IRC log for #asterisk on 20070626

00:00.06Qwell[]kombi: in the US
00:00.21dlucasit does not seem to affect operation (that I can tell) ...
00:00.22[TK]D-Fenderanonymouz666, If you want my help, don't put up road-blocks :)
00:00.23kombithere's one there, only a copy..
00:00.29kombi;)
00:00.29Qwell[]KSYR
00:00.36[TK]D-FenderSOSE?
00:00.37Qwell[]Where is that? :p
00:00.41kombiKayser!
00:00.54dlucasbut it may be indictive of an underlying problem inherent in the system I run that was not picked up with earlier verions?
00:01.05Qwell[]NY
00:01.18*** join/#asterisk dijungal (n=kdaniel@64.86.52.254)
00:01.25kombidlucas: maybe not because I had the same thing and heard of others
00:01.26dlucaskombi: running anything live?
00:01.38dlucasoh ok ...
00:02.01kombiQwell: well aware.. somewhere on the subway map I recall
00:02.05*** join/#asterisk dlynes_office (n=dlynes@d154-5-108-123.bchsia.telus.net)
00:02.26dijungalguys i get this error on my incoming h.323 call to asterisk, and i get NO AUDIO on the call also " RTP Transmission error of packet 43236 to [public ip removed]:16406: Invalid argument"
00:02.26Qwell[]kombi: yeah, I didn't know what state it was in :P
00:02.34dijungalany ideas what's going?
00:02.47kombiQwell: I like nassau there too..
00:02.57*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
00:03.29[TK]D-FenderI have family in Nassau.... I'm going to schedule a trip this fall....
00:03.46kombithe one in NY or Bahamas?
00:04.02[TK]D-Fenderkombi, BAHAMAS.... Oh Dear God.. you were talking about NY?
00:04.14kombisorry...;))
00:04.17dijungali guess not
00:05.22Strom_Mthey make quite spectacular messes, you know
00:05.31kombiyou are.. gross
00:05.44JTeh
00:05.48Strom_Mhey, you live in los angeles long enough and you'll see these things :)
00:05.48JTlife is gross
00:06.03*** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net)
00:06.58anonymouz666[TK]D-Fender: http://pastebin.ca/589446
00:07.08kombiwhat was the command again to trace segfaults and the like?
00:07.21kombistrace..
00:09.21kombiI am crashing.. have a pint/bud on my behalf!
00:09.39[TK]D-Fenderanonymouz666,  exten => 10,n,Mysql(Fetch fetchid ${resultid} var_row1 var_row2 var_row3) ; this line must be populated as much lines count returns   <---- No, you need as many vars as you have COLUMNS to retrieve, not ROWS
00:10.13*** join/#asterisk lindex (n=dex@66-162-134-242.static.twtelecom.net)
00:10.27lindexAnyone got a couple of minutes to help me out?
00:10.34[TK]D-Fender~ask
00:10.35jbotQuestions 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.
00:10.35lymeca[TK]D-Fender: And if I want to connect my VoIP softphone to the PSTN?
00:10.52[TK]D-Fenderlymeca, THEn maybe you'll have a reason to use *.
00:10.57lymecaCan I use Asterisk to make that connection for free instead of paying a company for a service?
00:11.02lindexWhat are some common causes to iax2 peer being unreachable
00:11.05lindexother than network issues
00:11.07[TK]D-Fenderlymeca, Because I'm pretty sure Ekiga won't do it by ITSELF
00:11.14[TK]D-Fenderlymeca, Sure
00:11.14lymecaNo it won't.
00:11.20anonymouz666[TK]D-Fender so it must be fixed length. but I need to loop the rows
00:11.51*** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com)
00:11.53[TK]D-Fenderanonymouz666, Yes you do, and EVERYTIME you do a fetch, a variable gets set to tell you if there is MORE to retrieve <-
00:11.59lindexi.e. iax2_poke_noanswer: Peer '<blah>' is now UNREACHABLE
00:12.02[TK]D-FenderMYSQL(Fetch fetchid ${resultid} var1 var2 ... varN)
00:12.03[TK]D-Fender<PROTECTED>
00:12.03[TK]D-Fender<PROTECTED>
00:12.03[TK]D-Fender<PROTECTED>
00:12.13lymecaOhh, okay so my server needs hardware that plugs into the PSTN and then I can run Asterisk on it as my PBX... would I connect Ekiga to my Asterisk machine which could then route the call to the PSTN?
00:12.40[TK]D-Fenderlymeca, Congratulations you have pieced EVERYTHING together. Far more than the average newb here.
00:12.41Jingleslindex: well, really, there's no *other* reason that an IAX2 peer is unreachable on your network  ** except ** for network reasons. ;)
00:12.57lymecaWell I'm definately a newbie...
00:13.07*** part/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net)
00:13.09lindexJingles: well see that's the strange part, machines are on the same switch/same network/ can ping each other etc
00:13.14Jinglesright.
00:13.17Jinglesdoesn't matter.
00:13.19[TK]D-Fenderlymeca, clearly one with an IQ and a concept of what plays what role
00:13.44Jinglesbecause every now and again, you'll have some kind of network chatter that causes peers (SIP or IAX) to become unreachable.
00:13.45lymecaWhat about the phone number?  Wouldn't I have to pay some company ultimately for a PSTN number for inbound calls to my softphone (again via Asterisk)?
00:14.00[TK]D-Fenderlymeca, Your remote party's Ekiga can connect to your * and lets you talk for free and if you want they may be allowed to access your pstn as well.
00:14.00lymeca;-]
00:14.05Jinglesnow - that having been said - I find it ONLY happens on networks with windows boxes on them.
00:14.13lindexJingles: what kind of chatter?
00:14.17[TK]D-Fenderlymeca, Well That depends HOW you want to get to the PSTN.
00:14.37Jinglesbecause I built a pure linux network with sip phones hanging off of two * servers trunked together (sip to sip) and never got 'unreachable'
00:14.43[TK]D-Fenderlymeca, If you have an existing phone line, you buy hardware to let * use that.  or you can sign up with an ITSP, etc.
00:14.44lymecaWell I mean say someone has a PSTN #... a normal cell phone for example.  I want them to be able to call me on my SIP softphone.  Where would I get that number from?
00:14.50lindexJingles: this is a server to server peer
00:14.53lindexnot phones
00:15.02obnauticus[TK]D-Fender with php AGI applications, can I have festibal say output through AGI?
00:15.09lymecaActually my server is at a college and I'm sure they have spare numbers to give me...
00:15.09obnauticusfestival*
00:15.10Jingleslindex: for example, windows boxes spam the network all the time with requests for the master browser, etc.
00:15.18[TK]D-Fenderobnauticus, don't see why not.
00:15.20Jinglesprinters, too - advertising themselves like cheap hookers.
00:15.22obnauticuswell umm
00:15.24obnauticusi was thinking
00:15.32obnauticushow would you get the shell commands to execute in the same shell as *
00:15.45lymecaSo when I get my number from my college I can somehow bind it to my Asterisk server, correct?
00:15.50Jingleslindex: actually, the phones and servers actually act about the same way as far as 'responsiveness and forgiveness for network latency'
00:16.10lymecaSo the thing that will be associated with that PSTN number is Asterisk, and definately not my softphone...?
00:16.19[TK]D-Fenderlymeca, depends HOW you get this "number" from your college.  You'll have to be more specific about what tech that is and how its delivered
00:16.33[TK]D-Fenderlymeca, Correct
00:17.11[TK]D-Fenderlymeca, * will talk to your PSTN connection by whatever means necessary, and Ekiga will talk to *.  Your * configuration will determine what you dial to get to the pstn.
00:17.34Jingleslindex: so, my advice? put a packet sniffer on the wire with your systems - and watch to see what's going on when your peers report that they timed out.
00:17.49*** join/#asterisk Greenbox (n=Brett@c-68-59-20-153.hsd1.sc.comcast.net)
00:17.49*** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
00:18.28lindexHmm
00:18.38*** join/#asterisk anthm_mobile (i=anthm@dhcp64-134-34-247.bwic.chi.wayport.net)
00:18.41lymecaYou say the * configuration will determine what I dial to get to the PSTN... so within Ekiga for example I have to do something different with the number/URI I am calling to let * know that the destination is on the PSTN?
00:18.46[TK]D-FenderJingles, timing out means you aren't GETTING any packets.... so you keep on spying on packets that jsut AREN'T HAPPENING and let us know how that turns out for you, ok? ;)
00:19.42[TK]D-Fenderlymeca, not with the URI so much.  You would dial a numbre like "100" and * can be told to dial your OTHER Ekiga soft-phone if its connected.  or if you dial a number that matches a PATTERN, to send it out your PSTN connection instead, etc.
00:20.02[TK]D-Fenderlymeca, * can be used to create a FULL PBX, including voicemail, etc.
00:20.09lymecaI'm not sure on the technology my school uses for doling out phone numbers, but I'm going to assume I can get one for my * server: when this happens, and others on the PSTN call my number, what will * do?  Obviously that's probably configurable but can I tell it to route the incoming call to my softphone and ONLY my softphone with an optional voice mailbox?
00:20.10[TK]D-Fenderlymeca, How much you want to implement is up to you.
00:20.39[TK]D-Fenderlymeca, You can make * direct calls from ANY device you connect to it in ay way you please.
00:20.49[TK]D-Fenderlymeca, Mine used to make me COFFEE.
00:21.09lymeca"any device you connect it to" includes SIP softphones, PSTN numbers, and other * servers?
00:21.25[TK]D-Fenderlymeca, So yes you can have it talk to your PSTN through whatever means necessary, ring your softphone, let people leave VM's if you're not around, etc.
00:21.32[TK]D-Fenderlymeca, yes
00:21.52lymecaAnd... what about ZRTP?
00:21.55*** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
00:21.57[TK]D-Fenderlymeca, but don't think of them as "PSTN numbers"
00:22.10[TK]D-FenderZRTP is not supported by * I'm sure.
00:22.21lymecahttp://en.wikipedia.org/wiki/ZRTP#Implementation
00:22.32lymeca"Ekiga, Twinkle and Asterisk also implement ZRTP and are available under GPL."
00:22.58[TK]D-Fenderlymeca, For PSTN you can either use an ITSP, or buy a hardware card to use a PHYSICAL line you may have.
00:22.58fileit is not in tree at least, there's a bug open for adding ZRTP but have seen nothing yet
00:22.58[TK]D-Fenderlymeca, True, THEY support it, but ASTERISK does not.
00:23.15lymecaBut... Asterisk is in that list...?
00:23.19[TK]D-Fenderfile, a bug open for it?  We don't have it yet, and ALREADY its a problem? ;)
00:23.24coppicebut it must support it. it says so on Wikipedia
00:24.04lymecaHaha coppice
00:24.04lymecaYes it could be that Wikipedia is wrong.
00:24.09lymecaOkay so no ZRTP support...
00:24.20lymecaMaybe I should edit Wikipedia
00:25.33[TK]D-Fenderlymeca, Everything you've asked is perfectly normal.
00:25.45lymecaOh that's good...
00:26.12*** join/#asterisk Hogie (i=blehblah@cpe-76-186-105-60.tx.res.rr.com)
00:28.04lymecaI have enough information now to know that I want to use this for sure.  I probably will be able to set up something that will be useful for my purposes that requires only the initial hardware investment for my server...
00:28.19lymecaBy the way how much do those cards usually cost?  They're PCI right?
00:29.46[TK]D-Fenderlymeca, There are other options as well.
00:30.03[TK]D-Fenderlymeca, SIP gateways that will talk SIP to your * box as well, etc.
00:30.27[TK]D-Fenderlymeca, Typically softphones SUCK, and we advise actual HARDWARE
00:31.04coppicetypically hardware phones suck too, so we suggest popping over for a chat.
00:31.46[TK]D-Fendercoppice, just have to shit on ALL of telecom, don't you? :)
00:32.13[TK]D-Fenderlymeca, Do you have an ANALOG LINE that you might like to let * use?
00:32.55lymecaPotentially yes.  I'm not 100% and would need to ask but I think that would be possible.  I'd need some sort of special expansion card that would add RJ-11 jacks to my server, right?
00:33.40*** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-6b3bc4357a088bd7)
00:33.46[TK]D-Fenderlymeca, if that is the case I would suggest the following unit : http://www.telephonydepot.com/product_p/105-054-312.htm
00:34.17[TK]D-Fenderlymeca, this will let * use your analog line, AND let you use a regular analog phone as a SIP phone instead of having to use Ekiga at all.
00:35.19lymecaI think I value ZRTP encryption more than the reliability brought upon by using a hardware analog phone.
00:35.50lymecaBut this unit... I mean my server that I'd run * on is no where near where I'd have a phone.  It's in a server room stowed away...
00:35.52[TK]D-Fenderlymeca, not a question of reliability (and the SPA's support encryption IIRC).
00:36.07lymecaI need this unit next to my * server, correct?
00:36.27lymecaThe picture is bad... can't see what ports it has.
00:36.31[TK]D-Fenderlymeca, And as * DOESN'T I guess you'll have to look to something else ENTIRELY.  Perhaps CISCO is up your alley.  I hope you have government funding....
00:37.02[TK]D-Fenderlymeca, 2 RJ11 (1 FXS, 1 FXO), 2 ethernet (WAN , LAN)
00:37.56lymecaAye... so the analog line into the FXS RJ-11 and a CAT-6 cable running from the LAN RJ-45 to my server's NIC?
00:38.21[TK]D-Fenderlymeca, if you want it wired direct on not on a general LAN with it, sure
00:38.51lymecaOh I could plug the RJ-45 LAN port into the same switch as my server is on...
00:39.02lymecaHow would that be beneficial?
00:39.14lymecaMultiple * servers on the LAN could use that same analog line?
00:39.40[TK]D-Fenderlymeca, Well you wouldn't need 2 NIC's in your server
00:39.57lymecaRight...
00:39.59*** join/#asterisk Op3r (n=op3r@121.97.208.81)
00:40.05lymecaBut what if I already do?
00:40.10[TK]D-Fenderlymeca, And you could remote admin if from a machine OTHER than your server
00:40.22[TK]D-Fenderlymeca, THEN WHATEVER!  Do the math! :p
00:40.23lymecaAny machine on that LAN then?
00:40.28JTa dedicated link to your ATA is generally overkill
00:40.36Op3rhello have anyone had this error?  [chan_zap.so]Jun 25 19:40:13 WARNING[5447]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory
00:40.47[TK]D-Fenderlymeca, its a web administered SIp gateway.  This is not rocket science.
00:41.10[TK]D-Fenderlymeca, If its only directly connected to your server then logic states you're doing that to ISOLATE it.
00:41.26[TK]D-Fenderlymeca, That means that you could only configure it via your server.
00:41.52[TK]D-FenderOp3r, You MIGHT want to actually compile libpri BEFORE Zaptel and Asterisk so that it WORKS
00:41.57*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
00:42.25lymecaYeah that would make sense for me I think... I mean the only person using this analog telephone line would be me and I am only going to be using it via my * server, and I have 2 NICs
00:42.25Op3r[TK]D-Fender, I thought u have to compile zaptel before libpr?
00:42.29[TK]D-Fenderanonymouz666, How's that coming along?
00:42.48[TK]D-FenderOp3r, No, it is DEPENDENT on libpri.  Read your instructions again.
00:43.03[TK]D-FenderOp3r, Zaptel USES libpri, not the other way around.
00:43.13Op3rerrr :(
00:43.17JT[TK]D-Fender: actually, i've been told ast depends on libpri and zaptel, and libpri and zaptel do not depend on each other
00:43.30JTbut both zap and libpri must be done before ast
00:43.50lymecaOnce I set up *, how would I tell Ekiga to useit?  BY configuring an SIP outbound proxy?
00:44.05[TK]D-FenderJT : You can't use libpri without zaptel, but you can use zaptel without libpri.  Math FTW!
00:44.10Op3rbeen installing asterisk and I always compile zaptel before libpri and never had any problem with me except this time
00:44.14[TK]D-Fenderlymeca, Yes
00:44.27Op3rok
00:44.30[TK]D-FenderOp3r, redo all3
00:44.33Op3rim getting this error now  [chan_zap.so]Jun 25 19:40:13 WARNING[5447]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory
00:44.36lymecaSo all my calls even to other SIP softphone users would first go through my *
00:44.36Op3rerr
00:44.36Op3rsorry
00:44.53Op3r[TK]D-Fender, while compiling im getting this error make: warning:  Clock skew detected.  Your build may be incomplete.
00:45.02lymecaSo who would the other person see is calling?  What address would they see the call coming from?
00:45.14[TK]D-Fenderlymeca, Correct and based on what you dial can be told to call other sip devices, etc.
00:45.14flenderslymeca: ever used asterisk?
00:45.25lymecaflenders: No
00:45.39flenderslymeca: even installed asterisk?
00:45.47[TK]D-Fenderlymeca, the IP of your * server (which in their eyes is probably the SAME), and whatever callerid you desire.
00:45.50lymecaHmm... not yet.
00:45.53[TK]D-Fenderflenders, of course not!
00:45.58[TK]D-Fenderflenders, DUH!
00:46.03lymeca0_o
00:46.20lymecaI would configure the callerid in *?
00:46.23flendersI don't know why people come here before even reading docs and trying to install it themselves
00:46.31[TK]D-Fenderlymeca, You know... you should just stop.  Right now.  Download *.  Download THE BOOK.  and get your hands DIRTY.
00:46.37lymecakk
00:46.48[TK]D-Fenderflenders, Ah, lay off him, at least his isn't a nit-wit....
00:46.50[TK]D-Fender~book
00:48.16jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:48.17[TK]D-Fender...
00:48.17flenderslymeca: I went through the same process a few months ago, and it isn't that hard
00:48.17lymecaflenders: I needed to make sure * could do what I wanted.  I had no idea truly what its capabilities were or how I could leverage its power to facilitate encrypted SIP-to-SIP communications and SIP<-->PSTN routing.  I've never worked with any PBX before.
00:48.17[TK]D-Fenderlymeca,  here : http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:48.17Op3rasteriskguru.com ?
00:48.17[TK]D-Fenderlymeca, Go read the book, go DL * and PLAY WITH IT.
00:48.17lymecaI have no problem reading a shitload now that I know I want this though.
00:48.29flenderslymeca: fair enough... you'll find all that on the first chapter of the book
00:48.45lymecaUnfortunately I can't play with the SIP<-->PSTN functionailty because I don't have a number or the necessary hardware.
00:48.54[TK]D-Fenderlymeca, Scratch encryption off your mind for the time being, the rest is good.
00:48.59Op3rflenders, have you experienced this error while doing make? make: warning:  Clock skew detected.  Your build may be incomplete.
00:49.18[TK]D-Fenderlymeca, there isn't mucht o imaginfor what * can do with the PSTN, you jsut need to see if you WANT what it offers.
00:49.52flendersOp3r: no.
00:49.53lymeca[TK]D-Fender: I really *really* want encryption though.  Specifically ZRTP.  How would that work with * if ZRTP is only for SIP-SIP communications?
00:50.07lymeca(I'm almost to the point where I'll shut up)
00:50.08[TK]D-Fenderlymeca, there is NO point in * then
00:50.21*** join/#asterisk guillote_GNU (n=guillote@host193.200-82-106.telecom.net.ar)
00:50.32[TK]D-Fenderlymeca, because * can't do ANYTHING with your calls.  no voicemail, NOTHING.  It cannot do that encryption.
00:50.41[TK]D-Fenderlymeca, Do without, or move on.
00:51.04Op3rflenders, any ideas? because I think thats the reason why asterisk wont start or wont even compile properly :(
00:51.10lymecaYou said earlier it could do voicemail...
00:51.29flenderslymeca: it can
00:51.36flendersit can't do encryption, though
00:51.50lymecadern
00:51.55flendersOp3r: sorry, no idea.
00:51.59Op3rok
00:52.19flendersOp3r: what distro are you running that on?
00:52.29Op3rflenders, centos 64bit
00:53.29Op3ri actually installed 4 VICIDIAL/asterisk today and they are working but this 2 servers arent because of that error :(
00:53.31flendersI don't know what the requirements are on centos, but if you can't compile, you're probably missing something
00:53.35[TK]D-Fenderlymeca, if asterisk can't do your encryption.. hten YOU CAN'T USE VOICEMAIL!
00:53.55[TK]D-Fenderlymeca, How can * talk to tyou .... if IT CAN'T TALK WITH THE ENCRYPTION?
00:54.05[TK]D-Fenderlymeca, My chicken just ate your egg.
00:54.19Op3r[TK]D-Fender, use pgp? ok lame joke
00:54.55*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
00:54.59flenderson softphones you could use VPN
00:55.02flenders:D
00:55.24flendersI use openvpn with x-lite and it works fine.
00:55.51flendersthough, only connections between phone<->* will be encrypted...
00:56.14[hC]do most of you guys run your ip phones (especially in customer networks) on a separate vlan from data when sharing one ethernet cable for phone and data?
00:57.04saftsacki have a maybe more philosophic question ... are numbers such as sip:max.example@domain.org the numbers for the futurue?
00:57.07techie/dns op3r
00:58.20JTlymeca: the only encryption i know built into asterisk is encrypted IAX2
00:58.23JTand it's beta
00:58.36JTotherwise it's a UDP based VPN
00:59.11*** join/#asterisk troy- (n=troy@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com)
00:59.15Op3rflenders, [TK]D-Fender : I found the solution. I was using ntpdate to correct and sync the time of my servers. I removed ntpdate on rc.local and it works now :D
01:00.01*** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net)
01:02.14*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)
01:03.23flendersOp3r: are you running this on a virtual machine?
