IRC log for #asterisk on 20070714

00:01.52*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
00:05.24De_Mongrrr, my phpagi is broke. its not reading the phpagi.conf but I duno why :(
00:05.52*** join/#asterisk PhilCiccone (n=pciccone@ip66-104-145-162.z145-104-66.customer.algx.net)
00:07.05PhilCicconeI am in fairly desperate need of help on a new (now production) asterisk server. I have a PRI negotiaton problem that after 6 hours now I cannot solve. Anyone avail who is good in this area?
00:07.14*** join/#asterisk ManxPower (n=manxpowe@015-797-116.area5.spcsdns.net)
00:07.35*** join/#asterisk plantseeker (n=chatzill@host81-154-189-124.range81-154.btcentralplus.com)
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00:12.17snuff-workmm sfb2, you could raise that with devs.. generally i thought it would be more the form of 'sip set debug' and sip set no debug
00:15.26De_MonFRACK
00:15.36De_Monstupid permission errors
00:15.39sfb2I guessed sip unset debug
00:15.53sfb2or sip set debug as a toggle
00:19.26ManxPowersip no debug
00:20.00*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
00:20.01ManxPowerat least for 1.0.x and 1.2.x
00:20.45Phrozen_Onehows everyone doing tonight
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00:24.03andresmujicahi, i'm hackig app_addon_sql_mysql.c so i can pass the port option to mysql_real_connect  but i'm getting problems trying to pass the port value to the MYSQL_exec function.. but it fails......
00:24.58[TK]D-Fenderbcnl, Still need help?
00:28.08*** join/#asterisk Tond (n=t@74.122.241.161)
00:28.08bcnl[TK]D-Fender: no thanks, it got figured out
00:28.29TondHi does anyone know why i am getting this error: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available ?
00:28.50Tondthe Mysql db is setup and all tables created and the config files are done correctly also
00:29.15Tondhowevere asterisk is having problems connecting to MySql for my sip users and peers
00:29.41*** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga)
00:30.28snuff-workyou tried following a guide from like http://voip-info.org
00:30.29snuff-work?
00:30.38TondYes
00:30.50TondI even did ./configure --with-mysqlclient=/usr since mysql was installed using yum
00:31.00Tondin astersik-addons
00:31.02snuff-workyou have unixodbc ?
00:31.15Tondno, MySql is installed on the localbox
00:31.44Tondin /var/lib/mysql/mysql.sock
00:38.07[TK]D-Fenderbcnl, Yeah, you can't just start a patter with a CID restriction on it like that in the middle :)
00:38.20[TK]D-Fenderbcnl, Need to start from priority 1
00:40.08*** part/#asterisk andresmujica (n=andresmu@190.24.227.202)
00:45.25Uatecoh that reminds me
00:45.35Uateci need to get my asterisk box logging to my mssql server
00:50.24Tondwhen i run ./configure in astersik-addons, it says checking for asterisk.h... no
00:50.34Tondhow come it can't find it?
00:51.25Tondi ahve asterisk instaleld
00:56.12[TK]D-FenderTond, and HOW did you install it?
01:01.34Tondtk> i did everything and anything, i did, ./configure, make, make install
01:01.44TondI did just make and make install
01:02.14Tondi even copied the asterisk.h file there since in the ./configure it was saying  asterisk.h ... no
01:02.24Tondi don't knwo what the hell is wrong here...  :S
01:15.51*** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga)
01:24.16*** join/#asterisk ta^3 (n=tacvbo@189.146.184.26)
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01:32.01_DAWis there a way to set sip t1 timer in polycom phones?
01:56.10*** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca)
01:56.24kiwonekagood eveing to all
01:57.43kiwonekais there a public place to get access to the latest software and bootrom for the polycom ip650
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02:08.13*** join/#asterisk Strom_M (n=strom@12.175.45.209)
02:09.31Sci_05there is one site, but it might not be the latest kiwoneka
02:09.56Sci_05they only want people that are register service people to have access to it
02:10.38kiwonekai know, i am trying to get certified
02:10.51Sci_05kiwoneka: check here http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+501
02:11.05Sci_05there is alink on that page to some polycom software
02:11.24Sci_05kiwoneka: I need to get certitifed only its a pain to get done
02:11.47Sci_05of course I don't have a lot of time to follow up and see whats gong on with it either
02:12.00kiwonekathanks
02:12.15kiwonekai ihave about 6 courses done
02:12.46*** join/#asterisk DaPrivateer (i=Privatee@66.92.79.20)
02:13.02Sci_05what one are you going for? I thought you only had to pass 2 tests to get access to the firmware?
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02:21.35kiwonekamore than that
02:21.56kiwonekasome cources have like 5 modules
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02:39.24ManxPowerPolycom has the one version before the most recent available to anyone via their web site
02:39.39ManxPowerIf you want the latest the you have to get it from a Polycom reseller
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03:47.22monstertruckanybody tried a grandstream gxw4008 ?
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03:59.31sweepermonstertruck: run away
04:05.05monstertrucksweeper, from grandstream?
04:05.13sweeperyes
04:05.44monstertrucksweeper, i dont need it for a long lasting application
04:05.52monstertruckis going to be in service for about a year
04:06.19monstertruckim considering it because of ilbc support
04:06.21sweeperhalf that time will be spent gettting it to work imo :v
04:06.43monstertruckreally?
04:07.10monstertruckwhich 8 port gateway would you recommend?
04:07.20sweeperwell, I had a really, really, really bad experience with their FXO dealy
04:08.02monstertrucki havent used any, only experience with GS are the budgetone phones
04:08.17monstertruckhave had 12 for about 5 months
04:08.48monstertrucknot too many problems, they do have issues from time to time and need to reboot them
04:09.20monstertruckbut then again those are at a remote clients site, so rebooting them is the easiest way to fix them
04:10.12monstertrucki would buy an spa 8000 but bandwidth is a problem
04:12.37coppiceall the ills of the VoIP world seem to be blamed on grandstream. :-)
04:13.14Strom_Mnot all of them
04:13.23Strom_Mmerely 99.999% of them
04:13.24monstertruckhaha
04:13.47Strom_Mthe rest can be conveniently blamed on superstring theory
04:14.28Qwell"I get bad audio quality on my SIP calls." "Do you have a grandstream?" "Yes" "Are you using it?" "No" "Is it plugged into the network?" "No" "Is it on?" "No..." "Replace it, and try again"
04:15.54monstertruckthats good to know .. then some of my problems havent been incorrect asumptions put into an agi script .. but instead superstring theory
04:15.59monstertruckim so relieved
04:16.10monstertruck:)
04:17.37monstertruckthats a bummer, given that i probably wont find any other gateway that supports ilbc or gsm
04:18.20Strom_Mwhat about......g.729
04:18.21monstertruckspa3102 supports g726-32, which is ok, but im not sure spa8000 supports it
04:18.23JuggieQwell, why is DIAL allowed from agi, does it actually work properly at all?
04:18.29Qwellno idea
04:18.34monstertrucktrying to stay away from licenses
04:18.35QwellI don't use AGI
04:18.41Strom_Mmonstertruck: it's eight channels
04:18.50Strom_Mand everything speaks it
04:19.00Strom_Malso passthrough
04:19.14Strom_Mbut not iced tea
04:19.28monstertruckdial works very well from AGI
04:19.37QwellStrom_M: You're in the South - it's sweet tea
04:19.50monstertruckmost of my implementation is based on agi, it works well
04:19.51Juggiemonstertruck, dial usually terminates the call when it exits though, does the g option work from agi?
04:20.23Strom_MQwell: cohujibuggle
04:20.27monstertruckJuggie, works exactly as it works from a dialplan
04:20.46monstertruckyou can call macros, transfers .. whatever
04:21.02Juggiemonstertruck, intreasting, i thought i had problems with it before but maybe that was a long long time ago.
04:21.37Juggiethe last thing that really bugged me w/ agi was the default build of php not being able to trap a sighup
04:21.43Juggieso you coudnt cleanup after a hangup
04:22.01Juggienot without starting the agi again in h as deadagi
04:22.13Juggieof course deadagi also works on an active channel for  some reason, but doesnt work well.
