server to run the internet connections and then connecting the 64 bit server to connect to it.  however, even when that local link is unavailable, chan_sip still appears to have issues.
15:55.16Qwellzoa: 1804
15:55.27Qwellor not
15:55.40zoacant you run 32 bit on the 64 bit machine for just a second to see if it will work ? That would confirm that it is an issue with 64bit and not the config or the machine
15:56.02*** join/#asterisk Jinxed- (93b128b1@gateway/web/freenode/ip.147.177.40.177)
15:56.06*** join/#asterisk paulc (~paulc@unaffiliated/paulc)
15:56.11Jinxed-what is the default login/password for asterisk now
15:56.14Jinxed-centos
15:56.16timeshell_atworkI haven't tried compiling 32 bit on 64 bit
15:56.30*** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227)
15:56.33zoahttps://issues.asterisk.org/bug_view_page.php?bug_id=0001174
15:56.35timeshell_atworkI'll look into it
15:56.36zoathis one :)
15:56.59QwellJinxed-: freepbx/fpbx - This is listed in the quickstart guide.  You read the quickstart guide...right?
15:57.09*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:57.09*** mode/#asterisk [+o leifmadsen] by ChanServ
15:57.59Jinxed-Qwell: where is the quickstart guide
15:58.21*** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net)
15:58.22zoaqwell, do you still remember that bug that causes all asterisk machines to die in the middle of the night ?
15:58.29zoai'd love to see some irc logs for that night :)
15:58.33Qwellhttp://www.asterisknow.org/AsteriskNOW-1.5-QuickStart
15:58.33zoapeople trying to wake up mark
15:58.40Qwellzoa: I remember it
15:58.44CrimsonXI was on the phone, sorry
15:58.53Qwellwas like right before astricon, IIRC
15:59.04zoai dont remember that or what the date issue was
15:59.19Qwellsome timestamp rolled over
15:59.21zoaaaah those memories :)
15:59.42traxxok, here's the pastebin: http://pastebin.com/Knq2G8gd - at the top is the context, bottom the output. if i dial extension 701 from ie 702, the callerid gets set, no problem, if i dial in extension 701 from outside, i get the base number of the pbx, and not the cid i want.
15:59.47zoathings are done a bit better now :)
16:00.00Qwellthere was an issue with a certain android phone.  the camera focus would work for 28 days, and then not work for 28 days
16:00.17CrimsonXMy setup is an h323 trunk to Avaya, and when an Avaya station call into the asterisk the CIDnumber shows up correct but the name is the IP of the Avaya instead of the caller name that is passed in the setup from the Avaya
16:00.43aidinbdahdi, oh dahdi!
16:00.50zoayeah i remember that one
16:01.01zoatraxx, what card is that ?
16:01.21zoatry grepping all your config files for the callerid that does show up
16:01.35zoao
16:01.43traxxzoa: from lspci: 01:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
16:01.46zoa701 from 702 fets the caller id because its all sip
16:02.09zoais that a 10$ modem ?
16:02.23zoaor is that a bri card ?
16:02.35traxxzoa: it's bri
16:02.50drmessano~wyd
16:02.51infobotWho's your DAHDI?
16:02.51zoaah its the wildcard
16:03.16*** part/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net)
16:03.22traxxi'm getting the impression that it might be a channel inheritance problem
16:03.25zoawhat country are you in ?
16:03.29traxxzoa: .de
16:03.30zoai doubt it
16:03.40zoahow many numbers can you use on the pri ?
16:03.43zoaeuh bri
16:03.44zoa8 ?
16:04.16zoaactually
16:04.22zoaso the call comes in on the bri
16:04.28zoaand gets sent to the internal number ?
16:04.53zoayour pastebin is the other way around
16:05.02zoathe way you describe will work
16:05.04WIMPytraxx: Tell your provider to enable "CLIP no screening".
16:05.05zoathe other way not so sure
16:05.29zoaWIMPy, they won't, they will only allow a few numbers on bri
16:05.42zoai think you can only pick callerid's on pri freely
16:05.58zoa(not a tech limitation, but a provider limitation)
16:06.11WIMPyzoa: Usually they will, but they will charge.
16:06.13zoabri is usually not for telco itnterop
16:06.16traxxlike i said, calling 701 from intern redirects to the mobile number. the CID gets sent. calling 701 from outside redirects to the mobile number. i get the CID of the pbx.
16:06.18zoanot in europe afaik
16:06.40zoaah
16:06.43zoathat makes sense
16:07.01zoayou will need to set the callerid manually
16:07.13WIMPyIt only available on ptp, however, not ptmp.
16:08.01zoaso the mobile number sees 701 in the first case and in the second case your pstn callerid ?
16:08.42traxxzoa: exactly
16:09.00zoaok
16:09.32zoagoogle for Set Callerid
16:10.30*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)
16:10.58traxxzoa: yeah tried that (asterisk callerid redirect) but i get others with the same problem and no solution
16:11.33zoahttp://www.voip-info.org/wiki/view/Setting+Callerid
16:11.45zoawill help you
16:13.13traxxtried that already. doesn't work. same effect. wouldn't know what to set the other things like name, ANI, etc.
16:13.44zoathat thing should be enough
16:13.51zoado a debug on the bri
16:13.56zoain both cases
16:13.58zoaand compare them
16:14.05zoawhen you see a difference, come as us again
16:14.31zoaask us i mean
16:14.45*** join/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net)
16:14.59LemensTSis libpri included in dahdi now?
16:15.09Qwellno
16:15.23LemensTSok seen it was libpri-1.4.X ...curious thx
16:15.34carrardidn't read the notes did you :)
16:15.46LemensTSdidn't even download it
16:16.48*** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227)
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16:22.22[TK]D-Fendertraxx: I'm not seeing CALLS
16:22.49traxx[TK]D-Fender: how would i output them ?
16:22.57traxxdebug ?
16:22.59[TK]D-Fender~pb
16:23.00infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
16:23.01[TK]D-Fender^^^^^^^^^^
16:23.08[TK]D-Fendertraxx: * CLI like everything else.
16:24.10traxx[TK]D-Fender: sorry don't understand. did you miss the calls in my pastebin ?
16:25.33[TK]D-Fendertraxx: No description of what tech is used, no debug for that tech itself.
16:26.47traxx[TK]D-Fender: i 'core set debug 20' on the CLI, but the output is the same as in my pastebin
16:27.20[TK]D-Fendertraxx: that is not a CHANNEL DEBUG option
16:30.02*** join/#asterisk zbyniu (~zbyniu@ip-62.181.188.13.static.crowley.pl)
16:30.35zbyniuhello
16:30.49carrarHARRO
16:31.21zbyniuI have problem similar to reported here: https://issues.asterisk.org/view.php?id=17693&nbn=8
16:31.37zbyniubut with versions 1.6.2
16:32.21zoai doubt it will be the bug
16:32.26zoapaste your config please
16:32.50zbyniuit looks like context is cutted somewhere in flow between chan_dahdi. and pbx.c
16:33.08zbyniuzoa: uh, it's big, but look
16:33.36[TK]D-Fenderzbyniu: PASTEBIN <--------------
16:33.38[TK]D-Fender~pb
16:33.39infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
16:33.41zbyniui have E1 configured with context=from-zaptel
16:34.00zoak
16:34.53zbyniubut in log it's destroyed: VERBOSE[2738] pbx.c:     -- Executing [s@tel:1] Ringing("DAHDI/2-1", "") in new stack
16:35.38zbyniumy analog lines context=from-internal are cutted to "ernal"
16:36.06zoahmm
16:39.10zoacan you make some noops
16:39.16zoaand some ${CONTEXT} printing ?
16:39.48zbyniusure, but where?
16:40.13Carp1on extension 11, 11 would be my sip username?
16:40.29[TK]D-FenderCarp1: #freepbx <-------------
16:42.00zbyniuCarp1: call executing (broken name) context doesn't have proper extension no more
16:42.05[TK]D-Fenderzbyniu: that bug report doesn't look like your descriptiona nd you aren't showing proper backup for it
16:43.17zbyniu[TK]D-Fender: https://issues.asterisk.org/file_download.php?file_id=26750&type=bug look here
16:43.29zbyniuStarting DAHDI/1-1 at ,s,1 still failed so falling back to context 'default'
16:43.50zbyniuit's almost the same, but context is empty here
16:44.13zoahmm true
16:44.37[TK]D-Fenderzbyniu: pastebin a SINGLE call and your configs
16:45.03zbyniuok, w8
16:47.29ChannelZok we w8 4 u lol!!1!
16:47.29*** join/#asterisk Asinus1223 (~MVCoon@adsl-190-81-134.asm.bellsouth.net)
16:47.57Asinus1223Hello everyone
16:48.14Asinus1223Is there a place to look up stoopid newbie questions
16:48.23[TK]D-FenderAsinus1223: bash.org
16:48.26ChannelZThe Google
16:48.56pabelangerzbyniu: pb a full debug log
16:49.12pabelanger~collectdebug
16:49.13infobotrumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue.  Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
16:49.16pabelangerzbyniu: ^^
16:49.30Qwellhmm
16:49.40Qwell~collectdebug @ pabelanger
16:49.45Qwelllame
16:49.52Qwell~collectdebug > pabelanger
16:49.55*** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender
16:49.57Qwelloh well
16:50.31drmessanoCan't he just tell you how it SOUNDS?