01:04.13Op3rflenders, nope im running it on a dedicated server from layeredtech
01:04.56flendersI've seen something similar on a virtual machine once. and it was on debian
01:05.41*** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk)
01:05.50Op3rI was thinking its the battery on the board or something
01:06.03Op3rbecause the clock is lets say messed up or something
01:06.27KDanUsing php-agi... with the server being called through IAX... how do I find out what the number is that was used to dial in? (there's multiple numbers mapped to the same dialplan)
01:06.53KDan(is there a way? or is that conceptually not possible)
01:10.00KDananyone?
01:11.29*** join/#asterisk anthm_mobile (i=anthm@dhcp64-134-34-247.bwic.chi.wayport.net)
01:11.33Strom_MKDan: depends if the provider sends you DNIS
01:11.47*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
01:11.55KDanI'm using a DIDX number..
01:12.08KDanhmm
01:12.23*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)
01:12.28KDani could always map each number to a different username, and each username to a different dialplan, and each dialplan to a different php script
01:12.54KDanbut it would be an understatement to say that that's not exactly ideal in terms of scaling
01:13.45[TK]D-FenderKDan, the EXTEN the call lands on my be a CLUE <--------------
01:14.15KDanhow so?
01:14.26*** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net)
01:14.36flendersI have a question, when people reach our main IVR, they can dial 9 to get hold of someone and 9 rings all phones, through Dial(SIP/01&SIP/02&...), however, if I enable call forwarding on my handset to forward calls to my mobile phone, not just the direct calls to my extension will be forwarded, but also the ones where someone dialed 9. any ideas?
01:14.53[TK]D-FenderKDan, What do you think the call lands on?  the targeted NUMBEr
01:15.30KDanatm it lands on guest@myserver.tld - which maps to a [guest] entry in iax.conf... which maps to the [iax] entry in extensions.conf
01:15.36KDanall that's under that entry is:
01:15.42KDan[iax]
01:15.42KDanexten => s,1,Answer()
01:15.42KDanexten => s,2,Playback(silence/1)
01:15.42KDanexten => s,3,AGI(record-simple.php)
01:16.00KDan(and then hangup, and a t,hangup as well)
01:16.13JTyou're matching with s.
01:16.20JTdon't do that if you want to be specific.
01:16.21KDanare you saying that i could replace the s with the specific number?
01:16.45KDanwell, i don't - i just want to know what number was used, when i'm in the php script
01:17.03KDan(i mean, i don't want to be specific)
01:17.04*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
01:17.16JT${EXTEN}
01:19.24*** join/#asterisk Exstatica (i=exstatic@redline.mednor.net)
01:19.39KDanok... worth a try!
01:19.48*** join/#asterisk paolob (n=donpaolo@196.3.84.214)
01:19.50KDan$agi->get_variable("exten") I think
01:19.55*** part/#asterisk paolob (n=donpaolo@196.3.84.214)
01:24.11KDanhaha... gutted
01:24.25*** join/#asterisk catch23 (n=catch23@69.60.124.109)
01:24.44JT?
01:24.47KDan$ext = $agi->get_variable("EXTEN"); log($ext);   >>> $ext = "s"...
01:25.15KDani mean log($ext['data']);
01:25.24JTso, you're gutted?
01:25.29KDanyep
01:26.20KDanotoh on the way i found this page: http://www.voip-info.org/wiki-Asterisk+variables ... which is kinda useful
01:26.41JTthere's README.variables in the asterisk source
01:26.55JTgutted means unhappy?
01:27.25KDansomething like that
01:27.45KDandifficult to translate to american
01:28.04KDanbut imagine you're a fish, and you've just tried to do something, and instead you find that you've been gutted
01:28.26Strom_Mi think by that point you'd be dead
01:28.27KDanit's like "oh crap.. didn't see that one coming"
01:28.30Strom_Mso what's the point?
01:28.34*** join/#asterisk jmacz (n=jmacz@190.24.97.36)
01:28.38KDannot if you're a magical thinking fish
01:29.09JTamerican? as if that's a language :P
01:29.21KDanit's not english ;-)
01:29.29KDanwe have to call it something else to differentiate..
01:29.31KDan:-)
01:29.34JTi have heard the term "gutted" used, it just didn't make much sense in this context
01:29.37KDan(no offence to americans)
01:29.38JTi don't call it american
01:29.44JTerr
01:29.45JTi mean
01:29.50JTi don't speak american
01:29.53JTweird typo
01:30.09Qwell[]JT: You speak Australian, right?
01:30.10JTamerican english is awful
01:30.13JTno, english
01:30.26Qwell[]Your english is pretty good for being from oz
01:30.31JTaustralian english is basically a copy of uk english, with a few extra words
01:30.40Qwell[]many extra Us?
01:30.48JTeh?
01:30.51Qwell[]I mean, we know they love them some Us :p
01:31.31KDanJT> australian english is basically a copy of uk english, with a few extra words << Mainly to do with beer, cricket and rugby
01:31.34KDan;-)
01:31.36JTheh
01:31.52JTso i guess you were gutted for feeling like a newbie then?
01:31.53JT:)
01:32.35KDanI was gutted that i thought "AHA! This is the answer I was looking for" - and then it gave me the same very exact but completely useless answer i had before
01:32.55JTi see
01:33.26KDanbah. {DNID} is blank and {CALLERID(all)} says "Guest IAX User"... not my lucky day...
01:34.44lindexhow do I reload my sip.conf ?
01:34.51Op3rreload?
01:34.57lindexjust sip?
01:35.01Op3rlindex, type reload on asterisk cli?
01:35.02Greenboxlindex, sip reload
01:35.06*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
01:35.08Op3roh just sip
01:35.16lindexperfect
01:35.17lindexthanks
01:35.24Qwell[]Strom_M: It's all your fault.
01:35.40*** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima)
01:36.19catch23are there any incoming DID providers that have rates cheaper than $4/month for just personal use?
01:36.34catch23but not free like ipkall
01:36.41Qwell[]catch23: depends on how much you use it.  Some charge a per minute fee
01:36.42KDancatch23: didx.net provides 2 free did's
01:37.14troy-is there any reason that asterisk would drop inbound/outbound calls at random intervals infrequently?
01:37.44catch23didx.net allows incoming calls to be on lines without CID?  ipkall doesn't allow incoming calls from unknown sources...  (since they make no dough from it)
01:37.47troy-i have an implementation where people tell me calls are dropped, but it doesn't happen often and when it does after different periods of time
01:38.34KDanhmm, dunno
01:38.39KDanyou could try calling my US number
01:38.46KDanand see if it lets you
01:38.48dijungal" RTP Transmission error of packet 43236 to 208.45.138.4:16406: Invalid argument" --- any ideas?
01:39.29catch23KDan: didx.net looks like wholesale, you think they'd just sell one DID number to me?  :P
01:39.34Strom_MQwell[]: of course it is
01:39.37Strom_Mit always is
01:39.46Qwell[]admission is the first step...
01:40.19KDancatch23: they sell any number, from the looks of it. I've had just the 2 free numbers for about 9 months now
01:40.27Qwell[]http://www.delta.com/flifo/servlet/DeltaFlifo?flight_number=4728&flight_date=Today&request=main
01:40.28KDanwill probably buy more, but so far haven't paid a penny
01:40.32Qwell[]^^ I'm pissed - very much so
01:40.50Qwell[]note that it's currently 9:40 in Atlanta
01:40.54troy-dix gives out free numbers?
01:40.57troy-*didx
01:41.07KDanyes. at least they did last summer!
01:41.34catch23KDan: good find...  i hope they don't start cancelling the incoming numbers though.  Seems like the 2 free numbers is part of their promotion type stuff
01:41.43KDanyeah
01:42.07catch23troy-: ipkall gives away free numbers too...  but there's a catch
01:43.00troy-ah
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01:43.10KDananyway... time for bed...
01:43.12Op3ru just have to open anonymous sip calling
01:43.13KDangood night people
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01:47.59catch23KDan: damn, it does look like they (Didx.net) have a catch too...  after "testing" the free numbers, you have to sign some agreement saying you'll always have 20 DID numbers purchased at all times or be charged $20 monthly
01:49.41anonymouz666is there a way to use arrays without AEL on dialplan?
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01:56.52*** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
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02:01.40Qwell[]anonymouz666: no, but BLAH[0] is a perfectly valid variable name, so you can easily fake it
02:02.13Qwell[]there is also the hash table stuff, but I don't really know how that works
02:04.11dijungalanyone has any idea what's causing my RTP erros?
02:17.23*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
02:18.40Exstaticawhat are the different database get commands?
02:23.18*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
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02:37.45*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net)
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02:46.28Exstaticaanyone running asterisk 1.2?
02:46.44Exstaticai need to see if you can run a cli command and paste me the output
02:46.50Exstaticato see if it changed from 1.2 to 1.4
02:47.12BSD_Techasterisk -r will tell you
02:47.23Exstaticawill tell me what?
02:48.16BSD_Techalot changed and not just waht to see in the show apps
02:48.37BSD_Techalot changed on the backend
02:49.19BSD_Techlike on 1.4 user exten moved from sip.conf and aix.conf to users.conf
02:49.28BSD_Techso did provider trunks
02:49.29Exstaticai need someone to run "database get CF xxx" and replace xxx with a valid extension
02:49.35BSD_Techloads of new switches
02:49.43Exstaticadoes it return 0 or does it return text?
02:49.48Exstaticai don't care what else changed
02:49.58Exstaticai need to see if a specific thing changed
02:50.22Exstaticain 1.4 it shows text
02:50.34BSD_Techwell if you where more spacific we could tell you
02:50.48Exstaticaum if you read what i said i think i was
02:50.52BSD_TechI have never seen show text
02:51.08Exstaticahttp://www.freepbx.org/trac/ticket/2021
02:51.10Exstaticaer
02:51.13BSD_TechI see where you asked if anyone was running 1.2
02:51.14Exstatica[19:49:28] < Exstatica> i need someone to run "database get CF xxx" and replace xxx with a valid extension
02:51.30Exstatica[19:49:42] < Exstatica> does it return 0 or does it return text?
02:51.32*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
02:51.37FuriousGeorgehey all
02:52.25FuriousGeorgefor some reason, when i use a programmed park-orbit button, versus just transferring to the park extension, i start getting "maximum retries exceeded" errors
02:52.41FuriousGeorgeim pretty sure that lead to some strange deadlock-like behavior today
02:53.00BSD_Techparkorbit never heard of that in asterisk
02:53.13BSD_Techyou sure its not a custom feature
02:53.40BSD_Techwhat gui you using
02:54.45FuriousGeorgegui?
02:54.53FuriousGeorgetheres no gui involved
02:55.05FuriousGeorgepark-orbit is a feature of the phone
02:55.19BSD_Techthen its not a asterisk feature
02:55.28FuriousGeorgenever said it was
02:55.31BSD_Techparking is but not park-orbit
02:57.05*** join/#asterisk InHisName (n=Administ@c-71-225-221-149.hsd1.pa.comcast.net)
02:57.23FuriousGeorgeparkorbit just parks the active call in the first spot.  if you are using local hints ala 1.4+ you dont need to listen for what spot the system picked, so you dont need to atxfer to the park extension
02:57.31FuriousGeorgeworks well, except i think its locking my server
02:57.37FuriousGeorgeguess ill make a bug report
02:57.58*** join/#asterisk switch (n=switch@saya.attrition.jp)
02:58.13BSD_TechExstatica, gateway2*CLI> database get CF 300
02:58.13BSD_TechValue: 6000I>
02:58.25BSD_Techthats in a 1.2.19 box
02:58.57BSD_Techparking works fine
02:59.22BSD_TechI have 1.4.5 and parking  at 700 parks it in the frist open slot fine
02:59.47BSD_Techand it calls you back and announces the spot
02:59.52BSD_Techjust fine
02:59.59BSD_Techthere is no bug
03:00.24Exstaticado you have cf enabled on 300?
03:00.25FuriousGeorgeat what point did i say parking wasnt working
03:00.33FuriousGeorgeeither you understand what im talking about or you dont
03:00.34BSD_TechExstatica, yes
03:00.37*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
03:00.45FuriousGeorgethere is no inbetween
03:00.49Exstaticacan you do it on one that is not enabled?
03:01.15BSD_Techgateway2*CLI> database get CF 1000
03:01.15BSD_TechDatabase entry not found.
03:03.13BSD_TechFuriousGeorge, I use 1.4 on a few server and being parked orbit is a function of your phone and not asterisk its not a asterisk bug
03:03.21FaithfulHey guys is there an app to let my dial numbers from webpages?
03:03.29FaithfulThat would be nice.
03:03.30*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
03:03.47BSD_Techthere is a page to have users have you call them
03:03.53*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:03.56BSD_Techits called call me
03:04.00BSD_Techits php
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03:06.13FuriousGeorgeBSD_Tech: what is this recurring theme about "a feature of asterisk"?  all parkorbit does is a blind transfer to the park extension.  so which part of that doesnt asterisk support?  the blind transfer, or the park extension
03:06.50BSD_Techif the function is in your phone its your phone thats the issue
03:07.06*** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com)
03:07.14BSD_Techif your disabling the phone function and using the asterisk function then it would be a asterisk issue
03:09.00FuriousGeorgeBSD_Tech: so, following my logic i'd say that its likely that anyone who blind transfers to the park extension might get this issue.  now that asterisk supports dev states for local channels, i bet there are a few of us.  so your position is that devs wont want to know about that?
03:10.03*** join/#asterisk ZX81 (n=matt@202.20.97.200)
03:11.08FuriousGeorgeand even if it doesnt work like i think it does the fact remains, i hit park and a call "somehow gets transferred" to a local extension defined in asterisk.  sounds like asterisk is at least trying to support it
03:11.21BSD_Techyou have to program it in your dial plan to do the channel status checking. and the fact stand that when the funstions are built into the phone per say dnd
03:11.37BSD_Techusing polycom to demostrate
03:11.45BSD_Techyou hit dnd on the phone
03:11.54BSD_Techasterisk does not know you have dnd on
03:11.58FuriousGeorgesure it does
03:12.10FuriousGeorgeits defined in the sip rfc
03:12.28BSD_Techand it will ring the phone and then after the ring time goto vm insted of the way asterisk does it if dnd is set via the asterisk dial plan
03:12.55BSD_TechI am polycom certified as of 2.1.1 the function did not work right
03:13.00FuriousGeorge-- Got SIP response 480 "Do Not Disturb" back from
03:13.02FuriousGeorgeblah
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03:13.09BSD_Techits supopost to be fixed in 2.1.2
03:13.10FuriousGeorge(23:12:04) BSD_Tech: asterisk does not know you have dnd on
03:13.16BSD_Technope
03:13.43BSD_Techif you hit dnd it only disables the ringer on all lines
03:13.59FuriousGeorge<PROTECTED>
03:13.59FuriousGeorge<PROTECTED>
03:13.59FuriousGeorge<PROTECTED>
03:13.59FuriousGeorge<PROTECTED>
03:13.59FuriousGeorge<PROTECTED>
03:14.03BSD_Techit does not tell asterisk to set dnd/1
03:14.24JTaron? what a weird name :P
03:14.33BSD_Techwhat firm war eyou have on your poly
03:14.37FuriousGeorgeyeah, usually its 2 a
03:14.49JTindeed
03:14.58BSD_Techit broke in 2.1.9
03:15.04wunderkinit would go to voicemail if it were the only person, not continue trying to ring
03:15.05BSD_Tech2.1.0
03:15.07FuriousGeorgei dont use polycom
03:15.24BSD_Techok I am using polycomes to point my fact
03:15.43FuriousGeorgewunderkin: i know, im just pointing out that asterisk does infact know the difference between congestion, busy, and do not disturb
03:15.51BSD_Techwhen a function is built into a phone asterisk does not always know how to handle the function
03:15.59BSD_Techok
03:17.46BSD_Techwell I would watch the cli and try to park a call
03:17.58BSD_Techand make sure to do a sipdebug
03:18.09*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
03:18.11BSD_Techand the put it up for review before calling it a bug
03:18.30BSD_Techusing your park-orbit button
03:18.42BSD_Techto make sure asterisk is seeing the full function
03:19.10BSD_Techbut wait
03:19.40BSD_Techalso did you have to program the buttons on your phone to use the asterisk functions
03:19.41FuriousGeorgei know how to file a bug report correctly, i was just hoping against odds that someone in here listening had experienced what I am experiencing
03:19.55BSD_Techis it setting dnd in the astdb
03:20.11*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
03:20.30FuriousGeorgewhat i showed you is the default behavior of asterisk when one of my phones is on dnd
03:21.24BSD_Techok but did you program the phones to use the asterisk function
03:21.51BSD_Techit helps to understand your phone more
03:21.57BSD_Techdont get bent
03:23.05FuriousGeorgeno
03:23.11BSD_Techeveryphone has diff functions and some can be remapped to use the asterisk dial plan and some cant
03:23.33FuriousGeorgethis is the default behavior of asterisk when one of my phones is on dnd
03:23.37FuriousGeorgeno setup involved
03:23.40BSD_Techwell do this get a paste bin while park-orbiting a call
03:23.47BSD_Techso we can review it
03:24.06BSD_Techwhat phone/model so I can look it up
03:24.31FuriousGeorgesip debug output is far to verbose for pastebin
03:24.37FuriousGeorgei think
03:24.53BSD_Techwe do them all the time
03:25.09BSD_Techand asterisk -vvvvvvvvvvgr would also help
03:25.26FuriousGeorgethe three second act of parking a call answering it and hanging up (no dialing) is thousands of lines of CLI output
03:25.33BSD_Techits a matter of seeing whats going on to troubleshoot it before calling it a bug
03:26.20FuriousGeorgeim looking at sip debug output, seeing a call being transferred, and getting a maximum retries exceeded error
03:26.23FuriousGeorgei know what is going on
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03:26.53FuriousGeorgei wish i had said "blind transfer" instead of "park orbit"
03:26.56BSD_Techit helps for others to see it
03:27.13BSD_Techasterisk has blind transfer
03:27.18BSD_Techenable it
03:27.29FuriousGeorgesigh
03:27.33BSD_Techbut it would mean dialing
03:27.39BSD_Technot just a button
03:27.58BSD_Techyour the one making it hard by not pastebing
03:28.09BSD_Techtho see it helps us more
03:29.22FuriousGeorgewhen you say "enable blind transfer" you must be thinking of setting asterisk to listen for dtmf so that zap channels, which obviously dont have a reak built-in transfer functionality, can approximate it
03:30.25BSD_Techno
03:30.30FuriousGeorgeon the other hand, transfer is part of the sip rfc, and you should know that the difference between blind and attended transfer is only if you REFER before calling or after calling the destination
03:30.32BSD_Techin the functions.conf
03:30.44BSD_Techthere is a blindtransfer function
03:31.09BSD_Techits disabled by default
03:31.17FuriousGeorgeand its for zap channels
03:31.20FuriousGeorgenot for sip
03:31.31FuriousGeorgeyou mean features.conf?
03:31.34BSD_Techno its for sip and zap
03:31.40BSD_Techyes features
03:32.03BSD_Techsorry been working hard at coding today
03:34.42*** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au)
03:34.43BSD_Techfeatures.conf is for sip/iax/zap/
03:34.49BSD_Techits not just for zap
03:35.26FuriousGeorgeeven if it does work for 100% sip calls, why would someone want to use that if they have a fully functional sip phone when the whole point of that is to achieve what the transfer button does anyway
03:37.39BSD_Techbecause not all phones work right with asterisk
03:37.55BSD_Techthis allows the functionalty no matter what phone your on
03:38.28BSD_Techit depands on the firmware/functionality built into the phone
03:38.37BSD_Techpoint and fact
03:39.04BSD_Techgxp2000 you have to disable the internal functions for *70/*71 to work
03:39.27BSD_Techbecause they built it into the firm ware and asterisk does not see the function activated
03:39.32BSD_Techor deactivated
03:40.33JTgxp2000, you're mentally disabled if using that in a business setting ;)
03:40.38FuriousGeorgeright, which is why i qualified my question with the phrase "fully functional".  SIP transfers blind or attended, work fine, unless they go to the park extension, at which point the cause asterisk to generate errors.
03:40.49FuriousGeorges/the/they
03:41.24*** join/#asterisk ldsjohn (n=darksage@c-67-171-248-120.hsd1.wa.comcast.net)
03:41.50BSD_Techwell I just tested with my cousin the blind transfer to parking and had no issue
03:42.03BSD_Techit did not generate errors
03:42.06FuriousGeorgewell, that settles it then
03:42.12FuriousGeorgethanks for your help
03:42.21BSD_Techso it goes back to a earlier question what ver of asterisk
03:42.40BSD_Tech1.4.5 has no issue
03:43.11BSD_Techthere is alot of info you did not give
03:43.18FuriousGeorgei got a lot of sip debug output to read through and filter before filing this bug report, seriously, thanks for your help
03:43.19BSD_Techso it makes it hard to help
03:44.39ldsjohnhey, Emergency type situation * for me at least *, but, say if I have 30 polycom phones atached to an asterisk box and one of them is an idiot and uses the forward button so our incomeing calls are forwarded somewhere, but nobody wants to admit to it, can I figure out remotely who is forwarding the calls?