04:23.34monstertruckJuggie, yeah, you are right about that, but having a clean up script running with deadagi is not so bad, except for having to start a new process
04:23.40Juggieyeah
04:23.50monstertruckif you are concious about resources
04:24.34Juggiewell, at the time i used php/agi to write ivrs
04:24.42Juggieso it was handy to be able to clean up, write stats etc.
04:24.51Juggieits someone elses project now, and its also in c/fastagi
04:24.55Juggieso thats no longer a problem
04:25.23Juggiefastagi is much nicer, handle multiple calls in a single thread
04:25.42Juggierather then one thread per call.
04:27.28monstertruckyup
04:27.52monstertruckon a different thread, spa8000 is not out yet
04:28.01monstertruckso im out of luck there
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04:52.51nclxCan anyone recommend a good quality VoIP provider in the US which will interface with my existing asterisk system, I am looking for multiple concurrent calls (at least 6)
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05:32.27remmomoan
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05:35.07kingsobmy iax providor says this should be the 1st line in my extensions.conf file for incoming calls.. exten => 6135551212,1,SetAccount(6135551212), but i'm getting this error in my logs...  pbx.c:1797 pbx_extension_helper: No application 'SetAccount' for extension ...
05:36.37kingsobany ideas?
05:39.48kingsobis it even necessary??  if i just rmeove the line it all works great
05:40.59snuff-workmm just use this..
05:41.17snuff-workset(CDR(accountcode)=${EXTEN})
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06:55.21[TK]D-Fenderkingsob, No.
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07:43.56bkruse_home~seen jcmoore
07:44.16jbotjcmoore <n=jcmoore@unaffiliated/tgrman> was last seen on IRC in channel #asterisk, 36d 3h 20m 28s ago, saying: 'GotoIf($[!${ISNULL(${BridgePIN})]?3:5)'.
07:44.16bkruse_homejbot come on!
07:44.17jbotoh alright then..
07:45.33bkruse_homehmm
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12:00.58twodeltahey guys - is this an appropriate channel for discussing branches/patches?
12:02.53twodelta(we're after a little help on how to merge a branch (oej/cancel_elsewhere_1.4) into the current 1.4.7.1 build (as a diff patch? - not very familiar with svn :)
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12:14.51_Raptor_hello
12:16.31_Raptor_can anyone tell me something about this problem:
12:16.41_Raptor_[2007-07-14 14:11:13] WARNING[20955]: chan_misdn.c:2917 misdn_request: Could not create channel on port:1 with extensions:0913191XXXXX
12:16.41_Raptor_[2007-07-14 14:11:13] NOTICE[20955]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'mISDN' (cause 0 - Unknown)
12:17.39*** join/#asterisk k31th (n=keith@cartman.nzsolutions.net)
12:18.41k31thhow can i check to see if a UDP port is forwarded ? I am attempting to forward 5060 and the other sip ports thru a nat router however i suspect its the router at fault here. Is there some way i can check? if this was tcp i would use telnet...
12:18.41Nuggettelnet is eeeeeeevil!
12:21.04fetcherk31th: you might try 'netcat' (nc) to send a test packet, and see if an ICMP Unreachable comes back (may need to run tcpdump at the same time), but that isn't foolproof.  Many routers will silently filter packets without returning an Unreachable
12:21.58k31thdamn
12:22.03fetcherk31th: there's nothing as straightforward as telnet, since UDP has no concept of a "connection" (anything beyond fire-and-forget has to be implemented by the application)
12:22.22k31thyeah its connectionless right
12:22.34k31thsurly udp is old hat?
12:23.38fetcherit has advantages for VoIP and other realtime streaming, where TCP's behavior (designed for bulk data transfer) is less than helpful
12:24.39fetchere.g. if a single packet is lost, TCP will keep retrying it, freezing up the entire connection until the missing one is acknowledged.  Not good for voice, where a brief drop-out would be much preferred
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12:25.55_Raptor_[2007-07-14 14:24:42] WARNING[21119]: misdn_config.c:664 _build_port_config: misdn.conf: "max_incoming=1" (section: default) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload".
12:26.14_Raptor_can anyone explain to me why 1 is out of range for maximum incoming calls?
12:26.24_Raptor_and the same for outgoing
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12:43.24tzafrir_homefetcher, what you say there is not accurate: SIP is only used for signalling. SIP's audio is passed as RTP in a separate connection which is indeed UDP for the reasons you mentioned
12:44.07tzafrir_homeSIP (according to specs) can be sent over TCP. Asterisk doesn't support this yet, though
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13:48.17rob0-- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Unable to do INITIAL ProSLIC powerup on module 0
13:49.03rob0This was working, only thing that has changed: no phone plugged in on FXS module 0.
13:49.21Strom_Mthat shouldn't make a difference
13:49.35rob0perhaps a failed FXS module?
13:51.05rob0hmmm, looking at the config, no, the failed one is the one with the phone.
13:51.14_DAWHell all, any polycom users know if it is possible to adjust the Sip T1 timer?  I am testing one over a satellite and it is an issue.
13:51.38Strom_Mrob0: is that the only fxs module?
13:51.48Strom_M_DAW: how so?
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13:52.19_DAWThe latency of the satellite is causing retransmits cause the timer is only set at 500ms or so.
13:52.32Strom_M"only"??
13:52.43Strom_Mtotal round-trip latency on a phone call should never exceed 400ms
13:53.18_DAWI know, but this is vsat so 500ms is actually a great round trip.  Its usually 6-700.
13:53.33_DAWand actually works pretty well excluding this timer issue.
13:54.53_DAWIve tried several other devices that allow changing the timer and that fixes it right up.  Just could not find it in the polycom.
13:58.08rob02 FXS, 1 FXO, 1 empty slot
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14:12.31Pengguhi all
14:13.08kiwonekagood morning to all
14:13.33Penggucould asterisk be used as a sip client to accept calls from say, another asterisk PBX server, for the purposes of accepting various inputs by callers (including audio) for further processing?
14:13.44Penggueg an IVR, but separate from the geenral PBX server itself
14:13.47Pengguwould be an over kill?
14:14.39Strom_Cyes
14:14.41Strom_Cthat's doable
14:14.44Penggueg. i was thinkig to make a 'bell timer' or 'paging' server (with a sound card), taht will for eg accept calls, take the recoding, and then play it over the sound card (to the PA system), or have ian interactive menu to set up a schedule, etc
14:14.57Strom_Cit's called a feature server
14:15.31Pengguwould asterisk itself be recommended for that, or are there (Easy to use, well documented) programs out there?
14:15.45Strom_Cuse asterisk
14:16.54Pengguthe other alternative i would have is to set up wnidoze xxx and use scheduled tasks (with all the remoteness that comes with it), but asterisk/linux with a web interface would be elegant
14:17.18Pengguand could be squashed into a small footprint
14:17.32Strom_Cyeah, i can't see why you'd want to do that with windows :)
14:18.28PengguStrom_C: u know anything about setting up an embedded device, small scale? eg buy a 'bare box' with a bit og grunt enough to be a server and just chuck on some things?
14:18.54Penggulike, a dsl modem/router without the dsl part
14:19.01Pengguand may be perhaps with a sound card
14:19.04Strom_Cfor your feature server?
14:19.07Pengguyeh
14:19.15Strom_Cwhy must this be a separate device in the first place?
14:19.50Pengguwell, in our case, the pbx is somewhere
14:19.56Pengguand the pa system is somewhere else
14:20.21Penggu(could prolly get a phone with auto-answer and hook up the ear piece to the audio input, hey?)
14:22.20MrMister2I have paging working fine with a extension that dials "console/dsp" but want to be able to control access to it. what would be my options?
14:22.46MrMister2only allow some extensions to dial the paging extension or allow all but ask for a code.
14:22.53Strom_CMrMister2: control access to it just as you control access to any other part of your dialplan
14:22.54MrMister2any ideas on how to do it?
14:23.23MrMister2I'm using Trixbox since it is easier for me but not afraid to mess with the dialplan if I get some hints :)
14:23.34Strom_Coh christ
14:23.38MrMister2LOL
14:23.39Strom_Cyou don't want to mess with that dialplan
14:23.54Strom_Cit'll make your head spin and then you'll mess it up anyway
14:24.03Strom_Cask in #trixbox or learn asterisk :)
14:24.12MrMister2ok. any ideas on how to go abou it then?