16:50.35drmessanoor LOOKS?
16:50.42drmessanoWhy all this "DEBUG" crapola stuff
16:50.51drmessano????11!!!1!!!???
16:51.02zoahe already told us how it smells
16:51.23drmessanoThen why hasn't some developer fixed it yet?
16:51.56pabelangero.0
16:52.03Asinus1223I have a TMD400 I can't get a dial tone off of the FXS port, and I was just wondering if I have the wires plugged in right.
16:52.16[TK]D-Fenderdrmessano: You'd love this one... a new girl in marketing called me over to look at her phone problem.  She says whenever she was calling this one person she'd get a EHN EHN EHN EHN sound. I looked at the phone for a second and told her "Every heard of a BUSY SIGNAL?"
16:52.19Asinus1223Should I plug the FXO prot prior to the demarq?
16:52.33zoaAsinus1223: if its an fxs, try to connect a normal phone
16:52.42[TK]D-FenderAsinus1223: did you connect the molex?
16:52.53Asinus1223should it give a dial tone without any fussing with the config files?
16:52.59drmessanoLOL
16:53.07Asinus1223yeah, the molex was the fiurst thing I did
16:53.42Asinus1223and my head is buzzing with channels and extensions and contextx
16:53.50zoawe need a new person to say NEXT!
16:53.58Carp1NEXT
16:54.02Qwell~NEXT
16:54.03infobotsomebody said next was NEXT!
16:54.19carrarPREVIOUS
16:54.24Carp1~infobot
16:54.25infobotcarp1, i love abuse, feed me!, or whack, yo
16:54.29Asinus1223CURRENT
16:55.00carrarbetter yet
16:55.04carrarPENULTIMATE
16:55.07[TK]D-FenderAsinus1223: Did you initialize DAHDI?  Confirmed the channels were configured and ack'd by * CLI?
16:56.28zoado you have fxs modules on that card ?
16:56.41zoaor am i completely mistaken ? :)
16:56.57Asinus1223I don't have the `dahdi show channel `command available
16:57.03zoaoh oh
16:57.16Asinus1223and yes, I have one green FXS module and one orange FXO module
16:57.28[TK]D-FenderAsinus1223: No DAHDI running in * = no dialtone
16:57.37[TK]D-FenderAsinus1223: Fix
16:57.38Asinus1223ah yes.
16:57.49Asinus1223lsmod shows dahdi runninng
16:58.07zbyniuhttp://pastebin.com/s0hAqndG
16:58.07Asinus1223I guess it isn;t runing "in asterisk"
16:58.11[TK]D-FenderAsinus1223: Irrelevant.  thats like having a working motor... that isn't installed in the car
16:59.53carrarWould the engine light come on without a engine?
17:00.07Naikrovekif the computer is still there, yes
17:00.11Asinus1223OK  is there some doc that shows how to have asterisk access the dahdi dirvers?  I thought this was taken care of by the `dahdi_genconfig` command
17:00.30[TK]D-FenderAsinus1223: did you look at the result?
17:00.39[TK]D-FenderAsinus1223: Did you check that chan_dahdi.so is loaded?
17:00.46[TK]D-FenderAsinus1223: Did you try to load it manually?
17:00.53[TK]D-FenderAsinus1223: Why is the sky blue?
17:01.09[TK]D-FenderAsinus1223: What is the average air-speed velocity of an unladen swallow?
17:01.19[TK]D-FenderAsinus1223: Who shot J.R.?
17:01.38Asinus1223Yeah, I looked at the result.  It produced a couple of files; /etc/dahdi/system.conf and the other /etc/asterisk
17:01.51[TK]D-FenderAsinus1223: And what did YOU produce? :)
17:02.00drmessanoIt was Kristen, Sue Ellen's sister
17:02.06carrar24 miles an hour
17:02.09drmessanoBitch.. hated her
17:02.11[TK]D-Fenderdrmessano: NO SPOILERS BITCH!
17:02.16Asinus1223ozone particles
17:02.16drmessanoHA
17:02.35drmessano[TK]D-Fender, I JUST RUINED THE 1981 TV SEASON FOR EVERYONE
17:03.01zbyniupabelanger: ok, i'll try to cut configs to some minimal set and pb it if http://pastebin.com/s0hAqndG is not enough
17:05.19*** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica)
17:05.41[TK]D-Fenderzbyniu: Poor.  unload the module. Load it.  Show us the channel definitions.  Retes
17:05.45[TK]D-Fenderretest*
17:06.24*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
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17:10.13*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
17:10.13*** mode/#asterisk [+o leifmadsen] by ChanServ
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17:16.06zbyniu[TK]D-Fender: chan_dahdi? nothing special: http://pastebin.com/xdZSczYE
17:16.46[TK]D-Fender[Jul 29 19:13:22] WARNING[3257]: chan_dahdi.c:17018 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35.
17:16.48[TK]D-Fender^^
17:16.49[TK]D-FenderPARDON?
17:17.00[TK]D-Fenderzbyniu: Go look in USERS.CONF while you're at it
17:17.02Qwellasterisk-gui
17:17.12[TK]D-Fender##$#$*$*$%#%(@&#^)!@#^#_&^@#$_*#^($(_#^&()^&*)@#^&^@%#$
17:18.13zbyniu[TK]D-Fender: and? hassip = yes
17:20.56Naikrovek[TK]D-Fender: watch this right up to the very last second, and laugh a bit: http://www.funnyordie.com/videos/ed36fa1ab6/between-two-ferns-with-zach-galifianakis-steve-carell
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17:45.03zbyniuhmm, myDebugLog generated via HOWTO_collect_debug_information.txt instruction has 311MB, are you sure I should paste it somewhere? :)
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17:48.36tzafrir_laptopzbyniu, the core file itself? no
17:48.45tzafrir_laptopthe trace from the gdb - yes
17:50.05tzafrir_laptopzbyniu, WARNING[3074] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (No such device)
17:50.30tzafrir_laptopCould you please pastebin the output of:  lsdahdi      ?
17:50.36zbyniutzafrir_laptop: i can enable it if you want, doesn't matter
17:51.28tzafrir_laptopzbyniu, you seem to have definitions in users.conf as well
17:51.59tzafrir_laptopIf you rely on chan_dahdi.conf and extensions.conf, I would suggest that you remove / move aside users.conf
17:52.05stixWhat does this reply mean? SIP/2.0 488 Not acceptable here
17:52.17tzafrir_laptopOr at least the parts of it that define DAHDI channels
17:52.37anonymouz666stix: could be a codec offer that result this
17:52.45stixhmm okay
17:53.28*** join/#asterisk RobH (~robh@wikimedia/RobH)
17:53.50zbyniutzafrir_laptop: http://pastebin.com/wxphDqjC  - users.conf
17:54.13stixmight be what it's saying here: Capabilities: us - 0x2 (gsm), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
17:54.27anonymouz666combined NOTHING
17:54.49stixtrying with alaw
17:55.01tzafrir_laptopzbyniu, if that's all, it's harmless
17:55.05stixwee! :)
17:55.06b11d`had a compiler warning (in 1.8-b2) about 'strudpa' in sig_pri.c -- modified strupda to ast_strdupa and that fixed the problem.. chan_dahdi loads now and so does pri.
17:55.16b11d`dunno if I should submit a bug report for such a small fix..
17:55.28b11d`it was causing chan_dahdi to not load in FreeBSD 8.1-amd64 anyway
17:55.29tzafrir_laptopb11d`, please do
17:55.36tzafrir_laptopOn what platform is this?
17:55.49b11d`FreeBSD 8.1-amd64
17:56.05zbyniutzafrir_laptop: yes, that's all
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18:04.26b11d`bug report submitted
18:07.00*** part/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net)
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18:12.58malcolmdb11d`: and a "real" diff attached.  thank you :)
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18:16.48b11d`no, thank you :)
18:17.55*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
18:17.55*** mode/#asterisk [+o leifmadsen] by ChanServ
18:20.36Kattyih
18:20.38Kattyhi
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18:24.18leifmadsenKatty: ohai!
18:24.26wcselbyo/
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18:27.10b11d`yay I contributed :)
18:28.15*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
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18:29.44Kattyhugs leifmadsen
18:29.51Kattyhugs wcselby :> o
18:30.23Kattyso what fragrance do you guys wear.
18:30.27Kattywhat's your favorite
18:30.33leifmadsenwow... my router has decided it wants to be chinese...
18:30.33wcselbywhatever soap my wife buys
18:30.42leifmadsenI wonder how the heck I switched the language on it....
18:30.57malcolmdleifmadsen: how do you know that someone in china didn't decide that for you?
18:31.16leifmadsenmalcolmd: I don't? :)
18:31.37zbyniu[TK]D-Fender,tzafrir_laptop,pabelanger: https://issues.asterisk.org/view.php?id=17753
18:34.06*** join/#asterisk theron (~theron@ip244.scolloc.lh.net)
18:37.16tzafrir_laptopzbyniu, hmm.... can you pastebin the output of:   dahdi show channel 1      ?
18:38.05wcselbyi've got a client that their dahdi pseudo device stops working with some frequency, preventing them from being able to access any meetme bridges.  they are still able to make and receive calls over their T1 lines.  where should I begin looking?  I'm getting a cli trace ready
18:38.42*** join/#asterisk evilbit (~evilbit@n1-20-152.dhcp.drexel.edu)
18:39.45tzafrir_laptopwcselby, what does dahdi_test show at the time of such a hang?
