IRC log for #asterisk on 20080122

00:01.57*** join/#asterisk NeonLevel (i=NeonLeve@200.52.142.184)
00:01.59*** part/#asterisk NeonLevel (i=NeonLeve@200.52.142.184)
00:03.52lmadsenanyone ever see a linksys spa942 just hang on "Checking DNS" and not allow you to factory reset the phone, and also, it won't save any settings... (this phone has worked fine in the past... but ... ya... it's a POS :))
00:09.18*** join/#asterisk entelechy (i=user@mail.beanproducts.com)
00:09.59entelechyhello
00:10.11entelechyi recently was playing with the 1.6 beta & asterisknow
00:10.25entelechyi have an old 1.4.11 instllation that works fine, backed up
00:10.42entelechybut after upgrading to 1.6 beta and correcting a few semantic issues, i am getting this error:
00:10.43entelechy[Jan 21 18:04:27] WARNING[1979]: pbx_config.c:1511 pbx_load_config: ==!!== Unknown directive: gui_ring_groupname at line 653 -- IGNORING!!!
00:11.14entelechyfor every line in extensions.conf that uses "gui_ring_groupname"
00:11.38*** join/#asterisk kimitaka (n=swiceje@cpe-065-184-219-014.ec.res.rr.com)
00:11.44entelechywhich asterisk module defines "gui_ring_groupname
00:12.07JunK-Ylmadsen: are you sure that phone has network connection?
00:12.33lmadsenJunK-Y: yes, it's plugged directly into the same router that everything else is working off of
00:12.38lmadsenit gets an IP, and I can get to the web interface
00:12.47lmadsenit just won't do anything I tell it to do
00:13.04kimitakaanyone using app_rpt or know much about it?
00:13.14lmadsenentelechy: show us the line in question
00:13.38JTkimitaka: i know a bit
00:13.41JunK-Ylmadsen: tell him to stop crack smokin'
00:13.42entelechygui_ring_groupname = FMP
00:13.47entelechygui_ring_groupname = BP AP
00:13.48lmadsenwhere is that?
00:13.54entelechyin extensions.conf
00:13.57entelechyCreated by asterisknow
00:14.04entelechypart of a voice menu
00:14.07lmadsenthat doesn't look like a valid dialplan line to me
00:14.07kimitakado you have to have a repeater controller or will asterisk do that part too?
00:14.16jjshoelmadsen 100% sure the port is good on the switch?
00:14.24lmadsenjjshoe: yep -- I've tried several ports
00:14.43entelechylmadsen: wanna see context? this was autogenerated by the 1.4 version of asterisk now
00:14.50entelechyi wouldnt think that much would break
00:14.59lmadsenit was plugged into a Cisco phone (I have a bunch of phones daisy chained), and I moved it just to check, but it's worked prior to tonight....
00:15.01*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
00:15.11lmadsenentelechy: nope -- I don't deal with auto-generated stuff
00:15.12entelechyi noticed the Goto(1|s|1) syntax is now completely deprecated
00:15.19entelechyyeah well
00:15.19lmadsenyou can't use pipes, only commas
00:15.28entelechyyep -- I KNOW -- as i said.
00:15.42lmadsenI know -- I was stating it for the room for those who didn't know
00:15.53entelechythe error message isnt comning from asterisk GUI. it is coming from the asterisk console. /usr/sbin/asterisk -- thats why i am asking here.
00:15.56lmadsenin case anyone was wondering WHAT the correct syntax was
00:16.49*** join/#asterisk Dr{Who} (n=mathewss@dev.null.nutech.com)
00:17.02entelechyi have editted extensions.conf by hand extensively in the past. i mostly know what i'm doing. thats why this error is so confusing
00:17.16entelechythis is a very simple voice menu
00:17.36Dr{Who}dho!.. anyone use ekiga? i just signed a week ago and i got a RANDOM call from someone saying they saw my # on some site.. blaa.. blaa..
00:17.46lmadsensomething other than standard dialplan logic must parse that line, because that is not standard
00:20.52nhuisman_workQuestion:  Say I have two contexts, [phones] and [inbound-calls]  My phones are all defined in [phones] and my pri is in the context [pri]  I don't want to have to write out hundreds of lines like ( exten => 342,1,GoTo(phones,342,1).  can I just use a pattern exten => _XXX,1,Dial(SIP/${EXTEN},) ?
00:21.09nhuisman_workand is that a best practice
00:21.10nhuisman_work?
00:21.26nhuisman_worksince you don't want your inbound context to be able to dial out
00:21.35[TK]D-Fender<PROTECTED>
00:21.36nhuisman_workmy [phones] context includes my outbound patterns
00:22.03[TK]D-Fendernhuisman_work, and no thoses Gotos are completely the wrong approach
00:22.10*** join/#asterisk Bleak (n=asdad@adsl-157-63.click.com.py)
00:22.24nhuisman_workhere i'll show you what I have to make more sense
00:22.31nhuisman_workhttp://pastebin.com/m3f414fa9
00:22.51nhuisman_worksorry i typed wrong i don't have gotos
00:22.55nhuisman_worki was just dialing the extensions
00:23.14nhuisman_worki think that's totally wrong.
00:23.42*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
00:24.20hmmhesays[TK]D-Fender: do you know if there is a way to make the microbrowser refresh every time the services button is pushed?
00:25.48nhuisman_worki guess one question is do includes include includes
00:25.57nhuisman_workso if I include a context that is including other contexts
00:26.02nhuisman_workdo I get those by including the first context
00:26.05[TK]D-Fenderhmmhesays, nope
00:26.38[TK]D-Fendernhuisman_work, exten => _XXX,1,Dial(SIP/${EXTEN},15) <- ugly becasue you can dial illegal #'s
00:26.53nhuisman_workso it's best to statically define all the numbers
00:27.01nhuisman_workand just deal with having lots of extensions in there
00:27.49[TK]D-Fendernhuisman_work, yup.
00:28.07[TK]D-Fendernhuisman_work, exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) <- "9" prefex... ew.. thats so... 80's...
00:28.16nhuisman_workthat's from the asterisk book
00:28.53nhuisman_workthe main reason it is forcing to dial 9 is because our current system does it and I have a feeling it would confuse the shit out of people to change now
00:28.55*** join/#asterisk RoyK (n=roy@91.149.19.19)
00:29.15nhuisman_workis that what you mean?
00:29.30nhuisman_workby that I mean, do you mean that dialing 9 to get out is 80s
00:30.09nhuisman_workI think I'll change it so you can not dial 9 as well
00:30.25*** join/#asterisk ManxPower (n=manxpowe@7.sub-70-218-140.myvzw.com)
00:30.57hmmhesays[TK]D-Fender: so you have to push refresh every single time?
00:31.06[TK]D-Fenderhmmhesays, yup.
00:31.39ManxPowerOr you can tell your polycom to check for new configs every night at X time and reboot and load them if they changed.
00:31.50[TK]D-Fendernhuisman_work, No system I depoly forces any kind of prefix.  in areas where 10-digit dialing is mandatory on standard phone lines I allow 7-digit and add the default area code
00:32.34Qwell[TK]D-Fender: comes to Huntsville.  Try to setup 7 digit dialing here...it's a PITA.
00:32.37nhuisman_workthat makes sense, I just need to allow people to add 9 since tha's how it is now, then I'll add other patterns for people who don't dial 9
00:32.40Qwells/s//
00:33.09Qwellsome prefixes require 7 digits, others require 10
00:33.13[TK]D-Fendernhuisman_work, sad.  tell them to change.
00:33.16jjshoecome to la and setup 7 digit :D
00:33.16[TK]D-Fendernhuisman_work, its for the best
00:33.18Qwellso you have to put all of the prefixes into extensions.conf
00:33.39Qwelljjshoe: people there expect to 10 digit dial
00:33.39[TK]D-FenderQwell, I'm sure a lookup table would do nicely :)
00:33.46Qwell[TK]D-Fender: no such thing exists
00:33.56ManxPoweror get a carrier that lets you dial all calls as 10/11 digits
00:33.57Qwelleven the telco couldn't tell us
00:34.03nhuisman_work[TK]D-Fender, I was going to tell them they no longer need to dial 9 but I have a feeling If I add the rule to allow 9 it's going to make it so much easier on myself.  Our other offices all still use dial 9 to get out, it would really confuse visitors
00:34.20[TK]D-FenderQwell, neither did the pyramids... now all I need is a gang of slaves to build this list just the same!
00:34.24nhuisman_workI see where you are coming from though
00:35.01jjshoeQwell yeah, it was in 91 when it split up
00:35.13kimitakawhat does alsa console driver do?
00:35.14[TK]D-Fendernhuisman_work, no prefix also means you can call "missed calls" on your phone with impunity as their CID won't have a "9" on it.
00:35.23[TK]D-Fendernhuisman_work, its a question of functionailty too....
00:35.57nhuisman_workwait, why would incoming calls have a 9 added to the caller id?
00:36.17nhuisman_workif asterisk strips them off before dialing out
00:36.25nhuisman_workwhich doesn't even affect incoming
00:36.36jjshoealthough it wasn't until 99 when changes we made to force 10 digit dialing, but still, almost 10 years ago
00:36.49[TK]D-Fendernhuisman_work, incoming calls DON'T have the "9", thats the issue
00:36.54jjshoeoh more likes
00:37.00jjshoeQwell 10 digit wasn't required until 2006
00:37.04nhuisman_work[TK]D-Fender, oh, i understand what you mean.
00:37.04jjshoelies*
00:37.14jjshoeHaving been staved off nearly seven years, the 424 overlay was finally implemented on July 26, 2006 and new telephone numbers issued in the 310 area code may now begin with either 310 or 424. Ten-digit dialing within the 310 area code became optional on January 1, 2006 and mandatory on July 26, 2006.
00:37.15Qwellit was required before 2006
00:37.19Qwellbecause there are many areacodes :p
00:37.23[TK]D-Fendernhuisman_work, if you miss a call passed on to your phone you wouldn't be able to hit the "callback" feature on the phone to call them because YOU have to have a "9" in front
00:37.33Qwell323, 626, 213, 310
00:37.41nhuisman_work[TK]D-Fender, yeah that dialplan wasn't complete, i'm adding the ability to also not dial 9
00:37.51nhuisman_worksorry
00:37.58QwellI know a lot of people who had next door neighbors with a different areacode
00:38.14jjshoeQwell not within your own area code though
00:38.30Qwellno, but nobody ever gives a number without areacode
00:38.41jjshoeyou haven't spent time in 310 :)
00:38.54QwellI still do it here..
00:39.07nhuisman_workso it would look more like this : http://pastebin.com/m5b05034b
00:39.09jjshoeI never give my area code out when I frequest buisnesses at home in the south bay
00:39.15jjshoebut I guess it's all personal and dependent on the area
00:39.32nhuisman_workunless i'm stupid and that won't work.
00:39.37Qwelljjshoe: you didn't grow up where that was necessary
00:40.22drmessanoTheres really no such thing as a 7 digit phone number anymore
00:40.36drmessanoArea codes today are what exchanges were 30 years ago
00:42.27*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f6a40703eb7540c0)
00:45.40drmessanoI guess I pretty well killed that convo :(
00:45.53nhuisman_workfun stuff
00:46.19nhuisman_workseppuku
00:47.34[TK]D-FenderI sliced my hand at the dojo just taking my sword out of its bag.
00:48.28*** part/#asterisk RoyK (n=roy@91.149.19.19)
00:48.49nhuisman_workheh
00:48.57nhuisman_workwhy do you have sharp swords at your dojo?
00:49.05[TK]D-Fenderit slid out of the saya as I was pulling the bad away and made an inch-long cut on my hand.  Ttok about a few seconds and then just kep bleeding.  fortunately nothing much to sop up
00:49.23drmessanommmm Shinchirin
00:49.26[TK]D-FenderI just let it cake on to dry and seal up
00:49.29nhuisman_workwould you just practice with wooden or shanias
00:49.40drmessano-n
00:50.14[TK]D-FenderI am a student of katori shinto so I use a katana for the iaijutsu component
00:50.41nhuisman_workah
00:50.47nhuisman_worki guess that wouldn't really work with a wooden sword :P
00:50.48[TK]D-Fendernhuisman_work, and the rest of my session (I did this right at the START) I did with jsut my boken with the person I came to teach.
00:51.02drmessanoSensei allowed us to practice with katanas the first week of class.. good way to eliminate overcrowing and filter out the weak
00:51.20*** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133)
00:51.36drmessano14 ambulances later, and we had a nice small, strong class
00:51.45[TK]D-Fenderdrmessano, OUCH
00:52.00nhuisman_workseriously?
00:52.08[TK]D-Fenderdrmessano, there is only 1 person other than myself with a real sword in the class, the rest are iaito.
00:52.32drmessanono, not really.. I cant even cook in a wok without requiring medical attention
00:52.35nhuisman_worki guess it forces you to not hit yourself with a sharpened oe
00:53.01drmessanoIm the second most dangerous person I know, next to my extremely clumsy wife
00:53.13[TK]D-Fenderdrmessano, and that other guy is a psycho and has cut himself numerous times in class.  mine was a careless un-bagging mistake which shouldn't have happened.
00:53.19nhuisman_workby dangerous you mean self endangering?
00:53.21drmessanoNiice
00:53.34drmessanoNo, we tend to harm others too in our wake
00:53.38nhuisman_worklaugh
00:53.43[TK]D-Fenderdrmessano, eek
00:54.13kimitakahow long shoudl asterisk take to compile on a 300mhz g3?
00:54.20nhuisman_workwhere can I find documentation on running a stability test call type of overload test
00:54.20[TK]D-FenderI've got my custom katana still on order.... awaiting the next shipment of blades for inspection.
00:54.30nhuisman_workie, make a whole bunch of calls to different extensions
00:54.32nhuisman_workand to the outside
00:54.35nhuisman_workto stress the system
00:55.14*** join/#asterisk gmcneish (n=gary@samclay.plus.com)
00:55.18jjshoeQwell I did actually, when they split chicago up into multiple area codes
00:55.22drmessanoWhen I am working, I am a calm, strong thinker, who puts in 110%.. For everything else, I give about 60%, which is where the carelessness comes in
00:56.01drmessanoI can wash a car and leave it dirtier than it was... do I care, not really.
00:56.03gmcneishhi can someone please look at my extensions.conf and see where im going wrong im using sip trunks and i need to delete the crap thats not related to sip  out of it
00:56.16nhuisman_workgmcneish, pastebin it
00:56.19gmcneishcool
00:57.15gmcneishhow do i select all in vim
00:57.51nhuisman_workyou in a window manger environment?
00:57.58nhuisman_workjust select it on your terminal screen
00:58.10drmessanoGood god... am I the only person that uses Nano
00:58.11nhuisman_workotherwise cat the file to your terminal window and then scroll up and select it all
00:58.12gmcneishi dont know ill just put it together my self
00:58.16drmessanoI feel like such a windows admin :(
00:58.25nhuisman_workyucky yucky nano
00:58.37nhuisman_workyou can select all in vim but it doesn't paste it for outside applications
00:58.46nhuisman_workshift + V and then press the up and down arrows
00:59.05nhuisman_workit's pretty sad
00:59.10drmessanolol
00:59.12nhuisman_worki'll be in a normal tex editor
00:59.24nhuisman_workand i'll be going into insert context and typing :w
00:59.29nhuisman_workthen go wtf, this isn't vim
00:59.41gmcneishits just im remoteing to another machine that only has comand line
00:59.59nhuisman_workgmcneish, remote in then cat the file
01:00.04nhuisman_workthen copy it from your terminal
01:00.14[TK]D-Fenderdrmessano, I uses nano.... when I don't have MC around :)
01:00.53nhuisman_worki use vim because all unix boxes have it
01:01.12nhuisman_worksorry *nix
01:01.46nhuisman_workit's not very hard to figure out nano though if you're stuck with out vim.
01:01.51NovceGuruI'm quoting a client for a asterisk appliance :D :D
01:01.51St1ckm4nI'm hoping to get some second opinions about our current asterisk situation, short version we're on an unstable asterisk@home install v1.2.3 and I want to rebuild for a call center should I use 1.4 or stick with latest of 1.2 please pm me if you are willing to discuss in more detail
01:01.53drmessanolol
01:02.00[TK]D-Fendernhuisman_work, the reverse can't be said...
01:02.07nhuisman_work[TK]D-Fender, yeah I know
01:02.15drmessanoIm gload, TK... I was getting the impression nano was some loser app
01:02.19nhuisman_work[TK]D-Fender, hence learning the harder one seemed like the way to go
01:02.25drmessanos/gload/glad
01:02.35nhuisman_workdrmessano, you gloader!
01:02.35[TK]D-FenderSt1ckm4n, 1.4.  1.2 is dead.
01:02.36drmessanoMaybe I need its spell checker
01:03.11drmessano1.2 is not dead.. it will be forked and live on FOREVER
01:03.33[TK]D-Fenderdrmessano, they mock us for it, but I like nano because I can edit in a "normal" way.  VI does become incredibly fast and efficient once you've put some time into it though.
01:03.45jblackFree software never dies. It just bit-rots eternally
01:03.46[TK]D-Fenderdrmessano, but its still cryptic shit!
01:04.08gmcneishdone http://pastebin.com/m1ef43ea8
01:04.26St1ckm4nI'm a little scared of 1.4 since we've been on 1.2 and I hear agentcallbacklogin has been depracated
01:04.36NovceGuruvi/vim is insane once you know some of the simple shortcut
01:04.37NovceGurus
01:04.50gmcneishi think i only need the bottom section as it says something about sip
01:04.56drmessano[TK]D-Fender, I know that merely hanging out here makes this statement complete hipocracy, but I don't think I am dorky enough for VI
01:04.59[TK]D-FenderSt1ckm4n, go download 1.4 read the upgrade notes, do the same with your 1.2 notes
01:05.01gmcneishalso im only going to be calling the uk
01:05.44*** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com)
01:05.52[TK]D-Fenderdrmessano, I have better things to focus on that learning a cryptic text editor.  I go in to get a job DONE.
01:06.00drmessanoYep
01:06.02jblackHere's a quote. "I bow down before you. I thought I had done some rather horrible things with gcc built-ins and macros, but I hereby hand over my crown to you. As my daughter would say: that patch fell out of the ugly tree, and hit every branch on the way down. Very impressive." -- Linus Torvalds
01:06.45gmcneishdone http://pastebin.com/m1ef43ea8
01:06.55drmessanoI think if you've ever kissed a real woman, you should not have to prove your worth by showing some knowledge of cryptic apps like VI
01:07.16drmessanoor knowing Pi to 15 places
01:07.50[TK]D-Fendergmcneish, tip : don't shove everything into a single ridiculous context making a psychotic mess.
01:08.01*** join/#asterisk kyron_ (n=kyron@211-217-static-ppp.3menatwork.com)
01:08.07nhuisman_work[TK]D-Fender, heh
01:08.10drmessano"I bet you dont know all the GCC compiler error codes!"  "I kissed a woman once"  "Ok, fine"
01:08.30drmessanoThats an XKCD in the making
01:08.36hmmhesaysbah my perl script is chomping the entire fscking line
01:09.11[TK]D-Fenderhmmhesays, bits, bytes, chomping..... all this computer talk is making me hungry!
01:09.33ManxPowerI know enough vi to edit and save a file.  Not fast, not efficient, but I can do it.
01:09.44gmcneishme to
01:09.47ManxPowerEVERYONE that works on *nix should know that much.
01:10.12nhuisman_worki would hope so
01:10.17riddleboxManxPower, thats about as much vi as I know too
01:10.42jblackOne of the best investments in time I ever made was to learn vim usage.
01:10.43hmmhesays[TK]D-Fender: I just ate
01:10.54hmmhesayschomp should only chomp \n by default in perl
01:11.06riddleboxgrep is something that I need to learn more of, as well
01:11.10ManxPowerjoe is the editor I use.  It uses WordStar keybindings.
01:12.05riddleboxI remember trying to use jove a long time ago
01:13.44drmessanoI think every windows user should learn edit.com, just in case
01:13.54nhuisman_workfuck it, use ed
01:14.06nhuisman_workdo you really need to learn edit.com
01:14.13nhuisman_workseriously what is there to learn
01:14.15nhuisman_workhehe
01:14.18drmessanoedlin
01:14.23[TK]D-Fendernhuisman_work, thats the point, its intuitive!
01:14.29drmessanoyes
01:14.31nhuisman_workit's also very limited
01:14.39riddleboxhehe I use nano a lot
01:14.56drmessanoIf I am gonna learn VI to be one of the cool kids, everyone should learn DOS EDIT for the same reason lol
01:15.01gmcneishso http://pastebin.com/m3ea05117
01:15.20ManxPowerdrmessano: you need to learn VI because it is the only editor that is ALWAYS installed.
01:15.22nhuisman_workI'm not advocating vim and saying all other editors suck, I'm just saying you better at least know the basics of vim and other default editors as well.
01:15.45gmcneishcan anyone help me first off theres a bit at the bottom about sip now im using siptrunks what part of the file will i need to delete
01:15.51nhuisman_workyou don't need to be an uber im geek
01:15.54nhuisman_workim=vim
01:15.57drmessanoI can open, edit, and save a file.. guess thats enough
01:16.03nhuisman_workyeah good enough then
01:16.03gmcneish:wq
01:16.05jblackgmcneish: Not one single include in that entire file?
01:16.17*** join/#asterisk fnordus (n=dnall@24.84.160.227)
01:16.21ManxPowerdrmessano: It is enough
01:16.22nhuisman_worklearning emacs on the other hand
01:16.54ManxPowerCtrunk
01:16.54gmcneishwhats an include
01:16.54nhuisman_worki dont' think you need to do :P
01:16.54ManxPower~trunk
01:16.55jboti guess trunk is is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term.
01:16.55drmessanonano loves me more
01:16.55drmessanoI have nano on windows..
01:16.55gmcneishyeah ill put the trunk bit in tommorow
01:16.55drmessanoJust because
01:17.08gmcneishi need to phone my provider i lost the details but its the bottom part thats got the ; before it
01:17.26riddleboxdrmessano, the only thing I wish nano could do is search, or jump to a line
01:17.36drmessanoBesides, if I run into any hardcore VI issues, jblack has my back
01:17.43gmcneishin my sip.conf file i have extensions 101 and 102 do i reference these in extensions.conf
01:17.48hijacked^w ?
01:17.58drmessano^w
01:18.05nhuisman_work^X
01:18.05drmessanosearch works awesome
01:18.14drmessanoand nano you can open a line from the CLI
01:18.21nhuisman_workso can vim
01:18.23riddleboxreally?
01:18.24nhuisman_work!command
01:18.29MihiNomenEstI think it's ^W ^g to search.
01:18.35drmessano^w to search
01:18.38ManxPowergmcneish: you do NOT have extensions 101 and 102 in your sip.conf file.  You have SIP accounts/users 101 and 102 in your sip.conf.
01:18.42nhuisman_work: then type !ls
01:18.47drmessanonano +120 /etc/blah
01:18.52MihiNomenEstsorry, I meant goto line.
01:19.04drmessanonano +187 blah
01:19.05ManxPowerYou might have done something utterly silly and newbieish and named your SIP account names and your extensions the same, but that would be short sighted.
01:19.11gmcneishok chers so users 101 and 102
01:19.20nhuisman_workwait i don't think i understand what you mean by open a line from the cli
01:19.26drmessanoLike
01:19.27gmcneishdo they get a mension in extensions.conf?