14:24.16*** join/#asterisk kickbackit (n=Assimila@24-116-182-58.cpe.cableone.net)
14:24.29Strom_Cstep 1: remove trixbox
14:24.31MrMister2well, I'm trying to learn * since that way I won't be limited by trixbox
14:24.44Strom_Cstep 2: learn how to use contexts
14:24.49Strom_Cstep 3: there is no step 3
14:25.11MrMister2I'm actually going about it the worst way since I'm testing stuff on astlinux :)
14:28.13*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
14:35.02kickbackitMy asterisk box is behind the corporate firewall along with 99% of the phones. I have 4 phones that are remote offices. I have pointed those phones at the firewall server and forwarded UDP ports 5004-5060 and 10000-20000 to the voip server. The external phones can get audio from the voip server, but cannot complete calls or hit keys when the ivr prompts for them. Am I missing a port?
14:35.54Strom_Cif they cannot complete calls, then how are they completing calls to the IVR?
14:36.15kickbackitIf they dial a land line they cannot get audio or send audio to the land line
14:36.38kickbackitI have an extention that sends them to the public ivr
14:36.43kickbackitthey can hear that fine
14:36.45Strom_Ccan they complete calls and talk to sip phones internally?
14:37.00kickbackitalong with anything coming from the voip such as music on hold etc
14:37.23*** join/#asterisk msetim (n=marcos@201-14-60-90.ctame706.dsl.brasiltelecom.net.br)
14:37.37kickbackitI have only tried remote to remote phone since no one is in the office yet
14:38.06Strom_Cgo into the office or wait till there is someone there
14:38.11Strom_Cthen continue asking
14:38.39kickbackitWe have a queue/agent ivr setup, with callback logins. It prompts for a passcode wich the remote phones cannot enter.
14:38.47Strom_Cyes
14:38.47kickbackitthis something with dtmfmode?
14:38.57Strom_Cit might be, or it might be a UDP issue
14:39.21Strom_Cwithout knowing whether they can successfully talk with a human on the inside, it's tough to narrow it down
14:40.00perf3ktis there going to be an update on the book?
14:40.15Strom_Cperf3kt: yes
14:40.17Strom_Cvery soon
14:41.08*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
14:41.09perf3ktcool, will it be free>? of course I don't mind buying I have the first oreily
14:41.27Strom_Cperf3kt: i would assume so, but I don't know offhand
14:41.29Pengguis that the tfot book?
14:41.41Strom_Cdidn't have a chance to ask Leif when I saw him yesterday :)
14:42.06perf3ktpenggu: that's what I was talking about so yeah I hope so
14:42.50Penggu2nd ed?
14:42.56Pengguor vol2?
14:43.22perf3ktso I have a theory question
14:43.31perf3ktthere is * cli - you guys
14:43.31Strom_Cask away
14:43.47perf3ktthen *now with the gui interface
14:44.03perf3ktthen a couple of others freepbx, elastix, trixbox
14:44.09perf3ktwhy?
14:44.40Strom_Cthe others were written before the digium gui
14:44.57perf3ktokay
14:45.08Strom_Cthey're mostly attempts to kludge a GUI onto a GUI-less piece of software
14:45.11perf3ktbut you guys the cli groups still dispise all gui
14:45.23Strom_Cthe digium GUI is a fundamental change in the asterisk architecture
14:45.40perf3ktusers.conf?
14:45.44Strom_Cyes
14:46.06perf3ktsee my months on the boards hav not been in vain
14:46.25Strom_Cheh
14:46.37Strom_Cwell, the digium GUI actually looks promising
14:46.44perf3ktbut the digum gui is the only one liek that
14:46.57Strom_Ci've been playing with the asterisk appliance recently, and it's actually quite usable
14:47.13tzangerStrom_C: howso?
14:47.18tzangerthe gui reads and writes native formats
14:47.27tzangerI I bet
14:47.28perf3ktit just seems that with the wealth of knowledge that the cli group has it would onl p  advance the digium gui
14:47.54perf3ktyeah they have the conf editor, sorts according to the context
14:48.15Strom_Cperf3kt: i'd like to play around with the gui more and see what I can do with it
14:48.20perf3ktbut the groups are so separate
14:48.22Strom_Ci just havent had the time to do so
14:48.29perf3kti understand
14:48.48Strom_Cperf3kt: well also, any GUI is going to limit your flexibility
14:48.59perf3kthonestly the only reason I went to the gui
14:49.19perf3ktis because I'm not a linux guy, and the install with regard to dependencies and all was mind-boggling
14:49.31Infestedheh
14:49.54perf3ktbut on the other hand you say I can pop is a cd that installs * and everything, okay
14:50.14perf3ktbut that is only offered with the gui guys
14:50.37Strom_Cwell, what do you want - flexibility or ease of use?
14:50.38Strom_Cpick one :)
14:51.03Strom_Cflatten the learning curve and you lose flexibility
14:51.07perf3ktwhat loss of flexibility is there?
14:52.10Strom_Cwell, the gui dictates a certain method of doing things
14:52.15*** join/#asterisk friedrich| (n=friedric@e177244140.adsl.alicedsl.de)
14:52.20mvanbaaka gui only allows you to do stuff that is implemented in the gui and it will only allow you to fiddle with it in the way the gui author decided
14:52.51*** join/#asterisk kickbackit (n=Assimila@24-116-182-58.cpe.cableone.net)
14:55.06perf3ktand I'm just asking to gain understanding
14:55.39kickbackitStrom_C, in my situation is there a better method then just forwarding the packets on to the voip? Maybe something like a Sip Proxy?
14:55.55*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
14:55.55*** mode/#asterisk [+o blitzrage] by ChanServ
14:56.03Strom_Ckickbackit: that might be a very good idea
14:56.09Strom_Chi blitzrage
14:56.14blitzrageyo yo
14:56.37blitzrageanyone know if I was right about the 's' priority for bcnl's issue?
14:56.50Strom_Cblitzrage: recap
14:56.52*** join/#asterisk tako-san (n=Tako-san@154.5.212.245)
14:56.55blitzragethat's what the issue looked like, and gave me an idea of how to use the 's' priority
14:57.40blitzragehe was trying to match on a CID at a certain priority, but had 3 different CID matches that he wanted to use at the same priority
14:57.44blitzragepretty sure I was right :)
14:58.46Strom_Cso something like exten => s/3115552368,n,NoOp(catsex) ?
14:59.03blitzrageyes
14:59.07blitzragebut replace 'n' with 's'
14:59.14blitzrage(for 'same')
14:59.55Strom_Coh, i haven't heard about that
15:00.14blitzrageya... I didn't know about that priority until I was writing the SECOND edition of TFoT :)
15:00.25Strom_Chaha
15:00.34Strom_Cwhen are you leaving town?
15:01.29Strom_Ci was thinking of a waffle house run before my flight out of huntsville :)
15:01.35blitzrageI'm already in Toronto :)
15:01.37blitzrageI left yesterday
15:01.42Strom_Chah, ok
15:01.48Strom_Coh that's right, durh
15:01.52blitzragehehehe
15:02.05Strom_Cyou were rushing out of the office at 4ish
15:02.42QwellStrom_C: when you leaving?
15:02.55Strom_CQwell: 2:50 IIRC
15:03.08fileStrom_C: Der Waffle Haus?
15:03.18Strom_CDAS WAFFLE HAUS
15:03.20mvanbaakoh no! ze germanz !
15:03.24Strom_CLA CASA DE WAFFLE
15:03.37Strom_CQwell: 2:49 !!!
15:03.47Qwelloff-by-one error
15:03.53Strom_Cyes
15:04.19mvanbaakI have that with the asterisk count here
15:04.31mvanbaakcame home after helping my lil bro move to his new house
15:04.42mvanbaakto find out on of my asterisk dev boxen has died
15:05.18*** join/#asterisk friedrich| (n=friedric@e177244140.adsl.alicedsl.de)
15:05.27*** join/#asterisk Op3r (n=op3r@121.97.214.210)
15:05.28*** join/#asterisk wunderkin (i=wunderki@ip68-2-61-64.ph.ph.cox.net)
15:05.32mvanbaakthose round big rolls on the motherboard. I have a row of them without roof now
15:06.28Qwellblown caps?