18:40.11tzafrir_laptopDo they have more than one T1 line on that box?
18:40.44evilbitif I'm in a context called internal and I want to dial a extension in a context of local-call do I need to include local-call in the context of internal?
18:40.48wcselbytzafrir_laptop - yes, 3 lines going into a TE420p, and a 4th line that runs into a brooktrout board on a separate hylafax server
18:41.00wcselbyit's a te420, don't remember the last number
18:41.28wcselby4 port T1 PCI-e card with echo cancel board
18:42.25[TK]D-Fenderevilbit: Yes
18:43.04evilbithmm, actually I have that set... I am unable to dial out via my iax provider. They are saying it's because I'm not setting the CID but AFAIK I am
18:43.36[TK]D-Fenderevilbit: If you say so
18:43.47evilbitSet(CALLERID(number)=${OUTCID})  should set it right? If in sip.conf for my user I have SetVar=OUTCID=<my did here>
18:44.59zbyniutzafrir_laptop: http://pastebin.com/psVUEWwK
18:46.26wcselbytzafrir_laptop - http://pastebin.com/z5TBpAqD requested output, plus cli from the failed meetme attempts
18:47.40[TK]D-Fenderevilbit: You might want to try actually looking at the call
18:48.13wcselbyI can bandaid the issue by stopping asterisk, then dahdi, the starting dahdi, then asterisk.  30 seconds of no calls though.  we usually have about 15-20 concurrent calls going during the day though
18:48.15evilbit[TK]D-Fender: well, I have and I can see it go out and * says it's accepted by the remote end but then nothing happens :-(
18:49.06[TK]D-Fenderevilbit: And you are showing us nothing.  Expect our ability to assist to be proportionate
18:49.17evilbitsorry... 1 sec
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18:50.35tzafrir_laptopzbyniu, there's something a bit unusual that you do: you don't provide an actual extension, and rely on falling back to 's'
18:50.56tzafrir_laptopWhat happens if you also provide a catch-all extension '_X.' ?
18:50.57evilbitit just stops here: http://pastebin.com/Ly48QQq4
18:52.08tzafrir_laptopzbyniu, btw: nice catch
18:54.43[TK]D-Fenderevilbit: Who are you calling?
18:55.16evilbitwell, right now I'm trying to call my cellphone (but it's just for testing... I can't call any number)
18:55.49evilbitit just stops at the "Format for call is ulaw"
18:57.01*** join/#asterisk sylar (~sylarrrr@bzq-79-177-48-232.red.bezeqint.net)
18:57.29evilbitbut if I dial another sip number then the CID that I set doesn't show up so I'm wondering if they are right and I have something wrong
18:58.25pabelangerzbyniu: Did you pastebin 'dahdi show channel 1' ?
18:59.29tzafrir_laptoppabelanger, he did. THe context shows there properly: "from-zaptel"
19:00.19evilbithmm, ok... so if I remove the SetVar from sip.conf then I can dial out
19:00.57[TK]D-Fenderevilbit: - Executing [12158872972@ip-solutions-internal:1] Set("SIP/hhoffman-desktop-0000000d", "CALLERID(number)=") in new stack <--- shouldn't have QUOETS here.
19:01.02[TK]D-FenderEQUOTES*
19:01.04[TK]D-Fendersjdfhjsdfgsdf
19:01.06[TK]D-Fendergah
19:01.54evilbithmm, ok
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19:04.42evilbithere are the relevant bits from extensions.conf: http://pastebin.com/JSHe7hQN
19:05.33zbyniupabelanger: http://pastebin.com/psVUEWwK  -  dahdi show channel 1
19:06.05pabelangerzbyniu: working on a patch for you to try
19:06.14zbyniutzafrir_laptop: if i remove ext tel it will go to 'default'
19:06.52pgracewhere does oej's srtp patch live nowadays?
19:07.05leifmadsenpgrace: in the 1.8 branch and trunk
19:07.06pgracethe http://www.e164.org/wiki/AsteriskSRTP link is wrong.
19:07.13leifmadsenpgrace: it's been merged already
19:07.14pgraceIs there a 1.6 patch still?
19:07.19tzafrir_laptopzbyniu, hmm... actually what I wrote is pointless. It found that extension in context 'tel' anyway, so we have to worry indeed about the context first
19:07.23leifmadsenpgrace: if there is, it's very old and broken
19:07.26pgraceok
19:07.27zbyniupabelanger: great, i've builder and pbx ready to test :)
19:07.31evilbitaha! It's because Wireless Caller was in quotes
19:07.37pgraceleifmadsen: 1.8 is still considered devel right?
19:07.37leifmadsenpgrace: it'll be on https://issues.asterisk.org as a closed issue
19:07.49leifmadsenpgrace: the fact we have 1.8.0-beta2.... yes :)
19:07.55pgraceleifmadsen: :)
19:08.00pgracedrats
19:08.03leifmadsenplease help test
19:08.39pgraceyeah, sounds like I gotta toss up a test vm
19:08.42pabelangerzbyniu: http://asterisk.pastebin.com/qkdrKFAg
19:08.47pabelangerzbyniu: untested
19:09.00pgraceleifmadsen: working on OCS and asterisk compatibility when not using a mediation server, it's... interesting.
19:09.09pgracepretty sure my problem is SRTP.
19:09.29wcselbyanyone have any thoughts on why my pseudo dahdi device isn't working - http://pastebin.com/z5TBpAqD
19:09.39wcselbyor have any other troubleshooting steps I can do other than restart dahdi
19:09.57pabelangerwcselby: permissions?
19:10.06LemensTSERROR[2097]: chan_dahdi.c:10180 mkintf: Signalling requested on channel 1 is ISDN PRI but line is in FXO Kewlstart signalling    <---I dont have fxoks set anywhere in /etc/dahdi/system.conf nor /etc/asterisk/chan_dahdi.conf
19:10.23wcselbypabelanger - same perms on /dev/dahdi/pseudo as any of the other /dev/dahdi/ devices
19:10.42tzafrir_laptopLemensTS, what version of Asterisk is it?
19:10.57LemensTS1.6.2.10
19:11.08tzafrir_laptopLemensTS, and it's actually fxo_ks in chan_dahdi.conf
19:11.27tzafrir_laptop(or users.conf)
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19:13.59wcselbydahdi version 2.2.1.2, vpm450 hardware echo can
19:14.15wcselbyasterisk 1.4.33.1
19:16.50LemensTStzafrir_laptop: http://pastebin.com/t40awzE9  here is pastebin of chan_dahdi.conf
19:16.55malcolmdwcselby: is it something that's resolved by restarting asterisk or only by restarting (unloading and reloading the kernel modules) dahdi?
19:17.12malcolmdwcselby: happens always, regularly, periodically (fixed or variable)?
19:17.53wcselbymalcolmd - only resolved by restarting dahdi
19:18.10wcselbyhappens more frequently than it used to (vague, I know)
19:18.18wcselbyi first noticed this when I installed oslec
19:18.24wcselbythen I removed oslec and installed the hardware board
19:18.30LemensTShere it is with system.conf at the bottom http://pastebin.com/Gj9WS2d1
19:18.30wcselbyand didn't see it for a few months
19:18.39wcselbynow it's happened three days in a row
19:18.46wcselbyeach time I had to restart dahdi
19:19.03malcolmdwcselby: this sounds like a "fun" one for our support team, i'm afraid.  http://www.digium.com/support
19:19.12wcselbyit happened once last week and maybe once the week before
19:19.15wcselbyyeah
19:19.33wcselbyi was afraid of that
19:19.42wcselbyi need to get the serial number of the card to get support, yes?
19:20.04malcolmdyup.
19:20.16tzafrir_laptopLemensTS, the 'group' setting at the end of chan_dahdi.conf has no effect, BTW
19:20.21tzafrir_laptopanything in users.conf?
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19:22.08LemensTStzafrir: added users.conf and output of trying to load the chan_dadhi.so from cli output   http://pastebin.com/XQN8aMTz
19:22.12LemensTS*at bottom
19:22.42bdheemanhello
19:23.17bdheemanis this http://svn.digium.com/svn/asterisk/trunk and 1.8.0 head?
19:23.39Qwellbdheeman: what?
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19:24.05malcolmdbdheeman: 1.8 branch is http://svn.digium.com/svn/asterisk/branches/1.8
19:24.15bdheemanok
19:24.37tzafrir_laptopLemensTS, I read that message backwards
19:24.42bdheemanand what's the above said trunk is?
19:24.46tzafrir_laptoptry running dahdi_cfg again
19:26.28malcolmdbdheeman: trunk is http://svn.digium.com/svn/asterisk/trunk
19:27.03LemensTSthat gets me: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22)  <--does 22 mean line number or is it an error code?
19:28.07bdheemanmalcolmd: which one you recommend should I try with a2billing 1.7.1~svn?
19:29.13bdheemanmalcolmd: for experimentation and testing
19:29.51malcolmdbdheeman: best to shy away from trunk as that's the source of all new development, and thus it's considered completely unstable and isn't guaranteed to compile.  1.8's a better place to go.  currently, we're in the beta process for 1.8 (we're at beta2 as of a few days ago).  so, what you check out of the 1.8 branch will eventually become 1.8.0 for general availability.