03:45.04wunderkinwell it will say in the cli
03:45.59ldsjohnwhen I dial in 30 extensions ring (* late night ring group *) and I can't figure out wich one is forwarding
03:46.35wunderkinit says if you have verbose on
03:46.46ldsjohnI have "set verbose 10"
03:46.50ldsjohnand I dont see it
03:47.33wunderkinwell if you are using forward on the phone
03:47.50*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
03:47.56FuriousGeorgecall all the phones till one doesnt ring, man
03:47.57*** join/#asterisk javar (n=javar@69.79.134.24)
03:47.59ldsjohnyeah on the phone, my thought is what would happen is a call woudl go in to phone, then phone would call out
03:48.01ldsjohn...
03:48.06ldsjohnGENIUS :-D
03:48.31ldsjohnsorry, I am dealing with the alarm company and a break in and this all at once.. so my brain isn't working, thats the obvious fix though :-D
03:48.35JTis the forward to another extension, or elsewhere?
03:48.45ldsjohnto an outside number
03:48.51JThrm
03:48.54ldsjohna number of a currently very mad IT guy
03:49.07JTsurely that'd hit the asterisk server
03:49.14JTgoing to the outside number
03:49.50FuriousGeorgeyou should see an outbound call
03:49.51ldsjohnyeah but I dont see it happening
03:50.04FuriousGeorgeshow channels when the mad it guy is on the phone
03:50.06ldsjohnim going to call down the extensions
03:50.08FuriousGeorgethat will solve the mystery
03:50.10ldsjohnhey
03:50.12JTor sip debug
03:50.15ldsjohnhrm
03:50.16ldsjohnk
03:52.42FuriousGeorgeJT:  you speak fluent sip debug?
03:52.49JTsure
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03:53.57FuriousGeorgenm, figured that part out ;)  im sure ill have another question momentarily
03:54.10ldsjohnim lost
03:54.16ldsjohnsip debug spews too much crap
03:54.25ldsjohnproblem is there are 30 phones in my incomeing ring group
03:54.55FuriousGeorgeldsjohn: you are sure that when someone calls in the call is getting forwarded to the it guy right?
03:55.10ldsjohnyes
03:55.13ldsjohnI am on the phone with him
03:55.17ldsjohnwhen I dial my companys number
03:55.21ldsjohnmy partner gets called
03:55.23FuriousGeorgejust show channels and show me what is says
03:55.24ldsjohnwe are an IT staff of 2
03:55.31ldsjohnit shows a bunch of ACK
03:55.38FuriousGeorgenot sip debug
03:55.41FuriousGeorgeshow channels
03:55.46FuriousGeorgeor core show channels in 1.4
03:56.21ldsjohnhow do I turn off sip debug
03:57.57JTsip no debug
03:58.02ldsjohnodd
03:58.09JTnot really
03:58.10ldsjohnwhen I am on the phone with IT guy
03:58.12JThelp sip
03:58.14ldsjohnthe call is liek this
03:58.39ldsjohnmy IAX phone > outside Pri, -> outside Pri -> outside Pri -> him
03:58.46ldsjohnI see 3 zap channels and my IAX channel
03:58.56ldsjohnit shows 4 active channels but none of them are  a sip phone
03:59.22JTcall using your cell
03:59.24JTless confusing
03:59.32ldsjohnk
03:59.33ldsjohnone sec
04:00.04*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
04:00.43ldsjohnwhen I do that
04:00.46ldsjohnthere are 2 channels
04:01.03ldsjohnone is Zap/1-1 and one is him
04:01.18ldsjohn0 active sip channels
04:01.22ldsjohnseams the handoff happens fast
04:02.40JTerr
04:02.47JTshouldn't it be in your CDRs?
04:02.59ldsjohnhrm
04:03.03ldsjohnit doesn't look like it
04:03.11ldsjohnthats the thing it looks like the forward doesn't happen
04:03.25ldsjohnbut that s what every forward from the polycom looks like
04:04.12JThrm
04:04.24ldsjohnthe cdr shows the incomeing like my cell or my iax phone
04:04.30ldsjohnand shows the dest as his extension
04:04.32[TK]D-Fenderldsjohn, Forward is damn near instant
04:04.36ldsjohnyeah
04:04.59[TK]D-Fenderldsjohn, basically onces the sip redirect is sent we know its accepted and the RTp is handed off.
04:05.06ldsjohnI havn't foudn any forwards in the logs yet and I dont know what one looks like, Ive been trying to find in the logs where the receptionist does it
04:05.09ldsjohnyeah
04:05.16[TK]D-Fenderldsjohn, No reason for it to take more than the blink of the eye
04:05.47[TK]D-FenderIINM it should look like a new call...
04:06.41ldsjohnhrm
04:07.10ldsjohnwell once it gets into the ring group is when I lose track of it I think
04:13.36ldsjohn...
04:13.39ldsjohnit was the CEO....
04:13.46JTnice
04:13.47ldsjohnnow I have nobody to yell at...
04:14.08rob0Yell at us. We're used to it.
04:14.49JThow did you find out?
04:15.00ldsjohnwell the forward caused the alarm company to call my partner
04:15.06ldsjohnand ther was a breakin
04:15.20BSD_TechFurios George what ver of 1.4 ?
04:15.23ldsjohnso the ceo is there clearing the building
04:15.38BSD_Techyou keep saying 1.4 but there is more then 1 version
04:15.40ldsjohnand I came on here to see if I coudl find it easy
04:15.53ldsjohnand in the end i just went down the list and called every extension,
04:16.08ldsjohnnow that his phone is reset its all good... cept for the breaking
04:20.50JTyeah
04:21.11FuriousGeorgeBSD_Tech: 1.4.4 im somewhat busy now
04:21.47BSD_Techtry updating to 1.4.5
04:21.54BSD_Techsee if the issue goes away
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04:27.18*** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net)
04:30.55*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
04:33.30*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
04:39.24*** join/#asterisk Jeffsey (n=jsmith@callgirl.alkaloid.net)
04:39.35Jeffseyanyone around tonight?
04:40.21Jeffseyjust bought a couple s101i's from digium and i'm trying to provision them ... i've done the process before but I can't seem to get the iaxyprov software to give me any output after it trys to provision
04:40.38Jeffseyjust wanted to verify with someone that hte code in the svn trunk for iaxyprov is working
04:40.52Jeffseyand its just an issue with me ..which is 100% probably the case
04:41.20*** join/#asterisk holiday42 (n=me@70-57-197-218.farg.qwest.net)
04:41.34*** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru)
04:41.46*** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca)
04:41.52_VoiceMeUp_COMcan we add header in AGI's ?
04:42.00_VoiceMeUp_COMor i need to set before
04:42.07*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
04:43.03[TK]D-Fender_VoiceMeUp_COM, In AGI you can do whatever you want.  Call whatever apps, set whatever values.  aside from a single dialplan pattern to dump you into AGI you don't even NEED the dialplan.
04:43.16_VoiceMeUp_COMah cool [TK]D-Fender
04:45.12mostyi thought about building my entire dialplan in AGI once
04:45.32JThah
04:45.39[TK]D-Fendermosty, Its painful, and adds system load, but you can do it.
04:45.52_VoiceMeUp_COMlol
04:46.04_VoiceMeUp_COManyidea on how to get a header value ? in cagi
04:46.08[TK]D-Fendermosty, Remember that FreePBX does its key work in dialparties.agi (Watch for it with the trash coming in here)
04:46.17_VoiceMeUp_COMwould it be considered as a var
04:46.21_VoiceMeUp_COMor its another exec
04:46.26_VoiceMeUp_COMah nevermind
04:46.30_VoiceMeUp_COMill do in dialplan lol
04:46.40_VoiceMeUp_COMwondering how mnay check ill need to add in c
04:46.45_VoiceMeUp_COMlike. if not present.. etc
04:46.51holiday42google not helping me here... trouble compiling asterisk-1.4.5 on cygwin.  see pb: http://pastebin.com/936498
04:46.53mosty[TK]D-Fender, i figured it would be painful, never got around to trying it (only for small parts of the dialplan)
04:47.00_VoiceMeUp_COMyeah
04:47.08_VoiceMeUp_COMwe have a few runing on fastagi
04:47.18mosty[TK]D-Fender, if you wrote it in something like C it wouldn't add much load would it?
04:47.18_VoiceMeUp_COMlike non important optional stuff
04:47.27holiday42... anyway would asterisk via cygwin properly "see" x100p clone anyway?
04:47.43_VoiceMeUp_COMhehe
04:47.50_VoiceMeUp_COMasteirsk on unix has problem seeing it
04:48.02_VoiceMeUp_COMimagining on wincrap its evern.. hmm 45:1 on 100% sucess
04:48.08holiday42urk
04:48.14_VoiceMeUp_COMunless its the new ultra ubber version
04:48.17[TK]D-Fendermosty, clearly not.  no extra interfaces, etc.
04:48.22_VoiceMeUp_COMthat one could be like 4:1
04:48.42holiday42never tell me the odds :/
04:48.50[TK]D-Fendermosty, apps are the most efficient way for sure, but typicaly less flexible because of compiling, etc.
04:48.55_VoiceMeUp_COMi rmemeber the old clone at my place.. on a old cpu.. it never hung up probperly
04:49.37mosty[TK]D-Fender, someone was asking me about writing a dialplan that couldn't be modified easily, that was my suggestion, i never went through with it
04:49.46holiday42it appears to be working fine on spare hardware currently... thought I would consolidate... sounds liek it's not worth the hassle though.
04:49.58*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
04:50.13JTare you crazy holiday42?
04:50.28JTpretty sure cygwin doesn't let you do low level hardware i/o
04:51.05[TK]D-Fendermosty, You could try to feel 1337 and do it all in AEL and ride the illusion ;)
04:51.50holiday42~ael
04:51.50jbotsomebody said ael was Asterisk Extension Language - a dialplan language with 'c like' syntax?
04:52.03holiday42ah
04:52.51[TK]D-Fenderholiday42, BS that gets parsed back to traditional EL.  AKA does NOTHING you can't do already.
04:53.16mostymaybe it's easier to write AEL though, i'm not sure, haven't tried it
04:53.27holiday42~el
04:54.41holiday42[TK]D-fender: rephrase please, as I didn't understand... what does "EL" stand for?
04:54.52JTextensions language
04:55.01holiday42oic
04:55.06[TK]D-FenderExtensions Logic (standard extensions.conf)
04:55.33JTah, you and all your acronyms ;)
04:56.11holiday42and asterisk jargon to boot
04:56.15*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
04:58.08[TK]D-FenderJT : STFU & GTFO ;)
04:58.18holiday42that i understood :)
04:58.48[TK]D-FenderI mince words.... with my KATANA :D
04:58.53JT[TK]D-Fender: WTF PEIAN
04:59.38holiday42x100p clone aside, is there a pathc to get *1.4.5 to compile under cygwin.  I see a precompile asteriskwin32 out there, but it's dated
05:01.52[TK]D-Fenderholiday42, why on earth are you trying to run * in windows?!
05:02.53JTwacky tobaccy
05:03.13holiday42to consolidate pc's
05:03.31JTto have a non-working pbx
05:03.49holiday42I would do it under vmware... but timing issues cause moh, etc to suck
05:04.00JTyou see
05:04.07JTthere's a slight problem
05:04.10holiday42ah ha
05:04.16JTasterisk was designed only to work under *nix
05:05.46holiday42A butterknife wasn't meant to be used as a tool.
05:06.45*** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net)
05:07.48_VoiceMeUp_COMi read butterfly
05:08.00_VoiceMeUp_COMand was about to agree.. but butterknofe youd be amazed..
05:08.10_VoiceMeUp_COMknofe even
05:08.14holiday42heh.
05:09.00_VoiceMeUp_COMif anyone triesd coutnerpath bria let me know
05:09.10_VoiceMeUp_COMif its going downwards or going up
05:09.25holiday42I see asterisk runing on mac, and *bsd... and I see some times in the past the asterisk code did have portability in mind
05:10.19JTholiday42: both of those are unicies
05:10.25JTwindows is not
05:10.56*** join/#asterisk misc-- (n=misc@122.2.118.233)
05:11.02holiday42sun then
05:11.19holiday42i take it back... dunno about that
05:11.19JTalso a unicie
05:11.28misc--hi.. how can I apply echo cancellation on an iax2 trunk or sip device? I don't use zaptel...
05:11.30JTa unix even
05:11.34holiday42sun == *nix?
05:11.43JTno shit, solaris is a form of unix
05:11.57JTand you can run linux or bsds on suns
05:11.58[TK]D-FenderSlowaris ;)
05:12.05JTonly on x86
05:12.08JTit's slow
05:12.51[TK]D-Fenderok, bed-time here.... later all
05:13.25*** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net)
05:13.44holiday42well damn.  maybe i'll just install linux on my pc, install asterisk , and run windows under linux-vmware
05:15.01holiday42or i could be lazy and use asteriskwin32 (asterisk 1.2) even though it was designed to for *nix
05:15.27JTi don't know anyone who runs asterisk on windows with success
05:15.33JTand i don't know why you'd want to anyway
05:15.50misc--amen
05:16.00*** join/#asterisk LoneShadow (n=a@c-76-103-55-28.hsd1.ca.comcast.net)
05:16.55*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)
05:19.01holiday42hmm. asteriskwin32 forums are a *little* ... vacant :]
05:22.15*** join/#asterisk DeL3e7 (n=DeL3e7@EV-ESR1-72-49-112-230.fuse.net)
05:23.36dlynes_laptopholiday42: amen to that, brutha
05:26.35LoneShadowI have installed libiksemel3 version  1.2-3 on my ubuntu. Still configure fails to find the library
05:26.46LoneShadowanyone know why ?
05:27.16DeL3e7dif ver?
05:27.30LoneShadowhmm
05:28.02DeL3e7libs can be hell
05:28.07LoneShadowI guess :)
05:28.15LoneShadowI am trying to avoid downloading all the sources
05:31.31*** join/#asterisk nephfl (n=nephfl@adsl-070-147-105-151.sip.gnv.bellsouth.net)
05:33.02nephflhello...i just fried a tdm2400 i need to switch to sip trunk and asterisk is failing startup...any ideas how to do this quickly?
05:35.29*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)
05:35.30zeeeshhi
05:36.11LoneShadowugh, I had install libiksemel-dev
05:36.13LoneShadowto*
05:47.45*** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net)
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05:59.15*** part/#asterisk misc-- (n=misc@122.2.118.233)
06:15.33*** join/#asterisk Zion800 (n=None@cpe-76-167-156-224.socal.res.rr.com)
06:17.13*** part/#asterisk holiday42 (n=me@70-57-197-218.farg.qwest.net)
06:17.23Zion800Hey, can someone help me fix a small Perl script for Asterisk 1.4?  It works with 1.2, but doesn't with 1.4...
06:18.17DragoraNis SRTP and TLS support for SIP included in Asterisk 1.2.13 ?
06:18.50JTno
06:21.17*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:23.06*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
06:25.07*** join/#asterisk gardo (n=gardo@121.97.196.53)
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06:34.22*** part/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com)
06:36.27obnauticusHey, for asterisk 1.4's embeded mini webserver, from within the CLI how do i reload it?
06:36.50JTmini webserver? :O
06:36.58obnauticus/etc/asterisk/httpd.conf
06:37.13obnauticushttp://www.voip-info.org/wiki/index.php?page=Asterisk+manager+API
06:43.50*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
06:44.45sergeeobnauticus: hi!
06:45.01sergeeobnauticus: thank you very much for your iinput!
06:45.23obnauticusnp sergee
06:45.27obnauticusi noticed it today
06:45.31obnauticusi think it sounds cool though
06:45.33obnauticusi don't want them to remove it
06:45.34obnauticusrofl
06:45.34*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
06:47.23sergeeobnauticus: to remove what?
06:49.17*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
06:49.31*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
06:52.37obnauticusthe loud squelch after the second user joins
06:52.40obnauticusi think i hear it after every user
06:53.28Strom_Mwhat, the sounds for meetme?
06:53.32obnauticusyes
06:53.40obnauticuswell
06:53.49obnauticusi don't think the one im talking about is supposed to be there.
06:53.55Strom_Myeah, that's the sound that's played into the conference when /every/ user joins
06:54.07obnauticusi guess so
06:54.13obnauticusi don't know any better :)
06:54.13obnauticuslol
06:54.15Strom_Mthe "join" sound is two repeating notes; the "leave" sound is two descending notes
06:55.24Strom_Mare those what you're hearing?
06:56.16*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:56.36DrAk0heya, im having big problems trying to record calls, when i recieve them by the TDM and I do transfer it hangs up and says it can't change the name of the recording file to null
06:57.11*** join/#asterisk psk (n=psk@golia.caltanet.it)
06:57.57Strom_Mhow are you recording the call?
06:59.34sergeeStrom_M: obnauticus: guys i have a patch for that loud sound in meetme, if you can test it and write your feedback in bugtracker, i would be very thankful !
06:59.49sergeeM9430
06:59.56sergee~9430
07:00.06sergeehttp://bugs.digium.com/view.php?id=9430
07:00.13obnauticusk
07:00.46Strom_MDrAk0: ...are you going to answer my question? :)
07:00.47sergee<Strom_M> the "join" sound is two repeating notes; the "leave" sound is two descending notes
07:01.05DrAk0Strom_M, sorry
07:01.54Strom_Msergee: yeah, so?
07:01.56Strom_Mit's been that way forever
07:02.00sergeeStrom_M: not exactly, as for me, i heare, a-crack-like sounds, a kind of nose
07:02.03sergeeStrom_M: not exactly, as for me, i heare, a-crack-like sounds, a kind of noise
07:02.16Strom_Mdon't repeat lines at me, please :)
07:02.24sergeesorry :)
07:02.29Strom_Ms/nose/noise/ would have sufficed
07:02.33obnauticusIs anyone here running webmeetme on asterisk 1.4 ?
07:02.50sergeeStrom_M: so instead of noise you heare some nice "ding"?
07:03.08Strom_Msergee: is this on 1.4?
07:03.11DrAk0Strom_M, http://pastie.caboo.se/73529
07:03.21obnauticusStrom_M I'm on 1.4 and I think i hear it too.
07:03.23sergeeStrom_M: yes
07:03.32sergeeStrom_M: on trunk actualy
07:03.41Strom_Mwell is it 1.4 or is it trunk
07:03.49Strom_Mbecause those are two different branches
07:04.10Strom_MDrAk0: uh, you have to specify a filename as an argument to Monitor()
07:04.37DrAk0Strom_M, i tried but i got the same error.
07:05.10DrAk0Strom_M, by now, i have disabled Monitor o the marcar-macro and it works.
07:05.11Strom_MDrAk0: what version of asterisk are you running/
07:05.20DrAk0Strom_M, 1.2.18
07:05.21sergeeStrom_M: trunk,
07:05.41Strom_Msergee: well i'm not running trunk, so I can't test your bug.  I'm running 1.4 branch
07:05.43sergeeStrom_M: i know, but this bug present in both
07:06.08Strom_Mo rly now
07:06.11Strom_Myou've tested?
07:06.12sergeeStrom_M: hmm so you are runing 1.4 and doesn't experience this bug?
07:06.32sergeeStrom_M: me - no, guys in bugtracker wrote about 1.4
07:06.32Strom_Mwell at least whatever month-old checkout of 1.4 I'm running
07:07.00sergeeStrom_M: do you have digium harware installed?
07:07.06Strom_Myes
07:07.19Strom_Mtdm844B / tdm31b
07:07.28sergeeStrom_M: very interesting,
07:07.38*** join/#asterisk nephfl (n=nephfl@adsl-070-147-105-151.sip.gnv.bellsouth.net)
07:07.47nephflhello, anybody awake still?
07:07.50Strom_Mno
07:07.52Strom_Mwe're all dead
07:07.57nephflI am
07:07.57Strom_Mgo away and let us rot in peace
07:08.07Strom_M:)
07:08.10nephflI killed a companies system
07:08.19nephflfried a tdm 2400
07:08.27Strom_Mhow'd you do that?
07:08.31sergeeStrom_M: can you write a feedback in bugtracker? describe your system (hardware, asterisk revision) and that you doesn't experience this bug?
07:08.31nephflnow im trying to get something up and running before daylight
07:08.53Strom_Msergee: im really not in the mood to go screwing with my system right now
07:08.56nephfli let "the new guy" move a module on the board...he put it back in one pin off and smoked it
07:09.06Strom_Mnephfl: ah
07:09.13Strom_Mwhere are you located?
07:09.21nephflin Gainesville,FL
07:09.25Strom_Moh ok
07:09.32nephfland this is an oil company that depends on their phones
07:09.45Strom_Mwhat were you doing moving modules anyway?
07:10.00nephfli think we had one bad port to begin with
07:10.08nephfli was going to rma just one module
07:10.46Strom_Mwell, set up an account with an ITSP and forward the inbound trunkgroup to that ITSP
07:10.52Strom_Mhow many analog lines are there?
07:11.02nephflso, im thinkin if i can get a few voip trunks, and buy some atas from the walmart, some duct tape and bailing wire...and maybe i can get something up
07:11.12Strom_Mhahahhahahahahahahaha
07:11.14Strom_Mwal-mart?
07:11.18nephfl4 fxo 8 fxs
07:11.27nephflwalmart here sells those vonage atas
07:11.36Strom_Mmaybe in 2012 you'll be able to buy unlocked ATAs in any corner market
07:11.50nephfli unlocked one and got it working on the system as a test
07:11.56Strom_Mhmm, ok
07:12.11Strom_Mnot something i would do on a production system, but i suppose these are desperate times
07:12.41sergeeStrom_M: it's a misunderstanding :) i don't ask you to touch your system :) i ask you to write a feedback in bugtracker - that this problem doesn't exist in your system.. if you have a few free minutes ofcause..