15:07.24perf3ktmust be a dell
15:07.25perf3ktlol
15:08.07Qwellit's not on fire
15:08.10Qwellmust be a gateway
15:09.28mvanbaakQwell: yeah, blown caps
15:09.48mvanbaakhate it when that happens
15:09.54mvanbaakand no, it was not a dell
15:10.06mvanbaakaopen motherboard (intel p4)
15:13.13*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
15:14.58perf3ktis the book dated on ways to obtain the * source code and dependencies?
15:15.09[TK]D-Fenderperf3kt, Yes
15:15.15[TK]D-Fenderperf3kt, www.asterisk.org
15:15.20*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
15:15.26Strom_Cdamnit, I completely forgot to go to chick-fil-a this week
15:15.33[TK]D-Fenderperf3kt, there are readme's in the source tarball as well.  READ them.
15:15.49QwellStrom_C: you have chick-fil-a in CA :p
15:16.02Strom_CQwell: like way the hell out in Ontario
15:16.05Qwellheh
15:16.22Strom_Cit's decent, but not worth driving an hour into the inland empire for
15:16.52Qwellthere's one in Torrence, and Redondo
15:16.57Strom_Coh
15:17.01Strom_Cthat's closer
15:17.04Qwella bit :p
15:17.09Strom_Ci only knew about the one in Ontario
15:17.28QwellCerritos, Long Beach
15:17.31perf3ktanyone use webmin?
15:17.36QwellOrange, Santa Ana
15:17.59[TK]D-Fenderperf3kt, I have and forget it as far as * is concerned
15:18.06Strom_Ci've only eaten at chick-fil-a once
15:18.10Qwellreally?
15:18.12Strom_Cat the philadelphia airport
15:18.12Strom_Cys
15:18.14QwellI love that place, heh
15:18.15[TK]D-Fenderperf3kt, You are going to have to really read and do this yourself.
15:18.22Qwellthere's one about...
15:18.26perf3kttk: just saw it on the ftp
15:18.30Qwell1/2 mile from me
15:18.38perf3kttk: and I have been reading and have been doing it myself
15:18.41Qwellon the corner of the main street
15:18.57QwellStrom_C: I'd recommend Zaxby's
15:19.01perf3kttk: dosesn't help when the book that paid 40 to be a rsource is dated
15:19.06Strom_Cwhat about chick-fil-b
15:19.06Qwellif you like chicken sandwiches and such
15:19.13Strom_Cwhere's zaxby's?
15:19.18blitzrageperf3kt: it IS 2 years old, and a new version is coming out in August
15:19.23Qwellthere's one on University
15:19.26*** join/#asterisk Chris-NB (n=chris@ip.tech.t-mobile.at)
15:19.29Qwellover by the parkway
15:19.43Qwell(and, of course, one on the same corner right here as Chick-fil-a
15:19.48blitzrageperf3kt: and it's not dated for 1.2, which is why the 2nd edition covers 1.4 (1.4 did not exist when we wrote the first edition)
15:19.51[TK]D-Fenderperf3kt, and the book is also available for FREE and yes new versions come out.  You consider * getting better as being BAD?!
15:19.55perf3kttk: and like I was telling the guys earlier its such a turn off when I wanna get into the program, but I'm stopped by installing, compiling, depencies and etc
15:19.58Qwellthere's a starbucks right next to the one on university :p
15:20.23perf3kttk: well sorry I didn't know it was FREE until after I got on here
15:20.28[TK]D-Fenderperf3kt, You should have simply gone to the SOURCE : www.asterisk.org Its all listed there.
15:20.33Strom_CEERF
15:20.38blitzrageFORZA 2!
15:20.41Strom_CFREEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE
15:20.41Qwellthere's also one right by digigraph, near the airport
15:20.44[TK]D-Fenderperf3kt, "Life sucks but rarely swallows"
15:20.51blitzrageI DON'T WANT TO MEET YOUR MOM
15:21.15[TK]D-Fenderblitzrage, I JUST WANT
15:21.15Qwell???
15:21.15blitzrage! ! !
15:21.15Strom_CQwell: you fail
15:21.15Strom_Csorry
15:21.29fileQwell always fails!
15:21.31perf3kttk: I'm not complaing at all, but don't try to treat me like some non reading n00b
15:21.47Strom_Cperf3kt: [TK]D-Fender can be a bit irascible at times
15:21.54[TK]D-Fenderperf3kt, s'ok
15:22.02perf3kttk: no wonder so many people go to the guis
15:22.07blitzrageperf3kt: ya.... gotta get thick skin to hang out in the asterisk community
15:22.11Strom_Cand by "at times" I mean "pretty much 99.999% of the time"
15:22.22[TK]D-FenderStrom_C, So who's playing the "kettle" today? ;)
15:22.33Strom_Cthe other .001% can be conveniently blamed on superstring theory
15:22.58perf3ktstrom: that's cool, I dont' want to be given anythign I wanna earn my stripes
15:23.04Pengguwill all these play nicely together: ast-1.2.21.1, zaptel-1.2.18, ast-addons-1.2.7, libpri 1.2.5 ? <-- i believe these are the latest version from the 1.2x branch..
15:23.44Strom_C[TK]D-Fender: i'm auditioning for the part of "KETTLE" but they may cast me as "POT"
15:23.46[TK]D-FenderPenggu, www.asterisk.org lists all the currect versions right next to each other, go check and you'll know for sure
15:23.48QwellStrom_C: there's also Lenny's, if you like subs
15:24.00QwellLenny's is pretty awesome
15:24.01filemmm Lenny's
15:24.03Strom_Ci think chicken sandwiches have caught my fancy
15:24.11[TK]D-FenderStrom_C, You "POT"?  I dunno... you're way too uptight ;)
15:24.13Strom_Cbut if even file is drooling over lenny's...
15:24.38Penggu[TK]D-Fender: ah yer.. didnt see that list on the RHS..
15:24.39*** join/#asterisk friedrich| (n=friedric@e177244140.adsl.alicedsl.de)
15:24.46Qwellin'n'out > lenny's though
15:24.54Qwellin'n'out > pretty much everything
15:26.19Strom_Cyes
15:26.29Strom_Cthe instant i get back home i'm going to in-n-out
15:26.59QwellI'm gonna tell Kevin that I need to go to Astricon, on grounds that I haven't had in-n-out in 9 months
15:27.14[TK]D-FenderStrom_C, but that'll only take a minute or so ;)
15:27.17Strom_Ci'd approve that
15:27.25Strom_C[TK]D-Fender: ?
15:27.47Strom_C[TK]D-Fender: you do know what in-n-out is, right?
15:27.59fileis there an in-n-out in phx?
15:28.03Qwellfile: yes
15:28.09[TK]D-FenderStrom_C,  perhaps not the one you're alluding to.
15:28.19filewell if that's not a reason to go I don't know what is
15:28.31Qwellfile: exactly
15:28.38QwellI should speak
15:28.53Strom_C"Asterisk asterisk ok let's all go to in-n-out"
15:29.00QwellThey cater
15:29.03Qwellhmm
15:29.04QwellThey cater
15:29.08Strom_Cooh
15:29.09QwellI should tell Sokol :p
15:29.11Strom_CThey cater
15:29.15Strom_C!!!
15:29.18fileThey cater?!?
15:29.24Strom_CThey cater !1111
15:29.32QwellThey have a semi-truck mobile restaurant :)
15:29.37Qwellit's pretty friggen awesome
15:29.47Strom_Calso, this year we should call the convention "Phoneix" just to confuse the dyslexics
15:30.04QwellHooked on Phoneix worked for me
15:30.16*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
15:30.21[TK]D-Fenderhukt on fonix werkt 4 me!
15:31.51Strom_Chttp://multigeeks.com/pics/strom.jpg
15:39.23*** join/#asterisk gardo (n=gardo@121.97.200.254)
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15:44.58asterisknerds<PROTECTED>
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15:45.23[TK]D-Fenderasterisknerds, you don't say....
15:45.27macTijn<PROTECTED>
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15:55.50asterisknerds2<PROTECTED>
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16:05.12Strom_CThey cater
16:07.39*** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com)
16:08.44QwellStrom_C: they totally cater
16:09.10Strom_Cok, seriously pulling myself away from the pc now
16:09.13Qwellhttp://www.in-n-out.com/cookout_trailer.asp
16:09.14Strom_Cshower tiems
16:13.04nclxI have two Snom320 phones registered as SIP clients on my LAN to asterisk, they have usernames of : 710 and 720 respectively.  I haven't altered anything else from the default dialplan aside from sip.conf, I can dial the demo from each phone, should I be able to dial 710 from 720 and ring that phone?  Right now I can't.  I'm just learning though.