19:30.33bdheemanmalcolmd: thanks
19:31.34zbyniuLemensTS: error code
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19:32.15malcolmdbdheeman: hrm..i'm looking at the a2billing page and it says that 1.7.1 stable was released in may of 2010.  when they released that, the latest asterisk was probably 1.6.2.  so, they may not yet be working on 1.8.  i'm looking at their documentation page (http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Documentation) and it says Asterisk 1.4.0+.  so, they may not even work with asterisk 1.6 (which was before asterisk 1.8).
19:32.38malcolmdso, if that's the case, then you can check out from http://svn.digium.com/svn/asterisk/branches/1.4
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19:33.22*** join/#asterisk davesullivan (44415202@gateway/web/freenode/ip.68.65.82.2)
19:33.30bakermdI put app_fax.so in my asterisk modules folder and restarted asterisk, however when I do `core show applications` I do not see fax listed... starting asterisk in non-forking mode does not display any errors - any ideas?
19:33.41malcolmdbdheeman: if you're planning to build something permanent, you might want to review the Asterisk Version release and EOL schedule here (http://www.asterisk.org/asterisk-versions)
19:33.45bdheemanmalcolmd: ok, I checked out http://svn.digium.com/svn/asterisk/trunk in around 2008
19:34.03pabelangerbakermd: *CLI> module load app_fax.so
19:34.20zbyniupabelanger: nothing changed with your patch
19:35.10bdheemanmalcolmd: thanks for the link, reviewing it right now
19:35.42bakermdpabelanger: Unable to load module app_fax.so  || Command 'module load app_fax.so' failed.  -- any idea how I can get additional info out of it?
19:36.41pabelangerzbyniu: Then you have a different issue then what I was thinking
19:37.05bakermdpabelanger: I see the full logging now - thanks
19:37.21bdheemanmalcolmd: from that page, seems better I try 1.8 branch
19:38.43malcolmdbdheeman: as far as asterisk goes, yes.  i just don't know the state of a2billing development related to asterisk 1.8 support
19:38.54pabelangerzbyniu: Did you update Asterisk recently?  Did this work previously?
19:39.13zbyniupabelanger: can you explain me in 2 words how incoming connection goes to extension resolve code?
19:39.20bakermdAny thoughts on error "undefined symbol: ast_register_application"
19:39.26zbyniufrom chan_dahdi.c:     -- Accepting call from <- here is ok
19:39.52zbyniuto: pbx.c:   == Starting DAHDI/2-1 at tel,0,1 failed so falling back to exten 's'
19:40.16zbyniupabelanger: i have oooold asterisk now, and working on new
19:40.31zbyniutried asterisk 1.6.2.6 and .10
19:40.44zbyniudahdi 2.2.1 and 2.3.0
19:40.45bdheemanmalcolmd: NP, I'll test and revert back to 1.6 or even 1.4 release if needed
19:41.07zbyniuwanpipe 3.5.10 and .14... :)
19:42.56pabelangerzbyniu: did you pb you chan_dahdi.conf?
19:43.23malcolmdbdheeman: cool :)
19:44.18bdheemanmalcolmd: :)
19:45.05zbyniupabelanger: one part here: http://pastebin.com/s0hAqndG
19:45.05Qwellbakermd: You aren't loading the module in the correct version of Asterisk.
19:45.14bakermdQwell: Aah - thanks
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19:49.44davesullivanAnyone have any experience running asterisk on ec2?  Seems like a physical machine works a bit better?
19:50.00Qwelldavesullivan: no reason it shouldn't work
19:50.06nightwalkFinally caught a capture of my asterisk installation acting up, though the output is sparse despite having set core debug to 99: http://pastebin.com/8NXmpiEs As near as I can tell, this dmesg output goes with it (but would seem to be useless): http://pastebin.com/k0K7raXn
19:50.58malcolmdb11d`: and...committed, issue closed.  thanks again
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19:52.15nightwalkBasically, what happened is a user called out to the POTS from an extension off of a TDM410, and was disconnected for reasons unknown ~1 minute later. The user hung up the phone, and it started ringing. No one was there when they answered it.
19:52.58davesullivanQwell: yeah, i've got one up and running, but I keep reading things about it being a bad idea due to some kind of timer thing
19:53.31davesullivanSeems to work well enough, but I get garbled audio issues occasionally and haven't tested it fully loaded w/ 20 calls going at once yet
19:53.59nightwalkI'm sure this ties in with the mysterious dropping call problem I've been trying to track down and fix, though the ghost ringing is new. Anyone have any ideas? I don't see anything in the trace that I didn't already know :/
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19:55.18nightwalkAsterisk 1.6.2.5-0ubuntu1, with DAHDI 2.2.1, btw
19:57.39malcolmdnightwalk: hit our support department up on this one, please; i think they're best-suited to help troubleshoot and resolve.  http://www.digium.com/support
19:58.06*** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net)
19:58.50nightwalkmalcolmd: Any particular reason you say that?
19:59.32citywokwhen a call is blind transferred to another person, the call has the same callid, and the same recording file. the problem i have is the first person that answered the call transfers, and a CDR is generated.  i process the CDR, encode the WAV file and delete it.  i have no way of knowing this is a transfer and the file is still being written to. is there any solution for this?
20:00.20hardwirehmm.
20:00.28nightwalkAlso, not sure Digium will support this card. It *does* appear to be a full-fledged Digium card ('Digium' *is* printed on it, and it *isn't* from china), but I didn't get it through them.
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20:13.20citywoknightwalk: considering how awful digium support is, even if you do get a response it could take weeks.
20:13.31b11d`no prob malcolmd... glad I could help for once :)
20:13.40wcselbylol, i've not had any problems with digium
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20:14.43nightwalkcitywok: No support from them anyway. According to malcolmd, my card is trademark-infringing knock-off.
20:15.02nightwalk...from canada
20:15.42wcselbydamn canadians
20:15.49wcselbyamerica's hat strikes again!
20:15.54wcselby:P
20:16.39citywokwcselby: 72 hour response times and they respond in the middle of the night b/c their email support team for the cheap support is done at night
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20:17.13citywokoh, and when their device has a crippling bug they admit in october, they still have not released a fix that keeps their PBX appliacnce from hard locking every week because they are working on adding new features.
20:17.20nightwalkActually, I knew what I was getting. I don't really care if it's Digium or not, so long as it works. Digiums are far too expensive to use in proof-of-concept implementations
20:17.21wcselbycitywok - called in at 7 am EST, answered on the second ring, guy helped me as much as he could (it was a hardware issue)
20:17.42citywokthanks, i dont want any freaking features, i want you to make it so my AA50 doesn't go read-only on the CF card every few days lighting the thing on fire
20:17.51citywokoh, i dont get phone support b/c the thing only came with email support
20:17.56pabelangernightwalk: what company in Canada?
20:18.05nightwalkpabelanger: nicherons
20:18.12wcselbycitywok - well, we've obviously had very different reactions
20:18.17wcselbyerm, not reactions
20:18.23wcselbyuh, experiences :)
20:18.33citywokyea. you are lucky for getting phone support :)
20:21.04*** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec)
20:22.35nightwalkFunny thing is, with all of the 'compatible' devices out there, Digium's prices seem to stay unreasonably high. I mean, I can get a whole populated tdm and still have money left over for what I'd pay for a bare Digium tdm
20:23.11citywoki have a pair of the 24 port cards loaded on a shelf
20:23.25citywokbought right off of digium's website like 2 years ago
20:23.38citywok(if you are looking to buy one or two that is)
20:23.45Qwellnightwalk: Ever heard the expression "You get what you pay for"?
20:23.59nightwalkqwell: I was waiting for that to come up, actually :)
20:24.14Qwellwell, if you're buying cheap garbage, you really shouldn't expect it to act otherwise..
20:24.27Qwellhowever, having said that.
20:24.35citywokwe used a couple knockoff T1 cards that worked fine, but no knockoffa nalog stuff
20:24.42QwellPrice similar hardware from other PBX vendors .  You'll cry.
20:24.48nightwalkLike other brands, Digium advocates always bring that up. Problem is, I don't have any experience with Digium, so they may well be as bad as the next guy.
20:25.10[TK]D-Fender[16:00]<nightwalk>Also, not sure Digium will support this card. It *does* appear to be a full-fledged Digium card ('Digium' *is* printed on it, and it *isn't* from china), but I didn't get it through them. <-- I've seen these fakes on eBay before.
20:25.14nightwalkPaying less means opening myself up to that much less of a rip-off
20:25.18[TK]D-Fenderalso reported them to Digium
20:25.20citywokQwell: no kidding. it's still 1/100th the cost per port of an off the shelf phone system
20:25.23Qwellnightwalk: countless people do.  you never see people praising the crap hardware.
20:25.44hardwireany good way to completely stop media progress?
20:25.48hardwireerr progress media?
20:25.53Qwellhardwire: ctrl-c :D
20:25.54hardwireI have it prohibited to the nth degree
20:25.57hardwireQwell: nono
20:25.58hardwirenonono
20:26.15nightwalk[TK]D-Fender: Yes, malcolmd already established that ;)
20:27.39nightwalkAnyway, I'm thinking the issue might be with the hardware the card is in. Problem is, I don't see anything in the traces I posted to support that. Any way to confirm without transplanting the card?