01:19.29ManxPowerexten => 101,1,Dial(SIP/101)
01:19.32drmessanoI can open line 160 from the CLI
01:19.32nhuisman_work!ls is external command
01:19.37drmessanoOr go to 160
01:19.45drmessanonano +160 /etc/blah
01:19.47nhuisman_workwhen you say cli do you mean running nano as a command?
01:19.56drmessanoyes, from the command line
01:20.08drmessanonano +160 /path
01:20.31gmcneishnow do i do that for every account
01:20.34nhuisman_workdisplaying a given line is a common thing you can do with linux utils
01:20.51drmessanonhuisman_work
01:20.54drmessanoI cant make this any clearer
01:21.03drmessanoI can open nano and GO TO THE LINE
01:21.18drmessanoif I told you go to line 160 in blah, I can nano +160 blah
01:21.28nhuisman_worki get it
01:21.50nhuisman_workvim has the same command
01:21.54drmessanook
01:21.59ManxPower^Kl is the command in jow
01:22.04nhuisman_worksame syntax
01:24.33drmessanobbiaf
01:24.41gmcneishhttp://pastebin.com/m3ea05117
01:24.49gmcneishwhat parts of this file can i delete
01:25.49*** part/#asterisk Dr{Who} (n=mathewss@dev.null.nutech.com)
01:26.25[TK]D-Fendergmcneish, bad question.  How are we supposed to know whats important or not in there?
01:27.21gmcneishsorry
01:27.53gmcneishim new to asterisk i have a sip fowarder account and i think there is stuff in that file that is not related to sip
01:28.20gmcneishor at the bottom of the page it says ;#### VDAD SIP UNREGISTERED TRANSFER ENTRIES ####
01:28.20gmcneish;#### Use these entries IN PLACE OF the entries above if you are using SIP trunks
01:28.21gmcneish;#### and are not registering your provider in sip.conf
01:28.30[TK]D-Fendergmcneish, Do you seem to understand the dialplan at all.
01:28.58[TK]D-Fendergmcneish, the dialplan controls what happens on EVERY call you make.
01:28.59gmcneishim trying to
01:29.17[TK]D-Fendergmcneish, it is not "sip" related unless you DIAL a SIP resource.
01:29.57gmcneishok is this the only section i need http://pastebin.com/m3e2ba04c
01:30.00[TK]D-Fendergmcneish, you may use * to take calls from a SIP ITSP and have NO reference to SIP in your dialplan at all if you are only an inbound call-center for example.
01:30.17[TK]D-Fendergmcneish, We still have no idea what parts aren't needed.
01:30.30[TK]D-Fendergmcneish, Go in there and remove stuff you don't need YOURSELF./
01:30.35gmcneishim outbound only to the uk
01:31.34gmcneishbut i dont want to remove parts that will affect my dialer
01:32.20gmcneishis an 800 number america only
01:32.32[TK]D-Fendergmcneish, Ok you have NO clue whatsoever.  Go ask a forum in your dialer community what parts you have to leave behind and then install * on another system and go learn it.
01:32.58gmcneishbut they all tel me oh thats an asterisk question go ask them
01:33.12gmcneishis an 800 number america only
01:33.20[TK]D-Fendergmcneish, then leave this system alone and learn * on another box.
01:33.45gmcneishlook my question is simple is 800 america
01:34.20nhuisman_workyes
01:34.26gmcneishthank you
01:34.45nhuisman_workthat was me googling, you better double check though
01:34.53[TK]D-Fendergmcneish, nobody here know what parts your dialer really NEEDS so we can't help you.
01:35.58gmcneishits just the extensions.conf file had zap stuff in it and iax and sip so i have to delete the irelivant stuff
01:36.41jblackWoot.
01:36.48jblackhttp://maps.google.com/maps/ms?hl=en&gl=us&ie=UTF8&lr=lang_en&msa=0&msid=100618373221056061016.000443665b744f9405cb3
01:36.58nhuisman_worknevermind :The toll-free prefix 800 has been widely adopted elsewhere, including as the international toll-free number. It is often preceded by a 0 rather than a 1 in many countries where "O for Operator" has no meaning in the national language.
01:37.02nhuisman_worki think it's not us only
01:37.17nhuisman_workjblack, what is that?
01:37.20jblackI think we may get a san diego node, and possibl even a colorado node.
01:37.21nhuisman_workasterisk trunks?
01:37.25jblackThat's the dundi network that I'm in.
01:37.30nhuisman_workcool
01:39.42nhuisman_workso http://www.nslu2-linux.org/wiki/Optware/AsteriskStdextenMacro those extensions below the macros allow you to hit *xx and then it starts forwarding to the number you specified to
01:39.55nhuisman_workbase on which one you chose
01:40.39St1ckm4nI just finished reading the upgrade notes on 1.4 but wanted to know if you guys feel that 1.4 is less or more reliable than 1.2.26?
01:41.44nhuisman_worki think the consensus is that 1.4 is ready for production
01:41.46[TK]D-FenderSt1ckm4n, 1.2 is DEAD.  No more bug-fixes coming, and 1.6 is is beta.
01:41.48drmessanojblack: thats hardcore
01:41.59nhuisman_workABE is now on 1.4 and that's digiums supported version
01:42.14tzangerABE's now 1.4?
01:42.21nhuisman_workthey just realized it a few days ago
01:42.26tzangeris there any official documentation to the effect/
01:42.27nhuisman_worki'm waiting for them to fix their installer iso
01:42.40nhuisman_workit's available in the be downloads
01:42.41jblackdrmessano: Well, we're trying to be hardcore
01:42.47drmessano1.2 is security maintainence only?
01:42.58nhuisman_worktheir iso isn't complete yet though
01:43.08nhuisman_worki opened a support case to find out what the deal was.
01:43.19nhuisman_workby not complete I mean there isn't one with rpath on it, the rpms are all there.
01:43.48tzangerthere was some rumour that digium's internal system was 1.2 while they were suggesting for everyone to use 1.4.  I don't much care, as I run 1.4 (svn trunk actually) for most systems
01:43.57nhuisman_workit was 1.2
01:44.07nhuisman_workoh you mean their own phone system
01:44.10nhuisman_worknm i don't know anything about that
01:48.43*** part/#asterisk beek (n=klinebl@65.211.106.243)
01:49.50gmcneishis the top part of this file related to sip
01:49.55gmcneishhttp://pastebin.com/m7a0a2f07
01:53.45gmcneishhttp://pastebin.com/m7e9992bc   if this is for a long distance uk number how would i change this to a local uk number
01:54.11gmcneishusing sip trunks
01:55.58[TK]D-Fendergmcneish, that garbage does not  mean ANYTHING.
01:56.18[TK]D-Fendergmcneish, AGI's from GUI's like that are unreadble trash
01:56.53gmcneishok so what txt would i need to put into my dial pattern if i wanted to dial only uk numbers
01:56.57[TK]D-Fendergmcneish, you have LOT of learning to do if you want to try and admin a box thats running a dialer like that and I don't anyone here is going to give you that kind of time.
01:56.58gmcneishon sip trunks
01:57.16[TK]D-Fendergmcneish, and stop using the term "sip trunk" like is a magic term
01:57.28*** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id)
01:57.36[TK]D-Fendergmcneish, noone can see how you are suposed to call out your provider based on that.
01:58.32gmcneishoh i thought there would just be a standard dial plan for calling local uk
01:58.49[TK]D-Fendergmcneish, if I were you I'd stop trying to mess with that system.  You don't userstand Asterisk at all and we can't help you with the junk your dialer throws in.
01:59.17[TK]D-Fendergmcneish, there IS no standard with *.  * is like PAINT, its whatever you want it to be and you don't even have a start.
01:59.27[TK]D-Fendergmcneish, you are in way over your head.
01:59.39gmcneishtell me about it but im getting there
02:00.24[TK]D-Fendergmcneish, You don't seem to be anywhere right now.  Stop, go read the book, and install * yourself on another machine and go learn it
02:00.33gmcneishok so how do i tell extensions.conf to call uk numbers because mine is set to american dialing rules
02:01.49drmessanolol
02:01.57[TK]D-Fendergmcneish, ok.  You just don't get it.  Those stupid AGI's are controlling everything and doing the actual dialing. NOBODY HERE CAN HELP YOU.
02:02.06[TK]D-Fendergmcneish, that stupid dialer OWNS YOU.
02:02.18[TK]D-Fendergmcneish, You have virtually no control.
02:02.23drmessanoSo you help me, no?
02:02.49*** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net)
02:04.35gmcneishi dont see what ur problem is my question is asterisk releated imaging i have a blank extensions.conf file and i want to make my asterisk box call numbers in the uk only using siptrunks
02:05.03gmcneishlike this
02:05.03gmcneish; dial a long distance outbound number through a SIP provider
02:05.04gmcneish; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
02:05.04gmcneish;; exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
02:05.04gmcneish; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,55,o)
02:05.04gmcneish; exten => _91NXXNXXXXXX,3,Hangup
02:05.19[TK]D-Fendergmcneish, do NOT spam in here.
02:05.42gmcneishsorry
02:05.43[TK]D-Fendergmcneish, and if you don't like the patter on that exten, go CHANGE IT.  Chapter 5 of THE BOOK
02:05.44mostyi have an odd issue, some of my sip phones will be on a call for a few minutes sometimes, then the sip phone hears music on hold, while the other end can still hear the sip phone, i can't figure out how or why this would happen, anyone have any ideas for things to look for?
02:05.46[TK]D-Fender~book
02:05.46jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
02:06.13[TK]D-Fendergmcneish, and no promises that your provider will even LIKE the number you give them
02:06.30mosty[TK]D-Fender, btw the polycom v3 firmware works ok with bootrom 4.0 and even 3.something on my test phone
02:07.03[TK]D-Fendermosty, I know.... I have them both and just haven't gotten around to installing yet
02:07.14[TK]D-Fendermosty, just picked up 2.2.2 that I didn't know was released either
02:08.06mostythe other night you guys said that it required bootrom v4.1, which doesn't seem to exist, that's all
02:08.16lmadsenI just finished upgrading my IP501
02:08.28lmadsenI think I'm runnign 2.2.0 now
02:08.37[TK]D-Fendermosty, I never said it needed it.
02:09.04mostyi forget who the other person was, the person that noticed firmware v3
02:09.23lmadsenlike... v3.x?
02:09.25[TK]D-Fendermosty, [hC] was the one who told me,
02:09.45[TK]D-Fenderlmadsen, yes.  SIP 3.0.0 is out with BIG changes
02:09.51lmadsenoh really.. interesting
02:09.53[hC]yeah i noticed v3
02:09.56[hC]its not on their downloads link
02:09.57lmadsenwhat are some of the more interesting ones?
02:10.03[hC]i found it in some weird other upgrade matrix page
02:10.09[hC]i dont think it was supposed to be out yet
02:10.14mostylmadsen, ldap contacts (with a specific license)
02:10.19[hC]the most interesting things are now it will do LDAP/active directory contacts
02:10.25BBHossawesome
02:10.28mosty[hC], it's available now, if you're registered
02:10.30[hC]headset offhook functionality using the Jabra protocol
02:10.41[hC]mosty: yeah i am, it just wasnt in the download list when i found it
02:14.39mosty[hC], well it worked fine with the old bootrom on my ip550 test phone, in case you're interested
02:14.54*** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211)
02:15.52*** join/#asterisk postconf (n=marquis@gw-corp.postconf.com)
02:16.39BBHossso you have to have a certain license to do LDAP?
02:16.44mostyyes
02:16.57BBHosstaking a page out of Ci$co's book
02:26.49*** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com)
02:29.42mpruettHello Everyone
02:30.27drmessanoWhat?
02:30.43PrayerHello mpruett
02:30.53mostyis there a way that i can see current channels that are on MOH in the cli?
02:31.07mpruettQuiet tonight
02:32.01mpruettDoes anyone know if the "g" option for the Dial cmd works in a macro?
02:32.16Prayertry CLI>sip show channels
02:32.47Mavviempruett: why wouldn't it work?
02:33.38PrayerThe "j" option doesn't work for me in 1.2   - the new s${DIALSTATUS] etc works better though.
02:33.41mpruettDon't know - I am having troubles exucuting further commands in the macro after the call has hung up
02:33.56mpruettI am on 1.2 also
02:34.14PrayerUmmm I think the macro is exited after call done ?
02:34.38*** join/#asterisk theron (n=theron@65.198.151.150)
02:35.07mpruettAnyway to prevent this or send it to another macro?
02:36.42mostyPrayer, was that directed to me? i tried 'show channels' while i had a call on hold, it just said that the two sip channels were bridged, nothing about MOH
02:37.41PrayerI don't have MOH much so wasnt sure about that.
02:39.15*** part/#asterisk postconf (n=marquis@gw-corp.postconf.com)
02:39.20*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
02:39.31*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
02:39.51theronHi all, got a firewall related issue I believe, and I'm wondering if I'm down to just replacing a cheap router.  I have an asterisk 1.4 server behind a netgear WGR614v6.  I'm forwarding ports 5060-5082 tcp/udp, port 4569 for my provider's iax connection, and ports 10000-20000 udp. to my asterisk server on my internal network. I can receive and place calls all day long from internal to my network, however from the outside I get a connection, but no audio.
02:40.10entelechyi am STUNned
02:40.16entelechy;-)
02:40.20entelechyhopefully someone else can help you.
02:40.21*** part/#asterisk entelechy (i=user@mail.beanproducts.com)
02:40.55jjshoetheron do you see the RTP traffic at the router?
02:41.03jjshoedoes the router give you the ability to see the traffic?
02:41.13theronI know that to get the rtp connection back to my client I need to use stun, I believe that I've got all the required settings in my sip.conf for that....
02:41.25theronjjshoe: I can't see it.
02:42.04drmessanoThats NAT the problem
02:42.07jjshoetheron what happens when you take the router out of the connection and plug the asterisk box straight in?
02:42.09drmessano~sipnat
02:42.10jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
02:43.16BBHosstheron: did you double check the rules to make sure you allowed tcp/udp instead of just tcp
02:43.23theron..
02:43.32theronBBHoss.  tcp/udp for all?
02:43.41drmessanoUDP
02:43.43BBHosswell actually UDP
02:43.52BBHossbut SIP CAN run over TCP
02:43.52drmessanoCheck your NAT settings
02:43.53[TK]D-Fendertheron, Read the guide...
02:43.58drmessanoThats likely it
02:44.11[TK]D-Fender* does not support SIP over TCP in 1.4 standard.
02:44.26[TK]D-Fenderand * does not support STUN either, nor does it need it.
02:44.58drmessano[TK]D-Fender, it can't be the SIP NAT settings.. that's too easy
02:45.15drmessanoHas to be some obscure firewall setting.. so start there first
02:45.23drmessanoUpgrade the firmware
02:45.27drmessanoWait a week
02:45.29drmessanoComplain
02:45.33therondrmessano:  running latest.
02:45.36drmessanoThen check the sip.conf
02:45.38theronnot interested in complaning.
02:45.49theronI "believe" it's correct.
02:46.01drmessanoIts probably not
02:46.12[TK]D-Fendertheron, Hitler thought he was "correct" too :)
02:46.17theronbet you a sandwich ;)
02:46.26[TK]D-Fendertheron, PASTEBIN is your friend.
02:46.27[TK]D-Fender~pb
02:46.28jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
02:46.48*** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com)
02:46.50drmessanoMost people with external audio issues that insist their settings are correct are dead fscking wrong
02:47.24*** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com)
02:47.26lmadsenwow... my polycom takes 2-3x as long to boot on the new firmware :)
02:47.45jjshoeof course this is why I asked him to plug directly into his source around his router
02:47.48jjshoerather then bitch at him
02:47.54jjshoejust ask him to take the router out of the picture
02:47.57jjshoeit's far quicker.
02:48.01*** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp)
02:48.03theronhttp://pastebin.ca/868040
02:48.12jjshoelmadsen polycom wins the race for slowest booting phone I've ever seen
02:48.32mpruettsorry guys I got disconnect - Prayer (anyone?) do you know a way for me to execute commands after hangup of a call in a macro?
02:48.39lmadsenI'm shocked at how much longer it takes on the 2.x from the 1.x
02:49.24drmessanoFar quicker until he puts the router back
02:49.30drmessanoYoure not solving the problem
02:50.15jjshoedrmessano you're proving to him where the issue is
02:50.45*** join/#asterisk krapper (i=krapper@wsip-64-58-154-130.oc.oc.cox.net)
02:50.51jjshoedrmessano until you can make him believe what you and I know, he'll never look at it, so this method is insanely quicker
02:51.07[TK]D-Fenderjjshoe, and Polycom wins the race for "needs rebooting the fewest number of times"
02:51.10Prayermpruett, I think there is a way I read yesterday but cant remember details.  Maybe executions after the macro() call ? Or a variable set to make that occur.
02:51.34drmessanojjshoe.. So?
02:51.44theroneven if I wanted to take the router out of the picture atm it's at a remote location.
02:51.46drmessanoIts his box, not mine... if he wants it to FAIL, thats up to him
02:51.55drmessanopaste your damn config pls
02:51.58therondid.
02:52.03[TK]D-Fendertheron, Where is 6000?
02:52.14theron6000 is my softphone...
02:52.25theronpasting extensions.conf ....
02:52.28drmessanouh
02:52.30[TK]D-Fendertheron, Ah.  for [6000] add "nat=yes", "qualify=yes"
02:52.37Prayermpruett, I was trying to use SET(TIMELIMIT(absolute)) to limit call times. I wanted a warning though and found something better. The macro ending was hurting the absolute sometimes.
02:54.21jjshoedrmessano nice attitude.
02:54.41jjshoe[TK]D-Fender I don't have to reboot an aastra more then once and I can make every change I want *shrug*
02:54.41mpruettThat gave me an idea to try Prayer - thanks for the help
02:55.06[TK]D-Fenderjjshoe, I have an Aastra 57i CT that likes to spontaneoulsy lock up...
02:55.28drmessanojjshoe, I dont need a lecture
02:55.39[TK]D-Fenderjjshoe, Aastra is fairly decent though... but the 5i Series manufacturing build BLOWS
02:56.03Prayermpruett, tell about idea and if it worked or not when you are done with experiment.
02:56.48theron[TK]D-Fender, adding now... question though, if they're in the general, dosen't it apply to all below?
02:56.57jjshoe[TK]D-Fender yes, we can reproduce lockups in house, we're working with aastra engineering to get it cleared up
02:57.12jjshoe[TK]D-Fender with the 5 series anyways, the 9 and 4 series we don't have issues with
02:57.25jjshoedrmessano You consider two words a lecture?
02:57.35drmessanothe beginning of one
02:57.38[TK]D-Fendertheron, just add it, and you should probably tell your phone not to do any kind of nat-keep-alive.  Also you should not be forwarding ports to your phone, and you SHOULD be forwrarding ports to *.
02:57.40*** join/#asterisk krapper (i=krapper@wsip-64-58-154-130.oc.oc.cox.net)
02:57.44drmessanoSo I halted it before it began :)
02:58.02theronok, I'm not, and I am.
02:58.09jjshoedrmessano I didn't plan on talking to you any more, *shrug* you have a shitty attitude, why would I want to continnue?
02:58.10theronchecking.
02:58.21drmessanoFair enough :)
02:58.31[TK]D-Fenderjjshoe, 480i was ok, but the 5i was a step backwards.
02:58.49theron[TK]D-Fender, sip reload enough for that?
02:59.33[TK]D-Fenderjjshoe, NO weight to the unit or handset, tinny speakerphone (wieght would help), shitty rubber buttons, pixel based screen they are still treating like char matrix, PITA call handling, and well.. the lokcing up bit...
02:59.38[TK]D-Fendertheron, yup
02:59.49theron[TK]D-Fender, same behavior.
02:59.56[TK]D-Fenderjjshoe, drmessano : calm down already.
03:00.04drmessanoIm calm dude
03:00.19theron[TK]D-Fender, i can see asterisk doing what it does..... vm-login....etc.
03:00.32[TK]D-Fendertheron, jsut no audio, right?
03:00.42theron[TK]D-Fender, correct.
03:00.44[TK]D-Fendertheron, What routers are you using?
03:01.00[TK]D-Fendertheron, Cisco PIX, and many D-Links are key-offenders.
03:01.45theron[TK]D-Fender, cisco something between me and internet, then WGR614v6
03:01.53*** join/#asterisk _ShrikE-cell (n=_ShrikE-@32.162.249.118)
03:02.01theron[TK]D-Fender, I had a coworker check from his end though.
03:02.05[TK]D-Fendertheron, that may be part of the issue.
03:02.13jjshoe[TK]D-Fender I don't like the 5 series either, look/feel is awful, but most everyone I know likes it more then the 9 look
03:02.15[TK]D-Fendertheron, PM.
03:02.17theron[TK]D-Fender, (no firewall that side)
03:04.49drmessanodouble nat
03:13.00jjshoe[TK]D-Fender I'm checking to see if they've gone public with a specific phone yet... sec
03:13.11drmessanojjshoe, who do you work for?
03:13.15drmessanoIm just curious
03:13.25jjshoedrmessano fonality.
03:13.35drmessanoOhh
03:13.38[TK]D-Fenderjjshoe, new model?
03:13.39drmessanoWhat do you do there?
03:13.50jjshoe[TK]D-Fender yeah, not in the 5 series
03:13.51[TK]D-Fenderjjshoe, the 5i series had a lot of promise, but came up quite short.
03:14.01Prayermpruett, experiment done yet ?
03:14.09[TK]D-Fenderdrmessano, .... don't go there!
03:14.27nhuisman_work[TK]D-Fender, I've created two small extension.conf examples and I was wondering if you would comment on which method is the proper way to setup inbound and outbound calls in concurrency with internal phones.  The pri is on context = inbound.  I read that you don't want outbound capabilities on your inbound because that would allow callers to call in and then dial out.  But I also wanted to find out if it was possible to not have to write
03:14.27nhuisman_workeach extension twice
03:14.36nhuisman_workhttp://pastebin.com/m4dda816f
03:15.07JTeww fonality
03:15.09nhuisman_workpart of the question is: if I include the internal context, does it in turn include the outbound context
03:15.22drmessanojjshoe: What do you do at Fonality?
03:15.30drmessanoSounds cool
03:15.43nhuisman_workeveryone else also could comment
03:16.03[TK]D-Fendernhuisman_work, You don't seem to allow 10-digit dialing...
03:16.25[TK]D-Fendernhuisman_work, also I don't see ${OUTBOUNDTRUNK} defined anywhere
03:17.09nhuisman_workoops forgot to paste that in, in the example that issue isn't what i'm worried about.
03:18.00nhuisman_worki'm wondering what the best way to go about it is, do you have double entries for all your phones, or just include the context.
03:18.08[TK]D-Fendernhuisman_work, and you should leave [internal] to contain ONLY extens related to ringing phones connected to your system
03:18.10nhuisman_workand does including the context leave me open to bad things since it includes the outbounds
03:18.36[TK]D-Fendernhuisman_work, and then make a NEW context that includes [internal] and [outbound] and point your phone's sip.conf entries to THAT.
03:19.00nhuisman_worki see
03:20.30nhuisman_workwhen you say phones sip.conf you mean where I specify what context each phone is in right?
03:21.35jjshoe[TK]D-Fender http://www.abptech.com/products/Aastra/sip_dect.html
03:21.39nhuisman_workadjusting the example now to see if i'm undestanding it properly.
03:22.06[TK]D-Fenderjjshoe, those available now?