16:13.36wunderkinprobably wrong time and day to ask but has anyone here ever compiled asterisk for openwrt or similar? i can't ever get it to find ncurses when it compiles in the editline directory
16:18.22*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
16:23.17[TK]D-Fendernclx, Noy you shouldn't be able to dial them with the"out of the box" sample extensions.conf.
16:23.17Penggunclx: you need to add those numbers to an accessable context in extensions.conf
16:23.33[TK]D-Fendernclx, You need to create extensions to do this
16:24.25Strom_Cwunderkin: i just used openwrt's asterisk package
16:24.35*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
16:25.14wunderkinall i see with openwrt is 1.0.x
16:26.10kickbackitI have been staring at sip proxy's for too long... Anyone have suggestions on one to use?
16:26.36[TK]D-Fenderkickbackit, OpenSER
16:26.49[TK]D-Fenderkickbackit, Do you really need one?
16:26.51Strom_Cwunderkin: i recall that a year ago there was a 1.2.x package
16:27.04Strom_Cso i suppose you're running a really old version of openwrt
16:28.02wunderkinok well there is 1.2.14 in backports... but still anything i find is old.. no its not really old
16:28.18kickbackitWell I wonder if its going to make my situation easier. My asterisk box is behind a firewall, but I have external clients. If I put the proxy on the firewall box I should be able to forward traffic to it without messing around witht he firewall's port forwarding.
16:28.22wunderkinthere has to be something wrong i'm doing compiling it
16:28.43Strom_Cyou're sure you have libncurses5-dev or equivalent installed?
16:29.01wunderkini do but this would be from the toolchain and not the host computer
16:29.28wunderkini even specified the directory to the library in the configure line for it
16:29.29kickbackitI installed openser, but the sit housing the documentation is timing out :S
16:29.46*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
16:31.02[TK]D-Fenderkickbackit, SO not workth it.  Just port forward.  You'll install all sorts of new work and problems for a "zero" problem
16:31.39kickbackitok, I have ports forwarded now, but have issues with the external client being able to dial out
16:31.42*** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-150-67.dsl.ipltin.ameritech.net)
16:31.56[TK]D-Fenderkickbackit, here : ...
16:31.58[TK]D-Fender~sipnat
16:31.58jbot[~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
16:35.00kickbackitthanks.... Looks like this is exactly the situation I have.
16:35.37[TK]D-Fenderkickbackit, I wrote it to cater to pretty much all of the usual circumstances and layed it out so each part of the example was seperate.
16:35.53Strom_Claid out, not layed out ;)
16:36.21[TK]D-FenderStrom_C, http://multigeeks.com/pics/strom.jpg
16:36.35Strom_Ccohujibuggle!
16:37.28Strom_Cmy god, [TK]D-Fender, can't you READ a DICTIONARY?!
16:38.08kickbackitDoes the bindaddress matter in this case? I note the internal and external ip, but no bindaddress in the example
16:38.08[TK]D-FenderStrom_C, Yes, I am your God, bow before me!
16:38.33shido6ZzZz
16:38.46Strom_CGNU/Boring
16:39.34Strom_Ccohujibuggle!
16:40.47*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
16:40.49*** join/#asterisk friedrich| (n=friedric@85.177.244.140)
16:40.49[TK]D-FenderStrom_C, you are indeed "incorrigible" ;)
16:41.20Strom_Ci didn't say incorrigible
16:41.24Strom_CI said cohujibuggle
16:41.41Qwellincohujibugglable
16:41.51[TK]D-Fenderlol
16:42.09Strom_Ck@-h@`-j@-b@g-@l
16:42.57kickbackit[TK]D-Fender, clients on the local lan, even though their external traffice is natted, traffic to the server should be direct. So in their sip.conf section nat=no.... Correct?
16:43.14[TK]D-Fenderkickbackit, local direct to *?
16:44.06kickbackitI have phones that are all behind the firewall, and 4 external phones that need to go through the firewall.
16:44.24kickbackitthe * server is sitting on the same network as the local phones.
16:44.58[TK]D-Fenderkickbackit, Ok, any phone OUTSIDE your LAN, that IS behind its own NAT requires "nat=yes", and "qualify=yes".  Clear?
16:45.11kickbackityep
16:45.14Strom_Cclear as mud
16:45.19kickbackitan bindaddress?
16:45.23kickbackitand*
16:45.31[TK]D-Fenderkickbackit, no, leave bindaddr=0.0.0.0
16:45.51[TK]D-Fenderkickbackit, that will let * talk on any IP that it has available to it
16:45.59kickbackitok
16:46.08kickbackitnow getting an error from a peer
16:46.10kickbackit<PROTECTED>
16:46.38[TK]D-Fenderkickbackit, you have multiple SIP phones behind the SAME remote NAT?
16:47.20kickbackitI have 85 phones on a 192.168.0.x address, the * is 192.168.0.5
16:47.42*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-135.ph.ph.cox.net)
16:47.50[TK]D-Fenderkickbackit, ummm... clarify please
16:48.02kickbackitI changed nat=no in sip.conf and I can call out
16:49.03kickbackitOk my remote phone issues are resolved
16:49.23kickbackitnow I have to drive in to the office and test the local phones
16:49.24[TK]D-Fenderkickbackit, those phones local to * should have "nat=no"
16:49.43kickbackitthat they do, and remote phones nat=yes
16:50.03kickbackitbut when I turned the nat=yes on in the general section I could not talk to my voip provider
16:50.47[TK]D-Fenderkickbackit, you need to set your PROVIDER's entry to "nat=no"
16:51.05[TK]D-Fenderkickbackit, and you should have "nat=yes" under [general]
17:00.02kickbackitAudio quality on the remote phone has improved a ton. Everything seems ok on the remote side of things. Thanks guys.
17:01.29coppicecan you purchase audio quality by the ton? it comes in kilos here
17:01.47kickbackitdepends on the codec
17:02.30coppiceyou mean Alaw would offer kilos of quality, and ulaw would offer tons?
17:03.25kickbackityeah I checked sip show channel, and in the report there I increased the audio quality from 1 ton to 2....
17:03.34kickbackitcodec was ulaw
17:05.33coppicerecent power supplies have lots of SATA power connectors, and not nearly enough good old molex connectors to serve the needs of these analogue telephony cards :-(
17:06.31[TK]D-Fendercoppice, You mean like.. 1?
17:07.28kickbackitthat reminds me, I am supposed to find a card for our * server to allow us to connect a t1. My boss is cheap and does not want to spend money on the ones that digum currently offers. What is an older card that I might pick up on ebay?
17:07.31coppiceif you have only 1 card, then 1 might be enough. still a problem though, since they come in bunches, with shorts leads between each connector
17:08.04[TK]D-Fendercoppice, And extensions/splitters cost like 50$.... oops.... /100
17:08.22[TK]D-Fenderkickbackit, BAD IDEA
17:08.36Strom_C~cheap
17:08.45jbotrumour has it, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
17:08.45[TK]D-Fenderkickbackit, Cheap out and you'll be hoping to win a lifetime supply of KY
17:08.46coppicewell, yeah, except I don't have any. I replaced a dead PSU, and find I don't have enough connectors. this is somewhat annoying me
17:08.47kickbackitagreed
17:08.47[TK]D-Fender~ygwypf
17:08.47jbotwell, ygwypf is You Get What You Pay For.  If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.
17:10.26coppiceygwypf is stupid. you don't get any mor than you pay for is far more accurate
17:10.42coppiceexpect for the "mor" bit :-\
17:11.09Strom_C~cohujibuggle
17:11.10jbothmm... cohujibuggle is gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb
17:11.29*** join/#asterisk mcab (n=mb@mostly-harmless.ca)
17:11.31[TK]D-Fendercoppice, I agree
17:12.34*** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net)
17:12.55[TK]D-Fender~gs
17:12.56jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
17:15.06*** join/#asterisk Strom_M (n=pocketir@m055e36d0.tmodns.net)
17:20.42*** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com)
17:21.00chiardonHello all!!