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20:35.48nnyI use the fake digium cards, but they say sangoma on them. Work better too!
20:35.52nny:D
20:36.34b11d`im not buying another sangoma card in my life
20:36.43b11d`i liked them, but zero support lately... zero.
20:36.45nnylol that's how I feel about digium's cards
20:36.55nnyreally? odd. I can usually call and hit a support tech almost right away
20:37.03b11d`yeah for like two weeks, calls and emails.. no reply.
20:37.13nnymy first t1 digium card with hardware ec sucked hard. Had to put a telelabs EC in front of it
20:37.16b11d`only bought six a104d cards
20:37.48b11d`im buying my first digium card next week
20:38.54b11d`after spending almost $10k on sangoma cards, I expect at least some kind of reply from their support people.. even a "we dont know, fuck off!"  would be fine haha
20:40.39Kobazsangoma support is really great
20:40.43b11d`it used to be
20:41.07b11d`i've had good experiences with them in the past as well, just not lately
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20:41.51Kobazwhat's the current issue?
20:42.41b11d`im experiencing a problem getting a D-channel operational on a fractional PRI, where only channels 1-11 are available, with D on channel 24.   My simple question to them is, whats the best way to configure wanrouter to support this setup?
20:43.07b11d`the card works flawlessly on a full 24-channel PRI in another location..
20:43.48b11d`and I've been over it a dozen times w/ the telco.. they say the PRI is good and in working order
20:44.07Kobazhave you checked at the smartjack
20:44.33b11d`yes, as I said, I've been over it w/ the telco a dozen times
20:44.55Kobazand nothing interesting from the sangoma logs?
20:45.08b11d`granted, all I can do is accept their statement of "its working fine" -- not much I can do to investigate myself.
20:45.23Kobazyeah... other than dialing into the smartjack
20:45.27wcselbywhat is the differrence between -nocana and -opteron with regard to the hpec binary?
20:45.35b11d`which I cant do
20:46.07b11d`no, not much interesting in the logs... i've tried manually setting the DCHAN and the AVAIL_CHAN options in wanpipeX.conf but to no avail..
20:46.14b11d`says the D channel is up, but its not..
20:46.42b11d`so, all im looking for from them is verification that im not doing something completely stupid and that my config SHOULD work, so I can go BACK to the telco and say its their end :)
20:46.43nightwalkI believe Sangoma is the one the who makes the fax card we use. They were the 'economy model' cards, but still cost like 10k. The 'replacement' cards were $20k+. Given that, and the fact that they don't seem to want to play nicely with linux, I'd imagine they probably won't be getting any more of our business.
20:47.55Kobazi've had no problems with sangoma on linux other than dahdi problems which isn't sangoma's fault
20:48.13nnyhmm guess opinions vary, i have had nothing but good luck with sangoma cards, and nothing but nightmares with digium ones
20:48.19Kobazlibpri/dahdi has just been really crappy for me lately... i've switched to external gateways
20:48.20b11d`and, two weeks of emailing them and leaving messages w/ support, and NOT A SINGLE REPLY IN ANY WAY.
20:48.35nnyyeah that sucks b11d, maybe you need to contact someone higher up
20:48.38Kobazgive them a call
20:48.41Kobazthey do phone support
20:48.43b11d`I have been calling
20:48.49b11d`I have been emailing
20:48.59b11d`as I said... my Digium card order goes in next week...
20:49.04b11d`they've lost me
20:49.15nnyheh well. Don't be suprised if digium isn't much better
20:49.16b11d`I'll try my luck w/ Digium
20:49.20Kobazheh
20:49.24b11d`well at least I can come here and get help :)
20:49.27b11d`no #sangoma
20:49.28b11d`:P
20:49.43nnyi have replaced more FXO modules than I can count, i got a box of them at the office
20:49.50Kobazyeah
20:50.07Kobazat the last place i worked.. fxos would fry pretty often
20:50.15b11d`poorly grounded wiring?
20:50.27NEEDINGHELP123Hi Guys
20:50.30NEEDINGHELP123hows it going?
20:50.34b11d`great, friend!
20:50.41NEEDINGHELP123good news
20:50.45citywoki'm going out on a limb here, and guessing you need help?
20:50.47Kobazno idea... went with sangoma and the modules stopped frying
20:50.55NEEDINGHELP123no man
20:50.55nnydunno, it's the telco 66 block --> card
20:50.58NEEDINGHELP123everythign good
20:51.05NEEDINGHELP123no help needed today
20:51.05nnylol
20:51.09cmendes0101Trying to add a register => into sip.conf but im getting this error: [Jul 29 21:50:02] WARNING[4266]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0x7f025445e9f0 (len 407) to 206.15.156.221:5060 returned -1: Address family not supported by protocol
20:51.10NEEDINGHELP123maybe tomorrow ;)
20:51.27citywoknny, we had a pair of 24 port cards that never ate a single module, all punched in to 66 blocks and SLC's in our old phone system
20:51.43citywokthey were the digium tdm2400p's i think
20:51.53nnycitywok: yeah guess it all depends on the telco wiring
20:51.58NEEDINGHELP123I may need to employ an asterisk tech on a permanant basis though
20:52.02NEEDINGHELP123anyone up for the job?
20:52.11citywokyou might want to tell everybody where the job is located
20:52.20NEEDINGHELP123remotely
20:52.24NEEDINGHELP123whever you want it to be located
20:52.26NEEDINGHELP123just be available
20:52.32NEEDINGHELP123and ready to work on demand
20:52.38NEEDINGHELP123with a minimum retainer fee
20:52.40b11d`im game :)
20:52.41*** part/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net)
20:52.47b11d`haha no there are much better people for the job
20:53.13citywoki'm imagine there are at least a few people in here that would be happy to work out a gig with you.
20:53.25b11d`yeah, no shortage of Asterisk consultants, thats for sure :p
20:53.25nny<-- always willing to do remote support
20:53.32citywokif you have more details feel free to PM me :)
20:54.22NEEDINGHELP123okay
20:54.27NEEDINGHELP123so'll be in touch
20:54.31NEEDINGHELP123I'm here and around
20:54.44NEEDINGHELP123I'd like to get a few cv's with some info if possible
20:54.49NEEDINGHELP123so that I can assertain who I need
20:55.15b11d`i'll be a phone sex worker, if need be
20:55.48NEEDINGHELP123where is Drmessano
20:55.53NEEDINGHELP123?
20:56.04NEEDINGHELP123he's a funny one where is here
20:56.07NEEDINGHELP123not seen him for some time
20:56.10NEEDINGHELP123:P
20:56.18NEEDINGHELP123not been flmaed by him for some time that is :P
20:56.26*** join/#asterisk Carp1 (~none@cpe-24-92-37-23.nycap.res.rr.com)
20:58.59nnylooks like 5 pm, and for once I have no work load for the evening. guess I can leave the computer alone for a bit... later all
20:59.20*** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com)
21:03.06*** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net)
21:13.37b11d`...
21:16.37b11d`just bought a new place... gotta go over to lowes, get a lawmmower..  weed eater... hedge trimmer..
21:17.03b11d`why do all new mowers come with the insufferable bag on them.. i hate bagged mowers
21:17.09*** join/#asterisk Benwa (~Benwa@unaffiliated/benwa)
21:17.23Qwellb11d`: you can use mine.  you have to mow my lawn though.
21:17.29b11d`you got it
21:17.58b11d`i asked one of the neighborhood kids to mow the lawn... their mother overheard me saying "so how much do you charge?" -- then the cops got involved..
21:18.28wcselbylol
21:18.54wcselbywait a week or two, you'll probably have a flyer hanging on your door with the name of a lawn service on it
21:19.07b11d`aye, you're likely right
21:19.46ChannelZget a mulching mower.. mine has no bag
21:19.54b11d`yeah thats what im gonna get tonight
21:20.30ChannelZis about to have 6 tons of landscape rock dumped in his driveway
21:20.41b11d`theres an 8-horse, 22" for $150 in town
21:20.42ChannelZcongrats on the new place btw
21:20.45b11d`thanks
21:21.00ChannelZfirst house or something?
21:21.08b11d`yeah...
21:21.12ChannelZcool
21:21.19b11d`got married, had a kid, needed a bigger place now
21:21.24b11d`working on kid #2 now
21:21.40ChannelZstill looking for parts? :)
21:21.44b11d`getting the 'hell years' done in short order :)
21:22.03pigpendoes anyone know of a way to find out what telco has a porting agreement with?  ie: I want to move my number.....and I want one friendly with asterisk.
21:22.21citywoki've ported to bandwidth.com and flowroute without any issue
21:22.33ChannelZIsn't it law any of them have to port?
21:22.53citywokyea, there's no "agreement".  the only requirement is the place you are porting the number to have a local presence in the area.
21:22.54pigpenthey must port, but not all can port to anyone.
21:23.19citywoki was unable to port my san francisco numbers to qwest b/c they dont have a presence in the sanfran area
21:23.22b11d`yeah I recently ran into this exact problem
21:23.38citywokif they are TFN's, then it's really easy and doesnt really matter at all.
21:23.45ChannelZI did Qwest -> Vitelity but they're both local.
21:23.55ChannelZQwest -> Integra too, same there though
21:24.04citywokChannelZ: where are you? PNW?