07:12.48nephflnow i logged into vtwhite and found out my account is locked and i have to get this up, so i need some DID's fast so i can get the local telco to forward their existing lines
07:13.07Strom_Msergee: but I'm not running the latest 1.4 branch checkout.  didn't you listen to me?
07:13.18Strom_Mnephfl: try teliax
07:13.24sergeeStrom_M: ok no matter :)
07:13.28Strom_MDID is $5 a month
07:13.35Strom_Mcall traffic 2c/minute
07:15.18nephflthink i can sign up get a few local dids set up, and setup 3 more wrtp54gs (from a store 25 miles away), in the next 3 hours
07:15.22nephfl?
07:15.28Strom_Mwhy do you need more than one DID?
07:15.41Strom_Mforward all the lines to the one DID and you're set
07:15.41nephfli need to forward 4 existing lines
07:15.55nephfli didnt know if that would work
07:15.56Strom_Myou can have more than one concurrent call on a DID, you know
07:16.52Strom_Mand yeah, 3 hours is GOBS of time to do that
07:18.08nephflok...i can do that, is signup and setup on asterisk easy with teliax?
07:18.16Strom_Mubereasy
07:19.25nephflok...and if this works i will be amazed
07:19.32Strom_Mi dont see why it wont
07:19.34awkhmm, asterisk not passing the callerid, could that have anything to do with switchtype= in zapata ?
07:19.45Strom_Mawk: are you using a PRI?
07:20.45awkStrom_M BRI
07:21.24Strom_Mawk: what BRI card and what BRI package?
07:21.27awkJust wondering what could be wrong for no callerid to be passed.. i'm using a generic zapata.conf from bristuff (zaphfc) wondering if its using setting that is wrong for my country
07:21.34Strom_Mbeats me
07:21.43Strom_Mwhat switchtype are you using, and what setting has the telco given you?
07:22.04awkStrom_M: hrm, to get that info out of my teleco is impossible. i'm using swtichtype = euroisdn
07:22.10*** join/#asterisk mkl1525 (n=qwertz@i59F739CB.versanet.de)
07:22.18Strom_Mwhat country are you in?
07:22.23awkSouth Africa
07:22.31Strom_Mhmm, dunno what telkom uses
07:22.32mkl1525Hi, does anybody know, if there's some timeout setting for connects to the * manager interface?
07:22.45JTawk: are you passing a legit callerid?
07:23.04*** join/#asterisk peanutb (n=paulb@c-24-16-243-186.hsd1.wa.comcast.net)
07:23.59*** join/#asterisk matsk (n=mk@194.68.102.172)
07:24.06awkYe, it is euroisdn.. There are two types of switches in the Telkom network namely the Alcatel E10 and the Siemens EWSD. The switch type you must select is Euro ISDN (ETSI) and the protocol type EDSS1.
07:24.26awkJT: ye, i'm calling froma phone next to me, if I call my cell it shows the callerid.
07:24.40JTokay
07:24.50JTthis phone next to you connects to what?
07:25.39awkSo it isn't the switchtype. I also have forced callerid in this sense. usercallerid=yes
07:25.42awkin zapata.conf..
07:25.53awkJT: umm, just a analouge line..
07:26.06awkif I call from my cell phone it doesn't pass that callerid either, let me double checkl
07:26.19awklet me cli the call quickly
07:26.31*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
07:26.35JTawk: pastebin zapata.conf and zaptel.conf
07:26.36JT~pb
07:26.38jbot[pb] a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
07:28.05*** join/#asterisk Pilko (n=pirch@213.80.169.119)
07:28.29awkhttp://paste.linux-vserver.org/2727]
07:28.44awkremove the ] from that string
07:28.51awkhttp://paste.linux-vserver.org/2727
07:29.09awklet me paste my zaptel and zapta too wait..
07:29.46JTumm yeah
07:29.50*** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg)
07:29.54JTbecause that's totally not what i asked for :)
07:29.58JTand doesn't help me
07:30.12JTexcept it tells me you have a very longwinded dialplan
07:30.40coldehi, im getting this error: WARNING[34816]: pbx.c:2450 __ast_pbx_run: Channel 'SIP/201-086ae000' sent into invalid extension 's' in context 'incomming', but no invalid handler
07:30.40JTwhat version of bristuff
07:30.48coldeany idea what it means
07:30.52awkhttp://paste.linux-vserver.org/2728
07:30.52coldeor how to solve it?
07:31.00JTcolde: spell incoming right?
07:31.04awkJT: using the latest. packed with asterisk 1.2.18
07:31.43JTprobably won't make a difference, but span=1,1,3,ccs,ami  should be span=1,1,0,ccs,ami
07:31.55coldeJT: of course, thanks
07:32.50JTawk: can you enable pri intense debug span 1 then make a call that should send callerid, and pastebin the whole call, including the debug?
07:32.59JTcolde: no probs
07:35.07awkJT: hrm, I can't get it to specify a file I want, where is the debug default file stored?
07:35.23awkor is it all passed to CLI ?
07:36.24JTit is passed to cli
07:36.28JTso you can enable full log
07:36.30JTin logger.conf
07:36.41JTand it will also be in /var/log/asterisk/full
07:37.12nephflteliax is $44/mo and you can only get 4 concurrent connections?
07:37.17Strom_Mno no no no no
07:37.21Strom_Mdo PAY AS YOU GO
07:37.27Strom_Mdurh
07:37.34nephflyeah..sorry it is early here
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07:38.36awkJT: http://paste.linux-vserver.org/2729
07:39.34awkJT: could it be a codec issue, ALAW ?
07:40.50sergeeStrom_M: one more question about your Asterisk & MeetMe :)
07:41.00Strom_Mk
07:41.04JTawk: no
07:41.07sergeeStrom_M: are you using 'I' option for meetme?
07:41.14Strom_Mno
07:41.19JTawk: damn the font on that paste site is horrible
07:41.19*** part/#asterisk harlequin516 (n=sham@styk.net)
07:41.33awk;/
07:41.39sergeeStrom_M: that's it! thank you very much! :)
07:41.49awkand there I thought my eyes where getting bad
07:42.06Strom_MJT: no, clearly looking cool is far more important than legibility
07:42.48nephfli assume i want an IAX account, right?
07:43.18sergeeStrom_M: i suppose that if you would enable "I" option - those "dings" will become a noises, reported by other users..
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07:45.21JTStrom_M: of course, die being cool, it's cool
07:45.34JTawk: umm, you've cut off the useful part of that pri debug
07:45.37JTawk: at the top
07:45.43JTso yeah
07:45.50JTrepaste with the useful stuff still there
07:45.54JTa WHOLE call :)
07:46.10awkJT: that was the whole call the only thing at the top was me saying set debug 10, etc
07:46.14awknothing from the call is missing
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07:46.41JTwell what number did you call?
07:47.12cy303sup
07:47.14awkJT: i'm not using DID i'm using any
07:47.18awkI only have 1 number :)
07:47.22JTerr
07:47.37JTok let's get one thing straight.
07:47.45JTare you RECEIVING or MAKING a call?
07:47.54awkmaking a call from a cell phone to this box
07:48.10awkthis box isn't getting the callerid
07:48.26awkso on my snom i just get Unknown the whole time
07:48.41JTwell you could've made that a hell of a lot clearer
07:48.50JTi thought you were making a call with the box
07:48.51awksorry I thought I had
07:49.06JTyou just said callerid wasn't been passed, that's it
07:49.16awksorry, my mistake
07:50.08JTso your telco isn't sending you callerid
07:50.12awkcalls going out, the callerid is working fine, so if i phone from the system to my cell i'm getting the callerid.
07:50.49awkJT: well thing is if I plug an analouge phone into 1 of the analouge ports on the NT I get the callerid
07:51.17JTi can't see it in the debug
07:51.23JTwhat's the number i should be seeing?
07:51.33awk0844045791
07:52.07awkbut you right it isn't showing it, so does that mean straight away its a Teleco issue not passing the callerid to me, or could it be a configuration issue?
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08:02.09awki've added this to zapata.conf usecallerid=yes hidecallerid=no callerid=asreceived
08:02.13awkbut still no callerid
08:08.59mostyawk, who is your telco?
08:11.00awkTelkom SA
08:12.08awkI will give them a call a bit later, but dam they difficult to deal with :D
08:26.17Uatecthey're all difficult to deal with
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08:28.55Fl1phi, my CLI says : Really destroying SIP dialog 'sahds8203...'@192.168.3.1 <- I'dont have any clue where this local Ip came from, this is also my reg contact shown by my sip provider.... Any idea ?
08:29.28Strom_Mit comes from your kitchen sink
08:33.07Fl1pStrom_M: Impossible, should i check the drain in my bathroom ?
08:33.39negativeduckno it's a trap! that's what *they* want you to do.
08:34.03Fl1pnegativeduck : thanks for the advice
08:34.18Fl1pany other clue anyone ?
08:39.39negativeduckit's not an IP on any of your local networks?
08:40.16nephfli am getting an all circuits busy with teliax now
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08:42.57Fl1pnegativeduck : no, thats the most confusing point
08:43.29Fl1pchecked all my .conf files with grep -S "192.168.3.1" -nothing
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08:46.00Fl1phmm if i set localnet=192.168.1.0/255.255.255.0 it causes this 3.1
08:46.14Fl1pif i disable it takes the local ip of the * server
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09:00.07*** join/#asterisk kombi (n=kombi@64.79.217.62)
09:00.44kombihow do I increase *'s log level to "insane"?
09:01.04JTset verbose 99999999999999999999999999999999999999999999999999999999999999
09:01.16kombithanks JT
09:01.29JTwell, i don't think it makes any difference above 10
09:01.43JTthere's debug levels, and specific debugs too
09:02.20kombican you help me debug meetme? I don't get any error message at all...
09:02.39kombithing just dies on me..
09:02.55JTyou have zap timing?
09:03.03kombi(verbosity is now 2178515698)
09:03.25kombiyip, got ztdummy & the works..
09:03.45kombishow application meetme nicely shows all the options
09:04.59kombiprompt "you are the first man in my life" (just kidding) is being played..
09:05.13kombibut nothing happens after that
09:06.37kombion a freshly started * CLI meetme says "No active MeetMe conferences"
09:07.09*** join/#asterisk teapot (n=tandrews@mail.grok.org.za)
09:07.16teapothi
09:08.00kombiwhen calling in it ask for pin and prompts but after that CLI does not take commandy anymore
09:08.04kombihi teapot!
09:08.10teapot:)
09:12.16teapotHas anyone here used CDRs ?
09:12.27snuff-workis the sky blue?
09:12.52teapotI can't understand why a call is marked as 'answered' two seconds after you've made a call
09:13.18teapotEven if the call is never answered, it still gets marked as such
09:13.30teapotSo as a result the billsec is totally wrong.
09:14.08snuff-workdepends what you are doing
09:14.29teapotWell, using an FXS to dial an FXO
09:16.41teapotexten => _8XX,n,Dial(200,30,o)
09:17.13teapoter
09:17.28teapotstrike that.
09:17.31teapotExecuting Dial("Zap/1-1", "Zap/g1/200||o") in new stack
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09:27.23teapotOh, of course. It's because it's an analog channel. There isn't any feedback to tell when it's actually answered. :(
09:27.39teapotheh
09:28.34kombifound the answer completely by himself, I like that!
09:29.28teapotyeah, clever chap ole' teapot eh ;)
09:30.37kombihere's one for you then: rtc: lost some interrupts at 1024hz <-- could that make meetme crash?
09:32.55mostykombi, disable acpi in your kernel
09:33.04mostyor use a hardware zaptel timer
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09:34.24kombimosty: good idea.. got spare digium hardware but it's completely not needed in this box
09:35.08tzafrirkombi, rmmod ztdummy; modprobe ztdummy
09:35.10tzafrirmaybe
09:35.27kombigood too, I try that first
09:36.09tzafrirkombi, I had that after suspending and reusming my laptop...
09:36.20tzafrirbasically a bug in rtc
09:37.43mostyi consider it a bug in zaptel, not rtc
09:37.56kombiI still wonder, do you actually need the full zaptel monty or just modprobe ztdummy?
09:38.41teapottzafrir: do you know if kapejod is going to have a version for asterisk-bristuff any time soon ?
09:39.06tzafrirteapot, don't know. Anything you need specifically?
09:39.14tzafrir1.2.19 ? 1.4?
09:39.14teapot1.4
09:39.30tzafrirI know he's working on that
09:39.33JTkombi: how do you know zap hardware is not needeed?
09:39.40tzafririncluding simplification of the patch
09:39.46teapotyeah I saw he has a beta of sorts
09:39.55tzafrirbut not much more
09:39.59kombiJT: sip only connection
09:40.37teapottzafrir: do you think asterisk-bristuff will die out ?
09:41.15kombiwouldn't want to stick in $800 worth of digium hardware just to make meetme work..;) (why is that so damn reliant on timing anyway?)
09:41.27JTkombi: umm yes meetme needs zapt timing
09:41.32tzafrirKapejod keeps promising he'll have a new site for it, with a better revision control system . I really hope for bristuff that such a thing will happen soon
09:41.34JTztdummy often doesn't stack up
09:41.36JTzap
09:41.56JTkombi: don't use meetme if it's a problem
09:42.04kombiI'll roll a nice new kernel next..
09:42.12tzafrirkombi, stick in a 10$/20$ worth of x100p...
09:42.24JTkombi: pre-empt and 1000Hz timing
09:42.55kombitzafrir: might make sense there..
09:43.19kombiJT: how do you mean?
09:44.11JTkernel options
09:44.46kombioh, ok good!
09:45.04JTmay or may not help
09:56.01*** join/#asterisk withnall (n=dave@60-242-0-60.static.tpgi.com.au)
09:57.35withnallhello. I have a question about installing the G729 codec. I purchased and installed it and it shows up with show g729 in the asterisk CLI. however now I get errors about translation when trying to make outbound calls
09:57.48withnalleg set_format: Unable to find a codec translation path from g723 to g729
09:58.05withnallI have not been able to find anything which tells me what I've done wrong through google
09:58.08withnallcan anyone here help?
09:58.18Strom_Mbecause you cant handle g723 with asterisk unless you have the tc400b card
09:58.32withnallwhat about slin
09:58.41Strom_Mwhy would you use slin on a voip circuit?
09:58.47Strom_Muse ulaw or something
09:58.55teapottzafrir: There's something I've been meaning to tell/ask you: Whenever I stop safe-asterisk I get this error:
09:58.59teapotStopping Asterisk PBX: asteriskstart-stop-daemon: warning: failed to kill 11532: No such process
09:59.20tzafrirteapot, don't use safe_asterisk
09:59.57teapotoh
10:00.03teapotwhy not tzafrir ?
10:00.13tzafrirthat said, you should probably be able to tweak the init.d script to make a pid file for safe_asterisk
10:00.25tzafriras safe_asterisk fails to create one
10:00.55tzafririt's not robust enough. Too easy to get into a loop, and hence won't really protect you from crashes of asterisk
10:01.05tzafrirthat's my personal opinion, at least
10:01.24teapotIt makes a pid file, but I think with the wrong pid. I've never gone into the details.
10:01.37teapotI've never had any other problems with it.
10:01.44teapotNever any loops
10:04.48withnallthank you for your help. turning off g723 fixed my issues.
10:05.40withnallquit
10:07.17tzafrirteapot, asterisk maintains a PID file. not safe_asterisk
10:16.06ai-aIn asterisk, how can i get the system to look up incomming caller ID's on a database / text file / ext. app so any matching company / customer is displayed on the phone which is ringing?
10:17.22Strom_Mai-a: there's more than one way to do it :)
10:17.30*** join/#asterisk tanacsdavid (n=david@office.axpnet.com)
10:17.36tanacsdavidHellello!
10:17.36snuff-workcreate urself a mysql table or somethin.. use the MySQL() and pass ${CALLERID(num)} to it then u can set(callerid(name)=result)
10:17.56teapottzafrir: ah, I see
10:18.06tzafrirI had many times problems where asterisk was killed and failed to restsrt. safe_asterisk happily kept restarting it.
10:18.06ai-ahow do i perform such tasks on an incomming call ?
10:18.18ai-aor any call infact would be greate,, internal => internal.
10:18.19tzafrirI had to kill safe_asterisk manually to stop the mess
10:18.30tzafrirthe init.d script was worthless
10:18.34tanacsdavidCan anyone help me with asterisk and clients behind nat? I have 2 clients in 2 nat, and cannot talk to eachother.
10:19.07ai-atanacsdavid: port forwarding / sip compatable routers
10:19.26*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
10:19.33tanacsdavidai-a: ok, but what ports to forward?
10:19.50teapottzafrir: set qualify=yes
10:19.51ai-athat sounds like a question for google
10:19.53teapotbah
10:20.05teapottanacsdavid: set qualify=yes
10:20.15tanacsdavidin the future I want to have more clients in the behind the nats
10:21.04tanacsdavidteapot: qualify is set to yes.
10:21.04tanacsdavidand nat=yes, too
10:21.04ai-aisnt it normal to have a pbx behind the nat, and communicate outside from one static ip.. and then have another pbx behind the other nat'ed network ?
10:21.04fordfroganybody can help me to make receiving of faxes using asterisk via sip? I followed "SpanDSP: Sending and Receiving Faxes with Asterisk" at http://www.voip-info.org/wiki-Asterisk+fax but it is not possible to set up communication between the faxes
10:21.14teapotYou need SIP connection tracking on your firewall then tanacsdavid
10:21.23tanacsdavidteapot: Thank You!
10:21.41tanacsdavidThat wil be a good way to start, and for finish. :)
10:22.02snuff-workai-a look up stuff on voip-info or even default extensions.conf samples come with * should be good enough to get idea
10:24.31*** join/#asterisk pietie (n=fdhjkdhj@dsl-242-7-209.telkomadsl.co.za)
10:24.36pietiegood day
10:24.41pietieand heeeeeellllppp
10:24.46teapotheh
10:24.59pietiei need help with a iax trunk setup
10:25.19pietiei am stuck and the bugger dont want to talk to eachother
10:25.33pietieimediate chan unavail
10:28.12pietiehelloooooooooooo
10:28.54tanacsdavidteapot: Isn't there any way to do the nated sip client with stun server?
10:33.27pietieis there anybody here that can give me some assistance i am busy loosin my pip
10:34.15Strom_Mpietie: ok, so calm down and pastebin your configs
10:34.33pietiethanks
10:34.36pietieeish
10:34.43pietieokey this is my anaolgue switch
10:34.57Strom_Mpastebin.ca
10:35.07Strom_Mor one of the others
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10:36.15pietiehttp://pastebin.ca/590104
10:36.55Strom_Mis that all from one box?
10:37.22pietiethis is my main swtich
10:37.25pietiehttp://pastebin.ca/590105
10:37.37Strom_Mwhy do you have two entries that conflict on the analog switch?
10:37.55Strom_Mgah
10:38.00pietiei have swtiched and changed them so many time i have kinda lost where i am
10:38.02Strom_Myour entries are all bonkers
10:38.32pietiekinda figured that
10:38.47Strom_Mjust have one for now
10:38.51Strom_Mone on each box
10:38.59Strom_Mor, even better
10:39.16Strom_Mhave two on each box: a peer entry for outbound calls and a user entry for inbound calls
10:40.12pietieokey basicly its just two switches passing calls to each other as when i put the misdn and the wctdm2400p card in one box i get horrible irq conflict
10:40.29Strom_Myeah
10:40.38Strom_Mlike i said: peer and user entries on both boxes
10:40.49pietiekay lemme try that
10:52.25nephfli cant get asterisk to accept incoming calls from teliax
10:53.04Strom_Mnephfl: did you set the DID up correctly on teliax's site?
10:53.50nephflyeah...i just caught the rejection in the CLI...ill take a look again
10:53.54nephflso close
10:54.02nephflpeople are starting to come into the office
10:54.29Strom_Mget the ATAs set up?
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11:28.18Fl1phow i can avoid all those <Really destroying SIP dialog '7729c1c8615a1dfd602307115a741767@83.43.12.3 Method: OPTIONS > on my CLI ?
11:30.11*** join/#asterisk ispireuk (n=ispireuk@cust-47-27.dsl.versateladsl.be)
11:30.27ispireukHello all
11:31.10ispireukCan someone tell me how many RTP ports a SIP client uses? If that one per client? Or does a client use several?
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11:31.27ispireukI'm having troubles with NAT and rtp
11:33.05Strom_Mone port for transmit, one port for receive
11:35.50teapottanacsdavid: Yes, you can use a stun server.
11:36.27tanacsdavidbut still not working
11:38.06tanacsdavidanyway, I try to convince my boss to use sip conntrack. :)
11:39.39kombiwho said turn off acpi?
11:40.55kombiyou are disco!
11:41.40Strom_Mdisco?
11:41.49kombiwas it you?
11:41.54Strom_Mno
11:42.04kombiyou are not disco then..
11:42.05Strom_Mi'm just trying to figure out what it means when someone is "disco"
11:42.05kombi;)
11:42.46kombiStrom_M: means he is entitled to feel like a million dollar (for a sec)
11:43.14kombiI just solved this meetme issue that was holding me up a whole das and night..
11:44.01Strom_Mso disco by association with the 1970s dance craze?
11:44.47kombiwell, sort of I guess, I actually don't now where the term comes from in this respect, just emerged I think..
11:45.21Strom_M......
11:45.22kombianyway, everybody: turn off acpi in you kernel and live in meetme happily ever after
11:45.26Strom_Mok then
11:45.57kombiI so am disco..