03:22.19[TK]D-Fendernhuisman_work, Correct
03:22.32drmessanoTrixbox Pro Engineer.. hmm
03:24.11jjshoe[TK]D-Fender dunno, but they rock.
03:24.37JTtrixbox pro is lame, they don't even "certify" polycom
03:24.41[TK]D-Fenderjjshoe, tease :p
03:25.21nhuisman_work[TK]D-Fender, ok so sip.conf phones are in the [phones] context, pri in zapata.conf is in [inbound].  http://pastebin.com/m3c7a9e3c
03:25.43drmessanoAsk jjshoe why, JT
03:25.52*** join/#asterisk Dayver (n=user@ip65-44-153-126.z153-44-65.customer.algx.net)
03:26.04nhuisman_worktrixbox also won't sell you support unless you use their hardware
03:26.09nhuisman_workor a list of two motherboards
03:26.15nhuisman_work....
03:26.18nhuisman_workfuck that
03:26.28jjshoenhuisman_work won't they? I just thought it was more *shrug*
03:26.31jjshoeI don't do shit with trixbox :)
03:26.43drmessanoYou dont?
03:26.54DayverI am looking for a good SIP provider with unlimited inbound lines. Pls help.
03:27.08jjshoe[TK]D-Fender range is INCREDIBLE
03:27.14drmessanohttp://www.trixbox.org/forums/trixbox-pro/trixbox-pro-general-information/new-reporting-engine-released
03:27.21drmessanoNot the same guy?
03:27.22[TK]D-Fenderjjshoe, Yeah I have a 5iCT... I know about the range...
03:27.38[TK]D-Fenderjjshoe, how many phones can an AP like that handle INDEPENDANT of each other?
03:27.54nhuisman_workdrmessano, snap.
03:28.48jjshoe[TK]D-Fender not sure, I don't test it
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03:29.09[TK]D-Fenderjjshoe, I'll have to investigate, thanks for the tip though, I'm in the market for a new one.
03:29.09nhuisman_worksomeone was trying to sell me a different wifi band of phone
03:29.22jjshoe[TK]D-Fender there's no "base" phone, just the white thingy you see, it's awesome
03:29.30jjshoenhuisman_work they won't sell you support but at a higher cost?
03:29.33nhuisman_worksupposedly worked better for in buildings.
03:29.49nhuisman_workjjshoe, i dunno they said they only supported their list of supported hardware
03:29.55jjshoenhuisman_work how long ago?
03:30.00nhuisman_worka month?
03:30.03[TK]D-Fenderjjshoe, will investigate.  Thanks.
03:30.28jjshoenhuisman_work pro or cce?
03:30.31nhuisman_workperhaps that just means no hardware support
03:31.10nhuisman_worktrying to find the links i was reading before
03:31.16jjshoenhuisman_work pro or ce?
03:31.28nhuisman_worki can't remember
03:31.56drmessanoDid I mention my favorite color is lime green?
03:32.01nhuisman_worki'll be honest I was looking at so many different choices I may have gotten them confused
03:32.05nhuisman_worki'll go find out
03:32.09jjshoenhuisman_work ok just asked, if it's not supported hardware you have to buy hourly
03:32.43jjshoelike we've all said a million times if it's not for you, stick with ce or whatever you already use *Shrug*
03:33.00nhuisman_workyeah that's fine
03:33.37*** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.mn.comcast.net)
03:33.57jjshoe[TK]D-Fender the only complaint about the wireless handsets, even on the 480ict, is that it doesn't take much to break the headset jack
03:34.14drmessanoTrix are for kids
03:34.19nhuisman_work[TK]D-Fender, any comment on that example I posted last? "[TK]D-Fender, ok so sip.conf phones are in the [phones] context, pri in zapata.conf is in [inbound].  http://pastebin.com/m3c7a9e3c"
03:34.30jjshoealright, I'm out, later home slices
03:34.32nhuisman_workspeaking of wireless handsets
03:34.40nhuisman_worki need to find some
03:34.47nhuisman_worknot 802.11 stuff
03:35.03nhuisman_workbut just a wireless handset that is sip
03:35.10nhuisman_worki guess the other option is to buy an ata
03:35.27jjshoenhuisman_work I just dropped a link...
03:35.41jjshoenhuisman_work down two floors and half way through the parking garage it sounds great :)
03:35.41[TK]D-Fendernhuisman_work, ATA is a good idea.
03:35.43nhuisman_workoh, let me scroll up
03:36.01jjshoe[19:21] <jjshoe> [TK]D-Fender http://www.abptech.com/products/Aastra/sip_dect.html
03:36.03[TK]D-Fenderthere is this too : http://www.telephonydepot.com/product_p/105-059-m3basic.htm
03:36.14nhuisman_workah yes
03:36.16nhuisman_workthose use dect
03:36.24jjshoeata is good if you have idjits who are going to drop them in a deep fat fryer
03:36.33nhuisman_workso it has dect in the handset and in the base?
03:36.36nhuisman_workor do I need aps everywhere
03:36.40nhuisman_workbleh aps everywhere
03:36.56jjshoebut yeah, out &
03:37.44nhuisman_workwhat do you mean drop them in a deep fat frier...
03:37.46nhuisman_workfryer
03:38.25drmessanoHmmm fried Trixbox
03:39.59nhuisman_workdeep fried.
03:40.06JTconsidering that fonality seems to never contribute back to asterisk
03:40.22JTthe question on when to buy fonality products seems like it has a clear answer
03:40.23JTnever
03:43.25*** join/#asterisk entelechy (i=user@mail.beanproducts.com)
03:44.29nhuisman_workcontext includes are one way right?
03:44.37*** join/#asterisk PepOSX (n=angeldav@190.78.221.19)
03:45.20[TK]D-Fendernhuisman_work, Correct.
03:45.52nhuisman_workk that makes more sense then, I kept getting afraid that by including outbound and inbound in the phones context it was letting inbound access outbound
03:46.20nhuisman_workthat makes it much simpler :)
03:47.01Olobolawhat is the going rate for asterisk setup/dialplan?
03:47.46[TK]D-FenderOlobola, Everything depends.  What do you need?
03:47.50nhuisman_work2 headaches for the first 10 hours
03:48.01nhuisman_workafter that it's only 1 headache every 72 hours
03:48.02[TK]D-Fendernhuisman_work, nice!
03:48.11nhuisman_work;)
03:49.17OlobolaSomeone wants to know to know the 'going rate'.. I suppose it could be any number.
03:49.26nhuisman_workthat's for a basic 10 extension plan.
03:49.50nhuisman_workdigium charges 400 for 2 hours
03:49.57nhuisman_workthat's for anything you could think to have them do though
03:50.28[TK]D-Fendernhuisman_work, 400 headaches?!?! OMG my skull would fracture!
03:50.46nhuisman_workdollars.  but there is a conversion from dollars to headaches, let me just look it up
03:51.45nhuisman_work1 U.S. dollar = 0.00683293475 headaches
03:52.17*** join/#asterisk entelechy (i=user@mail.beanproducts.com)
03:54.59nhuisman_workare includes transitive ?
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03:56.45[TK]D-Fendernhuisman_work, ?
03:57.08nhuisman_worka includes b, b includes c, does a include c
03:57.13[TK]D-Fendernhuisman_work, yes
03:57.15nhuisman_workkk
04:01.54nhuisman_workis there a way to play an all circuits busy message if the call can't be completed due to insufficient available channels?
04:02.24nhuisman_workwell obviously there is, should I just google for documentation or does someone already know where it can be found.
04:03.04[TK]D-Fendernhuisman_work, look at ${DIALSTATUS} after your call and play a message back if its called for
04:03.16nhuisman_workah yes that makes sense
04:03.25nhuisman_worki have a macro I can change to do that, thx.
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04:07.53nhuisman_workhmm i need an extra enter for 011 + country code + city code + number
04:08.00nhuisman_workenter=entry
04:12.52nhuisman_workhow do you do that? the country codes are not a set number of digits
04:13.56[TK]D-Fendernhuisman_work, you just do 011. and don't restrict the rest.
04:14.01nhuisman_workoh
04:14.12nhuisman_workso "011."
04:14.20[TK]D-Fendernhuisman_work, yup
04:14.26nhuisman_workah thanks.
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04:24.15ncampionI need some help with asterisk-gui's configure script, specifically the --with-zaptel option.  Anyone around?
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04:25.58[TK]D-Fenderncampion, GUI's are not supported in this channel.
04:26.08ncampionok
04:26.14nhuisman_workasterisknow?
04:26.15[TK]D-Fenderncampion, #asterisk-gui is their channel
04:26.16nhuisman_work#asterisknow
04:26.20ncampionthanks guys
04:26.20nhuisman_workah nm
04:26.27nhuisman_workdidn't know about asterisk-gui
04:26.48*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
04:27.23nhuisman_workit would be nice to see an addon for provisioning phones for asterisk
04:27.30nhuisman_workinstead of having to write all the scripts
04:27.57[TK]D-Fendernhuisman_work, GUI's do things like that.
04:28.10nhuisman_worki don't think asterisk-gui does
04:28.21[TK]D-Fendernhuisman_work, And the moment it configures your phones fo you it starts thinking its running the show, and not you.
04:28.47[TK]D-Fendernhuisman_work, yes, they has scripts for configuring Polycom's at least that I've heard of.  not sure of others.
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04:29.16[T]ankanyone know how to disable the do not disturb button on a polycom 301?
04:29.32nhuisman_worki know asterisknow has no provisioning, i thought that it used the asterisk-gui
04:29.34nhuisman_workdouble checking
04:30.32[TK]D-Fender[T]ank, you can remap the key in your provisioning.  Go check your admin guide.
04:32.14[TK]D-Fender[T]ank, in SIP 2.1.0+ you seem to be able to set it for server-side DND to which I presume you can send it to a dead-end as well
04:36.25*** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)
04:40.52nhuisman_workwhat are the cheapest atas for the price these days?
04:41.02nhuisman_worksorry i meant cheapest atas that aren't crap
04:41.08[TK]D-Fendernhuisman_work, Linksys.
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04:41.30nhuisman_workthey don't have 1 port ones eh
04:41.30[TK]D-Fendernhuisman_work, SPA-2102 is the norm.  PAP2's are a bit too low.
04:41.35[TK]D-Fendernhuisman_work, they do.
04:41.52nhuisman_worki only see 2 port ones
04:41.53PrayerSipura2000   I got several from eBay
04:42.11[TK]D-Fenderhttp://www.voipsupply.com/product_info.php?products_id=320
04:42.35nhuisman_workfunny that linksys doesn't have that on their site
04:42.42nhuisman_workah, discontinued
04:42.56[TK]D-Fendernhuisman_work, there's a LOT they don't have on their site, including a lot of current stuff...
04:43.12nhuisman_workwhat the hell is that, the replacement is a 2 port
04:43.35nhuisman_workoh well i'll just use those and leave them at the patch panel level
04:43.41nhuisman_workthen I can buy less
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04:44.28nhuisman_workeach interface is a separate phone line eh
04:44.36nhuisman_worki guess that's why there are 2 rj45 interfaces
04:44.44nhuisman_workeach is a different ip acting as a sip device
04:44.47jblackoh boy. Duck
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04:45.17[TK]D-Fendernhuisman_work, nope.
04:45.26dcmwaihello all.
04:45.32[TK]D-Fendernhuisman_work, the SPA-2102 can be a ROUTER.
04:45.44dcmwaiI would like to have your advise on linux-ha + Asterisk
04:45.47[TK]D-Fendernhuisman_work, and it does SIP out the single RJ-45
04:45.50jblackNugget: Nah. You're buying a stripped down version of an spa8k.
04:45.53nhuisman_workooh
04:46.04[TK]D-Fenderjblack, the SPA-8000 is *new* :)
04:46.22jblack[TK]D-Fender: Ok, then the 8k is a stripped up. :)
04:46.29nhuisman_workso if I don't need a router there is nothing less then this eh
04:46.30[TK]D-Fenderjblack, So its more like a duct-taped bunch of SPA-2102's by comparison :)
04:46.42dcmwaiMy coulleauge haev been configuring 2 Asterisk server + Linux HA but we have some problem.
04:47.01jblackpoint being, rj11-1 is ethernet in, and rj11-2 is a masqued network -- forced to 192.168.0.0
04:47.08dcmwaithe things work fine when primary fail...  backup will take over.
04:47.34jblackgah gah gah. rj45! rj45, I mean
04:47.55dcmwaibut when primay come back online... something wong, the backup seem are unable to send things back... anyone have setup things like that before?
04:48.20nhuisman_workwhen you say unable to send
04:48.22nhuisman_workwhat do you mean
04:48.26nhuisman_workheartbeat doesn't fork over the ip?
04:48.38nhuisman_workyou probably need to ask that question in another room if so, more of a linux ha question
04:49.11nhuisman_workLOL
04:49.13dcmwainhuisman_work, it does fork over the ip. However, it was to the backup Mac Address
04:49.35nhuisman_workon voipsupply you click on sipura spa-1000 it says discontinued
04:49.37dcmwainhuisman_work, so I also confused and wonder if that is the right way ..
04:49.42nhuisman_workwhich sends you do the spa 10001
04:49.46[TK]D-Fendernhuisman_work, then SPA-2102 it is.
04:49.54nhuisman_workwhich is then discontinued and sends you do the spa 2xxxx
04:50.17nhuisman_worki guess i'll just not use the router
04:50.42[TK]D-Fendernhuisman_work, exactly.  Its good for re-use later.
04:51.21[TK]D-Fendernhuisman_work, And supports T.38 which may come in handy later (esp having a 2nd port for it)
04:51.46nhuisman_workwhat about the pap2t-na
04:52.36jblackYeah, I suppose you can always get more on them... 8/250, or 6/240
04:52.39[TK]D-Fendernhuisman_work, no T.38, no router, wimpier CPU (not sure about multiple G.729, etc)
04:52.54nhuisman_workhonestly i'll never reuse them
04:52.57nhuisman_workis why i ask
04:53.06[TK]D-Fendernhuisman_work, PAP2 works... your call...
04:53.15nhuisman_workk
04:54.51nhuisman_workfax + asterisk = not worth trying?
04:57.56nhuisman_workwell thanks for the help today.  I'm out for the night.
04:57.59nhuisman_work*gone*
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05:08.59dkatz334Hello all.
05:09.19dkatz334I'd like to know recommendations for RX faxing with asterisk.
05:09.30dkatz334Not intereted in TX, just RX.
05:09.33*** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl)
05:10.02dkatz334Currently using spandsp with a digium TE212P and ecm, it doesn't work with one station, the fax never comes through.
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06:06.06UnixDogthis bites
06:06.31UnixDogsome of the patches wont work right on 1.6
06:06.40UnixDogthis is going to be painfull
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06:24.50UnixDogwow o far 4 patches can not be used
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06:36.48CCFL_Man2cli-msn seems to suck
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06:44.04*** join/#asterisk fozzmoo (n=fozz@166-70-238-250.ip.xmission.com)
06:44.22fozzmooGood evening/morning.
06:45.01fozzmooquick question: I'm trying to set some variables in AMI with the Set: parameter so that a dialplan that gets run by Originate has some data.
06:45.30fozzmooI'm using 1.2.x. Is "Set: variable=value" correct styntax?
06:45.40fozzmooOr do I need to use Action: SetVar?
06:46.35Corydon76-digfozzmoo: who told you it was ever Set in AMI?
06:47.41Corydon76-digfozzmoo: Try running Action: ListCommands
06:47.58fozzmooCorydon76-dig: jsmith. :)
06:48.13fozzmooI need to pass some variables to the dialplan from AMI.
06:48.13Corydon76-digfozzmoo: never has been and still isn't
06:48.31Corydon76-digThat's very risky
06:49.11fozzmoohere's what jsmith told me a couple days ago:
06:49.13fozzmoo(03:07:32 PM) jsmith: Action: Originate
06:49.13fozzmoo(03:07:54 PM) jsmith: Channel: Local/18012545677@longdistance/n
06:49.13fozzmoo(03:07:59 PM) jsmith: Context: somecontext
06:49.14fozzmoo(03:08:07 PM) jsmith: Exten: 1234
06:49.14fozzmoo(03:08:11 PM) jsmith: Priority: 1
06:49.14fozzmoo(03:08:29 PM) jsmith: Set: seller_id=51293851
06:49.28fozzmooAnd that "seller_id" line is the one I'm trying to work with.
06:49.35Corydon76-digAre you sure he was talking about AMI and spool files?
06:49.52fozzmooyeah- he explained how to telnet into port 5038 and all that before this.
06:49.52Nuggettelnet is eeeeeeevil!
06:49.54Corydon76-digerr, and NOT spool files?
06:50.03fozzmootelnet
06:50.18Corydon76-digI'll have to beat him down next time I see him
06:50.33fozzmooSo, I'm up a creek or what?
06:50.44Corydon76-digSetVar is the name of the action
06:50.50fozzmooOkay.
06:51.08fozzmooSo if I do a bunch of those before the Originate action, my dialplan will be aware of the variables?
06:51.35Corydon76-digThere is no way to set arbitrary variables from an Action: originate
06:51.54fozzmooHmm. I'll have to talk to jsmith about that inaccuracy. :)
06:52.04fozzmooFeeding me full of crap.
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06:52.36fozzmooCorydon76-dig: Thank you very much for your assistance.
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06:57.18*** join/#asterisk J4zen (n=Jvan4zen@a82-95-153-17.adsl.xs4all.nl)
07:03.40fozzmooAlright. Another AMI question: How can I use the SetVar: action to set variables for a local psuedo channel? In the realm of possible?
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07:10.17Corydon76-digfozzmoo: get on the Asterisk CLI and type:  show manager command setvar
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07:17.10citatsCorydon76-vcch: i thought you could use Variable: to set variables with originate, but maybe that has changed since i did it last
07:18.06fozzmoocitats: That's what I just read about
07:18.13fozzmooI tried it and it worked for the first Variable:
07:18.21fozzmooNow I've got to figure out what's wrong with the rest of my variables. :)
07:18.49citatsfozzmoo: iirc you could only pass one Variable: arg and there was some kind of seperator
07:19.04fozzmoocitats: that makes sense too.
07:19.08citatslike Variable: var1=this&var2=that
07:19.17citatsor maybe Variable: var1=this|var2=that
07:19.29fozzmooA HA!
07:19.31fozzmooVariable: var1=23|var2=24|var3=25
07:19.33tsabihi
07:19.35fozzmoo(voip-info.org)
07:19.45fozzmoocitats: THANKS!
07:19.46tsabii have a small problem with a grandstream BT101:
07:19.59tsabiasterisk writes this on console when i wish to make a call with it:
07:20.01tsabichan_sip.c:3670 sip_write: Asked to transmit frame type 4, while native formats is 0x1 (g723)(1) read/write = 0x8 (alaw)(8)/0x4 (ulaw)(4)
07:20.17tsabiwhat this means?
07:20.32J4k3tasbi: I'd guess your codec selections aren't the same on both sides
07:20.36J4k3does the call actually complete?
07:20.52tsabion asterisk i allow=all
07:21.04tsabithe call is done, but the phone is silcence
07:21.17J4k3I'd try forcing a setting in asterik
07:21.20Corydon76-digtsabi: one side negotiated g.723, but you don't have a translation path from G.723 to anything else
07:21.51tsabiCorydon76-dig: what thsi means? :)
07:21.57Corydon76-digThe only way to get G.723 working (legally) is to buy the hardware transcoder board
07:22.06J4k3your codec settings aren't the same on both sides.
07:22.14Corydon76-digotherwise, you CANNOT use G.723
07:22.27tsabiok, no G723, what other codec can i use?
07:22.41J4k3g711 ulaw/alaw and g729 (with license, of course)
07:22.42Corydon76-digTry ulaw
07:22.58tsabiulaw, that didnt worked last time
07:23.06tsabii tryed every coedec on the phone :(
07:23.17tsabibut i will try again, maybe the error message will be different
07:23.28J4k3make sure you've updated the firmware on the phone to the latest
07:23.36tsabiyeah i updated
07:23.56tsabii can make SIP calls with this settings, but no ISDN calls
07:24.00tsabivia ZAP channel
07:24.32J4k3try alaw
07:25.18tsabiok, i tryed alaw
07:25.35tsabinow i made a call, and it says no translation path from g723 to alaw
07:25.51tsabiso my ISDN is in g723 codec?
07:26.09tsabibut my linksys phone works well :S
07:26.10J4k3afaik its alaw except in north america and japan (ulaw)
07:26.12tsabii dont understand this
07:26.25J4k3I suspect you've got the codec settings on the phone all confused
07:26.34tsabiand i use 711 alaw on the linksys too
07:26.45J4k3and/or, you need to force codecs in asterisk (disallow=all, allow=[codec])
07:26.53J4k3thats how my bt101s work
07:27.05tsabihmm, i see
07:27.09tsabiok
07:31.25tsabihmm, it semms it works
07:31.56tsabibug in asterisk? :S
07:32.04J4k3bug in grandstream
07:32.17J4k3I'd trust asterisk a lot more than I'd trust a $30 pos phone
07:32.27J4k3don't get me wrong, I use gs101's every day, they do work
07:32.28jblackGrandstreams sure do have a poor reputation here
07:32.32J4k3but they're.... grandstreams
07:32.46tsabihmm, ok
07:33.07jblackI was actually chased down by people after they found out I had bought a gs, and they got me to return it
07:33.33fozzmooheh.
07:33.41hmodesasterisk still needs some serious work re: codec handling, grandstreams aside, imo
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07:34.05jblackwhat piece of free software is done... :)
07:34.33tsabiyeah i tested it in every way now, its works now
07:34.39tsabithny for big help :)
07:35.00hmodesyeah, truedat
07:35.05hmodesi'm not complaining, just saying :)
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07:35.48tsabiin my opinion, if i should force codec usage in asterisk, and asterisk says about codec confusion, it seems that asterisk confused, and asterisk have a bug
07:36.13hmodeseh, it's not a 'bug' per say, it's working as designed
07:36.25hmodesjust, the design could use some work
07:36.47hmodesonce you get used to it, it's not such a big deal except in a few very specific situations
07:37.23hmodesalso seems to be a big hangup for people who don't already understand how the codec negotiation 'works'
07:37.27bintutwhen to use realm on sip.conf ? is there a standard value/format for the realm?
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08:15.53awk~the book
08:15.56awk!the book
08:16.07awksomebody give me a link to that asterisk book
08:16.18J4k3~thebook
08:16.19jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
08:16.22awkthanks
08:16.41mort_gibhttp://www.asteriskdocs.org/
08:17.05awkthanks
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09:06.01yangIn grandstream phones, I have the option enabled when I dial the number it tells me in voice which number is calling me, where can i disable that, I looked all over the options
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09:21.31^shark_hi guys one question -- if i connect my asterisk box to another box that has got a PSTN connection, will my extensions be able to dial through to this PSTN network....? is this possible..?
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09:35.15dominic1Hi, anybody knows what happened to openhardphone.org?
09:37.50hi365it went soft
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09:45.57^shark_if i connect my asterisk box to another box that has got a PSTN connection, will my extensions be able to dial through to this PSTN network....? is this possible..?
09:50.56ronr^shark_: yes, that's possible
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09:57.23^shark_ronr: any guides to make this connection to the PSTN network from my box
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10:04.57*** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek)
10:05.22Zeeekso who's on 1.6 ?
10:11.13DarKnesS_WolFevening geeks
10:11.21Zeeekmorning!
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10:15.12DarKnesS_WolFZeeek: what with 1.6 ?