21:24.09ChannelZColorado
21:24.17citywokah. home of qwest. lol
21:24.37ChannelZyeah. Soon to be Centurylink I guess.
21:24.47citywokyea. i got that email from our rep a few months ago.
21:24.56b11d`what? they are changing their name?
21:24.59citywokwe run our entire company on qwest MPLS
21:25.03ChannelZCenturytel is buying them
21:25.03citywokb11d`: they got bought
21:25.06b11d`ohhh
21:25.13b11d`im surprised
21:25.30citywokwith any luck they will improve qwest support.
21:25.33ChannelZthey are a money toilet I think (qwest)
21:25.40ChannelZso not sure why they bought them either
21:25.43citywoki'm praying that they rip out QControl and beat it like a red headed step child.
21:26.08ChannelZbut they have q.com!
21:26.27b11d`never even heard of Century Tel..
21:26.36b11d`CenturyLink that is
21:26.57citywoki only have b/c my grandma's hosue had century-tel for phone service a few moons ago.
21:27.34citywokbut i thought centurytel was smaller than qwest, i had no idea they were capable of buying them.
21:28.10b11d`well, I'm gonna go get that mower.. talk to you all tomorrow :0
21:28.11b11d`goodnight
21:28.39ChannelZhave fun
21:29.08ChannelZCenturyTel changed their name recently to CenturyLink.  But I'd never heard of them either until I did a name-change video for them :)
21:31.38*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:32.16*** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc)
21:32.33booduhello
21:32.59lvlolvlocentury.... *shrug*
21:34.03*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
21:34.56IamTryingWhat can i change in host=localhost? [Jul 29 23:30:57] NOTICE[15053]: chan_sip.c:20603 handle_request_register: Registration from '"xxx" <sip:xxxx@127.0.0.1>' failed for '127.0.0.1' - Peer is not supposed to register
21:35.50[TK]D-FenderIamTrying: You do no specify a host for a device that is expected to REGISTER
21:36.33IamTryinghost=dynamic but i wanted to make a localhost test
21:36.59[TK]D-FenderIamTrying: Think of it like as if I was sure I knew your phone number, I would reject someone telling me where to contact you because I already know
21:37.20[TK]D-FenderIamTrying: Well your client is trying to register.  So if you want it to succeed it MUST be "dynamic"
21:37.34IamTryingright, ok
21:37.44IamTrying[TK]D-Fender,  thank you
21:38.44*** join/#asterisk NuclearLucifer (gavroche@gavroche.pl)
21:39.55*** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net)
21:40.14*** join/#asterisk rayk_sland (~rklassen@mail.mccscs.com)
21:41.21CoffeeIVI have some Polycom phones that were registering to a asterisk box, and the asterisk box was moved to a new IP and the DNS changed accordingly, and now they don't register.  I need some debugging tips -- is there a way I do a "udp telnet" to make sure port 5060 udp is open between the phones and the new server ?
21:41.21Nuggettelnet is eeeeeeevil!
21:41.46Kattychecks Nugget for signs of life
21:41.49ChannelZsipsak?
21:41.55CoffeeIVevil problems require evil solutions
21:41.58ChainsawCoffeeIV: You need to give up this telnet addiction.
21:42.06*** join/#asterisk amacgyver (~macgyver@ns0.calibre-solutions.co.uk)
21:42.07ChainsawCoffeeIV: Enable sip debugging for the phone IP address on the Asterisk console.
21:42.18ChainsawCoffeeIV: Soft-reboot your phone. Sit back and watch. The server will tell you.
21:42.21Kattyfriends don't let friends telnet.
21:42.22CoffeeIVok
21:42.33ChainsawCoffeeIV: Add beer for additional relaxation.
21:42.50CoffeeIVtelnet is awesome, we should have never replaced with these "browsers".  netcat is also good.
21:43.25*** join/#asterisk guilhermebr (~Guilherme@189.63.87.234)
21:43.51ChannelZGOPHER!
21:44.31NuclearLuciferCoffeeIV, Have you got some dnscache in your network? Maybe it hasn't update yet?
21:45.05*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)
21:45.18Kattyhi raden
21:46.15*** join/#asterisk guilhermebr (~Guilherme@189.63.87.234)
21:46.21raden_workhugs Katty
21:46.39Kattyhugs on raden_work
21:47.19*** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-170-204.mia.bellsouth.net)
21:47.36CoffeeIVI made sure the dns was good.  Using "sip debug ip <mylocalip>" I can see the SIP register messages coming from my phone to the asterisk.
21:48.50CoffeeIVbut "sip show registry" and the phone both indicate the registration is not happening.  I don't see any log messages as to why.
21:49.37*** join/#asterisk ltd_wk (~z@sixified.transact.net.au)
21:49.58ChainsawCoffeeIV: You can observe the traffic now. I'm sure Fender would love a pastebin of the dialog. He's fluent in SIP.
21:50.56raden_workCoffeeIV, set sip debug on
21:51.01raden_workthen have the phone register
21:51.08raden_workpaste debug
21:51.19CoffeeIVok, I'll do that and I'm setting up the pastebin
21:54.58raden_workHolly crap the dude actually paid me i helped the other day :)
21:55.04raden_workthere are honest people in the world
21:55.25NuclearLuciferha, I forced faxes to work with t38 today. And it works without any hangups. :)
21:56.20raden_workNuclearLucifer, really ?
21:56.50CoffeeIVhere is my pastie:  http://pastie.org/1066238  the IP of the phone that is trying to register is 10.0.2.9, the IP it is coming from is 66.93.217.226, the extension is 8449
21:57.07NuclearLuciferraden_work, Yes, looks stable. But it would go `on production' in 1-2 weeks. But as far as I made tests, it works fine.
21:57.23raden_workI can never get it to work
21:57.44*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
21:58.21raden_workhow can the ip of the phone and where it coming from be 2 different addresses ?
21:58.57NuclearLuciferraden_work, It's done with Asterisk 1.6.2.10.
21:59.18raden_workim on 1.6.0.10
21:59.23raden_workill have to try it out
21:59.31raden_worklemee check colorado co-los see what they running
21:59.40NuclearLuciferraden_work, I tested it from sending faxes from PSTN and to PSTN.
21:59.52CoffeeIVNAT, normally I would only use iax2 with NAT, but supposedly this was working previously.    It's a really fancy HP super-router that can proably be configured to handle that kind of stuff
22:00.08raden_work1.6.0.22
22:00.39raden_workCoffeeIV, what are you trying to do exactly and what exactly is the issue ?
22:02.00NuclearLuciferraden_work, I connected fax to asterisk via linksys spa3021 voip gateway.
22:02.06NuclearLuciferAnd it works fine.
22:02.23[TK]D-FenderCoffeeIV: Smart routers are usually fucking DUMB.  Just FYI
22:02.36*** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122)
22:02.40CoffeeIVI had an asterisk server on the 10.* internal network; a clone of it on an outside IP has been used as a backup.  The internal one died, and I switched the DNS to point to the other IP, and everything seems to work -- including incoming calls to the new box -- except my phone doesn't register to the new external box, and thus can't make or receive calls.
22:02.58[TK]D-FenderCoffeeIV: <--- SIP read from UDP:66.93.217.226:38572 --->
22:03.06[TK]D-FenderCoffeeIV: <--- Transmitting (no NAT) to 10.0.2.84:5060 --->
22:03.15[TK]D-FenderCoffeeIV: And you blatantly did your configs wrong
22:03.22CoffeeIVYeah, I hate smart routers too, and I would not have used SIP ever again if I had my way
22:03.25[TK]D-FenderCoffeeIV: its responding to the Private IP.
22:03.41[TK]D-FenderCoffeeIV: "no NAT" <- *cough*
22:03.46[TK]D-FenderCoffeeIV: Now go fix your peer
22:07.58*** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com)
22:08.26CoffeeIV[TK-]D-Fender: setting nat=yes; in sip.conf did it, thanks so much.
22:08.56CoffeeIVyou guys are awesome
22:10.04mmlj4keep studying, soon you can be promoted to CoffeeV
22:10.22raden_workCoffeeIV, configure your router properly
22:10.30*** join/#asterisk nettie (~nettie@stewie.freax.it)
22:11.46nettieHi guys, anyone using Sangoma BRI cards here? I'm having issues with one of them. I think it's related to clock source I just hear a very distorted noise when I try to place/receive a call.
22:11.52seanjohnI have been pulling my hair out trying to fix something that is my termination provider's fault. If I use one of their other three servers, call quality is fine but using the server I had as the DEFAULT for incoming and outgoing, call quality crashed after 4 seconds of connectivity. I had complained to them before when they, voip.ms, had  a problem (someone monitored several of my calls to toll free numbers) and they had the response "do
22:11.52seanjohnthe premium route?" and "we are a respectable company". Respectable company my ass! A respectable company based on Finland (I'm in the US) would abide by the law and make sure their systems were secure no matter how much we paid for the call; this is illegal!
22:17.42*** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com)
22:18.00seanjohnanyone else use voip.ms?
22:18.14seanjohnsorry, my workstation is REALLY failing (freezing)
22:18.23seanjohnwindoze
22:18.39seanjohnunlike the asterisk machine
22:19.36seanjohnasterisk was causing hard drive errors to display when the specific server was malfunctioning
22:20.04seanjohnwithout asterisk running, the hard drive errors didn't display
22:20.04Qwellhard drive errors?  VoIP provider?  what?