17:22.20*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
17:22.39Strom_Mhi
17:26.38*** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net)
17:26.55*** join/#asterisk dulzuroso (n=kvirc@eu85-84-137-219.clientes.euskaltel.es)
17:26.58dulzurosohiii
17:27.07dulzurosoanybody know what addon i should use to liseng an inbound call and redirect to the phone that the caller dial in that moment?
17:27.46Strom_M.....
17:27.55Strom_Mi dont understand you
17:28.00dulzurosowell
17:28.04*** join/#asterisk monstertruck (n=monstert@c-76-26-50-233.hsd1.fl.comcast.net)
17:28.10dulzurosoi want to call to a DID that i have
17:28.30monstertruckhey guys, im trying to implement a gateway to gsm
17:28.39dulzurosoand i wish asterisk take that call, and lisend to me to dial a numer where i want to call
17:28.42monstertruckanybody had any experience with PhoneLabs Dock-N-Talk?
17:28.49juuvamonstertruck: don't try, just bye one
17:29.09dulzurosothen i wish asterisk call this numer using a diferent voip acount, and joint it
17:29.16Strom_Mdulzuroso: look at the DISA app
17:29.23dulzurosothanx
17:29.42juuvaI've had nightmares about one voip-gsm -gateway solution we have tried to make work..
17:30.00monstertruckjuuva, using which hardware?
17:30.09dulzurosothere is no disa addons
17:30.10dulzuroso:(
17:30.50monstertruckjuuva, i found this Phonelabs Dock-N-Talk browsing .. it advertises GSM to Wired PSTN
17:31.13monstertruckwhich doesnt seem to be a bad idea, i can use a regular zap channel
17:31.21monstertruckand the DNT does the rest
17:31.35kiwonekaafternoon to all
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17:32.53chiardondulzuroso . . . they tell you to look at the DISA functionality in * to do what you want
17:33.00juuvafor xample eurotech sels sip-gsm -routers which would eliminate need for zap-channel
17:33.18dulzurosoou, oks
17:33.19dulzuroso:D
17:33.27kiwonekaplease please someone help and direct me to a place i can download the new bootrom and sip for the ip650
17:34.01juuvabut I haven't tried those.. and as far as I know , there is grandstream hardware inside + gsm module
17:34.23_DAWkiwoneka: your vendor
17:34.32kiwonekaebay?
17:34.38kiwoneka:)
17:35.09chiardonjuuva... you are right tha gran works so good!
17:35.46juuvasure.. and cows fly
17:36.26chiardonreally they fly???
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17:36.35juuvamaybe some of them ;)
17:36.49[TK]D-Fenderchiardon, on American Airlines, yes ;)
17:37.04chiardonI never seen that!!! so bad to me!!
17:37.10juuvaof course there abe proper gsm-voip gateways in the market, but prices are quite high
17:37.35juuva300+ euros/channel
17:37.52monstertruckjuuva, thats one of the reasons i was looking at the dock and talk
17:38.01monstertruckalthough 300E per channel is not so bad
17:38.19monstertruckthe one i saw was $3000 for 4 channels
17:38.24chiardonhaha thnx Fender but I have a vertigo issue!!
17:38.35juuvawith PSTN you could end up with poor sound quality and failures to detect remote hangup/answer
17:39.16*** join/#asterisk oej (n=olle@apollo.webway.se)
17:39.40blitzrageoej: !!!
17:39.46juuvathose eurotech routers are about 300E/channel.. for 1300E you could get 4 channel VoiSmart PCI-card
17:40.08oejThe one and only!
17:40.11juuvaWould like to test one of those
17:40.13blitzrageschweeeet
17:40.17blitzragehow goes?
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17:45.52monstertruckjuuva, will try one of the wurotech routers
17:45.54monstertruckthanks
17:46.26seele_hello, how can I balance the calls between two PRI (60 lines)
17:47.43[TK]D-Fenderseele_, on outbound you can look at channel occupancy, on inbound, you can't
17:48.29juuvamonstertruck: just make sure that you can return it if you don't like it..
17:49.08monstertruckyeah, thats why im buying one, if it works, then ill get a multichannel router
17:49.15seele_[TK]D-Fender, yes for outbound, i need to make the same number of calls for both PRI channels channels
17:49.51[TK]D-Fenderseele_, then before choosing which PRI to use go check how many channels are in use on each.
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18:35.48yakkophi, anyone know if its possible to have the Dial app play a sound file instead of a ring?  I see the music on hold option but that doesn't let you specify an exact sound.
18:38.30Juggiemake a music on hold class with only one sound
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18:40.04yakkopJuggie: is it possible to do that dynamicly? i.e. without restarting
18:43.21Juggieedit musiconhold.conf and reload moh yes
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18:45.50yakkopthanks
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18:50.37kiwonekamay someone direct me to a place to find the latest software for the ip 650
18:50.45Strom_Mkiwoneka: your reseller
18:50.58kiwonekano
18:51.13Strom_Myes
18:51.17kiwonekajust someone that has a bunch of phone that need
18:51.52Strom_Myour sense makes sentence not !!!
18:52.07wunderkin!1111
18:52.16kiwonekai have been at this for a while
18:52.48Strom_Mkiwoneka: contact the company which sold you the phone(s) and request the firmware
18:52.59Strom_Mor you can find the second-latest firmware on polycom's site
18:53.08kiwonekai bought them on ebay
18:53.49kiwonekathe one on the site does not work
18:53.55kiwonekai have been there
18:54.12Strom_Mthe one on polycom.com?
18:54.12kiwonekabut thanks for you assistance
18:54.18kiwonekayes
18:54.21Strom_Mthat's odd
18:54.30Strom_Mbecause i've used that exact version
18:54.36Strom_Mand it worked for me a hundred times
18:54.44kiwonekaat this link http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip650.html
18:54.59*** join/#asterisk RomDump (i=romdump@faeroes.freeshell.ORG)
18:55.10kiwonekacould be how i am downloading the zip file
18:55.11*** join/#asterisk oej (n=olle@apollo.webway.se)
18:55.33kiwonekai am on my mac
18:55.45Strom_Muse wget on your asterisk server :)
18:55.53kiwonekaok
18:55.57kiwonekai have tried that too
18:55.57_DAWr u click the link for 2.0.3 Rev B?
18:56.03kiwonekayes
18:56.21Strom_Mhow exactly does it "not work"?
18:56.34kiwonekait will not extract the zip
18:56.42*** join/#asterisk oej_ (n=olle@apollo.webway.se)
18:57.15Strom_Mjust for shits and giggles, let me try
18:57.33kiwonekayour a riot!
18:57.46kiwonekals
18:57.59Strom_MERROR: THIS IS IRC
18:58.35Strom_M-rwxr-xr-x omgwtfbbq.tar.gz
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18:59.15kiwoneka[root@tunes config]# unzip spip_ssip_sip_2_0_3b_sig.zip
18:59.15kiwonekaArchive:  spip_ssip_sip_2_0_3b_sig.zip
18:59.15kiwonekafile #1:  bad zipfile offset (local header sig):  0
18:59.16kiwoneka<PROTECTED>
18:59.16wunderkinmm bbq
18:59.52Strom_Mwell, hmm, it does the same on mine
19:01.23kiwonekaomg, your kidding
19:04.18*** join/#asterisk tako-san (n=Tako-san@154.5.212.245)
19:04.20_Raptor_P[ 1] stack_init: Cannot add layer 4 to this port. << mISDN gurus please tell me whats wrong
19:04.40Strom_Myour/you're homophone error
19:04.46*** join/#asterisk oej_ (n=olle@apollo.webway.se)
19:05.31kiwonekaso, you have a working sip that you can share Strom_M
19:05.49Strom_Mif you promise to learn the difference between "your" and "you're"
19:06.11*** join/#asterisk oej (n=olle@apollo.webway.se)
19:06.11kiwonekasure
19:06.33kiwonekayou're the one to teach
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19:20.53rob0ok, need to figure out my zap trouble ... http://pastebin.slackadelic.com/289
19:22.30Strom_Mrob0: did you swap the modules like I suggested earlier?
19:23.30rob0Oh I missed that. Put them where?
19:23.36Strom_Mswitch their places
19:23.50rob0hmmm ok, I'll try
19:23.52*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
19:23.53Strom_Mso put the one FXS where the other is, and vice versa
19:24.06*** join/#asterisk oej (n=olle@apollo.webway.se)
19:24.09rob0what about the empty module 2 slot?