11:49.03*** join/#asterisk Dovid (n=Dovid@bzq-82-81-34-115.red.bezeqint.net)
11:49.18Dovidhow do I see what channels are using what codecs ?
11:49.48corruptorsip show channels
11:49.54corruptoriax show channels
11:49.58*** part/#asterisk tanacsdavid (n=david@office.axpnet.com)
11:50.03Strom_Mzap show boners
11:50.04Dovidthanks. didnt get much sleep last night ;)
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12:40.33teapotgtg, cheers all
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12:48.41tzangerwow it's quiet today
12:49.17[TK]D-Fender*crickets*
12:50.37Jochemgrin
12:54.42*** part/#asterisk Kadran (n=mohammed@82.201.252.147)
12:55.23zeeesh1st time configuring asterisk realtime ... will anybody help me out
12:56.48zeeeshjust 3 commands want to enter by using realtime asterisk .. 1,answer 2,waitexten 3,dial(sip/${exten}@carrier)
12:58.25tzanger[TK]D-Fender: any experience with that "new" polycom phone?  430 or sommat?
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12:59.16[TK]D-Fendertzanger: 430 is a year old, and I've owned one and have them at work.  The 320/330/550/650 I have not gotten my hands on yet, but I've provisionined for a client (320/650) as they are identical to process.
12:59.26*** join/#asterisk sashion (n=sdgsdg@dsl-242-91-154.telkomadsl.co.za)
12:59.44sashionhas anyone worked on the astribanks?
12:59.47[TK]D-Fendertzanger: IP 320 = Category killer.
13:00.13tzanger... category killer?
13:00.19robl^Category Killer?!?
13:00.25tzangerther'es a 550 too
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13:01.55[TK]D-Fendertzanger: Term for a product that so severely trumps its competition as to devalidate them as a even being considered such.
13:02.02tzangerheh
13:02.11tzanger430 looks like a nice desk phone
13:02.48tzanger330 however also looks like a good phone
13:02.56tzangerwe don't need superduper phones on the desks
13:03.03tzanger501 is overkill
13:03.28tzangerat least I think so... the screen would be nifty but in all honesty we have the computers right there
13:03.37mvanbaakhhmm
13:03.41robl^the 320/330 is perfect for like a waiting room / break room.. or on a desk where somone traditionally had a single line phone... and you don't want to scare them with too many buttons, lines, or features ;-)
13:03.45mvanbaakwhere can I find all them polycom phones ?
13:04.17robl^at polycom.com ;-)
13:04.34tzangerrobl^: we've got the huge nortel phones now (73something?) -- I can't imagine needing a full-out 501 at the desk for normal people who call and maybe do a conference once in a while
13:04.41robl^I am replacing the last 5 Snoms with Polycom 320s
13:04.51[TK]D-Fendertzanger: if you have PoE, forget the 430.
13:04.59tzangerwe odn't have POE
13:05.05[TK]D-Fendertzanger: 430 is mid-volume 2 reg phone.
13:05.13[TK]D-Fendertzanger: then the 430 may have a place for you
13:05.25tzangeris the 330 not two-line as well?
13:05.48[TK]D-Fendertzanger: it is, but the 301/430 can do 5 calls per key, the 320/330 only 2.
13:05.55tzangerhmm
13:06.03[TK]D-Fendertzanger: Typically still plenty
13:06.03robl^the 320/330 are 2 line, low volume phone
13:06.12tzanger301 has no speakerphone so that'd be a big no-no
13:06.31[TK]D-Fendertzanger: I for one LIKE using multple calls/linekey.
13:06.32robl^yeah..  but puting more than 2 calls per key is cumbersome and confusing to many peple that don't spend their entire day on the phone
13:06.46tzangerok so 330 or 430 ...  what do you mean by volume (I know you mean number of calls) -- but how does that translate into the phone's usability?
13:07.00[TK]D-Fendertzanger: Just added for reference.  Personally I'd rather have my home IP 301 at my office desk than the Aastra 57i CT I have NOW.
13:07.25[TK]D-Fendertzanger: 0 difference in usability.  Polycom's function is consistent.
13:07.36mvanbaakSoundPoint® Pro SE-220
13:07.41mvanbaakthat one looks nice
13:07.47[TK]D-Fendertzanger: Any = awesome.  Join/split is the best call management feature EVER
13:08.02[TK]D-Fendermvanbaak: Analog waste...
13:08.17mvanbaakuhhuh
13:08.20mvanbaakbut still look nice
13:09.44robl^tzanger: My recommendation is 15 or less calls per day and answering only one line / DN...more than that, go 430.  I fyo have to answer multiple lines / DNs..  go with a 601 or 650
13:09.53*** join/#asterisk grEvenX (n=even@ti500720a080-8073.bb.online.no)
13:10.03[TK]D-FenderOther smart things Polycom's do : Lets you quit a 3-way call and leave the 2 other parties connected
13:10.16robl^err..  15 or less.. go 320
13:10.27tzangerrobl^: well these guys have lots of calls, but 99% is only simple call a guy or receive a call.  transfer to other dept sometimes.  very very very little conference
13:10.31tzanger[TK]D-Fender: yes, I've used that :-)
13:11.01robl^tzanger: then a 430 or 320 would suffice.
13:11.23tzangerok, well I know what I'll do
13:11.33tzangerI'll print out the specs, show hte guys and they make the decision, since it makes 0 difference to me
13:12.19filemmm Polycom
13:12.56robl^I prefer to use meetme over a phone's built-in 3-way conference.  As my Asterisk server is colo-ed.. and my phones are spread out over the country..  most phones are hosted in someone's home with just a DSL.
13:13.19*** join/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg)
13:13.45littleballhello, who can send me the cisco as5300 configuration file to me to connect to asterisk box?
13:13.50littleballi am new to as5300
13:13.55littleballover E1/PRI
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13:15.11JTrobl^: work from home telemarketing?
13:15.14[TK]D-Fenderrobl^: 2 DN is fine on 320/330 as long as you aren't expecting to be interrupted DURING a 3-way call all the time.
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13:16.07robl^[TK]D-Fender: right..  well they NEVER use 3way calls..  its either meetme or simple direct calls with transfer
13:16.32[TK]D-Fenderrobl^: Then the only point for the 430 is dual-powered inclusive.
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13:18.20robl^JT: nah!  It's a group of people that are volunteers that have normal full time jobs.. but need to keep up the image of being involved "full time" with the volunteer organization"
13:18.55JTi see
13:19.01robl^[TK]D-Fender: true.  but the power adapters are $20.  I just figure it in with the purchase.
13:19.03FreezeShello
13:19.14FreezeSit's possible to setup a minute counter in asterisk ?
13:19.28FreezeSI have a line with cheap mobile calls, but only 200 minutes/month
13:19.51FreezeShow can I setup the dialplan so that after the 200 minutes expire to route through the more expensive line ?
13:21.27[TK]D-Fenderrobl^: Yeah but I don't like add'l purchases, and I'd sooner buy PoE Injectors than bricks.  Injectors are recyclable
13:22.11[TK]D-FenderFreezeS: make your calls out that line count their duration from start to finish, and just store your usage in AstDB or something.
13:22.22*** join/#asterisk anthm_mobile (n=anthm@dhcp64-134-34-220.bwic.chi.wayport.net)
13:22.30[TK]D-FenderFreezeS: on hangup, add your billsec to the counter.
13:23.13FreezeShow can I make them count the duration ?
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13:26.08berktrhello
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13:26.17berktrhow can i enable jitter buffering for asterisk 1.4 SIP protocol
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13:29.02nephflit is official, you can unlock vonage wrtp54gs and use them to replace a tdm2400 that a dumbass at your company fried
13:29.31stoffell_wany advice on how to lower the volume DTMF-keys generate on Thomson ST2030 phones, since the last firmware? (also when dialing or transferring calls)
13:32.25[TK]D-Fendernephfl: How many ports does it offer?  TDM2400P is only if you know you're exceeding 4 lines, and I'd say 8 minimum....
13:32.27berktrhow can i enable jitter buffering for asterisk 1.4 SIP protocol
13:33.44*** join/#asterisk skeffling (n=andrew@andrew.1ec.aaisp.net.uk)
13:34.52nfi|ermeshow can i check if asterisk-addon is installed ?
13:35.15skefflingHi, I'm following the steps to install a B410P, and get to  dmesg | grep Digium and I get "HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring...". this is the first time I've installed this card, and I'm not sure where I've gone wrong! any ideas?
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13:40.18[TK]D-Fendernfi|ermes: Go see if the .so's are in your modules folder.
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13:45.50NovceGuruTodays task: Get asterisk running (never ever touched it before) so internal clients can call each other
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13:46.34filemmm Polycom
13:46.36fileeep
13:46.50Uatecskeffling, what OS are you installing it on, and which version fo asterisk?
13:46.50Uatecof
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13:48.16skefflingUatec: latest Centos with latest asterisk 1.2
13:49.01*** join/#asterisk TechBlazer (n=Tim@70.88.27.130)
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13:49.23skefflingUatec: asterisk-1.2.19 and centos 5 to be more specific
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13:50.14lesouvageTk dfender: yesterday you accused me of "nagging the channel". I actually don t understand why my question was qualified this way.
13:50.43TechBlazerHello all. Could anyone recommend a good place to purchase IP telephones?
13:50.48nephflwe had 8fxs and 4 fxo...i had to get something up last night..so i put in 4 wrtp54g's (8fxs) and a sip trunk
13:50.54[TK]D-FenderTechBlazer: Depends where you are located
13:51.11TechBlazerI'm in Alabama
13:51.29[TK]D-Fendernephfl: I wouldn't touch a previously locked device personally, esp as you can get new normal ones jsut about as cheap
13:51.44[TK]D-FenderTechBlazer: www.telephonydepot.com <- jsut about everything you could need
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13:52.05nephfli had to do this to get something up temporarily last night, im ordering a replacement tdm
13:52.17nephfland going with polycom ip extensions
13:52.19[TK]D-FenderTechBlazer: lowest prices on jsut about everything as well, except ofr one other place I HAVEN'T had dealings with before.
13:52.36[TK]D-Fendernephfl: Good choice.  What models?
13:52.46nephfl601s
13:53.06[TK]D-Fendernephfl: Dear God, what are the user's doing with them?
13:53.09Uatecskeffling, oh dear, well i only have experience with ABE 1.3 and Pound Key Linux i'm afraid
13:53.51nephflwhat do you mean?...they are switching from meridian phones with a key type system, so they wanted lots of line appearances
13:54.00filerun away
13:54.01filerun away now
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13:54.09skefflingUatec: ok... maybe someone else can help :-)
13:54.10[TK]D-Fendernephfl: Doesn't work that way, those phones are massive overkill
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13:54.30nephflthey also wanted heavier phones than what i first put in
13:54.32[TK]D-Fendernephfl: * & SIP are NOT meant to work like key-systems.
13:54.35TechBlazer[TK]D-Fender: Nice site. I've been looking at voipsupply.com and simlesingal.com for the Polycom IP 650, but the site you recommended has it for $20 less. Have you had any dealings with their customer service?
13:54.47[TK]D-Fendernephfl: And what did you put in first, GS?
13:55.11[TK]D-FenderTechBlazer: No more than "here's what I want" and I got it the next day
13:55.17nephflGS? analog aastra phones (the other end of the spectrum)
13:55.22TechBlazer(simplesignal.com)
13:55.23[TK]D-Fendernephfl: ICK
13:55.41bintuthello all..  anyone here uses the Starhub's Digital Voice PSTN line?
13:55.51[TK]D-Fendernephfl: 601 = massive overkill.  IP 320/330 is plenty for 99% of the users out there
13:56.01TechBlazer[TK]D-Fender: Great, thanks for the info.
13:56.05*** join/#asterisk ELBunce (n=erik@kde/developer/bunce)
13:56.16nephfli suggested the 320 but they wanted paging/intercom
13:56.25nephfland more buttons/line appearances
13:56.39[TK]D-Fendernephfl: EVERY Polycom supports that...
13:56.56[TK]D-Fendernephfl: And who actually HAS multiple IDENTITIES?
13:57.03nephfli wonder if i can show a busy lamp or appearance for the analog trunks without screwing Everything up
13:57.15nephflthey wanted visual call control
13:57.18[TK]D-Fendernephfl: Normally a user has 1 "extension"
13:57.24nephflyup...
13:57.45[TK]D-Fendernephfl: I'm not sure if you got this, but you can FORGET about the "line" concept on Meridian, we're "not there".
13:57.55nephflif they would have had a 1 line phone with a few lights that was big and heavy that would have been ok...but i didnt see that
13:58.47nephfland the boss kept saying that they dont want little girly phones they want a "mans" phone..and are willing to pay for it
13:58.56nephfllol
13:59.06NovceGuruadd some weight to the handsets
13:59.09[TK]D-Fendernephfl>i wonder if i can show a busy lamp or appearance for the analog trunks without screwing Everything up <- do your research first.  What little we have now is a half-baked attempt.
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13:59.16*** mode/#asterisk [+o anthm] by ChanServ
13:59.24[TK]D-Fendernephfl: Got PoE?
13:59.29nephflthats what i told them...i said i can get you a cheap aastra phone with lights and strap a brick to it
13:59.38[TK]D-Fendernephfl: Aastra = ick.
13:59.56[TK]D-Fendernephfl: Functional, but too damn light, poor call handling, etc, and thats their *IP* series
14:00.04[TK]D-Fendernephfl: And you gave them ANALOG.
14:00.07[TK]D-FenderICK!
14:00.25*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)
14:00.32nephflyeah...reasearch with asterisk is difficult to gauge...i thought setting up the tdm2400 would be a piece of cake from the docs, but 1 rma and 1 mb later i find it isnt
14:00.39*** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it)
14:01.20[TK]D-Fendernephfl: Should have asked in here first... oh well... batting 1000.....
14:01.29nephfli did
14:01.53fileif your client expects a key system that acts almost exactly like their Meridian... run away
14:01.59nephfland i suggested the 320 to the customer...they wanted bigger, more lines, so I said, we can do that
14:02.23[TK]D-Fendernephfl: Problem is you doo not even UNDERSTAND the concept of "lines" as they apply betweent he phone & *.
14:02.39nephflI told them that it would be like the meridian system, but they could get visual call control and not deal with the analog line flash crap
14:03.06_VoiceMeUp_COMlol waht you guys think of exceeded traffic: 0.60$/GB
14:03.09nephfli think im gettin a pretty good handle on it...they dont, but im doing what i can
14:03.22[TK]D-Fendernephfl: Phone's don't have "lines", they have call appearances, and speed-dials.  There is no key system functionality yet, SLA does not actually exist yet, and you are barking up the wrong tree
14:03.28_VoiceMeUp_COMthat expensive you ?
14:03.29*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
14:03.48*** join/#asterisk TalknTalk (n=targus@64.56.64.29)
14:03.51nephflok...so the "new" sla crap doesnt work...ok
14:04.14[TK]D-Fendernephfl: Correct.
14:04.15nephflcan i get a "buddy" or something that will show me the state of an analog trunk?
14:04.18bintuti don't know what's the problem with my setup. i subscribed for a caller id feature from my telco yet on my asterisk box, i don't get a caller id. anyone cares to check this? ==>  http://paste.debian.net/31423
14:04.22nephflor appearance
14:04.26nephflor lamp
14:04.34[TK]D-Fendernephfl: Yes, but you can't hijack the calls, so whats the point?
14:04.39*** join/#asterisk gerwinin (n=gerwinin@ip5457b30e.direct-adsl.nl)
14:04.52TalknTalkhello everybody
14:04.55nephflthey just want to see the analog lines to look for available lines
14:05.06TalknTalkneed help in zaptel-trunk
14:05.17*** join/#asterisk pigpen (n=mark@fw.seamans.cc)
14:05.17nephflso they know when to expect to have to wait..
14:05.20JTnephfl: why the bloody hell are you using analogue lines anyway?
14:05.27nephfllol
14:05.36Erreveryone likes to bang rocks together in the cave
14:05.37nephflthis company uses a TON of analog lines
14:05.55[TK]D-Fendernephfl: I'm not sure you've grasped this concept... YOU CAN'T PICK LINES WITH A STUPID BUTTON.  That concept DOES NOT EXIST.
14:05.56JTwhat knobs
14:06.00nephflthey have 10 locations with 2 analog lines each for credit card processing
14:06.10JTbut credit cards are different
14:06.12nephflevery computer in the place had a modem and separate line when i came in
14:06.18JThave nothing to do wirth asterisk
14:06.20[TK]D-Fendernephfl: Unless you are creating completely seperate registrations EACH.
14:06.22JTargh
14:06.25pigpencould someone describe "limitonpeer=" option in the sip.conf?  some of the doc's are a bit confusing...
14:06.25JTwhat clowns
14:06.30bintuti'm using the digium's dev kit with 1 fxo and 1 fxs here
14:06.38JTnephfl: how many CO lines on the meridian?
14:06.46nephfljust 4
14:06.46TalknTalkanyone experienced with zaptel-trunk ?
14:06.52JTi see
14:07.09bintutwhat do you guys think with my simple configs at http://paste.debian.net/31423 ?
14:07.21nephfli was just saying this is a long process of moving them to VPN's and faster digital connections
14:07.27[TK]D-Fenderbintut: in SG?  Could be your regions CID signalling isn't well supported...
14:07.35*** join/#asterisk javar (n=javar@69.79.134.24)
14:07.54JTnephfl: ideally phone lines move to PRI too
14:08.28nephflyeah, i really havent looked into changing their trunks...this phone system change came because of lightning strike
14:08.52JTdestroyed meridian?
14:09.12TalknTalknobody ?
14:09.30nephflnaa.. just killed their voicemail...so, we decided to just throw it out for asterisk
14:09.47*** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net)
14:09.53[TK]D-FenderTalknTalk: Sure.... based on WHEN?  And whats your actualy problem/qeustion?
14:10.11*** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg)
14:10.17nephfli know it would have been cheaper to go a bunch of other directions...but we were looking for an opportunity to replace the whole system
14:10.22skirmishaguys how can i flush menuselect?
14:10.27bintut[TK]D-Fender: yes, i'm in SG.
14:10.29bintut# cat /etc/modprobe.d/wctdm
14:10.29bintutoptions wctdm opermode=SINGAPORE
14:10.36[TK]D-Fenderbintut: What ver of * / Zaptel?
14:10.50*** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net)
14:10.55sci_05morning all
14:11.16TalknTalki m having problem in compiling zaptel-trunk
14:11.20[TK]D-FenderTalknTalk: pastebin the output of your failure (ALL OF IT.
14:11.22[TK]D-Fender~pb
14:11.23jboti guess pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
14:11.24[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^
14:11.43*** join/#asterisk NightTiger (n=derekm@ottawa-hs-64-26-171-99.s-ip.magma.ca)
14:11.44bintut[TK]D-Fender: the latest versions from debian unstable on debian etch. basically, i built the debian unstable sources of asterisk and zaptel on my debian etch
14:12.00[TK]D-Fenderbintut: ..... I want... NUMBERS!!
14:12.10fileuh oh
14:12.14file[TK]D-Fender is becoming angry
14:12.15TalknTalkmy problem is http://pastebin.ca/590312
14:12.18*** part/#asterisk NightTiger (n=derekm@ottawa-hs-64-26-171-99.s-ip.magma.ca)
14:12.22[TK]D-FenderHULK SMASH!!!!!!
14:12.40*** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com)
14:12.48rue_mohrany suggestions for a sip phone for linux?
14:12.50TalknTalkTKD : http://pastebin.ca/590312
14:13.11bintut[TK]D-Fender:  http://paste.debian.net/31424
14:13.15[TK]D-Fenderrue_mohr: Ekiga
14:13.23*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:13.24Mercestespolycom
14:13.50TechBlazerI'm going to have 12-20 extensions with 6 outgoing lines to a VoIP provider so we can call land lines. I'm probably going to purchase a Polycom phone based on the recommendations I've gotten from here. The 601 and 650 can handle from 6-42 lines. Are these the models I need to be looking at purchasing or would another model work?
14:13.51rue_mohrsoftphone
14:13.52[TK]D-Fenderbintut: exten => s,1,NoOp(Caller ID is ${CALLERID}) <- That var is GONE.
14:14.01nephflso, i do have another problem i know nothing about... we had 4 analog rollover lines i asked them to forward them to a did for our voip provider...we only get one call the second call rings busy
14:14.15[TK]D-FenderTechBlazer: Massive overkill.  You don't need a multi-line phone like that.
14:14.26[TK]D-FenderTechBlazer: IP 320 = $95 and will do the job.
14:14.42[TK]D-FenderTechBlazer: Its a question about how many simultaneous CALLS and IDENTITIES you need.
14:15.39bintut[TK]D-Fender:  that's why i don't know why the caller id doesn't appear
14:15.41[TK]D-Fendernephfl: Go bitch to your telco, its THEIR job.
14:15.46skirmishanephfl what is your problem exactly
14:15.54TalknTalkTK]D : waiting
14:15.56bintut[TK]D-Fender: is there any problem with my config?
14:16.03[TK]D-Fenderbintut: ${CALLERID} no longer EXISTS in 1.4! ${CALLERID(num)} <-------------
14:16.15bintutah..
14:16.22[TK]D-FenderTalknTalk: I have nothing to say for your error, perhaps someone else will.
14:16.32TalknTalkTK : ok
14:16.38bintut[TK]D-Fender: what's the right config for that line?