10:19.31Zeeekanyone using it?
10:19.39ZeeekI'd like to hear about it if so
10:23.38DarKnesS_WolFZeeek: not yet but yes i'll upgrade my asterisks to it
10:26.03Zeeekfrom 1.4?
10:26.16ronr^shark_: just connect the two boxes using sip or iax (see the book) and forward the call from the second box to pstn
10:27.35DarKnesS_WolFZeeek: yep
10:27.48DarKnesS_WolFIAX much better ^shark_
10:27.53DarKnesS_WolF~book
10:27.53jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
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10:46.23DarKnesS_WolFif have a sip phone and i'm dilaing using zap card what i the varibale that holds the sip phone number ?
10:46.26DarKnesS_WolFto use it in file ?
10:47.20*** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl)
10:47.27DarKnesS_WolF${CALLERID(num)}
10:47.29DarKnesS_WolFsweet !
10:47.31ZeeekCLI> show application dial
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10:51.46^shark_ronr: thanks man -- i had gone out for lunch
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11:03.55^shark_!dual servers
11:04.02^shark_sorry guys
11:08.13DarKnesS_WolFwhat is wrong wiht that ? exten => s,n,MixMonitor(${FILENAME}.wav|system(/usr/bin/lame -S -V7 -B24 --tt ${FILENAME} --add-id3v2 /var/spool/asterisk/monitor/${FILENAME}.wav /var/spool/asterisk/monitor/${FILENAME}.mp3))
11:08.32DarKnesS_WolFafter the call done it don't create the mp3 even it is executes the sytem function
11:11.00DarKnesS_WolFand even i did replace | with ||
11:11.08nebojsajsimichi all
11:11.10*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
11:12.12nebojsajsimiclittle help when i call 2N gateway i don't have callerID can it be fixed
11:12.18*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
11:12.33nebojsajsimicin asterisk i get only name of pear .....
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11:17.16DarKnesS_WolFset(callerid(num)=1234567890)
11:19.40nebojsajsimicwhere???
11:19.43nebojsajsimicto set
11:19.46DarKnesS_WolFin ur dialplan
11:19.56DarKnesS_WolFwhat is the problem i mean what if u don't have CallerID
11:19.57DarKnesS_WolFso what :D?
11:20.01DarKnesS_WolFwhat is wrong?
11:20.10nebojsajsimici have ID
11:20.31nebojsajsimiccall come from 2N gateway
11:20.35nebojsajsimicfrom GSM
11:20.56DarKnesS_WolFwhere is asterisk here?
11:21.00nebojsajsimicand i try to make something to put my Agents to queue
11:21.28nebojsajsimicbut y try to use CallerID for identification
11:23.31nebojsajsimicany idea???
11:25.26*** join/#asterisk ArchSSM (n=tommy@host-81-191-139-130.bluecom.no)
11:27.39DarKnesS_WolFnebojsajsimic: copy in pastebin ur errors and ur exntesions.conf part for this
11:27.47ArchSSMIf the channel.c returns "didn't get frame from channel: SIP/.... ".  Where do I start to debug?
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11:31.00DarKnesS_WolFArchSSM: i think codecs
11:31.15DarKnesS_WolFmake sure ur using compatible codecs
11:31.53ArchSSMHmm.. I see.
11:32.31ArchSSMThe thing is: It happens during a conversation. I can have an active conversation for everything from 10 seconds to 30 minutes, and then it just hangs up.
11:32.37ArchSSMWith the error mentioned.
11:33.04*** join/#asterisk BipBip (n=BipBip@194.65.5.235)
11:33.20BipBiphello all
11:33.31ArchSSMHello :)
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11:35.09BipBipii'm having trouble with one config wich i think it should be simple but i'm going nuts. anyone with some energy to waste in me?
11:35.36ArchSSMMaybe ;). Just ask and we'll help if we know.
11:37.08BipBipnice :) I'm trying to configure asterisk to register a sip account, the issue is that i need to use a proxy and it's not working as i thought it should
11:37.39BipBipregister => user:pass@proxy.voip.sapo.pt:5070
11:38.00BipBipthen [proxy.voip.sapo.pt] fromdomain=sapo.pt
11:38.15BipBipthe user should be user@sapo.pt
11:38.17mostyBipBip, use pastebin.com
11:38.27BipBipmosty: k
11:38.39mostyjust x out the password and username
11:39.56BipBiphttp://pastebin.com/mfc36ba0
11:40.40FlatFootmorning all
11:40.47BipBiphello FlatFoot
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11:41.12ArchSSMGood morning
11:41.25FlatFootgot a bit of trouble with res_odbc.so and res_config_odbc.so they are not present on my box
11:41.28mostyBipBip, what happens when you try to register?
11:41.37FlatFootusing FreeTDS do i need them installed ?
11:41.57mostyFlatFoot, i think you need unixodbc dev libraries for odbc, not freetds
11:42.11BipBipwith this config, he trie to use port 5060
11:42.20BipBiptries
11:42.31FlatFooti had this working on another box but this install does not seem as happy
11:44.30BipBipthis is the config that i need: http://pastebin.com/m4fdc766f
11:44.47FlatFootthe old box was running 1.4.11 this one is on 1.4.17 can't find too much info on this . Running FreeBSD can anyone suggest an answer
11:45.38BipBip... using 1.4.17 compiled
11:45.46FlatFootfrom ports
11:45.56mostyFlatFoot, so do you have the package i mentioned?
11:46.10ArchSSMBipBip: register => UUU:PPP@proxy.voip.sapo.pt:5070/sapo <--- should be correct
11:46.27ArchSSMBipBip: You had the username mentioned twice. And what does the cli say?
11:46.45FlatFootmosty: i have not installed it myself cos all the info about FreeTDS said it has some of the same files as unixODBC so i did not need it
11:46.56BipBipArchSSM: http://pastebin.com/m3e96da37
11:47.17mostyBipBip, tried setting outboundproxyport? and does it register ok?
11:47.23BipBiphumm, he is trying sapo.pt:5070 wich is wrong
11:49.21BipBipok, i changed host to proxy.voip.sapo.pt and now he tries to register, but with domain proxy.voip.sapo.pt and not with sapo.pt
11:49.27FlatFootmosty: i am installing it now
11:50.27*** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153)
11:51.16AlienPenguinhi, i noticed that in the 200 that asterisk responds to my client i do not have the INFO in the Allow header. Where can i change this behaviour?
11:54.57BipBipis there a way to force the realm? from what i'm seeing in the logs, the realm used is voip.sapo.pt and not sapo.pt. When i force the user to have the domain, the realm is changed to sapo.pt but i receive bad password
11:56.03*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
11:56.10mostyBipBip, there is an example at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
11:56.35BipBipmosty: thanks, i'll have a look
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12:01.26neuwaldHi folks. Does bindport=5060, 1020   will work in sip.conf ?
12:01.44neuwaldto sip listen in two or more udp ports...
12:02.23*** join/#asterisk vrtk (n=bb@189.21.178.20)
12:03.00mostyneuwald, i don't think so, do you get an error message when you try?
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12:03.27neuwaldI didn't try, first asking here, cause it's a production server
12:03.32ArchSSMneuwald: Why not use iptables for this?
12:03.57mostyneuwald, if you have a production server, you should also have a test server
12:03.58neuwaldI'm using pf on firewall, I think it's a good idea too
12:04.06neuwaldmosty ok, I'll test, just a minute
12:05.25neuwaldbindport=5060,1020
12:05.34neuwalddidn't worked. only 5060 is listen
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12:12.24AlienPenguini have seen some posts in asteriskguru that other ppl share my problem: asterisk responds 403 to any SIP INFO  message an UA tries to send to the remote party. Is there a way to change this behaviour?
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12:17.12troubledmornin's
12:23.14nebojsajsimici try to fix something in sip header i have callerID
12:23.22nebojsajsimichov can i catch it
12:23.25nebojsajsimic????
12:23.30nebojsajsimic*how
12:23.37*** join/#asterisk whymarkwhy (n=koos@nas-mid.nashuamp.co.za)
12:23.55whymarkwhyhi there
12:23.59ArchSSMHello
12:24.09nebojsajsimicHi
12:24.36whymarkwhyis there anyway you can edit you extension.conf file from the dial plan using one of aterisk's applications
12:25.48nebojsajsimici think YES you hav on Asterisk GUI file editor :)
12:25.55nebojsajsimicyou hawe
12:25.57nebojsajsimic***
12:27.43*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
12:27.45nebojsajsimicHow to Get parameter from sip header????
12:27.48nebojsajsimiceny help
12:27.53nebojsajsimicany help ......
12:28.01nebojsajsimicpoooooor my eng
12:28.28ArchSSMI'm afraid I don't know.
12:29.01ArchSSMAnd just a tips - The more question mark you have, the less of a chance you have of getting help ;)
12:29.53*** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk)
12:30.29^shark_hello
12:30.52^shark_what is the logic behind connecting dual servers, i have tried to check this out but have failed
12:31.21^shark_ok i know u have to connect dial peers... but  i dont know the full details and howto
12:31.28^shark_i am using SIP
12:31.47^shark_Oooops i meant dial plans
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12:33.57Zeeekso installed 1.66beta
12:34.12Zeeeks/1.66/1.6
12:34.19*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
12:34.51tzafrirThe computer has not gone up in smoke yet?
12:37.02Zeeekno and zaptel is working
12:37.19Zeeekbut I haven't seen anything register (network changes so I need to chg extensions and sip)
12:37.30Zeeektzafrir you tried it yet?
12:37.42tzafrirNot really
12:38.11ZeeekNot really? Is that a no?
12:38.29Zeeekso what can I do with 1.6 that I couldn't with 1.2?
12:38.42Zeeekprobably nothing with my antiquated equipment :)
12:38.59ArchSSMI'm actually a bit disappointed. It's not much new when you look at the change log
12:39.25tzafrirZeeek, you mean: compared to 1.4, right?
12:39.34ZeeekNO 1.2!
12:39.48ZeeekArchSSM I'm thinking a lot of the stuff is total re-write, though ?
12:40.04ZeeekIOW, maybe not a lot of new features, but lots of better code
12:40.20ArchSSMWell. both sip handler and the channel handler has increased performance and readability
12:41.28*** join/#asterisk the_5th_wheel (n=edd@dsl-242-89-144.telkomadsl.co.za)
12:41.57ZeeekSounds right out of the Digium flyer :)
12:42.15ArchSSMhehehe
12:42.18ArchSSMexactly :)
12:42.26ZeeekFunny how "legacy" used to me inheritance (good) and now it means crappy code (bad)
12:42.39ArchSSMtrue,true :)
12:42.42Zeeekas in "we've removed the legacy code"
12:42.57Zeeeklike "legacy application" = MS-DOS
12:43.10the_5th_wheelhi. are there any CLI checking avaliable for linux? My phone doesnt support CLI, but i the my predecessor got CLI working on his windows pc, but i dont use that.
12:43.25the_5th_wheel(and my predecessor doesnt return my calls)
12:43.31Zeeekheh
12:43.37Zeeekthey rarely do
12:43.44the_5th_wheel(caller line identification)
12:43.54Zeeekcallerid
12:44.05the_5th_wheelyeah
12:44.07Zeeekas against Command Line Interpreter
12:44.34Zeeekthe_5th_wheel I don't understand the question, though
12:44.44whymarkwhyif you have your sip client(xlite) connected to more than one asterisk server how do you dial out on say a zap one one server and sip on the other?
12:45.31the_5th_wheelon the windows pc next to me, they have a bubble popping up every when his phone rings, saying who is phoning
12:45.38the_5th_wheelis there such an application for linux
12:46.13ZeeekI hear Jabber can do that for you. On WIndows I use ...
12:46.28ArchSSMyep
12:46.29Zeeek"YAC"
12:46.31ArchSSMfreeswitch :)
12:46.39ZeeekYet Another Callerid
12:46.52Zeeekit's actually done for TiVo
12:47.16the_5th_wheelthat is what my neighbour has running on his pc
12:47.20the_5th_wheelbut i dont use windows
12:47.32Zeeeknice, eh? Linux, dunno. Try FOP
12:47.34DarKnesS_WolFwhymarkwhy: i think u do ad #1 or something like that not sure i don't use SIP since ages.
12:47.48DarKnesS_WolFthe_5th_wheel: which application is that ?
12:47.54the_5th_wheelYac
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12:48.38Zeeek...
12:48.43Zeeekmercilessly kicked
12:49.05^shark_hi
12:49.14^shark_in this scenario >>  exten => _8XXX,2,Dial(SIP/myserver:passwordB@SIPserverB/${EXTEN:1},30,r)
12:49.14Zeeekyes hit #1 #2 etc for multiple servers
12:49.46tzafrirZeeek, well one new feature is the ability to write dialplan logic in Lua, in case you're interested
12:49.54^shark_myserver: is an ip address and the password: is the userpassword!?
12:50.26tzafrirchan_zap has many small improvements
12:51.08ZeeekLua? Never hoid of it
12:51.31Zeeekso far I just realized I did a make config and blew away the old dialplan :)
12:52.14Zeeekbut it is running, with one FXS so I just need to find something to call
12:52.26*** join/#asterisk _ys (i=yuri@91.151.196.254)
12:52.32tzafrire.g: try:  zap set <tab><tab> with a recent Zaptel
12:53.53tzafrirYou can always call an echo test
12:54.06tzafrirOriginate Zap/1 application Echo
12:54.23tzafrirdid I mention originate is new as of 1.4? highly useful
12:54.36Zeeekhow do I reload extensions now?
12:54.59tzafrirdialplan reload
12:55.02Zeeekzap set - cool!
12:55.07Zeeekok, thx
12:55.40tzafrir(yeah I know: much worse that 'extensions reload' for tab completion)
12:56.09ZeeekIT WORKS!!!! asterisk 1.6beta1 WORKS!!!!
12:56.17Zeeekabsolutely amazing
12:56.18ArchSSMehm... shouldn't it?
12:56.26ArchSSMit's a beta... not pre-alpha0
12:56.27ArchSSM:)
12:56.32Zeeekthey don't always :)
12:56.45ArchSSMtrue
12:56.57Zeeeknote : http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/README?revision=99123&view=markup
12:57.05Zeeekrefers to 1.2 in the text
12:57.33Zeeek"To discover the major new features of Asterisk 1.2, please visit ..."
12:57.50ArchSSMhehehe
12:57.51Zeeekthat's the thing about the INternet
12:57.54ArchSSMnice
12:58.01Zeeekit's like an exploding star
12:58.20ZeeekYou can NEVER find and correct all those little oversights
12:58.50Zeeekanyway, so far I'm impressed.
12:59.03Zeeektzafrir what else can I try for fun, eh?
12:59.31Zeeektzafrir ext<tab>rel<tab> was a lot better
12:59.51Zeeekmaybe terminal function keys are the answer to that
12:59.53tzafrire<tab><tab>, actually, IIRC
13:00.22Zeeekyou're right, just tried it on 1.2
13:00.39Zeeekand all these years I've been tyoing 12 more keys than nec.
13:01.16Zeeekso, no more "core"? good, I hated that
13:02.00tzafrirwhy no more core? it's there
13:02.53Zeeeksip show peers works
13:03.04Zeeekwasn't it core sip.... in 1.4?
13:03.10tzafrircore show globals
13:03.18tzafrirno, it was sip
13:03.49hmmhesaystzafrir: your guys must not like emailing me back
13:05.04Zeeekok, get ready, I'm going to try to call from 1.2 to 1.6
13:05.07tzafrirZeeek, core set chanvar is also nice
13:08.15*** join/#asterisk lirakis (n=lirakis@65.200.191.241)
13:10.40DarKnesS_WolFtzafrir: what is chanvar?
13:10.46DarKnesS_WolFtzafrir: any updtes about the suse init ?
13:10.54*** join/#asterisk anonymouz666 (n=anonymou@201.19.235.168)
13:10.59tzafrirSet the value of a normal channel variable
13:11.06tzafrir(as opposed to a global variable)
13:13.40DarKnesS_WolFah i c
13:13.41DarKnesS_WolFnice nice
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13:31.24nebojsajsimicme again
13:31.44nebojsajsimichow to substring something
13:32.00nebojsajsimici fing way to logon into queue
13:32.20nebojsajsimicwhen i call ${CALLERIDNUM}
13:32.37nebojsajsimici get sip/123456-897946546
13:32.45nebojsajsimici need only 123456
13:32.54nebojsajsimichow can i do it ????
13:33.07ArchSSMregexp it :)
13:33.27nebojsajsimicplease help little
13:33.29nebojsajsimic:)
13:33.57penguinFunkregex is easy
13:34.06penguinFunkget your dog/cat to walk on the keyboard
13:34.07Kigh1,Set(FOO=${CUT(CALLERIDNUM,/,2)}
13:34.13ArchSSMhehehe
13:34.14Kigh2,Set(FOO=${CUT(FOO,-,1)}
13:34.26KighFOO contains 123456
13:34.28*** join/#asterisk af_ (n=getsmart@88-149-230-244.dynamic.ngi.it)
13:34.49Kighthis is no job for a regexp imho .. simple cutting
13:34.49nebojsajsimicthx Kigh
13:34.51Kighnp
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13:34.56ArchSSMKigh: Is that a 'better' way than exporting it and doing a real regexp of it?
13:34.58ArchSSMah... ok :)
13:35.17KighArchSSM: it is basic asterisk extension language, yes its 'better'
13:36.37ArchSSMOk.
13:40.38[TK]D-FenderAnd that isn't a callerid... thats a CHANNEL NAME
13:40.47^shark_i need some basic help -- if my asterisk box is not on the domain but only has a static local IP on the LAN and my mail server is also on the LAN but in the domain, how do i tweak postfix to send my voicemails to my inbox?
13:40.54^shark_do i just install postfix
13:41.19[TK]D-Fender^shark_: this is a question for #postfix , not here
13:41.39*** join/#asterisk qdk (n=qdk@85.235.253.139)
13:42.09^shark_[TK]D-Fender:  ok -- i was just wondering how you guys set this up
13:42.31[TK]D-Fender^shark_: most of us just let sendmail do its thing
13:43.09nebojsajsimicis there ${chanelname}
13:43.10nebojsajsimic???
13:43.10^shark_[TK]D-Fender:  so its juts a matter of installing send mail is that right>?
13:43.50nebojsajsimicand is there way to get Callerid from sip header
13:43.55nebojsajsimic???
13:44.14[TK]D-Fender^shark_: funny how you think that everything magically "jsut works" and will also "just work the way *I* want it to".  Do all of your corporate functions happen without any configuration as well?
13:44.48[TK]D-Fendernebojsajsimic: "show functions" <- go look at the list and see what stands out.
13:44.51nebojsajsimici receive call from GSM gateway and when i need CALLERID from real caller who call gsm gateway
13:45.04nebojsajsimicthx
13:45.23^shark_[TK]D-Fender: let me try and do something, will be back laters
13:45.27[TK]D-Fendernebojsajsimic: "sip/123456-897946546" is a channel name.  If that somehow ended up in the callerid its because someone overrode it manually.
13:46.15ManxPowerThe answers you seek are *within*, grasshopper.
13:46.39[TK]D-FenderManxPower: and coated with grease so the hard you grasp, the more slips through your fingers!
13:46.40Zeeekseek *this*
13:47.06ManxPowernebojsajsimic: ${CALLERID} contains the callerid.
13:47.12Zeeekso I tried 1.6
13:47.19ManxPowerIf it is not in ${CALLERID} then is is not Caller*ID.
13:47.28ManxPowerZeeek: What did you think of it?
13:48.05Zeeekso far, no make problems, but I need to make a dialplan as I deleted the old (unused 1.4) one
13:48.22ManxPowerZeeek: I have high hopes for 1.6.
13:48.23ZeeekI made one SIP call thru USA, worked fine.
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13:48.58Zeeekwhat I really need to do is delete the entire extensions.conf and start from scratch with the years of collected wisdom
13:49.18Zeeekbut playtime is limited
13:49.44Zeeekand now I have no fixed ip on that connex so it makes things a little harder
13:50.39ManxPowerOnce 1.6 hits RC status I might just take the pain and start testing it in semi-production enviroments.
13:50.56*** join/#asterisk hijacked (n=argh@66.255.220.17)
13:51.18[TK]D-FenderManxPower: I'll probably wait about 2-3 weeks after 1.6.0 and see if the body-count slows.
13:51.21ManxPowerAnd trust me, testing PBX upgrades in my environment means pain, usually caused by headaches caused by whiney users.
13:51.41Zeeekheh, well I don't have *that* problem
13:51.51ManxPower[TK]D-Fender: I'm making the effort to help with 1.6 if I can.
13:51.51Zeeekmy wife is used to it
13:52.01*** join/#asterisk nDuff (n=ccd@user-387ocuv.cable.mindspring.com)
13:52.52nDuffIs there a way to tell if I'm providing ringback on an outgoing call or if it's being provided by the remote end? I have "pri debug" logs, but don't know what to look for.
13:53.20ManxPowernDuff: if the call is not answered, you are not providing ringback
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13:54.09nebojsajsimicis there some good book with implemented functions and examples .....
13:54.22nDuffs/outgoing call/inbound call being forwarded to an outside line/
13:54.23ManxPowernow you might not be sending the Q.931 messages to make the remote telco provide ringback -- but that was not your answer.
13:54.54nDuffhmm.
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13:55.46ManxPowernDuff: is the forwarded call answered before it is forwarded?
13:56.31nDuffyes.
13:56.33mkl1525Hi, (* 1.2 and 1.4) haven't found it in voip wiki but is it possible to use extconfig.conf for agents.conf?
13:56.53[TK]D-Fendernebojsajsimic: "show function [FUNCTIONNAME]" and ...
13:56.55[TK]D-Fender~book
13:56.55jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
13:56.58[TK]D-Fender^^^^^^^^^^
13:57.07[TK]D-Fendernebojsajsimic: then there is the
13:57.09[TK]D-Fender~wikis
13:57.10jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
13:57.27ManxPowerthen Asterisk would have to provide INBAND (controlled by /etc/asterisk/indications.conf) ringback and you would not see that in pro debug messages
13:58.17nDuffahh; thank you.
13:58.50ManxPower"Don't know how to handle indication 15" or some sort of message like that is common if you don't have an indications.conf and you need it.
13:59.38nDuff(the remote end says they don't see my outgoing calls at all, though I'm receiving CALL PROCEEDING, ALERTING and DISCONNECT on them; they were wondering who was providing ringback on forwarded test calls -- but I can clarify that by changing indications.conf to make mine sound different).
14:00.16ManxPower"see"?
14:01.03nDuff"Not sure what to tell you.  I'm not seeing the inbound call attempt at all, but the system doesn't have a low level protocol debug option."
14:01.16ManxPowernDuff: Aha!
14:02.02ManxPowernDuff: We had a similar issue with outgoing leg of an automatically forwarded call failing.  Turned out we added a 9-1- to the CLID when the call came in and our telco got VERY upset when it saw calls being dialed with that callerid.
14:02.23ManxPowerfixed the callerid to be CORRECT (nothing but numbers, 10 digits only) and it started working just fine.
14:02.29*** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr)
14:03.04nDuffManxPower: hmm; I'll try that.
14:03.37ManxPowerand for good measure, overide the callerid on the outgoing leg to be a number on your PRI.
14:03.48*** join/#asterisk destructure (n=de@66.193.229.254)
14:04.07ManxPowerI guess I should get to work soon.