19:24.15Strom_Mif the problem follows the FXS module, then it is can be RMA time
19:24.23rob0ah
19:24.35*** part/#asterisk griels (i=user@60.67-246-213.ippool.namesco.net)
19:24.43rob0ok, will have to shut down for this, bbiab
19:24.53Strom_Myou're IRCing from your asterisk box?
19:24.57rob0nono
19:25.16rob0but it's my Internet connection to the place where I'm in irssi.
19:25.29*** join/#asterisk oej (n=olle@apollo.webway.se)
19:25.30kiwonekaStrom_M: you find that sip
19:25.40*** join/#asterisk cayorde (n=flexable@host245-111-dynamic.17-87-r.retail.telecomitalia.it)
19:25.47Strom_Mkiwoneka: I already sent you a privmsg
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20:05.13rob0Strom_M was right, RMA time. :(
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21:07.30*** join/#asterisk supjigator (n=sysgod@152.53.16.10)
21:08.00supjigatorAnyone mind helping me debug a IAX2 connection between two machine?  I have it working in on direction but no the other.
21:09.06sfb1how doesn't it work in the other direction?
21:09.47supjigatorIt won't qualify and looks like an auth problem.
21:10.01supjigatorI see POKE PONG INVAL
21:11.07supjigatorI have a Box A which has two interfaces. One public and one private. Phones are on private and I'm using IAX to a public server I have setup with PRI PSTN connection.
21:11.07supjigatorBoxB can send calls to BoxA.  BoxA cannont send any calls to Box B
21:11.07kiwonekagood evening to all does any one have a working zip file of  http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip650.html sip 2.0.3 revb?
21:11.51kiwonekathey zip off of polycom does not work
21:13.15sfb1is the public server box b?
21:14.05kiwonekayes
21:14.09supjigatorYes.
21:14.40supjigatorA also has a public address as well. Two nic cards.
21:15.02sfb1so the poke pong invalid is showing on box b?
21:16.04supjigatorI see the error on both sides but it looks like mirror
21:16.44sfb1is boxA firewalled on the public interface? any firewall log messages happening?
21:17.39supjigatorIt would appear that iax2 debug is telling me that A to b is failing and b to a is succeding becuase there are ACK's as well just not on box A.  Always POKE PONG INVAL on box a and iax2 show peers shows unreachable
21:18.04supjigatorON box B iax2 show peers shows reachable and can make calls to box A
21:18.43sfb1oops, thinking the wrong way around then - is B firewalled?
21:19.00supjigatorA iptables has defaults to accept
21:19.29supjigatorB as well.
21:20.24supjigatorIt looks like they are able to talk.  Does INVAL mean packets aren't making it to box b?
21:20.30supjigatorCause it looks like they are.
21:21.19supjigatorUm can't be a firewall issue cause B and A are recieving IAX packets from each other.
21:21.57supjigatorI've spent days on this including upgrading the older box to the same version
21:22.17sfb1http://tools.ietf.org/id/draft-guy-iax-03.txt suggests a POKE message must be replied to with a PONG
21:22.37supjigatorYep I see POKE and then a PONG then an INVAL
21:22.56sfb1but it doesn't suggest that PONG can be invalid - it's just like an acknowledgement
21:23.14supjigatorOn a working side I see POKE PONG ACK
21:23.31supjigatorso the RFC doesn't really tell us much as to what the actual problem is.
21:33.15justdaveare there any known issues with having a TE120P and a TDM400P in the same machine?
21:33.27justdavegetting lots of "NMI for unknown reason XX on CPU X" messages after booting the server with both in it
21:38.47justdaveok, just booted with only the TDM and they didn't go away.  must be RAM or something :(
21:42.17*** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus)
21:44.10sfb1supjigator: well, if it's not firewall or routing (both are talking and poke/pong are being transferred), and not asterisk itself as it's apparently working on both
21:44.58sfb1have you set verbose logging?
21:45.14sfb1for more detailed debugs (I think that's an option)
21:48.52justdavebooted without the TDM and it booted fine.  doesn't like the TDM card :(
21:49.16sfb1what's a TDM card?
21:49.32justdavehook up POTS phone lines or phones to the server
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21:51.23sfb1ah
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22:00.12sfb1my sip trunk establishes the connection, but wont pass any audio.
22:00.22sfb1I bet it's a NAT issue. :/
22:07.05*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
22:07.50hi365how can i get the voice mail time stamp to play in 24hr format? do i have to add the option to every line in voicemail.conf?
22:10.45snuff-workshould be a 'default' near the top of the voicemail.conf..
22:10.51snuff-workjust change that one
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22:13.51Stc884hello
22:14.00Stc884italiani cè qualcuno ?
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22:17.42tzafrir_homeStc884, try English, I guess
22:18.29Stc884no problem tzafrir_home thanks
22:18.47tzafrir_homeStc884, there's #asterisk-it , but it is empty right now
22:19.01Stc884yes, thanks
22:19.35tzafrir_homejustdave, it should work. Just make sure that the modules load in the right order
22:20.51justdavewhat order should they be?
22:21.10justdaveI saw a post on a forum saying the TDM cards don't like 64 bit slots
22:21.21justdavethe new machine is all 64 bit slots
22:21.28justdaveit was indeed in a 32-bit slot on the old machine
22:21.32[TK]D-Fenderjustdave, guess you should have looked first
22:22.13justdavehow would I have known to look for that? :)  it's an odd thing
22:22.44justdaveswapping machines was a convenience to keep the downtime minimal (we were staging the upgrade on the new box, then swapping everything at the maintenance window)
22:22.55justdavealready blown that screwing with the new box now. :)
22:23.04justdavegonna keep the old box and just swap the hard drives now
22:23.24[TK]D-Fenderjustdave, Sure, There's a new E10 gas out... go see if your old deisel VW Rabbit will run on it...
22:23.38justdaveheh
22:23.49[TK]D-FenderWhen the slot changes... find an IQ
22:24.30*** join/#asterisk Ebola (n=Ebola@host86-143-168-67.range86-143.btcentralplus.com)
22:25.56Stc884whereis asteriskNow livecd ?
22:27.03wunderkinjustdave, that can't be true
22:27.30justdavesounded funny to me, too, but it does fit the situation
22:28.01justdavethe two machines are supposedly the same model, but apparently the replacement one is a newer revision and the bus doesn't quite match
22:28.39justdaveboth machines are about the same age, theoretically, we thought we were recycling an old box :)
22:29.03justdavethey're HP DL385s
22:29.58[TK]D-FenderStc884, http://www.asterisknow.org/
22:30.15[TK]D-FenderStc884, And this is NOT AsteriskNOW's support channel.
22:30.23[TK]D-FenderStc884, please read the channel topic
22:33.23wunderkinjustdave, more like a motherboard incompatibility probably, contact digium support
22:40.27hi365tzafrir_home ping
22:41.56hi365snuff-work: got it. 1 question: does the Q option always default to ABdY or can it play just the yesterday/today file if apllicable?
22:42.51snuff-worknot sure.. i haven't played too much with it :)
22:42.59hi365thansk
22:43.20hi365in the custome time zones of voicemail.conf, does the Q option always default to ABdY or can it play just the yesterday/today file if apllicable?
22:45.19*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
22:46.02snuff-worki'm sure it does play the today/yesterday when applicable
22:47.13*** join/#asterisk [ViAjErO] (i=ViAjErO@201.103.53.73)
22:48.28tzafrir_homehi365, pong
22:49.23[ViAjErO]someone has analog lines to PSTN doing outgoing calls with SIP ? i have incoming calls to my sip Xlite from pstn .. but i can't make calls to PSTN trought Xlite :(
22:49.31hi365tzafrir_home: shavuah tov! i noticed that you had a couple of patches lined up for asterisk to enable "proper" hebrew numbering. what is the status of hebrew numbers as of now?
22:50.10tzafrir_homeNirS promismed me he had good patches. But then disappeared.
22:50.27[TK]D-Fender[ViAjErO], that makes NO sense.  Please be specific about the hardware and service providers involved.
22:50.45tzafrir_homeI'll guess I'll take a look at say.c again
22:50.50hi365snuff-work: it sure does, but you cant have a custom string defined and use Q/q PLUS what ever else you want. apperently Q defaults to ABdY
22:51.21tzafrir_homehi365, is this in say.c or in the voicemail?