14:16.45JTTalknTalk: do not use only VoIPoI for PSTN connectivity
14:16.51*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
14:17.00[TK]D-Fenderbintut: I just WROTE how you have to call the new function.  Go caffeinate....
14:17.37*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
14:19.48bintut[TK]D-Fender:  thanks. i got it working.
14:19.58[TK]D-Fenderbintut: It's a miracle!
14:20.13file[TK]D-Fender: Be nice to the general #asterisk population... they might rebel
14:20.13pigpenif I add limitonpeer=yes, will this affect anything other than let hints/presence work?
14:20.23sashionastribank's seem to be very temperamental
14:20.24[TK]D-Fenderfile: Come join me in my marathon!
14:20.34filepigpen: no, it only touches the hints stuff
14:20.40pigpenk...
14:20.42*** join/#asterisk murr4y (i=murray@valhall.samfunnet.no)
14:20.46[TK]D-Fendersashion: You shouldn't talk about them like that.... they might shatter on you....
14:20.56pigpenthanks...I was reading many entries that seem to be as confused as I am...
14:21.01Mercestesfile:  Let him be mean to all the n00bs.  They msg me later and tell me how much ass he rides and then they pay me to fix it for them.
14:21.11[TK]D-Fenderlol
14:21.14MercestesWorks for me, works for them, works for fender...i'd say it's a perfect setup
14:21.14sashion[TK]D-Fender, i beg for someone to prove me otherwise...
14:21.33[TK]D-Fendersashion: Please note that I did not DISAGREE with you ;)
14:21.53[TK]D-Fendersashion: Nor did I AGREE ;)
14:22.07*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
14:22.08sashion[TK]D-Fender: ofcourse not ;)
14:22.17skirmishaguys snmp on branch version, is it supported?
14:22.18MercestesWhat's an astribank?
14:22.29negativeducktalk, n if you haven't found it yet your missing a part of your kernel includes... /usr/include/linux/page-flags.h not existing... look at the source tree for your running kernel and you should be able to copy <kernelsourcebasedir>/include/linux/page-flags.h into /usr/include/linux/
14:22.30[TK]D-FenderMercestes: USB channel-bank/etc
14:22.37MercestesA usb channel bank?
14:22.42sashionyup
14:22.42MercestesVoUSB?
14:22.42fileskirmisha: what version?
14:22.50skirmisha1.4
14:22.52[TK]D-FenderMercestes: Get googling Lazy-Boy!
14:22.57skirmishai see agent.o is compiled
14:22.57sashionlol
14:23.04skirmishabut seems not loaded or something
14:23.05MercestesI am I am
14:23.07*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
14:23.11sashionwww.xorcom.com
14:23.14fileyes, it's supported... but you might have to configure it
14:23.35skirmishamy snmp server is configured
14:23.43skirmishabut snmp on asterisk seems to not work
14:23.49skirmishado i need to load something?
14:24.05skirmishaor how can i check that snmp on asterisk is running/supported
14:24.22Mercestesohh.
14:24.25MercestesThat's nothing.
14:24.27filethere's a document in doc, snmp.txt that has some information plus res_snmp.conf configuration file, and res_snmp.so has to be loaded
14:24.31MercestesI did VoString and Two cans.
14:24.55skirmishalet me see
14:25.36MercestesSounds like an overly expensive device based upon a retarded design meant to achieve the impossible task of interfacing an unreliable system with yet another unreliable system in a magically reliable way.
14:26.13sashionMercestes: You're very optimistic
14:26.20Mercestessashion:  I try to be.
14:26.34[TK]D-FenderMercestes: It has some added features and advantages, and offered affordable BRI in a modular way, but I have not personally found a way to validate its purchase HERE.
14:26.51sashionMercestes: Don't worry, secretly, I feel the same ;)
14:26.57Mercestessashion:  From my experience tho, what your going to see is the original PBX vendor going "it's not my equipment" and the astribank guys going "our stuff is fine" and everything on your end will look perfectly fine to you.
14:27.05skirmishafile i see res_snmp.so module when i type show modules within asterisk
14:27.10skirmishawhich means module is loaded
14:27.25sashionMercestes: lol, de-ja-vu anyone?
14:27.28Mercestes[TK]D-Fender, that whole, "plugs in via USB" tho...I can barely get my mouse to work over USB..dunno about voice.
14:27.37Mercestessashion:  ;)
14:27.56negativeduckusb is the devil! long live serial mice!
14:27.57sashionMercestes: truth be told, a single hook-flash brings an Astribank to its knees
14:28.00negativeduckerm *cough8
14:28.45*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net)
14:29.14*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
14:29.38[TK]D-FenderMercestes: It works apparently....
14:31.02bintutgtg
14:31.33Mercestes...
14:31.51*** join/#asterisk lorinc (n=ang@pool-2000.adsl.interware.hu)
14:31.53Mercestes>.>
14:32.29Mercestesnegativeduck, Yes!  Back when mice had it's own dedicated port to plug into!
14:33.16*** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-93c5c6db10be6608)
14:33.29*** join/#asterisk vlt (n=dm@port-87-234-95-227.dynamic.qsc.de)
14:34.58BSD_Techyou mean before usb
14:35.06BSD_Techand wireless laser  mice
14:35.27MercestesBSD_Tech, yes.
14:36.13Mercestes<negativeduck> usb is the devil! long live serial mice!
14:36.45*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
14:36.50BSD_TechUSB is serial you dummy
14:37.19BSD_Techthey just made it more of Universal port
14:37.52BSD_Techand it allows more then just 1 device at a time much much better
14:38.03*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
14:38.07BSD_Techdood been smoking to much crack
14:38.15MercestesBSD_Tech, and a compustion engine is the same as a steam engine...
14:38.22Mercestesexcept..it uses gas instead of water....
14:38.27Mercestesand it has better efficiency...
14:38.32Mercestesand doesn't use coal...or slaves...
14:38.35Mercestesand it goes faster.
14:39.03Mercestes=/
14:39.09Mercestespass the crack, please.  :)
14:39.10BSD_TechI want a car that runs on osmosis
14:39.43[TK]D-FenderBSD_Tech: They do.... they just are selective as to what they absorb for fuel ;)
14:39.45BSD_TechHamster Power forever
14:40.02BSD_Techlol
14:40.11[TK]D-FenderBSD_Tech: "The light is on, the wheel is spinning... but the hamster is DEAD."
14:40.20skirmishagot it working
14:40.31BSD_Techwhat about a care that runs on cocacola
14:40.38BSD_Techlol
14:40.46BSD_Techcare/car
14:40.52BSD_Techfat fingerd
14:41.10mvanbaakI want a car that runs on urine
14:41.23[TK]D-FenderBSD_Tech: ..... actually... thats not much of a SAVINGS....
14:41.37JTBSD_Tech: usb is not much much better
14:41.38BSD_Techtrue
14:41.41[TK]D-Fendermvanbaak: Living proof that people who are full of shit get it all...
14:41.42JTyes usb is serial
14:41.46JTit is not rs-232 however
14:41.52mvanbaaklol [TK]D-Fender
14:41.53JTrs-232 is very reliable
14:41.54JTusb not so
14:41.57FreezeSusb IS much better
14:42.10JTnot for reliable low bandwidth comms, FreezeS
14:42.30FreezeSJT: did it cause you any trouble ?
14:42.31BSD_TechI never said usb was better then rs232 seriel
14:42.41[TK]D-FenderSerial is good for super simple relay switch circuit building, and embedded projects.
14:42.43JTit's good for connecting toys temporarily
14:42.44FreezeSfor example, take the mouse serial port
14:42.46JTusb
14:42.52BSD_TechI just said it allows more devices
14:42.52FreezeSin windows at least, the implementation sucks
14:43.03FreezeSif you plug it out, you need to reboot to have mouse
14:43.06FreezeSsame for ps2
14:43.21FreezeSkeyboards work without restart though...
14:43.27mvanbaaknot on ps2
14:43.33FreezeSon ps2 they work
14:43.34BSD_Techwho uses windows
14:43.35*** join/#asterisk anthm (n=anthm@dhcp64-134-34-244.bwic.chi.wayport.net)
14:43.35*** mode/#asterisk [+o anthm] by ChanServ
14:43.42FreezeSwell, some people are gamers :)
14:43.51FreezeSand some are mostly windows programmers
14:43.55negativeduckBSD, hahaha ok yes if you want to get technical, I was more so making a joke :)
14:43.55mvanbaakgamers should use a game console
14:44.02JTsome are mostly windows programmers?
14:44.04JTwhat?
14:44.10FreezeSvisual c++
14:44.14JTwhat about it?
14:44.21FreezeSif that would run on linux, it would be marvelous :)
14:44.27negativeduck(sorry for the delayed response got pulled away) :)
14:44.29BSD_Techlinux
14:44.33BSD_Techlinux get real
14:44.34JTit's not the most used or anything
14:44.41BSD_TechRun Freebsd
14:44.44[TK]D-FenderYeah, I've got a USB wireless KB/trackpad for my HTPC server which DOESN'T WORK with my Linux install.  It works in the frggen BIOS, but not once the kernel boots.
14:44.54BSD_Techit will change your life forever
14:44.54UatecHey, does anybody know how to set distinctive ring on an SPA922?
14:44.57JTlinux has competant C and C++ compliants, no need for VC++
14:45.10[TK]D-FenderUatec: Its on the WIKI.  Go look.
14:45.21FreezeSJT: no, it's not quite true
14:45.37FreezeSvisual c++ has everything integrated, including the debugger
14:45.43JThooray!
14:45.53[TK]D-FenderFreezeS: Heck, for completeness, even the BUGGER ;)
14:45.54FreezeSwithout this integration, things that normally take 5 seconds take whole minutes
14:46.09*** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com)
14:46.10BSD_TechTK i have had that issue before and it has to do with linux not loading the usb kbd driver at boot time
14:46.24*** join/#asterisk alrs (n=lars@pozug.com)
14:46.29BSD_Technever had that issue with bsd
14:46.33JTFreezeS: 5 seconds... minutes, to an incompetant programmer
14:46.48JTBSD_Tech: no-one cares about your bsd biggotry
14:46.55FreezeSwell, a competent Visual C++ programmer is a noob on linux
14:47.06BSD_TechI am not a biggot
14:47.10JTso why do we care about VC++ users again?
14:47.15JTthey're not a majority
14:47.48[TK]D-FenderBSD_Tech: Do believe you'd be able to assist with my getting it operational?  I'm running CentOS 4.4 on that box.  Stock complete install.
14:47.51FreezeSbecause they use windows
14:47.53JTBSD_Tech: it's irritating when people are talking about something on linux and others come in and say "omg bsd, omg, it's so awesome"
14:47.59FreezeSto make programs for other people that use windows :)
14:48.09[TK]D-FenderI'm not a bigot.... I hate all people equally ;)
14:48.09*** join/#asterisk Waverly360 (n=Waverly3@dhcp64-134-34-220.bwic.chi.wayport.net)
14:48.15JTFreezeS: i don't think VC++ is even the most developed language under windows
14:48.30FreezeSno, it's not
14:48.33BSD_TechI am not racciest I hate all humans
14:48.38FreezeSVC# is
14:48.44[TK]D-FenderBSD_Tech: Umm... that doesn't work ;)
14:48.49JTjava too
14:48.51littleballhi, how to reset a specific zap span over cli?
14:48.54JTin corp software
14:48.55FreezeSbut VC++ is the language where you could make the fastest windows apps
14:49.15BSD_Techtk let me look the kbd/mice up
14:49.16*** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net)
14:49.17JTsure, but most people don't care about coding efficiently for windows these days
14:49.21JTthey want it to be fast
14:49.26BSD_Techwhats the brand name
14:49.30JTcoding it fast
14:49.37JTthe program, they often don't care
14:49.38FreezeSI know, but some care :)
14:49.45BSD_Technot in here
14:49.53FreezeSand a big problem of VC# is that you need a runtime
14:49.57FreezeSin VC++ you don't
14:50.03docelmoSay whats a good voip provider that offers solid termination with DID's and Toll Free?
14:50.03BSD_Techtake it to #windows-piss-poor-coding
14:50.05Corydon76-workIsn't "Hello world" on Windows currently taking 1MB on disk?
14:50.13skirmishaany idea if there is snmp support for registered users?
14:50.21FreezeSif you code it right, you don't need any non-windows dlls
14:50.21JTFreezeS: so any more reasons why usb is uber?
14:50.42BSD_Techbrb
14:50.47FreezeSyeah, I got carried away :)
14:50.49negativeduckIf you build the kernel again and update the ramdisk image you will probably find that you can resolve the USB problems, you can also try making sure the USB module and HID drivers are in /etc/modules incase the autoloader isn't picking it up.
14:50.51[TK]D-FenderBSD_Tech: its a http://www.adesso.com/products_detail.asp?productid=281
14:50.51BSD_Techtk I need a name an model number
14:50.54BSD_Techok
14:50.54sashionJT: Cause you can make a coffee warmer out of a USB port
14:50.58[TK]D-FenderBSD_Tech:  :)
14:51.02BSD_Techbrb
14:51.04FreezeSyeah, and you can charge your PDA :D
14:51.18JTit'd be more useful if usb provided -48VDC
14:51.24FreezeSit will
14:51.29FreezeSit's called powered usb
14:51.35JTeh?
14:51.44FreezeSlook it up :)
14:51.54FreezeSthere are several standards for powered usb
14:51.58FreezeSsome will provide 12v
14:52.12FreezeSand some more (I assume 48)
14:52.16JTi meant usb, that's actually available on a computer, a real computer :)
14:52.17*** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com)
14:52.25sashionlol
14:52.34sashiongotta be specific at times
14:53.08FreezeSwell, the jack is smaller
14:53.15FreezeSyou can power your cell/pda
14:53.15negativeduckNow if that were true that would be entertaining.
14:53.21FreezeSyou can have multiple devices on a port
14:53.22andydFreebie service worth every penny it costs - but please can someone check that the advice at http://www.tollfreeuk.com/ for asterisk is reasonably correct ?
14:53.25FreezeSyou can use hubs
14:53.50*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
14:53.52JTFreezeS: so in other words, this powered usb doesn't exist yet
14:53.56sashionyou think if I bring my astribank to Astricon, Xorcom will fix it for me ?
14:54.02FreezeSas I said: it will :)
14:54.28sashionJT: it does.. just might destroy your PC when you tap on a 48VDC power unit
14:55.01JTFreezeS: if my computers don't provide it, it's not useful
14:55.35JTseems that small amounts of power is one of the more useful things a usb port provides ;)
14:55.49FreezeSstill, in it's current form, USB is WAAAAY better than serial
14:55.59JTin what respects, for what uses?
14:56.00sashionJT:ooo those little USB lights are also quite useful...
14:56.07JTsashion: power!
14:56.17FreezeSjt: for example, I have a keyboard that has an usb hub
14:56.30FreezeSand I connect to it the mouse and the headset
14:56.37FreezeSand I have only one cable to the computer :)
14:56.38sashionJT: Yes, well power needs to be exploited by some device to do something useful
14:57.00JTusb headset, yeah i'll just go right ahead and avoid that :)
14:57.04FreezeScould I do that using a serial port ?
14:57.12FreezeSusb headsets are great
14:57.22FreezeSI work in a callcenter :)
14:57.48BSD_Techbbiab time to water the yard
14:57.56JTno, i'd rather use an audio headset
14:58.03JTthat connects to audio sockets
14:58.11vltHello. I'm running a 4xS0 card with the "bristuffed" version of asterisk. I'm currently running the module qozap with "ports=0" (TE TE TE TE). Now I want to change it to (TE TE NT NT). Is this "ports=2"? What about zapata.conf which now got "signalling=bri_cpe_ptmp"? How do I set signalling for the NT ports to "bri_net_ptmp" or "bri_net"?
14:58.34FreezeSour computers have crappy onboard soundcards
14:58.43FreezeSthe usb headesets are WAY better
14:58.54FreezeSfrom the sound quality point of view
14:59.00FreezeSand usability
14:59.20FreezeSyou only have one connector and the (stupid) agents can't put it wrong
14:59.23*** part/#asterisk TechBlazer (n=Tim@70.88.27.130)
14:59.25JTyeah using a hardphone would be better there
14:59.41FreezeSno because that would tie a hand and ear
14:59.51JTwith headsets......
14:59.53sashionhardphone with headset
14:59.59FreezeSwhy ?
15:00.05sashioncause its better
15:00.05FreezeSsoftphones are cheapre
15:00.06FreezeScheaper
15:00.15sashionLMAO
15:00.18FreezeSsince they have a computer anyway
15:00.20sashionthat explains everything
15:01.28*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
15:01.36JTso do you have a reason that usb is better, for a non peripheral device?
15:01.46nephflso, what do you have against unlocked linksys devices anyhow?
15:01.47*** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com)
15:02.02[TK]D-Fendernephfl: Doesn't USB IMPLY "peripheral" ? :)
15:02.04brettnemhey all.. can anyone recommend a decent VoIP Provider that can do IP authentication. ie: I don't want to have to register, I'll give them a static IP
15:02.24[TK]D-Fenderbrettnem: ACN <-
15:02.25nephfli think you meant jt
15:02.27FreezeSJT: have you seen "Monty Python's Life of Brian" ?
15:02.43JT[TK]D-Fender: i dunno, apparently some people connect channel banks and hard drives and other weird things with usb
15:03.04brettnem[TK]D-Fender: Thanks, I'll check them out
15:03.09[TK]D-FenderJT : And some people are morons.... amazing how the 2 demographics collide, no? :)
15:03.17FreezeSthere was a discussion about the romans
15:03.20JTheh
15:03.29Mercestes"Romans go home"
15:03.32FreezeSa guy was saying: The romans did NOTHING for us !
15:03.48FreezeSand someone replied: They built the baths
15:03.52Mercestes"You are all individuals" "I'm not"
15:04.02FreezeSfirst guy: Oh, and except the baths, what did the romans do for us ?
15:04.08FreezeSother: Roads !
15:04.12Uatec[TK]D-Fender, the wiki's explaination doesn't work :(
15:04.36FreezeSfirst: Oh, the roads are a given. So, except the baths and roads, what did the romans do for us ?
15:04.52FreezeSother: Schools
15:05.01[TK]D-FenderUatec: PASTEBIN <----------
15:05.03[TK]D-Fender~pb
15:05.04jbotpb is, like, a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
15:05.17FreezeSfirst: So, except baths, roads and schools, what did they do for us ?
15:05.25*** join/#asterisk anthm (n=anthm@dhcp64-134-34-244.bwic.chi.wayport.net)
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15:05.29stoffell_wdoes getting the error "Didn't get a frame from channel" mean the call is lost / hungup ?
15:05.30andydthey invented pastebins, freezes
15:05.32FreezeSand so on, until the list got really big
15:05.44FreezeSyeah, but this is supposed to be a _live_ discussion
15:06.06andydMonty Python is genius.
15:06.10FreezeSso, JT, you're doing exactly the same
15:06.29JTFreezeS: that was about the worst analogy i've ever heard
15:06.34*** join/#asterisk sof76 (n=chatzill@u92.num.math.uni-goettingen.de)
15:06.39JTfailing to see the comparison
15:06.55Mercestes"I will wead off all the pwisonews we have.  Wupert, Wupaul, Wowdewick, Wobewt"  "free Brian!"  "Yes!  Free Brian!"  "Ok, do we have a Bwian?"
15:07.22sof76Hi guys, is it possible to make asterisk listen on 2 ports for sip registrations, for ex 5060 and 80? Thanks
15:07.36MercestesJT:  well, your saying, "other than the power, speed, universaibility, etc. etc. etc, and perherpherials, how is USB better?"
15:07.52FreezeSwell, excepting the fact that you can connect multiple devices, can use it for power, has a smaller connector, it's A LOT faster (and probably many other advantages), how is it better ?
15:07.56[TK]D-Fendersof76: set your sip.conf entry to "pot=80"
15:07.59MercestesJT:  I guess you want a white sheet comparison of theorhetical technologies.  Also, granting, that USB is supposedly unreliable.
15:08.00[TK]D-Fendersof76: set your sip.conf entry to "port=80"
15:08.02Uatec[TK]D-Fender, http://rafb.net/p/takc6M80.html
15:08.03JTMercestes: it's not good for important applications
15:08.17andydnoone at all is free to take a look at www.tollfreeuk.com to see if the advice i put regarding how to configure the service against asterisk is safe ?  There's free uk calls to millions of uk / +44 destinations via it ;-)
15:08.19FreezeSJT: what would be an example of an important application ?
15:08.33sof76but then I can not register on port 5060, no?
15:08.36JTMercestes: yeah connect that mouse, mug heater, etc, but not something important
15:08.38MercestesJT:  Stating a fact using an opinion as a qualifier is a logical falacy. =/
15:08.39JTFreezeS: telephony
15:08.43JTFreezeS: remote management
15:08.48FreezeSUSB for telephony ?
15:08.51JTFreezeS: reflashing
15:08.52MercestesWe use USB Rariatans for remote management.
15:09.05FreezeSJT: usb IS USED for flashing microcontrollers
15:09.14JTit shouldn't be :)
15:09.19FreezeSwhy ?
15:09.22MercestesI also have USB wireless telephone headsets
15:09.32JTFreezeS: the astribank uses usb, so yes, telephony
15:09.44sof76do you mean I put bindport=5060 and port=80 ?
15:10.02JTFreezeS: because with some devices, if the connection fails mid flash, you're stuffed
15:10.04Mercesteskeyboards are important.