14:04.14ManxPowerI have to deal with end users today.  *sob*
14:04.31ManxPower(all non-VoIP stuff)
14:04.32*** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek)
14:05.24ManxPowerWe are VLANing one of the offices, did most of the work last night, but a few people locked their offices and I could not get in to renumber their printers, etc.
14:05.29Zeeekah.
14:05.35Zeeekof course.
14:07.17ManxPowerThere are other issues too like the electricition never putting the power outlets required for the security cameras (wondered  why they were not working...)
14:08.32ManxPowerI just dropped my cig lighter in my coffee.  8-(
14:08.54d3waynethat's not a good start to the day :-\
14:09.08ManxPowerd3wayne: no, it is not.
14:10.14Unihow goes brudda man?
14:10.17ManxPowerBTW, everyone, I am now accepting a limited number of clients.  VoIP, WAN, etc.  Preference is for clients in the SE USA, but not a requirement.  I do consulting.
14:10.24nDuffIt goes, it goes.
14:11.00ManxPowerBut I do require that a potential client is not a cheap bastard.
14:11.09hmmhesaysManxPower: don't we all?
14:11.15ManxPowerhmmhesays: apparently not.
14:11.38hmmhesaysI specialize in being macgyver
14:12.06ManxPower"I've got 7 SPA-2100s and my boss wants to convert the PBX to VoIP.  Oh, we have no budget!"
14:12.16shido6lol
14:12.50ManxPowerCall me selfish, but if you want me to get out of bed and help you, you had better compensate me for the pain of being awake.
14:13.27Zeeekthanks for that image
14:14.18ManxPowerI *LIKE* a challenging project.  I do not like futile projects. 8-)
14:14.54*** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek)
14:15.24ManxPowerWell, challenging technical projects at least.  I'm not fond of trying not to call someone a moron when they are connected to both the corporate network and the coffee shop's WiFi network at the same time and are having problems.
14:15.43*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
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14:16.38ManxPowerI compromised with saying something like "Don't tell me this, if HR finds out, you could be fired."  That shut them up pretty fast.
14:17.00*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:17.41nebojsajsimicok i get AgentCallbackLogin("SIP/81733-08215090", "81733#1234#")
14:18.01nebojsajsimicwhat i do wrong
14:18.40[TK]D-Fendernebojsajsimic: go read the instructions again.
14:18.54*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:19.42d3wayneManxPower: where in the SE ?
14:28.06*** join/#asterisk ncampion (n=ncampion@nat/ibm/x-19a4bc0c59812909)
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14:29.26*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
14:29.44flujanhi guys... :)
14:30.13*** join/#asterisk BaseJam (n=BaseJam@gw2.imperia.net)
14:30.18flujansimple question... Is the busy-level feature available in version 1.4.15 and above?
14:30.37BaseJamhave anybody a german howto for me ?!
14:30.54ManxPowerd3wayne: I'm based near Birmingham AL, but I have several clients in the New Orleans, LA area.
14:32.02ManxPowerBaseJam: Not really, but I think you have to start out by not having a sense of humor and speaking like you have a cold all the time.
14:36.01*** join/#asterisk sob0l (n=sobol@devel4.net)
14:36.37De_MonBaseJam a what?
14:36.37[TK]D-Fenderflujan: huh?
14:36.52flujanhi [TK]D-Fender . How are you doing?
14:37.02[TK]D-Fenderflujan: Muh.... need vacation.
14:37.16flujanI googled about a cal-limit issue that I am having...
14:37.51BaseJamDe_Mon, i am searching a german tutorial to configure an asterisk server, because i am a beginner in asterisk ;-) so i can learn faster and more abour asterisk
14:37.52flujan[TK]D-Fender: Wow, I recently passed through a surgery... Man I love to work.. It is too bad staying on a hospital.. :(
14:38.09sob0lI have a problem with CDR'a billsec > duration and duration=0
14:38.22*** join/#asterisk nejme_eddinne (n=nejm@41.225.251.170)
14:38.37nejme_eddinnehi
14:38.39*** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com)
14:38.42flujan[TK]D-Fender: http://svn.digium.com/view/asterisk/trunk/CHANGES?revision=62672&view=markup
14:38.44[TK]D-Fendersob0l: Perhaps you are billing for "ringing time" on a call that was never answered...
14:39.03sob0l[TK]D-Fender: the call was answered
14:39.16[TK]D-Fendersob0l: with a duration of 0?  odd.
14:39.19flujan[TK]D-Fender: There is the new call-limit/busy-level stuff that I need...
14:39.32sob0l[TK]D-Fender: and CLID=src=dst
14:39.38De_MonBaseJam oh a asterisk setup tutorial in german.. Nope none that I know of
14:39.39sob0lthat is stranger
14:39.43flujanguys who answers a queue need to have a call-limit of 1 to optimize the queue delivery....
14:39.47nejme_eddinneI have some questions can any one help me please ?
14:40.04flujanbut when the guys need to answer a call and placed another... it does not work ...
14:40.10De_Monflujan do you have a mantis ticket number for that feature?
14:40.27[TK]D-Fenderflujan: if you are using a direct channel interface you should be able to put "call-limit=1" in their sip.conf entries for example.
14:40.41[TK]D-Fendernejme_eddinne: ...
14:40.42flujan[TK]D-Fender: I am already doing it...
14:40.44[TK]D-Fender~ask
14:40.45jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
14:40.53[TK]D-Fender^^^^
14:41.09BaseJamDe_Mon, damn, do you have a good beginner setup tutorial in english ?
14:41.19[TK]D-FenderBaseJam: here :
14:41.21De_Mon~book
14:41.21jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
14:41.25[TK]D-Fender~jerjerguide
14:41.25jbot[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * :  http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/
14:41.25*** join/#asterisk nejme_eddinne (n=nejm@41.225.251.170)
14:41.28flujan[TK]D-Fender: but the guys who have the call-limit=1 needs to answer a call, and place another call in most of the cases... this is issue I am having...
14:41.44BaseJamthanks at all ;-)
14:41.54[TK]D-Fenderflujan: I think there was an "incominglimit" somewhere as well.
14:42.12flujanI need to allow the guys to receive just one call and place another... so call-limit=2 that is it...
14:42.13nejme_eddinnecheck ur trunk configuration...
14:42.16nejme_eddinnecall-limit
14:43.20flujan[TK]D-Fender: incominglimit and call-limit are equivalent. outgoinglimit is not supported anymore.
14:43.22[TK]D-Fendernejme_eddinne: So, just ask your question, don't ask to ask.
14:43.38BaseJamDe_Mon, and that is easy to use ? i hate to be a beginner in something
14:43.40BaseJam;-)
14:43.46[TK]D-Fenderflujan: Ah, well you could always run through Local and check if their on a call first
14:44.04flujan[TK]D-Fender: do you mean using the dialplan?
14:44.05[TK]D-FenderBaseJam: Go read the book.  Its the best thing out there.
14:44.25[TK]D-Fenderflujan: yup
14:44.30BaseJam[TK]D-Fender, sure, i'm downloading atm
14:45.15[TK]D-FenderBaseJam: the quick guide I linked you is a good small sample for some learning concepts onces you've gone through chapter 5 of the book.
14:46.06BaseJamthank you
14:48.15*** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net)
14:48.22whymarkwhyhow do you check witch linux version are you running suse or fedora?
14:48.54*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
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14:48.54whymarkwhyi know its not a linux question
14:49.14tzafrircat /etc/redhat-version
14:49.21whymarkwhythx
14:49.39De_Monflujan I'm not sure if http://bugs.digium.com/view.php?id=11180 is the right ticket for your question or not, if it is no it's not in 1.4
14:49.55tzafrirsorry: /etc/redhat-release . There should also be a similar for for SuSE
14:51.39whymarkwhycan't belive it the most stable system of them all is a fedora 4 box, never gave my a days problems
14:52.05*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
14:52.15whymarkwhycan one install yast on fedora?
14:52.17Dr-Linuxanybody tried vicidial?
14:52.31nejme_eddinnethank you, so I want to know if I can execute AGI script and making a DIAL() at the same time...
14:52.42tzafrirwhymarkwhy, why do you think it would still be stable after you install yast on it?
14:52.50tzafriryast is very intrusive
14:52.59tzafrirSo it surely changes things
14:53.05whymarkwhyso better not then
14:53.09whymarkwhythx
14:53.20*** join/#asterisk monkeytype (n=Bill@216.207.245.1)
14:53.47nejme_eddinneDr-Linux ca i Help you ?
14:53.58nejme_eddinnecan*
14:54.06Dr-Linuxsure
14:54.12Dr-Linuxnejme_eddinne: can i /msg you?
14:54.26nejme_eddinneyes of corse
14:54.27*** join/#asterisk lilalinux (i=e-trolle@fellatio.deswahnsinns.de)
14:54.30nejme_eddinnecourse*
14:54.30*** join/#asterisk AndyGraybeal (n=mind@casanueva.wifi.frognet.net)
14:55.12De_MonWoooo... backport of audio hooks to 1.4?
14:55.15lilalinuxis it possible to push a text instead of a phone number to a sip phone (e.g. SL75 WLAN) instead of the callerid?
14:55.37De_Monlilalinux I believe so
14:55.44De_Monlilalinux try it and let us know :)
14:55.53[TK]D-Fenderlilalinux: Sure
14:56.53nejme_eddinneto explain more, for exemple if I want to use the DIALSTATUS variable I have to wait until the call is hunged up to user it...
14:57.12nejme_eddinneI want to do it in real time...
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15:02.26nejme_eddinneno one have an idea ? :(
15:05.18lilalinuxthx
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15:08.26[TK]D-Fendernejme_eddinne: :"M" is the only important option.
15:08.56nejme_eddinneMacro ?
15:08.56*** join/#asterisk zuchmir (n=zuchmir@d58-110-29-90.meb8.vic.optusnet.com.au)
15:09.12nejme_eddinneI can Dial and execute a Macro at the same time ?
15:09.31zuchmircan anyone help me with this message: Registration for '61300xxxx@sip2.bbpglobal.com' timed out, trying again (Attempt #177)
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15:10.06skopiiI am trying to use txfax and get TIFF/F format is not compatible. I used the ghostscript command from voip-info (http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax) any advice?
15:10.09Dr-Linuxzuchmir: see if the host sip2.bbpglobal.com is up
15:10.29skopiinejme_eddinne: your macro can dial or goto if you want
15:10.30zuchmirit is, i think it's a nat issue
15:10.45Dr-Linuxzuchmir: try ip address instead
15:11.00nejme_eddinnethanks
15:11.49[TK]D-Fenderzuchmir: read up :
15:11.51[TK]D-Fender~sipnat
15:11.51jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
15:11.58[TK]D-Fender^^^^^^^^^^^^
15:12.05eric_hillskopii: I think the txfax uses a CCITT/G3 fax, not G4 for some reason.  You'll need to transcode before sending.
15:12.37eric_hillskopii: (And that's 1+year old information from my brain so it may have changed recently)
15:13.58skopiieric_hill: I will google for that...thanks
15:14.37*** join/#asterisk Teeli (n=tili@cm48.gamma244.maxonline.com.sg)
15:15.08*** join/#asterisk ThoMe (n=tm@keks.be)
15:15.10ThoMehi.
15:15.13ThoMekann hier auch wer deutsch?
15:15.33*** join/#asterisk CVirus (n=GoD@82.201.178.194)
15:15.51lilalinuxThoMe: koennen ja, duerfen nein ;-)
15:16.16ThoMelilalinux: hm. :-/ kurz #asterisk.de ?
15:18.43zuchmirdoes this make sense: http://pastebin.com/d27688dea
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15:20.03zuchmirmy * is behind NAT, but sip2.bbpglobal.com is presumably not, do i specify NAT=yes? in the [bbpglobal] section?
15:21.23skopiihey eric_hill the command on the tx/rx page says to use -sDEVICE=tiffg3 for ghostscript.....did I miss something?
15:21.35*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
15:22.44*** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com)
15:25.21km-how do I make the bot talk to me
15:25.23km-!tfot
15:25.32km-wanna find the free copy of the tfot :)
15:25.34km-~tfot
15:25.35jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
15:25.36eric_hillskopii: I don't think you missed anything.  The inbound format needed to be G3 (tiffg3), but you should be able to use imagemagick in a script to convert an inbound G4 to G3.
15:25.37km-there we go
15:26.01hmodeshey hey, it's a pete
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15:30.38skopiieric_hill: I am just trying to send a fax, I am converting a pdf to a tiff using gs, and then I made a callfile to call my fax machine and set the data to sendfax.agi (which just starts the txfax app with options $faxfile|caller
15:31.16*** join/#asterisk atop (n=a@oaktyres.force9.co.uk)
15:31.56skopiiI am wondering...does anyone actually use txfax?
15:33.06*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
15:33.11lmadsenno one uses fax :)
15:33.24eric_hillWe're using LightningFAX since it supports a hardware board, but we do 300+ faxes per day...
15:33.53eric_hillAnyone know if I can "clone" a voicemail after it's left into a second mailbox?
15:34.18atopThe telephone number shown for inbound calls is missing the first zero, so a call from 0151 2223333 shows as being from 151222333, this is using the latest version of asterisk 1.4 and anciliary files, calls are being presented over ISDN line connected to Sangoma card with latest version on wanrouter.  I've searched the log files, config files and web and I'm finding nothing.
15:34.24skopiieric_hill: you can just symlink two mailboxes together
15:34.26atopAnyone recognise the problem?
15:34.51eric_hillI need about 12 separate mailboxes that copy all of their voicemail into a central "logging" mailbox...
15:35.37eric_hillatop: Turn on debugging on your PRI and verify that the provider is indeed passing all of the digits to you.
15:35.50zuchmirtk-dfender: does my scenario fit into: "Asterisk as a SIP client behind nat, connecting to outside SIP Proxies "?
15:36.13skopiihmmm not sure about that one....sorry eric_hill I would say "make a script to rsync every 5 min or something" but that is surely not a good solution lol
15:36.30atopI did that, and I've checked with the provider who /claims/ they are sending it, but the asterisk debug does not show it.  Given that, should I assume the provider is wrong and go back to them?
15:38.13eric_hillatop: I would say that the provider may be wrong, but probably doesn't have the technical expertise to add that extra digit.
15:38.50eric_hillatop: Can you just work around it?  exten => 151XXXXXXX,1,Goto(realcontext,0${EXTEN},1)
15:38.58atopthought so.  I was getting frustrated looking for something at this side.  Is there a way to make asterisk....
15:39.02atopah, yeah something like that!
15:39.36atopnot sure how that would work tho, as the inbound number could be anything
15:42.11skopiicouldn't you just use a dialing rule like _X.?
15:43.06x86_ShrikE: you around?
15:43.22*** join/#asterisk zenobic (n=zenobic@174-204-116-85.dsl.manitu.net)
15:43.31x86anyone ever mess with Adit 600 channel banks?
15:43.54x86I seem to have a major alarm light on the TDM controller
15:43.54atopI'll get on the line to the carrier again and see what I can find out; thanks all.
15:44.04zenobichello:i have a voicemail problem.
15:45.30[TK]D-Fenderzuchmir>my * is behind NAT, but sip2.bbpglobal.com is presumably not, do i specify NAT=yes? in the [bbpglobal] section? <- for your peer/user, nat=no.
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15:47.39zenobicCan anyone explain me a the difference between Local Channel and SIP (Voicemail) .
15:48.29FlatFootanyone got a copy of asterisk-1.2.12.1-codec-negotiation-20060926.diff.gz  ?
15:48.50zenobichttp://pastebin.com/d4f237837
15:50.10x86ah ok, cleared the TDM controller alarm... only using one T1 interface on the Adit 600, but both were enabled...
15:50.32x86so now I've got all green lights, but for some reason I'm still not getting dial tone from asterisk
15:51.36[TK]D-Fenderzenobic: You clearly have a Dial(Local//..... line in your dialplan, go LOOK FOR IT.
15:52.13[TK]D-Fenderzenobic: And "Local" and "SIP" in this case have absolutely nothing to do with each other.
15:52.47[TK]D-Fenderzenobic: You just compared the originating channel of 1 call to a channel DIALED because of your dialplan.  These are apples & oranges.
15:53.15jpsharpApples & sausage
15:53.29[TK]D-Fenderjpsharp: Sure, why not...
15:54.12x86[TK]D-Fender: ever mess with an adit 600?
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15:54.19[TK]D-Fenderx86: nope
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15:54.38[TK]D-Fenderx86: Stop wasteing your time on channel-banks you schmuck! :p
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15:57.50skopiianyone using txfax?
15:59.21supjigatorwe tried but hylafax seems to just work.
15:59.51skopiisupjigator: can hylafax send faxes out?
15:59.53zuchmirtkd-fender: still getting Registration ... timeout
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15:59.58skopiiI just need to send faxes
16:00.03[TK]D-Fenderzuchmir: pastebin your sip.conf
16:00.24skopiisupjigator: nm me checks docs thanks =]]
16:01.24supjigatoryea hylafax does that well.
16:02.15tzangerAdit600 rocks
16:02.39tzangerI need to experiment with iaxmodem and hylafax
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16:02.52zenobic[TK]D-Fender: thx. i checked the dialplan: exten => 3847413,10,Dial(Local/413@default,240,tw) ...and later this: exten => 413,1,Voicemail(su413@default)
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16:03.53supjigatortzanger: hylafax can do t38 sip
16:04.34[TK]D-Fenderzenobic: well.. you put it there.  Not much more to say.
16:04.57zuchmirhttp://pastebin.com/d42e1f79e
16:05.58[TK]D-Fenderzuchmir: externip = my.fqdn.com <- NO.  this is for IP, not HOST.  use externhost / externrefresh for that.
16:07.06zuchmirtkd-fender: as far as i can see the only diff between the two is that exernhost gets updated more frequently (it does a DNS lookup more often)
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16:07.45tzangersupjigator: sure, but I still need to feed it t38 from somewhere
16:11.08zuchmirhttp://pastebin.com/d2fc724d1
16:12.17andrewnmaps.google.com working for anyone?  all i see is a man in a uniform
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16:13.04andrewnstrange, it's ok now
16:13.04zuchmirtk-defender: http://pastebin.com/d2fc724d1
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16:14.48zuchmirwould bbpglobal be initiating a connection (and thus not being able to get in)?
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16:15.35nny_1anyone who deals a lot with polycoms seen issues where they don't notify the user of new voicemails? (over a course of a couple of days..)
16:16.34*** join/#asterisk grantm (n=grant@kolob.wingateservices.com)
16:16.45penguinFunkmine work perfect
16:16.50penguinFunksnom300's
16:17.09nny_1yeah i have snoms out there too, they work well
16:17.15_ShrikEnny_1: I may have seen something like that way back.  What firmware are you running?
16:17.17penguinFunkdo you have mailbox parameters set in sip.conf ? that match the mailboxes in voicemail.conf?
16:17.31nny_1_ShrikE: let me check stand by
16:18.27[TK]D-Fendernny_1: Nope.
16:20.32*** join/#asterisk galeras (n=Martin@201.245.228.127)
16:21.22nny_1_ShrikE: I am assuming by polycom "Bootrom" is the firmware on the phone, in that case, it is 3.2.2.0019
16:21.29galerasDear Sirs, please which is de default set for  memberdelay parameter ?
16:21.40nny_1_ShrikE: let me see if I can peep the provision FTP server and see what version resides on there
16:21.41_ShrikEnny_1: No the bootrom is not the firmware
16:21.47nny_1_ShrikE: heh ok stand by
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16:23.16zenobic[TK]D-Fender: do i have to define: Local/413 => 413 in voicemail.conf (there's: 413 => 413), too?
16:23.26nny_1_ShrikE: is it under SIP app version?
16:23.34[TK]D-Fenderzenobic: that has NOTHING to do with voicemail!
16:23.35nny_1_ShrikE: in that case 2.1.0.2708
16:23.48[TK]D-Fenderzenobic: You don't even know why you're doing that in your dialpllan do you?
16:23.48_ShrikEnny_1: That's it
16:24.19_ShrikEnny_1: Thats way newer than what I was running back when I saw that.
16:24.46nny_1_ShrikE: yeah I have a polycom 501 here same model firmware that has no issues
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16:25.57zuchmirtk-defender: any ideas?
16:26.05nny_1anyone know of an easy way to drop a voicemail in someones box without calling? :)
16:29.19nny_1hmm maybe this error in messages might* offer a clue
16:29.35nny_1er nm
16:29.43nny_1thats a app_queue.c error :)
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16:34.13brodiemhas anyone done a chan_local style agentlogin()?
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16:36.10brodiemor better yet, pstn agents
16:38.14[TK]D-Fenderzuchmir: check your forwarding
16:40.18nny_1hmm.. I guess the only way to test voicemail is to leave one.. problem is right now their dialplan doesn't go to vm during the day.. it rings two sip clients, and if no one answers, autoanswers, puts the caller on hold, and continues to ring the clients...
16:40.46nny_1can you just cp a msg0001.WAV to the voicemail folder to test?
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16:51.01fedyawhat does the config line that sets up the cdr fields look like?
16:51.16fedyai'm trying to figure out where in my config the dst field is set
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16:54.42dishemit doesnt use system time?
16:55.56rrittenhouseIs it possible to use a vonage voip "box" as an ATA with asterisk?
16:57.44[TK]D-Fenderrrittenhouse: iF YOU CAN UNLOCK IT AND KEEP IT UNLOCKED, SURE
16:57.55nny_1heh well
16:58.05nny_1just left a voicemail here and the polycom ist silent
16:58.12nny_1so whatever I did there I managed to dupe here :\
16:58.15hescoIf a caller ID phone number I'm seeing on calls coming from the * server is not set in the call file and its not in extensions.conf, where might it be getting into the configuration?
16:59.10jpsharpDo you have a a "mailbox" set in the phone's sip.conf entry?
16:59.11rrittenhouse[TK]D-Fender, I would assume you could just "redirect" the request to get the config file to your own config file.. if its that easy
17:01.00dishemrrittenhouse: yeah but the configs are encrypted
17:01.12nny_1so is this right? 1.) asterisk notifies of new voicemails when the sip client re-registers, and you can add a checkmwi to sip.conf to change the message waiting indicator timing?
17:01.34rrittenhousedishem, ah.. darn. Know where I can read up on the subject?
17:01.43rrittenhouseNot sure if this is the right place to discuss it
17:02.25*** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
17:02.27dishemrrittenhouse: Not sure of any place specific unfortunately.. I'd google for how to unlock your specific device
17:03.12dishemalso voip-info.org wiki sometimes has info on devices
17:03.56dishemI think with a lot of vonage devices you had to have never connected the device to the internet
17:04.15dishemI have a pap2 that I hope is easy to unlock some day.
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17:06.03nny_1well crap
17:06.13nny_1it seems that default checkmwi is already 10
17:06.38nny_1although it seems that the host=dynamic means the client registers itself. not sure if this could be an issue with the voicemail NOTIFY
17:06.57nny_1wonder if there is a way to send the NOTIFY from console to test
17:11.39davevg-btwtechnny_1, see sip_notify.conf, probably what you are looking for
17:12.40skopiinny_1: it may be an issue with your firewall...I haven't messed around with polycom phones for a while. there's an option in the polycom config (sip.cfg or phone.cfg I forget) to make the phone reregister every 120S or something
17:13.04skopiihttp://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
17:13.35nny_1davevg-btwtech: thanks will look, i have the same configs here and our polycom works nicely
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17:13.44nny_1skopii: they are both on the local LAN
17:13.50nny_1skopii: but thanks for the suggestion
17:14.05nny_1I had qualify off on the system, wondering if that would do anything
17:14.55skopiiin the sip user? that just displays them differently in sip show peers
17:15.03*** join/#asterisk fozzmoo (n=fozz@66.7.122.158)
17:15.05nny_1skopii: yeah
17:15.07skopiiit will show the latency
17:15.16nny_1skopii: yeah thats what I figured
17:15.21fozzmooHey. Can someone show me an example of using func_curl in a dialplan?