22:51.25hi365tzafrir_home: im sorry, my memmory must of faild (it sd-ram...) i though they where your patches
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22:51.53tzafrir_homethey weren't good enough
22:52.04hi365tzafrir_home: currently im working with voicemail. it actualy seems ok to me (IF you have the correct recordings)
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22:54.34tzafrir_homerob0, isn't morse code good enough?
22:54.50rob0hmmm smoke signals!
22:55.06rob0Does * support smoke signals?
22:55.13[ViAjErO][TK]D-Fender: I have one TDM22B 2 FXS and 2 FXO, I have my FXS ports connected to PSTN, I have Xlite SIP client on my computer, right now i'm receiving calls from FXS-PSTN lines to my Xlite client (SIP extension) . But when I want to make an outgoing call dialing 9+8numbres as DialRule establish it rings in analog extension instead to call out PSTN
22:56.00tzafrir_home[ViAjErO], can you make calls to analog phones connected to the FXS ports?
22:56.07sweeperrob0: sure,  just hook it up to a webcam and a usb fog machine
22:56.18[TK]D-Fender[ViAjErO], go to www.pastebin.ca and paste your zapata.conf and your relevent sections of extensions.conf
22:56.32tzafrir_homehmmm.... FXS ports connected to the PSTN?
22:56.48tzafrir_homecan you accept incoming calls?
22:56.48[ViAjErO]yes ...
22:57.04rob0Dial DLV-MCCXII
22:57.07[ViAjErO]yep
22:57.17[ViAjErO]i have made som calls from public lines ...
22:57.26[ViAjErO]then asterisk answers
22:57.35kusznirHi all:  I've got an asterisk system set up for my home.  Incoming DID calls are directed toward a few extensions in the house.  From my perspective, things work fine.  However, the caller is presented with silence until someone answers or voicemail takes it.  Config: http://www.pastebin.ca/619679
22:57.36crayz_is there any way to add/remove calls from a "queue" which simply holds calls in asterisk without trying to connect them? I want to use AGI/AMI to place calls into such a queue and then use AMI to connect them to an operator. I don't want to be tied into the asterisk distribution strategies in the queue though
22:57.45[ViAjErO]i dial extension and my Xlite rings and stablish the call
22:57.56tzafrir_homerob0, apart from XXX, what is this good for?
22:58.20rob0toga parties?
22:58.57[TK]D-Fendertzafrir : he's obviously poorly grouped his channels
22:59.08[ViAjErO]hmmm
22:59.16[ViAjErO]ports are inverted ?
22:59.20[TK]D-Fender[ViAjErO], please provider the pastebin requested
22:59.31[ViAjErO]ok thank you
22:59.33[TK]D-Fender[ViAjErO], No, you have grouped your FXS & FXO togther wrong
22:59.42[TK]D-Fender[ViAjErO], in your configs
23:00.05[ViAjErO]i see in the card tdm22b this ... 1,4 are FXS 2,3 are FXO
23:00.22[ViAjErO]I think so
23:00.54[ViAjErO]j1,j4 are fxs right? and j2,j3 are fxo
23:00.55rob0The card is physically Green-Red-Red-Green ?
23:01.40[ViAjErO]i'm going to pastebin
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23:07.23[ViAjErO]pastebin 1 -> zapata.conf http://www.pastebin.ca/619687
23:07.39[ViAjErO]pastebin 2 -> extensions.conf http://www.pastebin.ca/619690
23:09.18[TK]D-Fender[ViAjErO], You are using the GUI and I do not see enough to tell what lines(s) need ot be fixed
23:09.43[ViAjErO]then ?
23:09.54[ViAjErO]what do I need ?
23:09.58[ViAjErO]ok
23:10.05[ViAjErO]i'll review the groping
23:10.07[ViAjErO]:)
23:10.10[ViAjErO]thank you
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23:13.09[TK]D-Fender[ViAjErO], Right no I belive you have set all your channels to be in 1 group.  That is what is mixing up your channel selection
23:19.13kiwonekagood evening to all does any one have a working zip file of  http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip650.html sip 2.0.3 revb?
23:19.19kusznirIn the dialplan, does one need to do anything special after Answer to provide ringing to callers?
23:19.26kiwonekathe zip off of polycom does not work
23:19.52_DAWSpeaking of polycom, is it possible to change the sip timers?
23:21.34Sci_05_DAW: what do you mean sip timers?
23:22.03_DAWthe Sip T1 timer.
23:22.39Sci_05_DAW: what is it used for?
23:23.46kiwonekai am kinda stuck on this one
23:24.12kiwonekai just got a bunch of 650s and i need to upgrade
23:24.32_DAWSci_05: T1 timer is round trip time estimate.  I am using satellite so the 500ms default is too low.
23:24.43kiwonekaany one have a copy of 2.0.3 revb that works?
23:27.15Sci_05hmmm thats one I am not sure about, did you look thru the provisioning scripts? Are you using a provising script?
23:31.31kiwonekathanks wunderkin
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23:32.29kiwonekaawsome, its comin wunderkin
23:32.35kiwonekai really appreciate it
23:36.08_DAWSci_05:  I really dont think they support it.  Surprising since it is not an uncommon option on other phones.
23:36.50Sci_05what phone is it? 501, 301?
23:37.27crayz_if I set MusicOnHold and then answer a call in AGI(without dialing through to an operator), will the music play for the call or is there something special I need to do to start the music?
23:37.54[TK]D-FenderSci_05, Didn't you see him repeat himself like a broken record a doze times like the rest of us?
23:38.25Sci_05na its the weekend, I don't look at irc that much on the weekend....lol
23:38.40[TK]D-Fendercrayz_, No, MoH will not play jsut because you answered.
23:39.00crayz_how would I start it playing from AGI?
23:40.28[TK]D-Fendercrayz_, There is no practical way to leave background music while doing other things.
23:40.43[TK]D-Fendercrayz_, dialplan processing is entirely linear.
23:42.12crayz_could I just play the music indefinitely and use AMI to interrupt & connect a call? I'm somewhat basing this on instructions from here:
23:42.13crayz_http://www.orderlyq.com/asteriskqueues.html?winId=6382#orderlyq
23:42.41crayz_which seem to indicate it's possible to setup MOH, transfer the call to AGI, and then later connect the call to an operator
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23:48.07[TK]D-Fendercrayz_, What are you doing in this AGI?
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23:50.44crayz_basically I want to be able to use a very customized call queuing system where a call comes into AGI, checks for operators in a database & connects if available, and otherwise queues the calls up and connects them to operators based on rules in the AGI script. also needs ability to "dequeue" calls and push them to an operator via AMI
23:52.02crayz_from what I've read about the built-in queuing features in asterisk they're not going to work, or would be a huge pain to make work.... all the other dialplan decisions are in AGI, but doing any kind of queuing in AGI seems impossible or at least undocumented
23:53.27crayz_ideally AGI would just shove the calls into a queue where they'd hear music, and then the background AMI script would decide which calls to connect to which operators - but I can't see any way of doing this either
23:53.59[TK]D-Fendercrayz_, Oh boy..... this is looking like you should make your own Queue app
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23:55.37crayz_yea, there's about zero chance of me being able to write this in C.... that's why I was hoping to use some combination of AGI/AMI to do it :/
23:57.09[TK]D-Fendercrayz_, Sounds like an ugly hack (more like large series of...)  if possible
23:57.21Juggiecrayz_, write what
23:58.07[TK]D-Fender<crayz_> basically I want to be able to use a very customized call queuing system where a call comes into AGI, checks for operators in a database & connects if available, and otherwise queues the calls up and connects them to operators based on rules in the AGI script. also needs ability to "dequeue" calls and push them to an operator via AMI
23:58.07[TK]D-Fender<crayz_> from what I've read about the built-in queuing features in asterisk they're not going to work, or would be a huge pain to make work.... all the other dialplan decisions are in AGI, but doing any kind of queuing in AGI seems impossible or at least undocumented
23:58.07[TK]D-Fender<crayz_> ideally AGI would just shove the calls into a queue where they'd hear music, and then the background AMI script would decide which calls to connect to which operators - but I can't see any way of doing this either
23:58.48crayz_the one thing I was looking at is WaitMusicOnHold indefinitely and then using some AMI command to steal the call and ring to an operator

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