15:10.16JTnot as important as telephony
15:10.23FreezeSJT: that happened a lot to me using serial
15:10.24Mercestesoh yes...
15:10.35Mercestes21st century, keyboards are not important. =/
15:10.41FreezeSso, from the reliability point of view, for me they're the same
15:10.46JTFreezeS: i guess you need to do a lot of reflashing in the callcentre?
15:10.52JTyour experience is clearly limited
15:10.57FreezeSno, I don't work with microcontrollers anymore
15:11.07JTso what about remote management? why does everyone avoid usb like the plague?
15:11.12FreezeSbut I made an mp3 player for my graduation project
15:11.22MercestesWe don't.  We use USB Raritans
15:11.31JTMercestes: crack :P
15:11.42JTMercestes: what servers has USB LOM?
15:11.45JThave
15:11.48MercestesJT:  I'm not on a anti-USB campaign.  :P
15:11.57Mercesteswhat is LOM?
15:12.00JT...
15:12.05JTLights Out Management
15:12.29MercestesUm, all of our UPS's have a USB connector to them.
15:12.49JTa ups is not a server
15:12.51Mercesteshell, all our servers don't even have PS/2 anymore, they've been replaced with USBs.
15:13.04JTall decent servers still come with rs-232
15:13.07cy303Anyone have any experience with the Nokia N800 and Asterisk?
15:13.10JTfor LOM and general
15:13.15Mercestesthat's another opinion.
15:13.22Errsince USB is a master-slave link (host-side polled) instead of a peer-to-peer like serial, there are certainly applications in which it isn't as suitable as others
15:13.22JTit's an absolute fact
15:13.22coppiceUSB is used for UPS, yet when USB on a server doesn't work most people say "why would you need USB on a server"
15:13.38MercestesI haven't encountered that.
15:13.40Errhowever, in fixed datarate applications like telephony, I don't see why it would be insufficient
15:14.00JTMercestes: i haven't encountered a server with USB LOM
15:14.10ErrI'm sure that many (most?  perhaps even all?) OSes don't provide a mechanism to properly provision the bus sufficiently - but that's unrelated to the bus's abilities
15:14.11MercestesI haven't encountered anyone saying USB sucks either.
15:14.12cy303USB LOM?
15:14.13cy303wtf
15:14.14Mercestesuntil now
15:14.22negativeduckUSB LOM?
15:14.29JTcy303: never seen one
15:14.31Mercestesbut this entire discussion has degraded into retarded so...whatever.
15:14.33cy303JT: me either
15:14.48JTMercestes: dude, you don't even know what LOM is, so yeah
15:14.56MercestesI mean, basically your saying 1985 technology is better than 2007 technology.
15:15.03JTErr: also, usb connectors don't positively lock
15:15.10JTMercestes: for some applications, most definitely
15:15.22JTand there's others where neither rs-232 or usb is suitable
15:15.58MercestesI mean, yea, it's true for Asterisk but...for most other technologies, development tends to get better over time.
15:16.17vltI think I solved the ports= part: TE TE NT NT should be represented by "ports=12", but what about zapata.conf and its line "signalling"?
15:16.31coppiceI've now see a server with USB and no serial or parallel ports.
15:16.39Uatechttp://rafb.net/p/takc6M80.html <-- am i trying to set the distinctive ring on my SPA922, but it's just not setting. what's wrong with what i'm doing?
15:16.42JTvlt: read the provided config examples in bristuff
15:16.49JTcoppice: brand name?
15:16.56Mercestescoppice:  Yea, our Dell PE's now a days just have USB, no serial/paralell/ps/2
15:17.09JTwhat model PE?
15:17.20Mercestessome Dell Pcs come that way too, well, with Parallels'
15:17.34MercestesI think our 2950 is just USB.  I'd have to go look adn tell you which one for sure.
15:17.36JTpcs are different
15:17.43JTto servers
15:17.51Mercesteswell, yea...I was just saying.
15:18.06MercestesYour just made because you have to compile extra modules in linux. :P
15:18.10Mercestess/made/mad/
15:18.16*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)
15:18.16Mercestes~botsnack
15:18.16jbotMercestes: thanks
15:18.19JTit's annoying that laptops don't often have serial now
15:18.30JTi had to get a pcmcia rs-232 card
15:18.40[TK]D-FenderUatec: Try this style : http://forums.digium.com/viewtopic.php?t=3329&highlight=&sid=a4295324b3dfc30d5e07c6ee853e0608
15:18.52coppiceserial is no great loss. USB serial works fine. a lot of things require a real parallel port, though
15:19.09Qwell[]coppice: anything made in the last 10 years? :)
15:19.32negativeduckmy portmaster 200!
15:19.36negativeduckerm not.
15:20.00JTcoppice: i program 2-way radios with my laptop, if the usb adapter fails, it would cost over USD$1000 to fix the radio, not worth the risk
15:20.03coppiceyep. lots of development tools, industrial gadgets and other stuff will only work with a real parallel port. USB to parallel converters won't do
15:21.20awannabehey guys, on FXO ports what is the best setting for echo cancel to start wuth?
15:21.23JTalso, i thought the primary purpose of a laptop was to emulate a serial dumb terminal ;)
15:21.34Errthings that abuse the serial and parallel port lines (that is, they don't send real asynchronous serial out the serial port, or they bit-bang the parallel port data and status lines) require Real Ports - many devices do this, because people are lazy and wires are cheaper than interface logic :-P
15:22.14Waverly360Hey anthm, you paying attention in here? :P
15:22.17Uatec[TK]D-Fender, nope, that doesn't work either :(
15:23.00Waverly360Anyone at ClueCon paying attention in here? :)
15:23.22coppiceErr: if you want a laptop to talk to specialist things, a parallel port was ideal until they disappeared
15:23.40[TK]D-Fenderawannabe: "yes"
15:23.57Errcoppice: I wouldn't say 'ideal' - I might agree with 'sufficient' ;-)
15:24.43*** join/#asterisk berktr (n=cn@teknopet.com)
15:24.50berktrwhat do you think is the best jitter buffer time
15:24.51berktrin ms
15:24.53nfi|ermeshow can i check if asterisk-addon is installed ?
15:24.56Qwell[]berktr: 0
15:24.59MercestesI still think ISA was far better than EISA and a light year leap better than PCIx or PCIe
15:25.00berktrfor perfect quality voip conversations
15:25.02berktrif i need to
15:25.02Qwell[]oh, buffer
15:25.02coppicenow everyone has to build a USB gadget, and the latency if often a serious problem
15:25.04awannabe[TK]D-Fender: any other options to start with as well though? that helped, but still needs work, we are getting echo mostly
15:25.13Qwell[]it depends entirely on your setup and the jitter you're seeing
15:25.16[TK]D-Fenderawannabe: What hardware?
15:25.39awannabetdm400p
15:25.47[TK]D-Fender~wglwat
15:25.47jbotwglwat is probably well, good luck with all that
15:26.06berktrQwell, is there a way to analyze how much jitter i need?
15:26.24[TK]D-Fenderawannabe: Time to try another EC routine, cross your fingers, say a prayer, then put your head between your knees and kiss your sorry ass goodbye!
15:26.36[TK]D-Fenderberktr: You need all the jitter you can get!
15:26.42coppicei think you need lots of jitter. its the very soul of a voip call
15:26.48awannabehaha
15:27.38nfi|ermeshow can i check if asterisk-addon is installed ?
15:30.18*** join/#asterisk gbodemantv (n=gbodeman@corpex.pivotal.televerde.com)
15:31.07Mercestesawannabe, you can also dink with your tx/rx values as well.  ztmonitor can help diagnose issues.  I have intermittent echoproblems on a tdm400p as well.
15:31.53[TK]D-Fendernfi|ermes: Already answered this for you....
15:32.23awannabeMercestes: thanks ill take a look at that as well
15:32.34[TK]D-Fendernfi|ermes: Go check if you've got the .so's in your modules folder.
15:34.04lilalinuxWhat's the difference between zaphfc and vzaphfc (bristuff)
15:37.46*** join/#asterisk d-phaze (n=phaze@pcbcu420a.unil.ch)
15:40.36sci_05anyone have any good links on setting up h323 on *
15:40.42tzafrirlilalinux, basically different drivers for the same hardware
15:46.04vltJT: Aah, I got it. You start a new "block" of TE or NT ports by adding a new "signalling" line ... Ok, thanks.
15:47.44berktrwhen i put a caller to hold
15:47.54berktri can only hear the ringing tone 10 seconds
15:47.58berktrthen i hear the error tone
15:48.07Mercestessci_05, man rm
15:48.11berktrhow can i fix this
15:48.53[TK]D-Fenderberktr: Would help if we knew what HARDWARE you were using....
15:49.13berktrpap2
15:49.39berktri want to put a caller on hold and wait for at least 60 seconds
15:49.43lilalinuxtzafrir: which one would you suggest, it looks like zaphfc in debian/etch is with florz patch
15:49.52[TK]D-Fenderberktr: And how are you putting them on hold?
15:49.56berktrduring this time i can talk to people in our office about the topic and other stuff
15:50.06berktrby pressing to hook button once
15:50.16[TK]D-Fenderberktr: That is not HOLD.
15:50.17*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
15:50.20berktri see started music on hold on asterisk
15:50.22*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-17-38.bflony.east.verizon.net)
15:50.24[TK]D-Fenderberktr: Go read its manual again
15:50.37[TK]D-Fenderberktr: Thats to begin a transfer / 3-way call.
15:50.48[TK]D-Fenderberktr: Which is why it times out.
15:52.17irulehow may I add zaptel to debians init? I know about redhants chkconfig but this is differente thanks
15:55.11SuPrSluGi'm getting calls dropped from zap channel. It also happens when callers are leaving voicemail. What could cause these issues?
15:55.41SuPrSluGcanreinvite=no on all phones
15:56.59[TK]D-FenderSuPrSluG: Poor disconnect supervision, etc.  pastebin your zapata.conf
15:57.10SuPrSluGk
15:58.06*** join/#asterisk kn0x (n=pinochle@76.76.10.159)
15:58.38*** join/#asterisk Fulk (n=fulk@87-194-176-39.bethere.co.uk)
15:59.15SuPrSluGhttp://pastebin.ca/590457
15:59.38SuPrSluGits a tdm04b card
15:59.50SuPrSluG4 fxo
15:59.59*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
16:00.49[TK]D-FenderSuPrSluG: ;callprogress=no <- uncomment, and remove your extra redundancy.
16:00.53*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
16:01.03[TK]D-FenderLunch time, bbiab
16:01.13tzafrirlilalinux, not sure which is better
16:01.17tzafrirtry both
16:02.09*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
16:02.21*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
16:02.58SuPrSluGthe 2x callerid=asrecieved
16:03.59SuPrSluG[TK]D-Fender:thanks
16:04.28*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
16:09.22lilalinuxI get this error when using vzaphfc and starting asterisk: ioctl32(asterisk:5411): Unknown cmd fd(17) cmd(40044a26){00} arg(ffcca5e8) on /dev/zap/channel
16:10.11*** join/#asterisk nso (n=checco@host184-88-dynamic.55-82-r.retail.telecomitalia.it)
16:10.18lilalinuxAnd I get this error when using zaphfc: "sync lost, pci performance too low. you might have some cpu throtteling enabled."
16:10.36lilalinuxI have 2 hfc cards btw
16:10.42nsosalve
16:10.43*** join/#asterisk mellen (n=thansen@85.235.250.110)
16:10.55lilalinuxnso: ave
16:11.03nsoa te:P
16:14.48tzafrirlilalinux, where do you get that error message?
16:14.59tzafrirah, from asterisk
16:17.46*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
16:17.56Dr-Linuxfrom where i can download asterisk-im plugin?
16:18.25*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
16:18.38shmaltzanybody knows whats up with telephonydepot.com?
16:19.21*** join/#asterisk galeras (n=root@200.31.204.42)
16:20.38[TK]D-Fendershmaltz: What about them?
16:21.03shmaltz[TK]D-Fender, can you access their site now?
16:21.38[TK]D-Fendershmaltz: not currently.  could earlier though...
16:21.49shmaltzlooks like a DNS issue
16:21.55lilalinuxtzafrir: I get the error in dmesg
16:22.08shmaltzprovider didn't receive the check ;)
16:23.11MercestesIt's in the mail.
16:23.29tzafrirlilalinux, unimplemented ioctl?
16:23.31shmaltzD-Fender you got a phone number for them?
16:23.41Dr-Linuxfrom where i can download asterisk-im plugin?
16:26.20*** join/#asterisk legis (i=1001@unaffiliated/legis)
16:26.30[TK]D-Fendershmaltz: 1-800-337-1358
16:26.39shmaltzD-Fender thanks
16:26.50[TK]D-Fendershmaltz: Nothing you couldn't have found out jsut as fast :)
16:27.31lilalinuxtzafrir: maybe, problem is, that I'm running a vserver, it worked in the past with 1 card and a different system, but with 2 cards (now in dual amd64) it doesn't work anymore
16:27.41legisHi, is the TFoT book a good start for learning VoIP or should i get into a more general VoIP book and the read TFoT to setup * ?
16:27.52tzafrirhmmm... cards on the vserver host?
16:28.42[TK]D-Fenderlegis: To learn VoIP, no, to learn ASTERISK, yes.
16:28.59lilalinuxtzafrir: the modules are loaded inside the host, asterisk is inside a guest
16:29.15[TK]D-Fenderlegis: How about we slow down and confirm what you actually want to DO before what route is the best way to get there...
16:29.24lilalinuxthe host runs debian/etch, the guest debian/sid
16:29.44tzafrirthe device files are generated statically?
16:29.50lilalinuxtzafrir: yes
16:30.05lilalinuxcopied from within the host
16:30.26tzafrirfd17 of the asterisk process is indeed /dev/zap/channel ?
16:31.14legis[TK]D-Fender: I want to setup an * system but know nothing about voice.
16:31.21lilalinuxtzafrir: crw-rw---- 1 root dialout 196, 254 2007-06-26 18:08 /dev/zap/channel
16:31.39[TK]D-Fenderlegis: Ok, what kind of lines are you planning on using? (PSTN connectivity)
16:32.10legis[TK]D-Fender: yes
16:32.11lilalinuxtzafrir: or did you mean something else?
16:32.15shmaltzD-Fender, can you access telephonydepot. now?
16:33.18[TK]D-Fenderlegis: That was not a yes/no question.
16:33.25tzafririrule, basically with something of the sort of:  update-rc.d zaptel defaults # however, you should mind the order of zaptel and asterisk
16:33.31[TK]D-Fenderlegis: I asked what KIND of PSTN connectivity you were planning on.
16:33.39legis[TK]D-Fender: also to connect connect offices
16:33.55*** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)
16:34.06[TK]D-Fenderlegis: no, that WHAT you are going to use it for.  I asked WHAT KIND OF PSTN CONNECTION <------------
16:34.18iruletzafrir oh, thanks
16:34.33legis[TK]D-Fender: gsm
16:34.35tzafrirlilalinux, that is what I meant
16:34.51legis[TK]D-Fender: is that what you meant?
16:35.02[TK]D-Fenderlegis: Wireless access to the PSTN?
16:35.12*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
16:36.39legis[TK]D-Fender: no
16:36.49legis[TK]D-Fender: gsm is the PSTN
16:37.03[TK]D-Fenderlegis: that is not making sense.
16:37.16legis:D
16:37.23*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
16:37.24JTthis looks fun
16:37.25[TK]D-Fenderlegis: GSM i s a wireless protocol, or a codec.
16:37.51lilalinuxtzafrir: sometimes the fd is 18 or 19
16:38.11BSD_Technope dont wanna
16:38.16BSD_Techcant make me
16:38.19BSD_Techno way
16:38.23BSD_Techforget diont ask
16:38.29legis[TK]D-Fender: i can use the gsm card, put it in a device with a r11 port
16:38.47JTlegis: ok, you are talking about a GSM GATEWAY
16:38.48ai-aHow easy is it to set up a ISDN card in Asterisk ? - we have a sangoma isdn based in UK.
16:38.53JTlegis: gsm is not the pstn, btw
16:39.09[TK]D-Fenderlegis: Sorry I'm not sure wht kind of hardware it will take to support that.
16:39.14JTai-a: congratulations, i think you're the first person in the world to use that card
16:39.36[TK]D-FenderJT : Language and hierarchy communication problem.
16:39.38JTai-a: unfortunately, i have no idea how easy it is to make work
16:39.42JTai-a: i assume you mean BRI
16:39.46shmaltzusing FXO without callerid on zap channels, it still takes around 2 rings to ring my phones using app_dial, is there anyway to shorten that?
16:39.57BSD_TechJT its easy
16:40.08ai-aJT: yep.
16:40.10JTBSD_Tech: have you used an A500?
16:40.22ai-a1st person ? - we havent bough it yet.. but will next week.
16:40.31BSD_TechI have setup 1 9 months ago
16:40.33ManxPowershmaltz: Yes.  Easily
16:40.35[TK]D-Fendershmaltz: make sure you say "usecallerid=no" in zapata.conf
16:40.39JTai-a: it's been on their website for only a week or too
16:40.48JTBSD_Tech: err, it wasn't released that long ago
16:40.51ai-aoh i see ;)
16:41.00JTs/too/two/
16:41.07BSD_Techthe beta card was I worked with it in beta
16:41.16BSD_TechI do beta for sangoma
16:41.17JTto connect to what network?
16:41.20shmaltzsilly me just commented out usecallerid instaed of setting it to no
16:41.35shmaltzthanks, ManxPower, and [TK]D-Fender
16:41.43BSD_TechI never connected I just tested to make sure drivers saw and activated the card
16:41.49ai-aBSD_Tech: what level is easy ? - im using asterisknow (i didnt mention that in here honest!) and novice linux user.
16:41.54JTgood testing <_<
16:41.55BSD_Techthen passed the card on to a line tester
16:42.01shmaltzanybody here can access telephonydepot.com ?
16:42.02*** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com)
16:42.20JTBSD_Tech: so do the bri channels access just like pri channels?
16:42.23JTjust need wanpipe?
16:42.25[TK]D-FenderBSD_Tech: I haven't seen one yet, but are they working on an ADSL2/2+ S518 card?
16:42.35BSD_TechI would say on a scale of 1 to 10 its a 6 to install
16:42.56watchyi got a asterisk 1.2.9.1 server and sometimes it seems to miss dtmf tones when dialing out.
16:42.57ai-a10 being easiest ? ;)
16:42.57BSD_TechI have heard but not seen one yet
16:43.04BSD_Tech10 being hardest
16:43.08ai-aoh ;(
16:43.08BSD_Techlol
16:43.23JTBSD_Tech: so do the bri channels access just like pri channels?
16:43.28ai-awill the dist detect the hardware, and get teh dev in place ?
16:43.38JTlol hardware detection
16:44.04[TK]D-FenderBSD_Tech: I love my Current S518 (LP 2nd gen), and am just thinking to keep it current, and generate an extra for a router PC.
16:44.06BSD_Techyou have to build and install the i4l drive and or the bri driver
16:44.15JTeww
16:44.21JTi4l fucking sucks donkey balls
16:44.28BSD_Techand the sangoma driver to go with it
16:44.29JTexplain on the "bri driver"?
16:44.42BSD_Techlet me find my how to on it
16:45.21BSD_TechI have a cd for each card with the driver and the howto and the pdf
16:45.43BSD_Techbe it I have only installed on BSD
16:45.45BSD_Technot linux
16:45.53BSD_Techbut it has the howto for both
16:47.27*** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com)
16:49.28lilalinuxtzafrir: du you have an idea?
16:50.16tzafrirlilalinux, not right now. Later when I get home
16:51.14lilalinuxk
16:51.35BSD_Techlooking for my cd's brb
16:51.53BSD_TechI know they are here but my office looks like a junk yard
16:52.54*** join/#asterisk YonahW (n=kvirc@IGLD-83-130-71-223.inter.net.il)
16:52.55tzafrirBSD_Tech, which asterisk channel does it use?
16:53.29*** join/#asterisk angryuser (n=Miranda@df01t2-212-194-219-100.d4.club-internet.fr)
16:53.33angryusergood day
16:54.27*** join/#asterisk centrex (i=centrex@nat/digium/x-64092c6ba962f1db)
16:54.47angryuseri have a simple question, the timeout defined directly in queue(name|||300) command is the same parameter which is defined in queue.conf as timeout = 300 ?
16:55.53BSD_Techtza dont recall off the toop my head but I  think it was chan capi
16:56.12BSD_Techlet me find the cd
16:56.35angryuseri want 2 different actions on 2 timeouts , timeout of agent being called and timeout of person waiting in queue, how to separate that ?
16:56.42JTthat's terrible
16:57.01JTif the A500 doesn't use zaptel or something equal, no point buying
16:59.02[TK]D-FenderJT : Supports Asterisk®, Yateâ„¢,   FreeSwitchâ„¢,   CallWeaverâ„¢, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications.
16:59.07*** join/#asterisk sashion (n=sdgsdg@dsl-242-91-154.telkomadsl.co.za)
16:59.14JT[TK]D-Fender: tells me nothing
16:59.40JT[TK]D-Fender: "we copy and paste open source pbx names from our other cards' pages"
16:59.57tzafrir[TK]D-Fender, right. And yet the recent drivers from their site don't seem to have that support. So "supports" is probably in future tense
17:00.05[TK]D-FenderJT : BRI is practically PRI when yuo get down to it.  I'd be betting that they wedge themselves in the same way
17:00.17JT[TK]D-Fender: i'm only hoping
17:00.21JT[TK]D-Fender: i'm wanting to see
17:00.27[TK]D-FenderJT : No, you're NEUROSING :)
17:00.34tzafrir[TK]D-Fender, Asterisk's chan_zap has fine support for PRI. BRI is more complicated
17:00.35JTit's the big A500 mystery
17:00.48[TK]D-FenderJT : go have your little embolism and be quiet! ;)
17:00.50JTwhich way will the A500 lottery go
17:01.26[TK]D-Fenderangryuser: Go read the sample queues.conf again, and "show application queue"
17:02.26*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
17:03.03angryuseri found it, it' the timeout for switch the agent, but, let's sayt i want somebody being redirected if agent is timed out, of the queue , how do i do that ?