17:15.21skopiiat least that's what I thought it does heh
17:15.22nny_1need to find a way to test it
17:16.05nny_1can I just cp a new voicemail message to their INBOX?
17:16.05*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
17:16.05skopiinot sure, my guess is yes though
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17:18.20nny_1yeah it works
17:18.21nny_1cool
17:18.30nny_1easy way to test it at least
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17:20.25_ShrikEWhen did polycom release SIP 3.0?
17:20.47zenobicnny_1: asterisk 1.2? Theres also a txt.
17:21.34nny_1zenobic: nah 1.4
17:21.43nny_1zenobic: er but yeah I copied both
17:21.48nny_1is the txt file for 1.2?
17:22.31zenobicnot only 1.2 (i just asked). tried also with both :)
17:22.41nny_1ahh thanks
17:22.46skopiiisn't the txt file just the callerid?
17:23.02nny_1yeah it works here.. sad thing is the issue I am chasing is only* on that system.. methinks may be pebkac somehow
17:23.13nny_1skopii: and time called, duration, etc
17:23.30*** join/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk)
17:23.52skopiiwell if I was still at my old job I could dig up the * and polycom configs for ya =\
17:24.10nny_1hehe nah i think this problem isn'
17:24.27nny_1t related to the configs, I think it may be a user error or similar ilk
17:24.29*** part/#asterisk fozzmoo (n=fozz@66.7.122.158)
17:24.44nny_1but now that I can force a voicemail...
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17:40.13bsdwarriorI can call out with a .call file fine, but If I send the same info to the manager (using originate), I get this error.Starting SIP/104 at mycontext,,1 failed so falling back to exten 's'
17:41.52De_Monbsdwarrior mycontext,,1 isn't valid so that would make sense. What *Exactly* are you sending with originate?
17:42.28bsdwarriorde_mon,  mycontext - its really outbound, i will pastebin
17:43.16outtoluncbsdwarrior: use Exten: not Extension:
17:43.48bsdwarriorhmm
17:43.58bsdwarriorhttp://pastebin.com/d238ed8eb
17:44.25bsdwarriorthats awesome
17:44.26bsdwarriorthanks man
17:44.30outtoluncnp
17:45.01bsdwarriorouttolunc - Its correct in the docs too. I can't read
17:45.23outtoluncit is one of those 'differences' that has gotten a LOT of people over the years
17:46.43*** join/#asterisk [T]ank (n=ckwall@206.71.78.158)
17:47.02[T]ankis there a dialtone sound file?
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17:48.13[T]anki am doing a section of an auto attendant where an outbound call may be placed... i want it to do something like: exten => blah,1,Background(dialtone)
17:48.28[T]ankbut i do not see any sound files that are named what I would expect for dial tome
17:48.29[T]anktone
17:48.56[T]ankexten => 9833899,1,Answer()
17:48.56[T]ankexten => 9833899,n,Background(tt-monkeys)
17:48.56[T]ankexten => 9833899,n,Set(TIMEOUT(digit)=5)
17:48.56[T]ankexten => 9833899,n,Set(TIMEOUT(response)=10)
17:49.12bsdwarrioris there any way to show on the phone who they are calling? I.E. Im sending calls to the phones but the callerid gets cutoff
17:50.27outtoluncbsdwarrior: the easiest way is to set a var, then set that var as teh callerid on its way back in
17:50.39outtoluncyou can do that with 'Variable: ...'
17:51.03outtoluncremember to use __VARS
17:51.08bsdwarriorim setting the caller Id as "Callback phonenum", but the phone number is cut off ?
17:51.43outtoluncyour callerid should be complete as in 'Some Name <xxxxxxxxxx>'
17:52.13bsdwarriorthanks. fixed
17:52.59x86[TK]D-Fender: hey i forgot the value in the polycom config file to have the phone auto-check for updates (firmware updates), do you know it off the top of your head?
17:54.58jjshoe[T]ank why not play dtmf digits?
17:55.42kyronQ: does * follow the even/odd version release scheme such as the Linux kernel?
17:56.04bsdwarrioronce I send a call with the manager and originate, is there any way to get the status of the call besides using cdr ?
17:56.22[T]ankjjshoe: found DISA, seems to do what I am after.
17:56.30*** part/#asterisk [T]ank (n=ckwall@206.71.78.158)
17:56.48skopiibsdwarrior: maybe AGI...
17:57.58outtoluncbbiam nature
17:58.05jjshoebsdwarrior what exactly are you doing?
17:59.06x86kyron: 1.0 == stable, 1.2 == stable, 1.4 == stable, 1.6 == development...
17:59.12x86so i dont think so
17:59.49kyronhmm...weird.. so 1.5 == twilight zone... (wooOOOooooOOooo)
18:00.03bsdwarriorjjshoe, im sending outbound calls , I want to know if they dont answer, etc so I can flag it in the db and try again later.
18:00.06kyronx86, also, aren't 1.0==1.2==deprecated?
18:01.08x86kyron: now they are, but at one time they were stable releases
18:01.26*** join/#asterisk whymarkwhy (n=koos@196.211.34.2)
18:01.37*** join/#asterisk shinao1 (n=shinao1@41.222.65.129)
18:02.06kyronof course...
18:02.30skopiiI thought 1.2 was supported?
18:02.31*** join/#asterisk atisss (n=atisss@193.238.212.171)
18:02.36outtoluncbsdwarrior: most use async: true so it does not hangout waiting for originate to end, and add exten => failed,1,Hangup to the context.. this is create either OriginateSuccess/Failure OR OriginateResponse events
18:03.04kyronFrom what I am reading on the * web site, I am better to start off *Now than trixbox (does this imply that *now is more compatible in the way it generates/manages the config files?)
18:04.03whymarkwhyhi there could anyone please help me i dont grasp the consept if i dial a sip channel(x-lite(Horaaaaaaaaaaaaaaaaa) finaly got that out of the way, my brain tells me if a phone is busy it must be ingaged, how do you send a ingaged signal if sip phone is busy currently i get more than one call ringing on my sip phone ps: did i tel you i got my first sip call
18:04.05kyron(yeah, I know this ain't #asterisk-gui nor #asterisknow but I want the pov from #asterisk people ;) )
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18:04.37skopiikyron: try them both?
18:04.50kamanashisroyhi, anyone initiated a call from manager and tracked that ?
18:05.43kyronskopii, uhm...well...am running trix at the moment but was considering asteriskNOW for better support/compatibility/flexibility+manual configurability...
18:06.03*** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2)
18:06.11whymarkwhyit says you can specify to jump to exten +101 if busy how does one implement that and could some please set some light on this
18:06.16skopiifreepbx has all that...there should be a foo_custom.conf file that is empty
18:06.29ArchSSMIs either trixbox/asteriskNow really something for a 'professional' installation ? ...
18:06.33whymarkwhyis this a freepbx channel?
18:06.48skopiiisn't asterisk the freepbx?
18:06.54[TK]D-Fenderwhymarkwhy: BOOK... and priority jumping like that is deprecated...
18:06.57[TK]D-Fender~book
18:06.58jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
18:06.58fiXXXerMetI have asterisk working just fine, but as soon as I install asterisk-addons, asterisk stops working.  Stops as in when I do an asterisk -r, it doesn't bring me to the command line.  It shows the copyright info and then puts me on a blank/empty line.
18:07.03skopiii mean it's free and it's a pbx right?
18:07.12whymarkwhythey told me the best way to learn asterik is from the command line
18:07.28[TK]D-Fenderkyron: All GUI's own your ass, and IMO are to be avoided.
18:07.44whymarkwhyi installed asterisk 1.4 does this book cover it? [TK]D-Fender?
18:07.58fiXXXerMetwhymarkwhy: yes, give the book a read.  It's good.
18:08.00[TK]D-Fenderwhymarkwhy: because as I said, priority jumping is DEPRECATED
18:08.00skopii[TK]D-Fender: I think gui's are like training wheels
18:08.04*** part/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
18:08.08[TK]D-Fender~zeeek
18:08.08jbotwell, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."
18:08.10[TK]D-Fender^^^
18:08.17kyron[TK]D-Fender, hehe, so not even worth looking into then...nuff said ;)
18:08.31kyronLOOL
18:08.31whymarkwhygood one
18:08.34kyronlmao
18:08.36jjshoebsdwarrior I'd sent it all to a context and do the work there *Shrug*
18:08.37*** part/#asterisk whymarkwhy (n=koos@196.211.34.2)
18:08.52[TK]D-Fenderskopii: Training wheels?  No... at least YOU are riding the bike.  FreePBX is like having a private chauffeur.  You SIT in the car while someone else does everything and you learn NOTHING.
18:09.06*** join/#asterisk [T]ank (n=ckwall@206.71.78.158)
18:09.37skopii[TK]D-Fender: well in all honesty I learned * by setting up freepbx
18:09.45skopiiit was all very overwhelming at first for me
18:10.00[T]ankis it possible to call into a telephone number and capture digits entered in to the phone and use them as a variable?
18:10.05[TK]D-Fenderskopii: and then when your driver runs into a wall you come out screaming and trying to fix the car but you know nothing, and nobody wants to touch the car your driver though was "well desinged and easy to maintain"
18:10.22[TK]D-Fenderskopii: its doesn't teach anything about dialplan... theirs is a psycho mess.
18:10.30[T]ankwhat I am trying to do is call a number enter my desired caller id and then use it when dialing out to a new telephone number via disa
18:10.31jjshoe[T]ank do you mean Read ?
18:10.41skopiithey have some cool macros..
18:10.42[T]ankmaybe, let me go to the wiki and see that one.
18:10.44jjshoe[T]ank Read.
18:10.48[TK]D-Fender[T]ank: "show application read" ,_
18:10.53[T]ankawesome
18:10.54[T]ankthanks
18:10.56bsdwarriorouttolunc - I have that setup, so then I can just lookup what happened to the call in the cdr db ?
18:11.08[TK]D-Fender[T]ank: Don't use the WIKI for this stuff until you have gone through the * CLI
18:11.16[T]ankok
18:12.18outtoluncjust parse that Event: and use the Reason: x code
18:13.01bsdwarriorforgive me but parse it from where?
18:13.33outtoluncumm.. from the manager interface.. where else would an "Event: .." be
18:13.58bsdwarriorok
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18:20.37fiXXXerMetIs asterisk-addons-1.4.5 broken?  As soon as I install it, even with just the cdr_addon_mysql and res_config_mysql modules, asterisk stops working.
18:21.22fiXXXerMetOr should I just use odbc with mysql instead of mysql directly?
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18:25.10De_MonfiXXXerMet look at asterisks logs or start it in console mode and find out why it "stops working" an fix it.
18:25.48fiXXXerMetDe_Mon: The problem is that I can't get to the console and that the logs are empty, even with debug turned on.
18:26.00*** join/#asterisk prabu^ (i=prabu@prabu.hitbsecconf.org)
18:26.25prabu^Hi guys, i just wanted to know does SIP/SIMPLE messaging work better in 1.6.0-Beta1
18:27.06De_MonfiXXXerMet -f doesn't work?
18:27.32fiXXXerMetDe_Mon: Doesn't just 'asterisk' start it then 'asterisk -r' to get to the console?
18:27.38*** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com)
18:29.49De_Mon-f tells asterisk not to fork, so there is no need to re-attach
18:30.01De_Mon(-r)
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18:39.23fedyahttp://pastebin.com/d75e29e4f
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18:39.32fedyadoes this mean that the context isn't found?
18:41.50[TK]D-Fenderfedya: no exten that matches the number is in a context with that name
18:42.44fedyahttp://pastebin.com/d275ba4ea  <-- this is in sip_custom.conf
18:43.25ZPerteeI have Linksys Spa8000.  I have setup sip extensions in asterisk w/ voicemail.  Also I have setup SPA8000 to give phones a stuttered dial tone.  The SPA detects messages as I can tell from the web interface info page.  However no MWI or VMWI.  Any ideas?
18:43.31[TK]D-Fenderfedya: so?
18:43.50UnixDogok I have a question about 1.4 and 1.6 with respects to users.conf
18:44.22UnixDogI like the idea of users.conf but why did they not make a trunks.conf for trunk setting to match
18:44.37UnixDogthat way you seperate the 2 and have better control
18:47.18fedyaas far as i understand i want to go into this context to extract the DID, then go to from-trunk like it was doing originally
18:47.34[TK]D-FenderUnixDog: And we used to have "type=peer/user/friend" for sip.conf, but thats being consolidated as well.  2 files would eb a step back for them.
18:47.42*** join/#asterisk RoyK (n=roy@91.149.21.205)
18:47.51[TK]D-Fenderfedya: well that context you showed is clearly not being used for anything.
18:48.09[TK]D-Fenderfedya: So you can go ahead and create 500 more just like it and it won't get you any farther.
18:48.19UnixDogwhy would users.conf and trunks.conf be a step back ?
18:48.39UnixDogits just putting trunks in 1 file to better control them
18:48.42UnixDogbut ok
18:48.55[TK]D-FenderUnixDog: because Asterisk is CONSOLIDATING account-types, not splitting up.
18:48.56fedyai guess that's why it rings busy, i dont know why it doesn't do anything though...
18:49.16[TK]D-Fenderfedya: it doesn't because the context that is being refernced doesn't have a match.
18:50.23pagecdoes asterisk support NI1 or NI2 better for PRI lines?
18:50.48[TK]D-Fenderpagec: Either jsut as well
18:51.17[TK]D-Fenderpagec: I think NI2 is offering 2BCT soon for * though, so if you can, thats probably a better choice.
18:52.32jpsharp2b call transfer?
18:52.43*** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk)
18:53.11[TK]D-FenderUnixDog: And a thing to understand : SIP is SIP.  * has no clue about HOW you want to use this "account".  What makes one a "phone" versus a "trunk" (another term that should NEVER be used)
18:53.27[TK]D-Fenderjpsharp: 2 B Channel Transfer
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18:53.44pagec[TK]D-Fender: tnaks
18:53.49pagec*thanks
18:57.43UnixDogwell then tk it is trunks vs user exten vs devices
18:58.27[TK]D-FenderUnixDog: users.conf = flaming pile of SHIT.
18:58.45UnixDog?
18:59.09UnixDogit allows you to set a single exten as both iax and sip is a plus
18:59.46UnixDogit allows you to create a single exten and set it up insted of having to waste time and do it twice in 2 files
19:00.00UnixDogand the have to write dial plan to ring both
19:01.07UnixDogand the idea of doing the same for trunks that would be put in sip and iax.conf into a trunks.conf would allow the same if you have a provider that allows both type of trunks for fail over.
19:01.39UnixDogbut I was just asking
19:01.47UnixDognot starting a flamewar
19:02.45*** join/#asterisk uluatu (n=deg@200.195.161.164)
19:03.46pagecare there any good pages out there on asterisk and fax setups?
19:03.57hmmhesaysI'm having a hell of a time compiling gnutls on this blackfin
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19:36.49d-techanyone here have access to a live ccm5 or ccm6 system ... I need 'LdapDialingRules.xml' config file ... TIA
19:39.59*** part/#asterisk myiagy (n=Jose@200.215.59.133)
19:41.41[TK]D-Fenderd-tech: thats like asking for a Whopper at McDonalds.....
19:42.30SwKcan asterisk do do CSD on NI?
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19:45.16hmmhesaysuclibc
19:45.28*** join/#asterisk The_X (i=chris@true.fiberpimp.net)
19:45.46The_Xanyone ever used an adtran TA 90x with asterisk for voicemail?
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19:49.04*** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk)
19:49.20[TK]D-FenderThe_X: Asterisk already does VM, why would we use something external?
19:50.01*** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net)
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19:52.18The_Xcustomer doesn't have an asterisk box
19:52.22The_Xonly an adtran with analog ports
19:52.38The_XI want the adtran to fwd voicemail to a centralized asterisk server
19:53.09jjshoeThe_X I'm sure someone has, are you having a specific issue you need help with?
19:55.04[TK]D-FenderThe_X: And how would this Adtran be "forwarding" on voicemail?
19:55.41*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
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19:57.04The_Xfwd on busy?
19:57.18The_Xor after 5 rings, I'm sure it's doable
19:57.30[TK]D-FenderThe_X: HOW is it doing it?  What is * expected to do?
19:57.46[TK]D-FenderThe_X: Where do those ports go now?
19:58.40The_Xever used an adtran TA?
19:59.57*** join/#asterisk dho_ragus (n=dho_ragu@cup1.sugarcrm.net)
20:00.21dho_ragusdoes anybody know a good polycom vendor?  dell is being stupid with polycom right now, and my other polycom vendor is totally giving me the run-around.
20:00.54*** part/#asterisk kamanashisroy (n=root@202.56.7.142)
20:01.04jblack[TK]D-Fender: Still not today.
20:01.05Qwelldho_ragus: I always see [TK]D-Fender recommend telephonydepot
20:02.58*** part/#asterisk jjshoe (n=jjshoe@72.37.252.50)
20:04.45dho_ragusthanks Qwell
20:06.56*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
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20:14.12*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
20:14.12*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta1 (2008/01/18), Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org)
20:19.20*** join/#asterisk nick-temp (n=spid3r@229.87.modemcable.oricom.ca)
20:21.50*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
20:21.50*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta1 (2008/01/18), Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org)
20:22.16*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
20:22.48*** mode/#asterisk [-b %#phix!*@*] by russellb
20:22.52*** mode/#asterisk [-b %phix!*@*] by russellb
20:23.04phix:D
20:23.19phixThat's better
20:25.33phixok, I want to hook up an ATA or TDM to an existing PBX system (as extra lines for it).  Since it will be providing extra lines I will need FXO modules right?
20:26.09jpsharpFXO modules for the PBX, yes.
20:26.48*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
20:27.00phixnice, now my next question (I have a few cached up :)), ATA or TDM? I have 3 lines left over on my existing PBX system that I can use.
20:27.15jpsharpDepends on how much you want to spend.
20:27.50phixwell the only ATAs I can find are 2 or 4 fxs :( not fxo.
20:28.13_ShrikEphix: audiocodes MP114-FXO
20:28.16phixa TDM with 3 FXO modules would set me back around $300 or so I think
20:28.28_ShrikEpretty decent ata
20:28.39jpsharpNo, you'll need FXS ports on the ATAs, FXO ports on the PBX.
20:28.54phix_ShrikE: ok, does it produce an echo? I am having echo / feedback issues on my linksys supara
20:29.06jpsharpAssuming that you want the ATAs to show up as lines not phones.
20:29.47_ShrikEecho is not really a product of the ATA.  I have sipuras that work just fine.
20:30.03jpsharpAll 8 of the lines on the Lucent PBX I have here at my office are driven off of 4 Cisco ATA-186s.
20:30.19phixjpsharp: 4 or so PSTN lines plug into the PBX atm, they are PSTN lines so they are fxo?
20:30.37jpsharpYes.  They're FXO ports on the PBX.
20:30.43phixoh ok
20:30.54phixlol I keep getting them mixed up
20:30.55jpsharpSo, a couple of plain, dumb ATAs will plug into the other 3 lines on your PBX.
20:31.17phixjpsharp: I am not happy with my supara :( but I can get them on special
20:31.21*** join/#asterisk UnixDog (n=unixdog@adsl-69-234-190-155.dsl.irvnca.pacbell.net)
20:31.25flujanhi guys... I have a wav file working as a music on hold file...
20:31.26phix$60 each
20:31.40phixflujan: I have that too on a system, but it chunks :(
20:31.42flujanI have two queues, One receives calls from a sip trunk and the musiconhold is working...
20:32.10bsdwarriorI want to prevent 411,911 ,etc to be called back. does anyone have a good list of blocked outbound numbers?
20:32.12flujanthe other queue receive calls from a pri/e1 digium board.
20:32.20flujanguys that call on the sip trunk hear the music.
20:32.36flujancalls incoming from the queue does not work. The caller hear nothing...
20:32.38jpsharpphix:  Look at other vendors for ATAs?
20:32.41flujanany idea?
20:32.59*** join/#asterisk fnordus (n=dnall@24.84.160.227)
20:33.06jpsharper, rather, looking?
20:33.41flujanphix: why it chunks?
20:34.00phixjpsharp: maybe, but I don't know which ones are good
20:34.04phixflujan: nfi :(
20:34.19[TK]D-Fenderbsdwarrior: its your dialplan, what are you LETTING them dial those numbers?
20:34.53bsdwarriortkd-fender, Im just looking for a blacklist of numbers.
20:34.55[TK]D-Fenderbsdwarrior: You don't seem to be following the big picture.  * accepts those calls because you made extensions for them.  Simply don't DO it for those.
20:34.57jpsharpI'm happy with the ATA-186s I have.  I've also used Grandstream HT-486s.
20:35.46[TK]D-Fenderphix: I've never had any issue with Sipura/Linksys
20:35.53phixI have heard bad things about grandstream
20:35.57[TK]D-Fenderphix: then tend to be the most predictable and flexible.
20:36.19phix[TK]D-Fender: ok. well maybe it is my asterisl system producing the echo / feedback
20:36.28bsdwarriortkd-fender forget it your just talking over me
20:36.32phix[TK]D-Fender: how can I tell?
20:36.45flujan[TK]D-Fender: any idea about the issue I am experiencing?
20:37.06[TK]D-Fenderflujan: pastebin it.
20:37.08flujan[TK]D-Fender: some time ago you said to me use the wav file to avoid the mp3 problems of transcoding...
20:37.12phix[TK]D-Fender: I have an analog phone connected to sipura, I can hear my voice intermittently
20:37.14flujan[TK]D-Fender: ok
20:37.26[TK]D-Fenderphix: Sounds like your gains are shot
20:37.39phixwhere are gains set?
20:37.52phixsipura or asterisk?
20:38.06[TK]D-Fenderphix: Sipura
20:38.24[TK]D-Fenderphix: also on those older models be careful which firmware you use.
20:38.32[TK]D-Fenderphix: the stock one was pretty decent.
20:38.47[TK]D-Fenderphix: I set up a friend of mine on my old SPA-2000 and its working 100% fine
20:39.58flujan[TK]D-Fender: http://pastebin.com/m1ebb4959
20:40.01ddunavantIs there any reason why * wouldn't read the priority labels I've assigned?
20:40.23*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur)
20:40.24[TK]D-Fenderddunavant: if you showed us your dialplan and the CLI output of your failed attempt perhaps we could.
20:40.26[TK]D-Fender~pb
20:40.26jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
20:40.28[TK]D-Fender^^^^^^^^^^^^^^
20:40.41phix[TK]D-Fender: Product Name:SPA-3102, Software Version:3.3.6(GW)
20:41.26[TK]D-Fenderphix: go to www.voxilla.com and check out their forums.  There are plenty of threads about which FW's work best and how to tweak them
20:41.37[TK]D-Fenderflujan: pastebin something COMPLETE
20:42.33flujan[TK]D-Fender: but that are my zapata.conf and musiconhold.conf files... which file do you need me to pastebin?
20:42.35phix[TK]D-Fender: ok thank you!
20:42.57[TK]D-Fenderflujan: full call showing the failure, queue config, etc.