22:20.13seanjohnyep Qwell
22:20.15Qwellsecure?  illegal?
22:20.36seanjohnI've been backing up everything for an hour and trying to fix anything on Centos for the past day
22:20.49Qwellhow are any of these 4 things related?
22:20.51citywokseanjohn: why dont you describe your problem in a way that makes sense. so far your ranting makes none at all.
22:24.36seanjohni'm trying to find the log lines
22:24.43seanjohnwhy doesn't asterisk rotate them more often
22:24.51*** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca)
22:25.31seanjohnI wish there was an admin here cause this is a bug in asterisk to asterisk sip (or callweaver)
22:25.59seanjohnmy system was fine and my termination's "asterisk" was malfunctioning.
22:26.06seanjohncausing my system to malfunction to
22:26.22seanjohnthe admin could tell me what I posted 3 entries into the room ago
22:26.36seanjohnfender???
22:26.37[TK]D-Fenderenables defragmentation on seanjohn
22:26.43seanjohndefrag?
22:27.12seanjohnsorry for pressing enter so much but I'm so pissed
22:27.15[TK]D-Fenderseanjohn: 2 words summarize you right now : mixed nuts
22:27.31[TK]D-Fenderseanjohn: Its coming out a garbled mess
22:27.43seanjohnfender, I came in here about 2 pm eastern and posted an exact error I was getting on asterisk
22:28.16[TK]D-Fenderseanjohn: That is 4.5 hours ago.  Long gone.
22:28.56[TK]D-Fenderseanjohn: And the other side being callweaver... well fear not, there is already a special circle in Hell for that (Dante missed a few)
22:29.16seanjohnsomething about when the call hangs up on chan_sip.c "method not supported" and when this error appeared on asterisk, because of the remote server's malfunction, it would cause my system to produce fake write error messages.
22:29.43seanjohni'm grepping all over the place
22:30.04Qwellfake write errors?
22:30.31seanjohnwell, Qwell, e2kfts and badblocks said no errors, just fragmented
22:32.42seanjohnbingo
22:32.43seanjohnchan_sip.c: Remote host can't match request CANCEL to call '2245d2e93e4501a50e27c20d0b3bf4dd@173.50.101.11'. Giving up.
22:33.39Qwelland how does that have anything to do with disk write errors?
22:33.49Qwellor being secure, or...illegal?
22:34.11*** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com)
22:34.25*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
22:34.56seanjohnit actually made the drive think it was having errors itself, storing a count of 9 errors in SMART, and producing the error ata2.00: cmd c8/00:08:8d:f7:b6/00:00:00:00:00/e5 tag 0 dma 4096 in
22:34.56seanjohnJul 29 16:36:41 server1 kernel: ata2.00: cmd c8/00:08:8d:f7:b6/00:00:00:00:00/e5 tag 0 dma 4096 in
22:34.56seanjohnJul 29 16:36:41 server1 kernel:          res 51/40:08:8d:f7:b6/00:00:00:00:00/e5 Emask 0x9 (media error)
22:35.00seanjohnsorry
22:35.24*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
22:35.25*** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br)
22:35.32citywokare you saying a SIP message made your hard drive have a smart error?
22:35.34Qwellit IS having errors.
22:35.48seanjohnno, all other diagnostic tools say its fine
22:35.59seanjohnthe 9 errors were produced between today and yesterday
22:36.13Qwellyour diagnostic tools are wrong.
22:36.27seanjohnthe same time as this started with their ONE, not the other three though, servers
22:36.31citywoksmart errors dont always mean the drive is failing.
22:36.36citywokit could be remapping sectors
22:36.56seanjohnyes, citywok, but this is a raid 5 array
22:36.57citywokseanjohn: did you pass 3rd grade english?  i'm having a hard time believing you did.
22:37.00Qwellseanjohn: log files write to disk.  if you are logging errors, you are writing to disk.  and those writes are causing *actual* errors.
22:37.24seanjohnDevice: /dev/sda, 3 Currently unreadable (pending) sectors
22:37.29citywokseanjohn: i suggest you look up the difference between Causation and Correlation
22:38.12seanjohni have had this happen before citywok with completely different hardware caused by a faulty LOCAL sip device.
22:38.26seanjohnits an exploit waiting to happen for dos of asterisk
22:38.35citywokwhat kind of hardware are you using?
22:38.44seanjohni WAS using an spa2102
22:38.52citywokpentium 133? software raid5?
22:39.04seanjohnthat one went in the trash and no errors until now
22:39.26seanjohncitywok: q 8600 4 gb ddr2
22:39.37*** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-170-204.mia.bellsouth.net)
22:39.38seanjohnthe 3 raid drives are western digital
22:40.22seanjohnthe problem with the device that went in the trash was on an Athlon 3000 512 ddr
22:40.34seanjohnsame error different hardware, completely!
22:41.20seanjohnthis is asterisk 1.4.31
22:41.42seanjohnx86_64
22:42.07citywokwell, i suggest you submit a ticket in the asterisk bug tracker if you believe this is a bug in asterisk.
22:42.24QwellIt's not a bug in Asterisk.
22:42.29citywokBefore doing that i'd recommend you get a grasp of your actual problem, get all of the logs you can, compile with DONT_OPTIMIZE as htey suggest, and post the bug that way
22:42.46citywokQwell: yea, i would put up a lot of money at very high odds that you are right :)
22:42.47seanjohnits not a bug, its caused by an idiot on the other end using a faulty build of asterisk and my asterisk doesn't counteract the issue
22:42.57Qwellcitywok: No, I am definitely right.  Period.
22:43.07QwellApplications *CANNOT* cause write errors.
22:43.11QwellThey are caused by *ACTUAL ERRORS*
22:43.17*** join/#asterisk l337ingDisorder (~l337ingDi@S0106000c425431b2.gv.shawcable.net)
22:43.20citywokyea.  Causation and COrrelation as i suggested he look up.
22:43.23QwellYour disks are broken.  Fix them.
22:43.30citywokCorrelation being the errors happen on writes. lol.
22:43.40seanjohnwhen I disabled this server and used the other 3, call quality is fine
22:43.49QwellI wonder why.
22:43.59seanjohnnot my server, one of their 4 servers
22:44.07citywokmaybe during the time you tesetd against the other servers the disk errors were not being hit.
22:44.19Qwellhow does call quality have anything at all to do with disk errors?
22:44.23Qwellyou're making no sense.
22:44.28seanjohntests were for 30 minutes, within 4 seconds of EACH call this would start
22:44.31citywokyou should try writing a ton of garbage data to your disk and see if it throws smart errors
22:44.43seanjohnok, command?
22:44.50seanjohndd if= size=
22:44.51seanjohn?
22:45.00citywokhowever you'd like, i dont give a shit
22:45.10QwellHire a consultant.
22:45.24l337ingDisorderI think I may have borked something. I'm following this guide: http://hubpages.com/hub/Installing-Asterisk-NOW-and-Configuring-Soft-Phones   and it says after the AsteriskNow installer finishes, I should see a console screen where I can configure Asterisk. Instead I just get a login prompt and when I log in I get a standard bash prompt. I can access the web interface but it only has options...
22:45.25citywoki'd suggest hiring somebody that has a clue what they are doing. you clearly do not.
22:45.26l337ingDisorder...for Voicemail & Recordings, Flash Operator Panel, and FreePBX Administration (According to the guide there should be a menu with a 'users' option). Can anyone help?
22:45.30seanjohncitywok, you act like I'm being arrogant to you or something, like you're the owner of voip.ms
22:45.44citywoki'm acting like you have no idea what you are talking about whatsoever
22:45.48*** join/#asterisk joako (~joako@opensuse/member/joak0)
22:45.49citywokbecause you sound ignorant, not arrogant
22:45.53seanjohngo try it for yourself
22:46.01seanjohnput 5 bucks on voip.ms
22:46.06seanjohnuse their chicago server
22:46.32citywoki'd bet if it were a problem with their server any of their thousands of other customers would have found it, and they would have resolved it within an hour.
22:46.48citywokso i'm going to save my time, and save you the hassle of paypalling me the $5.
22:46.49seanjohnits just the ONE server
22:47.15seanjohnyou don't realize you can ORIGINATE to each server simultaneously but "i don't have a clue"
22:48.53seanjohnbtw, the chicago and atlanta are the newest ones
22:49.09seanjohndifferent providers never used before
22:49.10QwellSo stop using the Chicago server
22:49.20Qwellproblem solved.  until your disk dies.
22:49.25seanjohnI did but I"m trying to help you in case this may be an explioit
22:49.27citywoklol, i really like that.
22:49.33Qwellan exploit?
22:49.41Qwellyou're going to have to explain that
22:49.58seanjohnin case the way their service is malfunctioning, someone else doesn't repeat the same packets in an attempt to dos/ddos
22:49.59citywokqwell, he already did! the other user can send a SIP CANCEL that will cause your hard drive to fail. it's  a DDoS.  DUHHH!!!! JEEEZZZZ
22:50.17Qwellhow is that going to DoS anything?
22:50.37seanjohncitywok, you couldn't have any sense other than computers if god touched you himself
22:50.49TJNII"SIP CANCEL that will cause your hard drive to fail."  lol whut?  I'm abviously late to the party, but....
22:50.49seanjohnbash people, see where it gets you
22:50.56Qwellseanjohn: Is that what you're suggesting?
22:51.09Qwellis that a provider can cause a disk to fail by sending a bad SIP message?
22:51.20seanjohni didn't post the messages online
22:51.31seanjohnfrom others having the same error and results
22:51.41QwellTJNII: I've been here the whole time, and I still don't know wtf he's talking about.
22:52.02QwellI'm pretty sure citywok nailed it though.
22:52.05citywokTJNII: neither do i. scroll up a bit if you have history. it's like a 3rd grader trying to explain what is going wrong.
22:52.21*** join/#asterisk Arsenick (~y@fedora/Arsenick)
22:52.34Qwellseanjohn: Answer please.  Is that what you're suggesting?   That a provider can cause a disk to fail by sending a bad SIP message?
22:52.36citywokyea, i think i deduced what he was trying to get at. sip cancel causes logging that blows up the hard drive as a DDoS. it's a bug. maybe a cockroach.
22:53.57[TK]D-Fenderhttp://tinyurl.com/37vuqsl <-------------
22:54.02TJNII...interesting.
22:54.13citywok[TK]D-Fender: that's perfect.
22:54.15Qwellseanjohn: I'm trying to understand...
22:54.39[TK]D-FenderQwell: Your FIRST mistake ;)
22:54.40seanjohni guess you'll say that viruses never had access to CMOS rom
22:55.08Qwellseanjohn: IS that what you are suggesting?
22:55.39joakoI setup chan_mobile ok call comes in and asterisk answers... but no audio.
22:56.09Qwellseanjohn: we can't fix it if we don't know what the problem is.
22:56.26seanjohnhow secure do you think RTP is, especially when both the disk and dahdi, of asterisk, require the system clock to time theirselves. If asterisk threw off the timing somehow, it would cause disk errors
22:56.37Qwelloh good lord.  I'm going home.
22:56.47citywokno kidding
22:56.58citywoksoon we'll be hearing about conspiracy theories. did you know that 9/11 didn't happen?
22:57.46TJNIII have a coworker who is into conspiracy theories.  He tried to tell me that the standardization of A 440 in music was due to space travel.  I laughed in his face, couldn't help it.
22:58.47*** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)
22:59.28TJNIIseanjohn: If you believe that, then reproduce it.  Sniff the traffic that causes the failure, and create a reproducible testcase.
22:59.44*** join/#asterisk edguy3 (~edguy@c-98-221-27-224.hsd1.nj.comcast.net)
22:59.56citywokmake sure you read the asterisk bug tracker guidelines on posting a bug.
23:00.23Qwellseanjohn: issues.asterisk.org - please open an issue with a *CLEAR AND CONCISE* description.
23:00.31QwellIf it is not clear and concise, I will close it.
23:00.55citywokQwell: if he does post it, can you PLEASE send me the bug#? it's going to be like comedy gold or something.
23:01.48TJNIIIf he can reproduce it, it is going to be a bug on many levels.
23:01.55Qwellseanjohn: I don't want to see any ranting about the provider, or Chicago, or fake errors.  Give us detailed information, precise errors received, etc.  No opinion.  Mmk?
23:01.58citywokEBKAC?
23:02.41QwellNobody is going to take you seriously if you're just ranting and making no sense.
23:03.19citywokhe still hasn't confirmed if my summary was what he was getting at
23:03.31citywokprobably because when you make it 10 words long it sounds absurd
23:03.40radenwalks in the room
23:05.59citywokwalks in to a wall and bangs forehead chanting *it hurts to listen to seanjohn* over and over.
23:07.47Qwellseanjohn: Once you've reported the issue, /msg me the issue number so I can look at it.
23:07.59TJNIIEh, I'd just lay back and encourage him to reproduce it.  He'll either give up, wise up, or prove us all wrong.
23:09.16radenQwell, people with less intelligence than others usually do become an annoyance
23:11.42*** join/#asterisk BlackBishop (dexter@d3xt3r01.tk)
23:12.30BlackBishopHow can I give a busy signal if a user is already in a conversation ? ( like .. when I call myself ! )
23:12.34*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)
23:13.12*** join/#asterisk chazzam (~chazz@173-24-236-101.client.mchsi.com)
23:14.10joakoBlackBishop, you can set call-limit=1
23:15.01BlackBishopI don't want all users not to be able to call anybody if one is calling ..
23:15.24BlackBishopI just don't understand why I don't get a busy signal if I call myself !
23:16.16beardyYour phone has more lines than you currently use?
23:17.19BlackBishopI don't have an actual phone .. I'm just using SIP ..
23:17.41beardyphone/sip device/client/doodaa
23:18.12BlackBishopwell, I guess not !
23:18.28BlackBishopbut that still doesn't explain why I don't get a busy signal when I call my extension ..
23:18.56beardyI think it does.
23:19.22Kobazbusy schmizzy
23:19.37BlackBishopnope
23:19.38BlackBishopon my mac
23:19.41radenwhen sip show peers is done does the port mean the port the device is registered on or the port its available at ?
23:19.41BlackBishopI can call myself !
23:19.46raden302/302                    69.130.250.249   D   N      26622    Unmonitored
23:19.46raden303/303                    66.168.15.100    D   N      5060     Unmonitored
23:19.46raden304/304                    66.168.15.100    D   N      1024     Unmonitored
23:19.55BlackBishopone sais ringing and the other says incomming call
23:20.13BlackBishop( one window I meant )
23:20.20NuclearLuciferBlackBishop, Set call-limit=1 under your sip context.
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23:20.33NuclearLuciferBlackBishop, As joako wrote.
23:20.43beardyYour sip client, whatever you want to call it, doesn't consider itself busy, and accepts another call (think two, or more, lines).
23:20.49beardyWhat they said.
23:22.08BlackBishopdoes call-limit limit outgoing calls ?
23:22.15citywokeither you can make multiple calls, or yuo are busy. if you can make multiple calls, you aren't busy until you hit your call-limit
23:22.18BlackBishopwhat if in the future I'll try configuring "put on hold" ..
23:23.22BlackBishopthere should be a way to only give the busy tone if I call myself !
23:23.47citywokwhy would you need a busy tone for calling yourself?
23:23.53citywokthat's the most retarded feature ever
23:24.00beardyYou, and people should be clever enough not to do that.
23:24.12BlackBishopyeah, I did it just to see what happens ..
23:24.35BlackBishopand it doesn't happen like it does on my landline or cell phone
23:25.07beardyBecauese, like I said, you have more virtual incoming lines.
23:25.09citywokthose don't allow more than one line to be in a "ringing" state
23:25.12BlackBishopcitywok: I wouldn't consider it a feature .. but calling myself shouldn't be allowed either ..
23:25.27citywoka sip client doesnt mind if 5 lines are ringing at the same time
23:26.12beardyBlackBishop: Then you do some ugly hacking in your dialplan.
23:26.35beardyIF from own ext. KILL
23:26.56citywokyea, if CDR(SRC) = CDR(DST) playback(busy)
23:27.11BlackBishopgoes read about dialplans
23:27.31citywokthough i would consider that a non-issue and not waste my time with it
23:29.40BlackBishopwell, it'll give me hours of "fun" to play with the dialplan ! :)
23:29.54BlackBishop'till I get bored and go to sleep ( 2:30AM )
23:32.30beardyWhat you want to do is just dumb it down to simulate a one-line POTS phone.. dumbing things down is almost always not very smart.
23:33.04BlackBishophow do I dumb it down if I only block calling myself ?
23:33.31citywokyou are removing the ability for something to work
23:33.37citywokthat's what i'd consider dumbing down
23:35.01BlackBishopI just don't see why I should be able to call myself ! and for the sake of fun at 2:30 AM because I can't sleep .. I'm just trying to see if I can do this ..
23:35.04beardyConfigure your softphone to only use one "line" instead, if that's what you must have.
23:35.41BlackBishopthe softphone doesn't have that "feature" ...
23:36.12BlackBishopwouldn't having "one line" also block putting calls on hold ?
23:37.26beardyDepends what you mean with that. Putting on hold and making more outgoing calls, no..
23:37.58beardyBut in general, lift your mind from POTS telephony limits.
23:38.24BlackBishopnot only more outgoing, but more incomming ..
23:39.06BlackBishopwonders why irssi doesn't have a spellchecker by default
23:39.44beardyBecause it would annoy 99% of its users, which would become 3 people if it did.
23:39.59TJNIIBecause some of us can spele properly, damnit!
23:40.59BlackBishopI didn't mean actually enabled by default .. but have the .perl script there ..
23:41.05BlackBishopsorry for the off-topic thingy :)
23:46.13BlackBishopexten => _X,1,GotoIf($["${CDR(src)}" = "${CDR(dst)}"]?Busy)
23:46.28BlackBishopnow I gotta learn the syntax :)
23:47.39*** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com)
23:49.17*** join/#asterisk NuclearLucifer (gavroche@gavroche.pl)
23:54.55BlackBishopreload gets extensions.conf reloaded too, right ?
23:55.27NuclearLuciferright.
23:55.43NuclearLuciferBut you can reload only extensions witt `dialplan reload'.
23:57.07BlackBishopnoted, thanks :)
23:57.36BlackBishopnow to figure out what exactly I'm doing wrong .. ( besides everything I'm trying to do ! )

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