20:43.18ddunavant[TK]D-Fender: http://pastebin.com/d1ccdbd0a
20:43.21flujan[TK]D-Fender: the queue is configured in the realtime table
20:44.07[TK]D-Fenderddunavant: Jan 22 15:36:45 NOTICE[17677]: pbx.c:1753 pbx_extension_helper: No such label ' CallID' in extension 's' in context 'macro-IntMenu' <-- see this?
20:44.40[TK]D-Fenderddunavant: No such label ' CallID' <- this means stop putting whiespace after the "?" in your GotoIf's
20:44.59ddunavantAhhhh
20:45.00[TK]D-Fenderddunavant: exten => s,n,GotoIf($[${Option} = 2]? CallerID) <- Whitespace = BAD
20:45.17ddunavantgotcha
20:45.17[TK]D-FenderNEXT!@!@ (c) BKW
20:47.04flujan[TK]D-Fender: here is what is happening:http://pastebin.com/m58d23327
20:47.14flujan[TK]D-Fender: the cli output from the file
20:48.01flujan[TK]D-Fender: as you can see, asterisk shows that it is playing the moh but I hear nothing... :(
20:48.32flujan[TK]D-Fender: with the mp3 files, I hear it well. :(
20:48.56_ShrikESome of the features in the new polycom 3.0 sip firmware look neat.
20:49.35*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
20:51.10nick-temphi, is there a way to *block* incoming calls to an agent when the extension he is registered to dial out on pstn ?
20:51.37*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
20:51.42Kattyhi!
20:51.55Kattyhewwoes?
20:52.00[TK]D-Fender<PROTECTED>
20:52.04Katty[TK]D-Fender: i have NEWS!
20:52.06Kattyalso, mews.
20:52.10Kattymew mew.
20:52.12Kattyetc.
20:53.40lmadsenwem
20:53.47[TK]D-FenderKatty: :O
20:54.06flujan[TK]D-Fender: do you think it could be a problem with the .wav file? But if the file was the problem the callers from the sip trunk should hear nothing too right?
20:54.46Kattyi officially have a timeline for leaving the united states.
20:54.53[TK]D-Fenderflujan: no idea, that doesn't tell me much...
20:54.55Kattyas in work visa leaving.
20:55.00Kattyhopefully.
20:55.14[TK]D-FenderKatty: Looking to move permanently and this is the first step?
20:55.25Katty[TK]D-Fender: *nod*
20:55.35[TK]D-FenderKatty: Neato... Whereabouts?
20:55.35flujan[TK]D-Fender: which information may help you help me??? I just loose some hair trying to discover what happened. :(
20:55.45Katty[TK]D-Fender: narrowed it down to europe.
20:55.54Katty[TK]D-Fender: due to speaking english and such.
20:55.56[TK]D-Fenderflujan: can't concentrate on this now, only much later tonight
20:56.11flujan[TK]D-Fender: ok
20:56.15[TK]D-FenderKatty: wow... big.  Solo?
20:56.18flujan[TK]D-Fender: thanks anyway for the help
20:56.19flujan:)
20:56.22Kattyi'm not sure how work visas work, tho i think a talk with Junky will help
20:56.37Katty[TK]D-Fender: mostly, solo.
20:56.58Katty[TK]D-Fender: no Big Move can be done entirely alone
20:57.36flujanhi jblack
20:57.43jblackhi
20:57.47jblackwhat's going on?
20:59.21jblackflujan: So, can you recap for me what's up?
20:59.32*** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net)
20:59.42flujanjblack: sure...
20:59.59fiXXXerMetThe ODBC -> MySQL works so well that I'm having trouble seeing why I need asterisk-addons at all.
21:00.09flujanI have two queues with musiconhold class=music1
21:00.32flujanone queue receive calls from a sip trunk... This queue is working and all users here the musiconhold while holding...
21:00.32Kattyanyone here have experience with work visas?
21:00.56flujanthe other queue receive calls from a pri/e1 line... these guys hears nothing while waiting... :(
21:01.13jblackOk.
21:01.18flujanjblack: asterisk shows that started music on hold on the channel, but the end point just do not hear it... :(
21:01.49jblackOk. First, let's make sure you're playing the same moh channel to both incoming contexts.
21:02.21jblackCan you pastebin your sip.conf (don't forget to XXX passwords) and your zapata.conf ?
21:02.31*** join/#asterisk atisss (n=atisss@193.238.212.171)
21:02.52*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
21:03.05jblackor, if you feel comfortable, make sure mohsuggest is appropriately set.
21:03.51[TK]D-Fenderflujan: go prove that class even works normally
21:04.19[koss]don't say XXX passwords you're getting me excited
21:05.00fiXXXerMethah
21:05.09jpsharpcoldshower.conf
21:05.14jpsharpor app_coldshower
21:05.16jblackSince he's claming that moh is working on sip, but not on zapata, I'm thinking he's setting different contexts; either suggesting an old, boken music class in zapata, or overriding a broken default class in sip.conf (which, iirc, can override the default moh)
21:05.33[TK]D-Fenderjblack: he has 3 classes, none of which I trust
21:05.49flujanjblack: http://pastebin.com/m4e66feb5
21:06.11jblackflujan: Then while I'm reading this pastebin, paste your musiconhold.conf as well.
21:06.31jpsharpand your extensions.conf too.
21:06.51flujanjblack: http://pastebin.com/m6c7d4370
21:07.05*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:07.35jblackYuck. Do you really need 6 different moh channels?
21:08.11jblackHmmm.
21:08.35jblackaccording to your sip.conf and your zapata.conf, you are sending them to two different places.
21:08.45jblackmusicclass=default
21:08.51jblackmusiconhold=music1
21:09.20jblackYou're sending sip to directory=/home/totalip/sounds/music, and zapata to directory=/home/totalip/sounds/music1
21:10.34flujanjblack: fixed it.. the file to the default and music1 is the same... the others are department specific mohs
21:10.37jblackSo, if the mp3s in /music/ are close enough to acceptable, they'll play... but the music1 ones may be loaded with id3tags and all sorts of stuff that the builtin format_mp3 inappropriately chokes on.
21:10.53jblackflujan: What do you mean "fixed it" ?
21:10.54flujanboth files are .wav
21:11.06flujanjblack: put both to the same directory.
21:11.25jblackNot according to the config files you pasted. You're definitely pointing to two different places for the channels
21:11.30lesouvageis it technically possible that two inbound lines spontanious get bridged to each other. A costumer claims that that happened today but the loggings and cdr look quite normal. It is an Asterisk 1.4 box and snom 320 phones with scopserv as gui. Any suggestion?
21:12.24jblacklesouvage: It's technically possible for two inbound lines to get bridged (there's tools to do it on purpose). I've never heard of it happening magically on it's own.
21:12.39filethere was actually a discussion on the -users list about that...
21:13.04flujanjblack: yeap... I saw it and changed the sip.conf to the /home/totalip/sounds/music1 location... musicclass=music1
21:13.04jblacklesouvage: file to your rescue. ;)
21:13.10file"Calls Being Randomly Bridged" specifically with Snom phones
21:13.39hmmhesaysI hate cross compile never fun, EVER
21:13.42filehad to do with transfers and the way Snom handles them on the way, people accidentally transferring the wrong legs
21:13.49jblackflujan: Um,... you changed the one that works to the one that doesn't?
21:13.58lesouvagefile & jblack: thanks
21:14.13*** join/#asterisk fnordus (n=dnall@24.84.160.227)
21:14.20jblacklesouvage: Don't thank me. I unintentionally misled you
21:15.05jblackfile: Ok... well then, reload your sip.conf, call in, and see if things are now broken on sip too.
21:15.24jblackIf so, it's time for you to examine the permissions for your audio files more carefully, the format of those wavs, etc.
21:15.43Kattyhmmhesays: hai
21:16.08jblackfile: No, I was talking to you.
21:16.10flujanjblack: reloaded and it works for the sip don't work for the zap
21:16.15flujanI will check the files now
21:16.16jblackfile: Sorry.
21:16.21jblackI am talking to flujan.
21:16.30flujanjblack: but they are exactly the same.
21:16.42fileif I'm asking for help in here something is wrong with me
21:17.01jblackNah. It's me. I'm only on my 3rd cup of coffee
21:17.34Kattyfile: ask for help. i dare you.
21:17.36hmmhesaysHello Katty
21:17.42fileKatty: nooooooo
21:17.47flujanjblack: http://pastebin.com/m2f1a8604
21:17.49jblackflujan: So you're telling me that you ahve changed things so that both sip and zapata are both pointing (this time, really?!?) at teh same moh channels, and sip still works, and zapata doesn't?
21:18.05flujanjblack: yeap.
21:18.07jblackflujan: You did do a sip reload, and called back in?
21:18.16hmmhesaysI can't get libgpg-error compile on this blackfin for the life of me
21:18.16flujanI reload asterisk
21:18.22flujanstop now and safe_asterisk again
21:19.31jblackOk. Let me see your sip.conf and zapata.conf again
21:19.50jblackand if you can run "file fila_espera.wav" for me, that would be nice
21:20.01*** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.ca.comcast.net)
21:20.05jblack(the file fila...) can be pasted here, since it's 1 line
21:22.09jblackbe back in a few minutes
21:23.20Kattyfile: :>
21:24.05[TK]D-Fenderok, heading home, later all.
21:24.07defsdoorare there any freely available english accent recordings available ?
21:25.02*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
21:25.13jameswfthere are some ausie prompts never heard uk
21:25.50jameswfsome southern hillbilly redneck prompts would be cool
21:25.59Kattyjameswf: :<
21:26.01jameswfhey yall press 1
21:26.05jpsharplol
21:26.06Katty:<<<
21:26.08Katty)_=
21:26.19Kattyjameswf: you're redonkerus!
21:26.25jpsharpHere, hold my beer and press 1
21:26.40defsdoorhmm found some
21:27.02russellbAsterisk: brought to you by the deep south.
21:27.07russellbseriously, it is.
21:27.15defsdoorhttp://www.enicomms.com/cutglassivr/
21:27.43flujanjblack:
21:27.44flujanhttp://pastebin.com/meb77495
21:27.57jameswfI set up a test dial plan and thought all-your-base might be cool, but it looses something when allison says it in broadcastereze
21:28.14jblacklooking
21:28.46jblackThat's as far as I can take you, flujan
21:29.26flujanjblack: thanks anyway... I just have no idea why it is happening... :(
21:29.28flujandanmed.
21:30.43*** join/#asterisk lelandg (i=leland@24-116-151-196.cpe.cableone.net)
21:31.42lelandgHi, I get "configure: error: *** termcap support not found" when I run './configure'; running debian
21:32.15*** join/#asterisk fedya (n=fedya@75.112.143.226)
21:32.44fedyaexten => _.,1,Set(CDR(accountcode)=4949494) <-- is there any reason why this won't work, i tried userfield too, and i have it enabled in cdr_mysql
21:33.00*** join/#asterisk RoyK (n=roy@91.149.3.9)
21:35.01*** join/#asterisk janinge (i=j@ninge.net)
21:37.10*** join/#asterisk Strom_C (n=strom@208.127.172.112)
21:37.36*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:38.36jblackfedya: I'd try using external (say, madplay) on music1, and see if mp3s play ok.
21:38.44jblackfedya: Never mind. someone else
21:39.03jblackflujan: I'd try using external (say, madplay) on music1, and see if mp3s play ok. The internal moh player has plenty of problems.
21:39.07*** join/#asterisk RoyK (n=roy@91.149.3.9)
21:39.33jblackflujan: I imagine that its also possible that in extensions.conf, you're not really sending people into moh like you think
21:39.41jblackor into the wrong one
21:40.21jblackfedya: _. is really, really buggy. It tells you that on the console. Try _X.
21:40.52fedyawhat does that part mean, where can i read something about it?
21:41.12fedyai put s now, i saw that in some other guides
21:41.38jblackIn the console.
21:41.56jblackhttp://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns explains some though.
21:44.25flujanjblack: sure with madplay and mp3 it worked for both... :D
21:44.33flujandoes madplay supports .wav files?
21:44.38jblackflujan: No idea.
21:44.45*** join/#asterisk atisss (n=atisss@193.238.212.171)
21:44.49jblackmadplay does support a variety of formats, though.
21:45.53*** part/#asterisk [T]ank (n=ckwall@206.71.78.158)
21:46.40jblackAt least on output it does. I don't know what it supports on input.
21:47.17*** join/#asterisk tripps (n=sean@72.20.150.196)
21:47.28fedya.X,1 does the 1 indicate the instruciton number?
21:47.33fedyai'm still here
21:47.59jblackdonadie: Please take me off your list of bot recipients.
21:48.53jblackfedya: the 1 represents the priority for that extension.  Please reread chapter 5 of the * book
21:53.56*** join/#asterisk Mavvie (n=edwin@ppp121-44-71-49.lns10.syd6.internode.on.net)
21:55.19fedyashould i use s, or _X?
21:55.29jblackYou should use the book.
21:57.26jameswf~book
21:57.27jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
21:57.34jblackI'm not being an asshole. There's a lot there you have to read and understand. walking it through will take longer, be less comprehensive, and take longer than for you to read the chaper yourself
21:57.55jblackmuch longer, considering how i said longer twice. =)
21:57.57jameswfyes you are jblack but its cool we still love you
21:58.14jblackYou callin me an asshole?
21:58.25jameswf:)
21:58.32jblackLet me get this right. You. are calling. Me. An asshole?
21:58.39JTjameswf: btw, it's "aussie, not ausie
21:58.48jblackI'm gonna come down there and kick your parakeet's ass!
21:58.52jameswftechnicaly I was agreeing with you in an inverse manner
21:59.06*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
21:59.42*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
22:00.15jblack[TK]D-Fender: I think flujan is basically covered. By getting him in the same moh class and convincing him to use an external player, he's much happier
22:00.49flujanjblack: I will test it... :D
22:01.12jblackflujan: Testing something.... That's a good idea!
22:01.43flujanjblack: I really dunno what this damned this is happening... I will stop asterisk and try to use the .wav with madplay..
22:01.48defsdooranyone use debian ? I trying to listen to some .gsm files and can't find a player that will play them
22:01.52flujanI dont have the mp3 file right now...
22:02.01jblackdefsdoor: I found one.
22:02.03jblacklet me find it again
22:02.42jblackhere it is. the play command from sox plays them.
22:02.44defsdoorI've got libgsm installed so figured sox might
22:03.01defsdoorjblack: play doesn't recognise them here :o
22:03.09defsdoorplay soxio: Failed reading `0.gsm': unknown file type `gsm'
22:03.18defsdooraah libsox-fmt-gsm
22:04.12jblackAnd that makes 3. I think I have fulfilled my civic duty for the day and can spend the rest of my day with obnoxious-but hilarious-trolling
22:04.26defsdoorjblack: nah that don't count
22:04.55defsdoorjblack: you were no help to me at all - I'd already tried play - I used my aptitude search skills to find the lib needed
22:04.58jblackYou're at death's door. Your mind is muddled. So I'm right, and you're wrong.
22:05.04defsdoorsorry - troll later :)
22:05.50fedyawoo i got it to set the account code, using _XXXXXXXXXX
22:06.02jblackfedya: Figured.
22:06.06defsdoorfedya: :|
22:06.09jblackfedya: Congrats
22:06.24fedyais that the solution?
22:06.34fedyahopefully it works the same way for userfield
22:06.38jblackfedya: Avoiding _. ? Yup.
22:06.45fedyathe guide i was looking at did it with _.
22:06.53fedyai don't understand the 's'
22:07.03*** join/#asterisk ZX81 (n=ZX81@202.49.106.158)
22:07.04jblackfedya: that's why I told you to read chapter 5 of the book.
22:07.25jblackfedya: Buy the book if you can. If you can't, use the pdf (type ~book) until you can afford it.
22:07.47defsdoorfedya: _X. no good ?
22:08.21defsdoorI bought the book - never read it :)
22:08.23fedyainteresting assumption
22:08.37fedyai've got the book right here, i didnt realize you were talking about the same book i have
22:08.40defsdoorread the pdf instead as I left the book at home
22:08.59jblackfedya: No assumption at all. There's only one book, as far as I know.
22:09.07defsdoor_the_ book
22:09.11fedyaafford it :0 :)
22:09.32*** join/#asterisk SteveTotaro (n=root@pool-70-16-26-249.balt.east.verizon.net)
22:09.50jblackOhhh, you can afford that. That's great. Can you afford to send me a check for $2,500?  Daddy wants a new computer.
22:09.53[TK]D-Fender_. = evil
22:10.15defsdoor_X. is ok though isn't it ?
22:10.17fedyainteresting assumption because i can't afford it
22:10.24[TK]D-Fenderdefsdoor, much better.
22:10.25ZX81hey, anyone know how to get ${CDR(duration)} in a deadagi run in h exten?
22:10.26jblackdefsdoor: Much safer.
22:10.29fedyabut i do have the book, company property
22:10.51*** join/#asterisk echosyp (n=echosyp@75.111.175.135)
22:10.57fedyajblack: a macbook air/
22:11.00fedya?
22:11.07defsdoorjblack: btw - still can't play .gsm file :)  play writes to some odd default audio device :)
22:11.13echosypcan someone clarify the difference between a sip server and a pbx
22:11.18fedyai'll try the _X. see if it works
22:11.34echosypim assuming the sip server can't be terminated to the pstn
22:11.38defsdoorechosyp: (possibly) a sip server just routes sip to sip
22:11.50echosypthat was my thought
22:11.51jblackfedya: I didn't assume you could afford it or not. I gave you a simple code path  while (toobroke) ReadPdf(); BuyBook();
22:11.57SteveTotaroa pbx might not even do VoIP
22:11.58echosypi want a pbx
22:12.11echosypi see
22:12.12defsdoorechosyp: I can sell you one
22:12.13jblackechosyp: Asterisk is your char.
22:12.14fedyaHAH!!
22:12.26echosypi don't pay for things i can do myself
22:12.27echosypsorry
22:12.31fedyathx for helping me with that :0, weird glitch with the _.
22:12.39*** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com)
22:12.46SteveTotaroyou must be young echosyp
22:13.05jblackechosyp: Asterisk is free software. that said, do you bake your own bread? Sew your own clothing?  Make your own movies?
22:13.06echosypno
22:13.12defsdoorechosyp: paying me to do it is much more fun (and helps my mortgage)
22:13.30echosypjblack missed the point
22:13.37echosyphah
22:13.38SteveTotaroi can do plenty of things myself but my time is better spent doing things that make money
22:13.47defsdoorthough I am still a novice - building my 3rd * install now
22:13.49echosypits fun for me learning something new
22:13.50[TK]D-Fenderjblack, Just that once, but Paris' lawyers are trying to block its release...
22:14.03echosypim installing asterisk on my router
22:14.07echosypdd-wrt
22:14.08jblackNope. Not at all. I just happen to know that in a world like ours, that people can often do things for you cheaper than one can do it for oneself.
22:14.13defsdoorSteveTotaro: I can do some things myself but I much prefer my wife to do them ;)
22:14.25echosypusing gizmo to connect, or so the plan is
22:14.30SteveTotarolol def
22:14.36jblackI can make my own cheeseburgers. But if I give mcdonald's a dollar, they'll give me one that's nicer than what I can make.
22:14.37SteveTotarothat is a good point too
22:14.38echosypeventually i'll terminate it
22:14.49fedyaok i can set the userfield now, but the call never makes it through now.. 1 problem fixed but now the call doesnt come to the phone
22:14.58jblackYeah. And don't forget children. They're virtually slaves until they're old enough to get a job.
22:15.02[TK]D-Fenderjblack, You clearly suck if yours doesn't kill McD's :0
22:15.11echosypyeah
22:15.24SteveTotarowendy's makes a killer triple quarter pounder
22:15.31echosyplisten to this guy
22:15.42defsdoorfedya: trixbox isn't it - where have you added your bit ?
22:15.42echosypsaying children are slaves
22:15.51echosypi bet he runs a sweat shop in his attic
22:15.54SteveTotaro~trixbox
22:15.55jbot[trixbox] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
22:16.00jblackMcDonald's can make a better burger for a buck than I can.
22:16.29defsdoorjblack: dunno - you'd have to make a pretty bad burger
22:16.44SteveTotarofor $1 is his point
22:16.51echosypin any case
22:17.03echosyphas anyone ever setup asterisk on a dd-wrt router?
22:17.09SteveTotaroi have
22:17.20[TK]D-Fenderechosyp, yes, a whole bunch of people have
22:17.23jblackHave you seen how much food costs these days?
22:17.24SteveTotaroyou should use an old pc or something
22:17.32fedyayes tb
22:17.40defsdoorI used a laptop with a busted screen
22:17.42echosypwhats sad is my old pc's are weaker than my router
22:17.48echosypi have a wrt350n
22:18.06echosypwith usb
22:18.12SteveTotaroyeah but you cannot stick a modem in it and plug a phone line in
22:18.29SteveTotaroif you really want to explore asterisk, i would go with a pc
22:18.36phixhi
22:18.41jblackThe 1/3 of a pound of beef costs 66 cents alone. Add the cheese, and that's probably another 30 cents.
22:19.07echosypwhy can't i plug a modem in it
22:19.09echosypits got usb
22:19.10[TK]D-FenderSteveTotaro, you just said "really want to explore asterisk" and " stick a modem in it".... LOL.
22:19.13jblackThat leaves 4 cents for bun and cooking. I'll either have to skip the bun, or eat a raw hamburger.
22:19.13echosypthere are usb modems
22:19.33[TK]D-Fenderechosyp, those will not work with *.
22:19.35jblackechosyp: That's not the funny part.
22:19.41SteveTotarohe can use a "modem" as an FXO
22:19.46jblackasterisk is software. there's no plug to stick a modem into.
22:19.52[TK]D-Fenderechosyp, You are working with a toaster, don't expect more than toast.... BAGELS if you're lucky
22:20.08defsdoorjblack: got a 5 star name badge ?
22:20.11Corydon76-digYou are not a member of the A-Team.  You cannot take a pile of junk and make it into a mansion in 30 minutes.
22:20.20echosypiv already done taht
22:20.26[TK]D-FenderSteveTotaro, There is only a specific chitset that can be used, the other 99.999% of them are 100% worthless.
22:20.31echosypasterisk is installed on it
22:20.35[TK]D-FenderSteveTotaro, and even then the X100P SUCKED.
22:20.39echosypi just need to configure it and test it
22:20.39jblackdefsdoor: Heh. my technical skillset has ensured plenty of jobs. My lack of a personality skillset ensures that I don't keep them. =)
22:20.47echosypi can get an fxo for it
22:21.07[TK]D-Fenderechosyp, well if * is installed go get the book, and start reading.
22:21.10defsdoorjblack: if you don't mind me asking what are you doing now then ?
22:21.13SteveTotaroyou could buy a linksys or something i suppose
22:21.21SteveTotaronot much room for VM on a router though
22:21.24jblackdefsdoor: I breathe for a living.
22:21.37defsdooryou got a good deal there then
22:21.41SteveTotaroand eat $.99 burgers
22:21.53defsdooroccasionally a happy meal ?
22:22.06SteveTotarosticks to the dollar menu
22:22.08defsdoorjust - for a treat ?
22:22.12*** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com)
22:22.14jblack(/me takes a deep breath in). See that? I just made ten cents
22:22.39lesouvageI found the info about the romdomly bridged calls on the user list (http://readlist.com/lists/lists.digium.com/asterisk-users/14/70908.html) and I'm astonished about the bug. How to explain to the costumer?
22:22.58bsdwarriorim calling a perl script with agi(myscript.pl) . the script uses Asterisk::AGI; when I call $AGI->set_variable($variable, $value) in the script, I should be able to read the variable in asterisk?
22:23.21SteveTotarolesouvage, just tell them it is their imagination
22:23.30SteveTotarountil the bugfix comes out soon
22:23.37SteveTotarouser error
22:25.25lesouvageSteveTotaro: To be hones, I did that a couple of month ago because I couldn't believe that it was caused by a software bug. It happens now 4 times with 60 phones in 5 month.
22:25.45SteveTotarosame here
22:26.01tzangerlesouvage: I saw that on the list too, never experienced it though
22:26.33SteveTotarothe best thing to do is to show them the bug report or at least explain it to them and that a fix is on the way
22:26.34fiXXXerMetIs there a way to have the Directory() application read back the names to you, maybe using Festival?
22:26.50lesouvageSteveTotaro: How did you deal with it, the imagination option?
22:27.07SteveTotaroi seriously thought it was their imagination
22:27.08fedyahttp://pastebin.com/d75ae2124, before changing to _X. it would go through this context, not set the userfield and return to the previous context and ring, now it sets the userfield but doesn't ring
22:27.22[TK]D-FenderfiXXXerMet, your users should be recording their names for the directory....
22:27.37SteveTotaroit did not happen enough to recreate
22:29.21lesouvageSteveTotaro: did you have contact with Snom?
22:31.47defsdoorfedya: I'm talking to you in #trixbox :O
22:31.58*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
22:32.12SteveTotarono, i don't do work for that company anymore
22:33.53*** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net)
22:36.24[TK]D-Fenderfedya, it wouldn't ring or do anything proactrical... because there is nothing more to do!  You ran out of priorities and aren't sending it off or doing anything.
22:40.35fiXXXerMetTesting out Directory().  When I type the 3 digits of the person's last name, asterisk hangs up.  console and extensions @ http://pastebin.com/m1f2f9001
22:42.17fedyaif i take out the CDR() line, it goes on to the next context from the context that called from-pstn-custom, and rings, but when the CDR() line is in there it rings busy
22:42.46[TK]D-FenderfiXXXerMet, voicemail.conf plesae...
22:43.27[TK]D-Fenderfedya, doesn't matter.  that exten is a dead-end
22:43.53*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
22:43.53*** mode/#asterisk [+o anthm] by ChanServ
22:43.58[TK]D-Fenderfedya, and the end of that last priority your call will terminate
22:44.01ZX81anyone here used the maxtnt or the lucent apx8000?
22:44.20fedyaa. howcome it works without the line? b. where do i tell it to go?
22:44.31[TK]D-Fenderfedya, Show me.
22:44.33fedyahttp://pastebin.com/d43187118
22:44.44fedyafirst 3 lines show what it does without the CDR() line
22:44.52fedyaeven with the deadend
22:44.55defsdoor[TK]D-Fender: it's included from [from-pstn]
22:45.04fedyaso i guess it needs to go to ext-did
22:45.30[TK]D-FenderExecuting Goto("SIP/64.2.142.30-088e9d08", "ext-did|s|1") in new stack <- where is YOUR goto?  You showed me a DEAD-END
22:45.34fedyafrom-pstn: http://pastebin.com/d1dbaf65
22:45.45ZX81I need a box that can handle ~1xds3 of fxs
22:45.55[TK]D-Fenderfedya, And you do NOT merge contexts with confliscting extens in them!
22:46.15fedyawhat does that mean?
22:46.21fedyawhat is merge contexts?
22:46.28[TK]D-Fenderfedya, INCLUDE.
22:46.28fedyaand what conflicts?
22:47.52defsdooryour _X. hits everything - so the following stuff included in ext-did is never reached
22:48.49lelandgHi, I get "configure: error: *** termcap support not found" when I run './configure'; running debian; I know its pretty basic but if someone has a suggestion to get past this... very appreciated
22:48.58fedyanot sure what you mean
22:49.02fedyawhy is it never reached?
22:50.05fedyaah
22:50.10fedyas,1  then s,n
22:51.29*** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
22:51.29defsdoorthis netgear poe switch is louder than my pcs
22:51.56SteveTotarofedya if you use _X. you need to include that INSIDE a context with with things you want to match on first
22:52.02ZX81lol just got a job to do 4400 analogue extensions
22:52.10ZX81with 8 port gateways would be a little insane
22:52.11ZX81:)
22:52.13SteveTotarootherwise it will always hit _X. first
22:52.40ZX81:) ~500 power sockets
22:52.41ZX81mmm
22:53.01SteveTotaroZX81, look at the tenor AX by quintum
22:53.10ZX81sweet, thanks man
22:53.12SteveTotarohighly recommended
22:53.15ZX81cool
22:53.33fedyahm, i changed it to s,n and s,1; it rings now, but doesn't record the CDR() again
22:53.40fedyait's either one thing or the other
22:54.05ZX81that 48?
22:54.29ZX81still 91 power sockets :)
22:54.33SteveTotaroi only used 24s not sure if they have 48 yet, they didn't when i bought
22:54.47SteveTotaroyeah, you have a big install there!
22:54.52ZX81:)
22:55.04ZX81university
22:55.07ZX81funnnnnn
22:55.08ZX81:)
22:55.24SteveTotarothere must be a higher density product out there
22:55.28SteveTotarojust not sure what it is
22:55.38ZX81yeah I was looking at maxtnt and apx8000
22:55.49SteveTotarohow many ports?
22:55.51ZX81but I used the tnt before for a ds3 for a predictive dialing customer
22:55.55ZX81ds3
22:56.01ZX81but high cps killed it
22:56.02fedyahttp://pastebin.com/d34046678
22:56.20SteveTotarocps = calls per second?
22:56.21fedyausing this setup: http://pastebin.com/d488825c3
22:56.23ZX81yeah
22:56.43SteveTotaroi wouldn't expect that to be a problem with a university
22:56.44ZX81the uni would be pretty normal CPS, so I'm kinda looking at it again
22:56.48ZX81exactly
22:56.52SteveTotarohow much $$$
22:57.03ZX81don't remember - customer bought it direct
22:57.12ZX81have put out a quote request
22:57.17ZX81so will have to wait and see
22:57.29ZX81had another install of 600 to satisfy as well
22:57.33SteveTotaroprobably not many of those on ebay ;)
22:57.36ZX81so could be used for that too
22:57.38ZX81:) yeah
22:57.55fedyaso when i use the s it doesn't even execute
22:57.58SteveTotarohow are you getting these large jobs?
22:58.03SteveTotaroif you don't mind me asking
22:58.44SteveTotaroi did a ds3 call center for inbound
22:58.46ZX81:) we got investment from a guy who was working with the incumbent telco for 41 years
22:58.56ZX81was the account manager for all the large accounts
22:59.01SteveTotarooh, it's who you know, should have figured
22:59.08ZX81he left the telco and now works full time
22:59.10ZX81yeah 100%
22:59.16ManxPower(consulting)
22:59.20ZX81:)
22:59.23ZX81cool
22:59.51ZX81http://cgi.ebay.com/Lucent-Max-TNT-for-VOIP-with-spares_W0QQitemZ110215871831QQihZ001QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
22:59.52ZX81:)
22:59.56ZX818 pri
23:00.42lesouvageI'm reading Snom320 issue posts and there seem to be a problem with "engin calls". What is ment with "engin calls"?
23:01.06fiXXXerMet[TK]D-Fender: http://pastebin.com/m445a890b  I have the actual accounts stored in a mysql database
23:01.27ManxPowerZX81: I'm located in the Birmingham, AL area (close to Atlanta, GA),  but I'm also in New Orleans frequently for customers.  I do mostly WAN, LAN (VLAN), and VoIP (Asterisk, Polycom,etc) consulting.
23:01.32SteveTotaromy ds3 implementation was seven servers with quad port sangoma cards running asterisk
23:01.35mostylesouvage, engin is a voip provider
23:01.41SteveTotaroan NFAS
23:01.59ZX81sweet
23:02.01SteveTotaroso seven trunk groups as the telco called them
23:02.11SteveTotaroit worked flawlessly
23:02.11ManxPowerZX81: So if you have need for my skills...
23:02.13lesouvagemosty: thanks
23:02.30jpsharpOr if you need someone *in* Atlanta....
23:02.58ManxPowerjpsharp: I suspect I've been doing this sort of stuff a while longer than most.
23:03.27SteveTotarolonger than me?
23:03.42*** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net)
23:03.47ManxPowerI've been doing the non-VoIP stuff for a little over 10 years.
23:04.00lesouvageManxPower: You were around when I asked my first x100p related question on this forum 4 years ago <;-)
23:04.02ManxPowerthe voip stuff since before Asteris, 0.65.
23:04.10SteveTotarosix years traditional voice
23:04.16jpsharpI've had my hands on * since 2001ish.
23:04.37ManxPowerSteveTotaro: I've only doing voice for 4 - 5 years.  Not sure.
23:04.48jpsharpregular telco stuff since 1995ish.
23:05.08SteveTotaroccna since 96
23:05.24ManxPowerAnd here I thought almost everyone here was a total noob 8-)
23:05.34*** part/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net)
23:05.56SteveTotarowell i see alot of names but only a few speaking up
23:06.07jpsharpWe just like to lurk and glare at the newbies.
23:06.18hmodesmmhmm
23:06.24SteveTotarowe should form a conglomerate
23:06.30SteveTotaroand blanket the US
23:06.56jpsharpThe Asterisk Strike Force
23:07.07SteveTotaroseparate companies under one umbrella company
23:07.20SteveTotaroor at least one name
23:07.38SteveTotarofor marketing and passing leads around
23:08.19jpsharpI'm a little out of practice with bleeding edge *.  I just use it to handle IP phones for our VSAT customers.
23:08.25SteveTotaroi just setup a bunch of regional self storage companies up like that so they can compete with PODS and the like
23:08.58SteveTotaroI don't like bleeding edge * because it is not stable enough for my customers
23:09.00ManxPowerI speak far too much
23:09.19SteveTotarothat is good because i am a man of few words
23:09.24SteveTotaro;)
23:09.53fedyathanks for all your help, i have to pause for today
23:10.26*** join/#asterisk goodmove (n=yves@209.88.71.201)
23:11.06goodmovehello asterisk community
23:14.18cappizsomeone knows of a fax-client that uses SIP/IAX?
23:14.34mostycappiz, fax over voice over ip is stupid and unreliable
23:14.58mvanbaakit's not stupid
23:15.04mvanbaakbut it is unreliable
23:15.10drmessanolol
23:15.10cappizmosty, sometimes i have to
23:15.24drmessanoThats damn well put
23:16.14mostycappiz, why is that?
23:16.25SteveTotaroit can be made reliable but you have to eliminate as much IP as possible
23:16.29goodmoveI am relatively new to asterisk. I have a production system using 1.4.13. Whenever an IP phone develops a problem while a call is in progress, Asterisk does not delete the bridged connection. This connection will remain bridged until a "soft hangup" is issued. The symptom of this problem is that only one half of a subsequent telephone conversation from the problem extension is heard. How to fix this?
23:16.37cappizcause i dont have a real faxmachine
23:16.41mvanbaakSteveTotaro: indeed
23:16.50mvanbaakwe do fax over ip in lan setups
23:16.54mvanbaakno problem there
23:17.06mvanbaakbut you dont want it over the public internet to unkown connections
23:17.51lesouvagemvanbaak: I read your name on the mailing list with the snom320 issue and randomly bridged incoming calls. How often does that happen (every hour, every day, once a month)Edw ooo
23:18.09mvanbaakmore like every minute
23:18.10SteveTotaroi suggest a crossover cable or a T1 channel bank for fax solutions
23:18.21goodmoveHow to fix this problem so that Asterisk will realise that one of the channels is dead and therefore teardown this bridge..?
23:18.28mostycappiz: there are email to fax providers that are pretty cheap
23:18.41SteveTotaroyes, i personally use www.trustfax.com
23:18.58mvanbaakour ITSP's offer fax2mail
23:19.00SteveTotarobut when you are talking about thousands of pages a day, it ain't cheap
23:19.16mvanbaaklesouvage: every minute. We replaced the snom phones with aastra's and all is fine now
23:19.22mostygoodmove, probably worth trying the latest 1.4 version to see if it's been fixed already
23:19.45mvanbaakgoodmove: SIP ?
23:20.13mostymvanbaak, fax over ip is not stupid, fax over voice over ip is
23:21.10cappizmosty, okey... still, i want to check it out :)
23:21.32mvanbaaktry, and be disappointed by it :)
23:21.50goodmovemosty, I noticed that this problem was apparent since version 1.4.9. I also noticed this in version 1.2 as well. I also searched the bugs.digium.com but came up empty.
23:22.13mvanbaakgoodmove: is this with SIP channels ?
23:22.58goodmovemvanbaak, yes with SIP channels bridged to Zap Channels using a Sangoma A400 card
23:23.31mvanbaakgoodmove: try 1.6 beta. It has SIP session timers
23:23.37mvanbaakthat will help with this stuff
23:23.54mvanbaakwe noticed it on 1.0, 1.2 and 1.4 and the new session-timers stuff fixes it
23:24.14SteveTotarobut what else doesn't work in 1.6?
23:24.19SteveTotaroscary
23:24.34mvanbaakSteveTotaro: I run -trunk in production
23:24.48mvanbaakif you follow the commits mailinglist it can be rockstable
23:24.55goodmovemvanbaak, is 1.6beta stable enough to use in my production environment without disruption?
23:25.20mvanbaakgoodmove: depends on the setup. try it in your test setup
23:25.23SteveTotaroi wouldn't bet on it unless you get a call an hour
23:25.26mvanbaakyou do have a test setup right
23:25.46mvanbaakSteveTotaro: we run like 1500 calls an hour
23:25.48SteveTotarorockstable under what load?
23:25.49mvanbaakwithout trouble
23:26.01SteveTotarodoing what?
23:26.04SteveTotarojust a pbx?
23:26.12mvanbaakiax to sip and vice versa
23:26.17goodmovemvanbaak, i will try upgrading a mirror system of my production system tonight
23:26.28SteveTotaroso you just transcode and put packets on the wire
23:26.32mvanbaakwith media handling, queues, mixmonitor, voicemail
23:27.08mvanbaakadaptive odbc for cdr handling
23:27.34mvanbaakwe use the manager interface for originating calls and querying device status
23:27.35SteveTotaroare these real calls or some kind of test calls?
23:27.43mvanbaakSteveTotaro: production
23:27.46mvanbaakreal calls
23:28.00SteveTotarosomehow i think you are exaggerating
23:28.35mostymvanbaak, how many iax/sip clients approximately?
23:29.00mostydefsdoor, FXS? you always need a power connector for that
23:29.02defsdoorunfortunately the internal power connector doesnt reach
23:29.07defsdoormosty: fx0
23:29.27mvanbaakmosty: 6 IAX ITSP's and like 300 sip clients
23:29.32mostyi've never had anyone use that many FXO ports in a PBX :)
23:29.32mvanbaakSteveTotaro: why ?
23:29.39SteveTotarothe power supplies ring voltage to the phone
23:30.05defsdoormosty: got 6 analog lines
23:30.05mostymvanbaak, i have similar numbers of calls, but many more sip/iax clients, and we have big threading issues with 1.4.17 :(
23:30.37mostydefsdoor, that many channels usually means BRI here
23:30.39SteveTotarobecause 1.4 is a disaster and i cannot imagine 1.6 being better than 1.4 at this point
23:30.45SteveTotaroit is BETA
23:30.49mvanbaakso ?
23:31.00defsdoormosty: yeah - but it's not - 6 separate lines
23:31.14defsdoormosty: I'd would have liked to move to PRI but they didn't want to spend
23:31.23mvanbaakI started running trunk at home 1.5 years ago with the skinny phones, and moved our production platform to it 4 months later
23:31.28SteveTotaroyou asked why and I told you, that's why
23:31.34mvanbaak1.4 is not a disaster
23:31.43mostydefsdoor: can you get BRI in the uk? might be cheap enough
23:31.46mvanbaakmaybe 1.4.0 and 5 following releases
23:31.47SteveTotarois ABE using it?
23:32.02mvanbaakbut latest 1.4 versions are fine
23:32.11defsdoormosty: yes - but would need 3 - too expensive
23:32.30SteveTotaroi subscribe to the bug list and my folder fills every day with major bugs
23:33.00SteveTotaroagain, i just don't buy it
23:33.08mvanbaakfine
23:33.12mvanbaakcall me a liar
23:33.34SteveTotaroyou said first that you were just bridging calls
23:33.39mostyi think there's plenty of bugs in 1.4, but not everybody encounters many of them
23:33.40SteveTotaroi believe that
23:34.13SteveTotarothat is my point, if you are just bridging calls then it's probably plenty stable
23:34.26mvanbaakbridging calls is easy, and you dont need asterisk for that. ser does that way better
23:34.43SteveTotaroser does not transcode
23:34.55mvanbaakwho needs transcoding ?
23:35.06mvanbaakI dont
23:35.19mvanbaakall traffic here is alaw
23:35.21SteveTotarodoes ser bridge iax to sip?
23:35.44mvanbaaknope, not that I know
23:36.27SteveTotaropstn connectivity?
23:36.44mvanbaakwe dont have that
23:36.44Shaun2222I'm using dial()+GOSUB and when i set GOSUB_RESULT=CONTINUE for some reason the call is bridged.
23:36.48mvanbaakpure voip setup
23:37.01drmessanoGive your users a Cell/SIP handset and you can force them to GSM.. they wont know
23:37.02mvanbaakwe use ITSP to do the PSTN stuff
23:37.07drmessanoBam... worldwide GSM
23:37.30mvanbaakWe do SIP, IAX2 and Skinny
23:37.32mvanbaakthat's it
23:37.52mvanbaakIAX2 to ITSP and local pbx-en
23:37.59mvanbaakSIP and Skinny for phones
23:38.16mostycan anyone recommend a good way to to iax load testing?
23:38.32mvanbaakmosty: 2 asterisk boxen with loopbacks
23:38.40SteveTotarodoesn't iax2 have a limit of 254 calls?  i thought i read that somewhere
23:38.52mvanbaakSteveTotaro: erm, no
23:38.58mostymvanbaak, i've thought of that, i was hoping for something more lightweight on the client side
23:39.12ManxPowerSteveTotaro: I don't know, we don't use SIP, but I'm pretty sure if it ever was (maybe for an iax2 trunk?) it's been fixed.
23:39.21mostySteveTotaro, we have thread iax issues at around that point
23:39.45ManxPowerlater versions of 1.4 has MASSIVE IAX2 fixes, including threading issues, IIRC
23:39.46SteveTotaroyeah, i thought there was an issue there
23:40.05mvanbaakwe have had issues with 1.2
23:40.19mvanbaakso might be that that was the issue there
23:40.27mostyManxPower, yeah but this issue is with 1.4.17
23:40.38SteveTotaroyes, that is what i thought
23:40.41mvanbaakwe switched from 1.2 to trunk around the time 1.4.7 was released
23:40.53SteveTotaroi believe you are living in a dream world mvanbaak
23:41.16mvanbaakmihau did find and fix a lot of issues in chan_iax2.c
23:41.21SteveTotaromaybe 1,500 one minute or less phone calls an hour
23:41.45goodmoveI am contemplating setting up an IPPBX with two SATA drives using RAID 1(mirroring). I have been searching for an adapter that is compatible with RedHat Enterprise Linux 5.0. Can anyone recommend a suitable product that provides a good performance/price.
23:42.02mostygoodmove, what kind of adapter?
23:42.04mvanbaakSteveTotaro: or we do it the right way
23:42.24SteveTotarook guy
23:42.31goodmovemosty, sata raid adapter
23:42.51mostygoodmove, the best value in terms of performace/price is linux software raid, mdadm
23:42.59SteveTotarowant a mid range server for a good price with a 3yr warranty?
23:43.05*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
23:43.25mvanbaakmosty: get a sun X2200
23:43.40mvanbaakthey are nice and supported with redhat
23:44.16mvanbaakSteveTotaro: I really cant see why you think we are not able to do stuff like this
23:44.25SteveTotarohttp://www.dell.com/content/products/features.aspx/pedge_r200?c=us&cs=04&l=en&s=bsd
23:44.34SteveTotaroi would go for the $799 deal
23:44.42goodmovemosty my experience with software RAID is that the boot partition is not mirrored. It therefore means that should there be a failure on the boot disk, then the system will not boot, even though you may be able to save the data.
23:44.48Shaun2222is this a bug... I'm using dial()+GOSUB and when i set GOSUB_RESULT=CONTINUE for some reason the call is bridged.
23:45.02ManxPowergoodmove: Have you considered asking on the correct channel?
23:45.17ManxPowerShaun2222: using M() option to Dial, I assume.
23:45.19mostygoodmove, i do software raid1 for boot here, works fine
23:45.26mvanbaakgoodmove: the boot partition only needs the kernel. It's easy to copy a couple of files everytime you install a new kernel
23:45.41Shaun2222ManxPower: no that would be a macro... i'm using U()
23:45.48mvanbaakand with newer grub/linux kernel you can even boot from raid1
23:45.54mostygoodmove, any modern linux kernel with initramfs can mount a raided root/boot/anything
23:45.55goodmovemosty are you using RedHat?
23:46.06mostygoodmove, no, debian
23:46.11mvanbaakI dont like redhat
23:46.20ManxPowerShaun2222: You know the issues with M(), I would not be surprised if they happened with U() as well.
23:46.41Shaun2222what issues with M() do these issues exist in trunk/
23:47.11goodmovemvanbaak, Redhat was the first distro I leaned linux therefore I have a sentimental attachment. I am also fairly familiar.
23:47.50mvanbaakguess I was lucky with being introduced to Linux with Debian
23:48.00ManxPowerShaun2222: I don't know about trunk.  There are various issues with M() macros jumping out if it gets DTMF, etc.  I assume they are design decision rather than a bug.
23:48.11mvanbaakas far as I've tested it's the least worst of them all
23:48.14codefreeze~pb
23:48.15jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
23:48.21mvanbaakstill, it is linux so it's flawed by design
23:48.49SteveTotarofreeBSD rulez
23:49.03SteveTotarosolaris as well
23:49.04mvanbaakthat's a lot better then linux indeed
23:49.15*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
23:49.20mvanbaakbut I prefer OpenBSD
23:49.42Shaun2222ManxPower: well i suppose that could make sense since the callee is hitting 3 to send the CONTINUE
23:50.24Shaun2222ManxPower: but actually... i see the dialplan run for the Set()... so it didnt jump out.
23:50.42Shaun2222not unless it decided to run one more exten before it bails
23:51.27ManxPowerShaun2222: what happens if the caller presses a DTMF that you do NOT trap for?
23:51.28*** join/#asterisk MaliutaWrk (n=nikolai@kiev.lusan.id.au)
23:52.11ManxPowerI always liked CPM
23:52.30Shaun2222ManxPower: not sure, right now the caller and callee is me just on seperate phones... the caller isnt hitting any DTMF's while the gosub is running.
23:52.45Shaun2222http://pastebin.ca/869203
23:53.19*** join/#asterisk [hC] (n=hardcore@vpn.voxter.com)
23:54.16Shaun2222also i dont understand this either.... == Spawn extension (app_dial_gosub_virtual_context, s, 1) exited KEEPALIVE on 'SIP/306-ac0331f0'
23:55.37mvanbaakSteveTotaro: did you report your issues under load to the bugtracker ?

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