17:03.09coppicethe A500 is certainly a mystery. The blurb says every free telephony platform supports it, yet none of us knows a thing about :-\
17:03.16JTrofl
17:05.00[TK]D-Fenderangryuser: Whats the point of having a queue when you don't cycle through your agents?
17:06.16ManxPower,cnhjbl;kfhgdfg
17:06.19ManxPowerhmm, that didn't work
17:06.57angryuser[TK]D-Fender:  i will cycle once, and if he is out, that hapens i want him to timeout on ringing and directe a person out, let's sat to a secretary
17:08.03[TK]D-Fenderangryuser: If you don't care about your queue logs, then just have it use a Local channel, Answer, Dial, then Dial the secretary
17:08.16Dovidwhats is this an erorr of ?
17:08.16Dovidchan_sip.c:2273 update_call_counter: Call to peer '201' rejected due to usage limit of 2
17:08.56*** join/#asterisk AtomicDawg (n=atomicda@74-206-0-80.static-ip.m.telepacific.net)
17:09.02[TK]D-FenderDovid: Clearly you have assigned a call limit on that device
17:09.04ManxPowerDovid: that means that there are already 2 calls in place for that device, and you have a max of 2 calls configured for that device
17:09.19ManxPowerDovid: Oh, and stop using a GUI to manage Asterisk
17:09.54*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)
17:11.40DovidManx: I AM OFFENDED !!! You know my feelings on GUI's.
17:11.41Dovidthey suck
17:11.54Dovidi didnt set a call limit. (i did work on the box at 5 am so anything can happen
17:11.56Dovid)
17:12.14ManxPowerDovid: So you just "forgot" you created call limits?"
17:12.30ManxPowerDovid: magical pbx fairies didn't set them
17:12.51Dovidit was a clients box
17:13.07ManxPowermaybe a magical client fairy set them
17:13.10robl^fairies?!?!!?!  eeek!!  *jumps up on the chair and shrieks!!*
17:13.16Dovidand now i am in there and i see it. i didnt set up. i guess client has a gui. they told me it was a clean install !! argh
17:13.40robl^Clean install of TrixBox?
17:13.51ManxPoweryou can tell if there is a GUI.  All the config files are so totally screwed up you can't even tell what is set and what is not.
17:14.14[TK]D-FenderManxPower: Don't talk about queers like that! ;)
17:14.39ManxPowersorry,  faeries
17:14.44Dovidlol
17:15.05lilalinuxtzafrir: gotta go, if you find something pls leave me a note here or query
17:15.30Dovidi owe it to the users list 3+ years ago. i wanted a gui and ev1 yelled at me for it. it was good advice. or i would be an asterisk expert **Cough Cough - GUO Expert**
17:15.35DovidGUI*
17:15.36*** join/#asterisk diskfree (n=bve@www3.datarack.nl)
17:17.06diskfreehello. is there a way to change to overall volume level in asterisk (1.2.x)?
17:17.44ManxPowerdiskfree: no.  Only on Zap channels, as VoIP channels have their volume set by the VoIP device
17:18.31diskfreeok thanks
17:18.46diskfreetoo bad... I have users complaining about the overall volume level
17:18.55ManxPowerdiskfree: then fix it
17:18.56diskfreethey're quite noisy themselves
17:19.03ManxPowerdiskfree: it's not hard to do
17:19.27ManxPowerBut without know anything about your VoIP devices, PSTN channels, etc there is not much else I can suggest
17:19.34diskfreeproblem is I'm only using ztdummy for timing purposes
17:19.45diskfreewe're using cisco media gateways with sip
17:19.48ManxPowerdiskfree: no, that is not the problem
17:19.53[TK]D-Fenderdiskfree: What phones are you using?
17:20.05ManxPowerdiskfree: then set the volume on the Cisco
17:20.36ManxPowerWhatever devices do the analog/digital conversion (phones, gateways, etc) should be setting the gain.
17:20.39diskfreepolycom IP 600's... I've already looked through the admin guide, I only found stuff on persistency (which I already implemented as the customer wanted it)
17:21.09ManxPowerdiskfree: Then you didn't look very hard.  The polycoms have like 500 options for setting audio gains
17:21.42diskfreemanxpower, ok, english is not my first language, I searched on volume
17:21.43ManxPowerI would go thru and make sure the gains on your cisco are correct before screwing with the polycoms
17:21.53diskfreeI know polycom has tons of settings
17:22.02ManxPowerdiskfree: search on "gain"  txgain, rxgain, etc
17:22.05[TK]D-Fenderdiskfree: When in doubt now you can up the gain, not just the volume.
17:22.23diskfreeOk
17:22.36ManxPowerBut you will have major problems if you change the gains on the polycoms without making the cisco gains correct first
17:23.03ManxPowerPSTN/IP Phone calls might sound fine, but Phone/Phone calls would be too lound or too soft.
17:23.10diskfreeThey have issues with internal voip calls
17:23.16diskfreeNot with PSTN calls
17:23.27ManxPoweronce you get the gains correct on the cisco, then fiddle with the gains for the polycoms
17:23.36diskfreeBut it's a broker firm, they tend to speak loud to eachother themselves :)
17:23.45diskfreeOk
17:23.49ManxPowerdiskfree: and when you up the gains to make the phone/phones sound good then PSTN call gains won't be correct.
17:24.25ManxPowerI have NO idea how to calibrate the gains on the Cisco, but Cisco has plenty of documentation
17:24.26diskfreeSo they'd need to be balanced both, I can't just screw with one part of the infra
17:24.35[TK]D-Fenderdiskfree: Ok, go change your EMPLOYEES.
17:24.40diskfreeAt least you gave me something to dive into :)
17:24.43ManxPowerdiskfree: correct.
17:24.50*** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca)
17:24.54Slingkyhi guys!
17:25.03Slingkysomebody know spa2102 ata config ?
17:25.14Slingkyi hook up a fax on this device
17:25.22Slingkyi route fax to that extension
17:25.38[TK]D-FenderSlingky: Bonne chance....
17:25.49Slingkyit rings but when the fax try to pick up the line, i hear a dial tone instead of it starting receiving ???
17:25.57ManxPowersorry, I don't help people trying to do FaxOverVoiceOverIP
17:26.01Slingkytu parles français ?
17:26.23diskfreeSlinky: Il faut chercher un autre solution
17:26.28*** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-220.bwic.chi.wayport.net)
17:26.39*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)
17:26.51Slingkytout ce que je veux c'est que le fax décroches comme il faut. j'imagine que c'est dans le dial plan du ata ?
17:27.01*** join/#asterisk rootfield (n=rootfiel@200.103.96.98)
17:27.09rootfieldhi all
17:27.19Slingkysi je fais hook et start receiving , ça fonctionne !
17:27.28Slingkymais je ne veux pas avoir à faire cela...
17:27.42diskfreeSlingky: Ouais, mais tu auras de problemes quand memem
17:27.43rootfieldthe sintax SetAccount is changed on asterisk 1.4 ?
17:28.32diskfreeManxPower: FoIP by ActiveVoice is pretty neat though, works well with cisco media gateways. I agree that cheap ata's suck.
17:28.46rootfieldi'm using "EXEC SetAccound ..." on my python script on asterisk 1.2.. but not work on asterisk 1.4...
17:29.27ManxPowerrootfield: Try spelling it correctly
17:29.45ManxPowerWhat error do you get?
17:30.20ManxPowerChances are SetAccount is depricated (it would be listed in UPGRADE.txt in the asterisk source) and has been replaced by a function, but you can easily find out.
17:31.17*** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
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17:31.59BSD_Techsorry still digging and cleaning at the same time my office is a shamble
17:32.06rootfieldManxPower, yes.. is deprecated.. but i dont find new sintax using agi script.. only using on extensions.conf =/
17:32.55ManxPowerrootfield: I suspect a search of the -users list would help you find out
17:34.20*** join/#asterisk jaguiar (n=jaguiar@189.142.86.124)
17:38.32*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)
17:40.17irulemake config, ....install: cannot stat `init.asterisk': No such file or directory, make: *** [config] Error 1
17:40.19irulewhats up?
17:40.38mercestesare you in /usr/src/asterisk-?
17:40.39irulethis is the latest 1.2.19
17:40.42iruleyes
17:40.49iruleasterisk:/usr/src/asterisk-1.2.19#
17:41.00ManxPowerdo you have an init.asterisk?
17:41.14Qwell[]irule: what distro?
17:41.15ManxPowerirule: are you running as root?
17:41.18Qwell[]slackware perhaps?
17:41.20iruledebian sarge
17:41.25iruleyes root
17:41.31Qwell[]huh, do you not have /etc/rc.d/init.d/?
17:41.46irule/etc/init.d/
17:41.58irulethat is its place in debian
17:42.10mercestesirule:  Did you ./configure?
17:42.10ManxPowerirule: I think make config is for readhatish and /etc/rc.d/init.d based distros
17:42.19iruleyes
17:42.23Qwell[]yeah, that's a bug
17:42.26ManxPowermercestes: 1.2 DOES NOT HAVE ./configure!
17:42.35mercestesHuh?
17:42.44mercesteshow would I have not noticed that?
17:42.52ManxPower"make clean"  "make config"
17:42.53mercestesI've done a make ./configure and most of my experience is on 1.2.blah
17:42.53iruleoh exactly, there is no ./configure lol
17:43.20iruleI ran my build asterisk sscript, I can execute it correctly with asterisk -cvvvvvvvvvvvvvvv
17:43.39Qwell[]irule: cp contrib/init.d/rc.debian.asterisk /etc/init.d/asterisk
17:43.57angryuserhm, cant find how to generate a event when agent is called and timed out in his queue
17:44.02angryuserany help?
17:44.21angryuserit is stupid agent and sometimes he forget to log out
17:44.25ManxPowerangryuser: "generate an event"?
17:44.38angryuserpass to another extension
17:44.38iruleawesome! thanks a lot!
17:44.56ManxPowerangryuser: I thought there was an option to remove the agent of they don't answer
17:45.28DrukenLPYanyone know how to fix amp/freepbx from trying to auth incoming calls?
17:45.42angryuserManxPower:  no no , i need if he dont answer for 80 sec "exten do something else"
17:45.44ManxPowerDrukenLPY: No.  That would be a different channel
17:46.02BSD_Techgot #freepbx
17:46.07DrukenLPYManxPower: yes i'm awear... just figured i'd throw it out there :)
17:46.09*** join/#asterisk mtaht4 (n=m@55-85-62-200.enitel.net.ni)
17:46.15ManxPowerangryuser: no way that I know of.  I just recall an option for timeouts, removing the agent from the queue, retry, etrc
17:46.30angryuserManxPower: oj thanks for help
17:46.39ManxPowerDrukenLPY: the last time someone did that we tied them to a tree and made them admin Cisco Call Manager.
17:46.51*** join/#asterisk peanutb (n=paulb@c-24-16-243-186.hsd1.mn.comcast.net)
17:47.09DrukenLPYbah, that's cruel intentions...
17:47.14YonahWisn't there also an option to through the call to another extension or context if there are no longer any agents in the queue?
17:47.29mercestesyay, a road trip!
17:47.33ManxPowerYonahW: no.  There is an optioon to exit the queue
17:47.37*** join/#asterisk bkruse_home (n=kruz@pool-71-115-138-15.gdrpmi.dsl-w.verizon.net)
17:47.37YonahWs/through/throw
17:47.50ManxPoweronce the call exits the queue you can do anything you want with it
17:47.55YonahWand what happens after you exit the queue
17:48.12mercestesYonahW, Whatever you preordain
17:48.16*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
17:48.17ManxPowerYonahW: uh, you go to the next priority of the dial plan just like every single other application
17:48.21mercestesexercise the powers given to you
17:48.22YonahWright thats what i figured
17:48.44ManxPowerI'll bet app_queue sets channel variables with information in them
17:48.47YonahWso assuming this agent that timesout is the only agent then the call can be passed to another option
17:49.02YonahWand if hes not then the next agent can get the call
17:50.10ManxPowerThere might be an option to set that
17:50.34*** join/#asterisk peanutb (n=paulb@c-24-16-243-186.hsd1.mn.comcast.net)
17:50.36ManxPowerWhen was the last time you read "show application queue", README.variables, queues.txt, and the Wiki information on agents and queues?
17:50.47ManxPoweroh, and the sample config files for queues and agents, of course.
17:51.04YonahWbeen awhile but then again im not trying to do anything with queues at the moment
17:51.40ManxPowerI don't use queues very much, as they do not work the way my users need it to work
17:51.46*** join/#asterisk santiago (n=santiago@debian/developer/santiago)
17:52.12YonahWi like the idea but found them to be a little too restrictive
17:52.30mercestess/restrictive/broken/
17:52.44ManxPowerI use Local/ channels and clever dialplan stuff to do what I need.
17:53.50ManxPowerThe biggest problem I face is that 1) users forget to log in and log out and 2) There is nobody to answer the phones full time.  They always have several other jobs like copy, fax, office manager, etc
17:55.10DrukenLPYManxPower: get someone a wirless headset
17:56.12*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-218-123.dsl.irvnca.pacbell.net)
17:56.50*** join/#asterisk seva (i=seva@sevatech.com)
17:57.24sevaany suggestions on a free (or commercial with source) sip/iax2 java applet or activex control?
17:57.30*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
17:57.51*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
17:59.17*** join/#asterisk ibob63 (n=james@bb-87-82-21-236.ukonline.co.uk)
18:00.53ManxPowerDrukenLPY: every single one of those I have seen suck.
18:01.29ManxPowerThe Polycoms are somewhat limited with what headsets work with them
18:02.50*** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net)
18:04.01ibob63i know this is a common problem but I am having problems with making a call over sip. The sip registration and call connection works perfectly. but the rtp gets sent to local ip address rather than the public ip address. any pointer would be helpful. thanks
18:04.19DarylVOIPHey all...quick question.  I've got several DIDs coming in from the same provider going into the same context via sip.conf.  I'm trying to figure out how to make them all do the same thing in extensions.conf.  Seems like using s,x,<whatever> woudl work, but its not.  I have to use <did>,1,<whatever>.  Any ideas?
18:04.33ManxPoweribob63: you need localnet= and externip= set in sip.conf [generall]
18:04.38ManxPoweror [general]
18:05.04ibob63thanks ManxPower. I'll check it out
18:05.12[TK]D-FenderDarylVOIP: jsut use GOTO and dump them in the same place.
18:05.13ManxPowerDarylVOIP: extension "s" will match calls for which there is NO destination number
18:05.23ManxPower<did>,1,<whatever>. is the correct way to do it
18:05.59ManxPower"s" stands for "stupid" not "start".  stupid == port or remote device is too stupid to send the dialed digits
18:06.04DarylVOIPYeah, I'm just trying to avoid having to put each DID in.
18:06.14DarylVOIPGot it...makes sense.
18:06.16[TK]D-FenderDarylVOIP: It is my opinion you should not do anything specific in that context, but rather jump out to another where you'd implement IVR, etc.  As little processing against the oridinal EXTEN as possibl
18:06.30ManxPowerFXO ports and some VoIP providers like IAXTel do this
18:06.53[TK]D-FenderDarylVOIP: You should WANT to do it per DID, but tune it down to a single Goto each.  that way you pick one out of the pack easily.
18:06.56DarylVOIPIt's not even going to answer...this is for a callback.  It's going to dump the call as soon as I cat the Contact field from the SIP invite.
18:07.04ManxPowerNEVER EVER put incoming calls from untrusted sources in a context from which they can dial out
18:07.26DarylVOIPNo, I really shouldn't WANT to do it for each DID.  They all do the same thing, and provisioning new ones is simplified if I don't have to change extensions.conf.
18:07.31DarylVOIPThis is a very specific purpose.
18:08.03[TK]D-FenderDarylVOIP: Depends if you have any expectation to change that purpose or have others mised in there.
18:08.06ManxPowerDarylVOIP: exten => _XXXXXXXXXX,1,Noop(wildcard because I'm too lazy)
18:08.22ManxPowerassuming your DIDs are NANPA 10-digits
18:08.28ManxPowerif not, then adjust accordingly
18:08.33DarylVOIPNo, they won't all be.  But I can work with that.
18:08.33ManxPoweror better would be _NXXNXXXXXX
18:08.39DarylVOIPThanks....let me give it a try.
18:08.42*** part/#asterisk seva (i=seva@sevatech.com)
18:09.10ManxPowerDarylVOIP: The correct way is to list each one seperatly with just a Goto to send them to the right place
18:09.42ManxPowerDon't worry.  Using wildcards for this will eventually bite you in the ass.
18:10.02ManxPowerYou just hope that the bite is not a massive LD bill you can't billback
18:10.11DarylVOIPExplain how....these DID's are never answering.
18:10.28DarylVOIPI'm just grabbing the contact information from a call being proxed through an SBC.
18:10.38ManxPowerDarylVOIP: there are a million small mistakes you can make in your dialplan to cause things to not do what you want them to.
18:10.41DarylVOIPWhich already authenticated the inbound DID.
18:11.06ManxPowerThe most common is being sent to an IVR, but that is not the case in your configuration
18:11.33DarylVOIPOf course...I wouldn't want to do that if I were actually answering the call or hadn't authenticated the inbound from another device that I control.
18:12.13ManxPowerFor example if you miss a priority and have another extension pattern that matches then that pattern will be run for the missing priority
18:12.16DarylVOIPI don't use Asterisk (in this circumstance) as anything more than a feature server.  All calls are handled through a pair of SBCs.
18:14.20ibob63ManxPower: I have the localnet and external ip set... Is there any other things I should check?
18:15.01DarylVOIPExecellent...that works great.  Thans Manx.  Now I just have to figure out how to grab that contact field and pass it to my agi.
18:16.23*** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-218-123.dsl.irvnca.pacbell.net)
18:18.26*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net)
18:21.54ManxPoweribob63: no.
18:22.55*** join/#asterisk kombi (n=kombi@213.160.14.18)
18:23.57*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)
18:24.30kombiwhat might be meant by "trunk version of asterisk"? (found on http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO)
18:26.37kombi"trunk version" = >1.2?
18:27.06*** join/#asterisk hi365_m (i=HydraIRC@212.199.22.26.forward.012.net.il)
18:27.22hi365_mhello, does 1.4 put the queue_logs in to sql?
18:28.41Corydon76-workNo, it does not
18:28.54Corydon76-workNo logs are put into a database at this time
18:29.19hi365_mhm,
18:29.26hi365_many plans for the future?
18:29.34Corydon76-workWell, CDRs might be
18:29.38Corydon76-workNo plans currently
18:30.11*** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net)
18:30.11hi365_mcdr can be put in the sql allready
18:31.24kombiIs manager_redirect part of 1.2?
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18:33.05obnauticusIf anyone here has the pro version of idefisk...i'll borrow it ROFL.
18:34.29DarylVOIPQuick one....can I echo something to the console from the dialplan?  I'm looking for a quick way to debug the output of things like: exten => _.,1,NoOp(${CALLERID(num)})
18:36.38DarylVOIPHmmm...maybe I'm already seeing it.  It's just not what I expect.
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18:37.20*** part/#asterisk hi365_m (i=HydraIRC@212.199.22.26.forward.012.net.il)
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18:43.41obnauticusTo setup PHP on asterisk, you don't really need to make any asterisk modifications, all you need to know is like what you can and can't do with PHP and how to do it on asterisk...correct?
18:43.50obnauticusBecause I'm looking at this guide and it looks too easy
18:44.20flujanhi guys. I am searching for a nagios plugin to monitor asterisk. Which one are you using?
18:46.05[TK]D-Fenderobnauticus: Adjust your terminology.  You do not setup "PHP on asterisk".
18:46.15obnauticuswell i mean use it for AGI
18:46.20[TK]D-Fenderobnauticus: BETTER
18:46.47obnauticusSo am I right?
18:46.47[TK]D-Fenderobnauticus: And yuo can do no more with it for AGI than any other language.  Go read up on AGI itself
18:46.54obnauticusI know
18:47.00obnauticusI was using PERL on AGI earleyer
18:47.30*** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-220.bwic.chi.wayport.net)
18:50.31*** join/#asterisk rantsh (n=chatzill@201.210.16.238)
18:50.37rantshhello everyone
18:51.13obnauticusperl* [TK]D-Fender lol
18:51.28rantshhow's everyone doing?
18:51.31rantshin case anyone remembers I did get to do the transcoding, and I've succesfully managed to record calls
18:51.45*** join/#asterisk Supaplex (i=supaplex@166-70-62-199.ip.xmission.com)
18:51.59rantshand that was all thanks to you people
18:53.21rantshbut I have an issue, and really need your help, yet again...
18:55.09[TK]D-Fender~ask
18:55.09jbotQuestions 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.
18:55.13obnauticus[TK]D-Fender I have a quick question about IP phones, people have been telling me that you have to pay for firmwares and stuff and that they aren't that good...
18:55.34rantshfor some reason, sox is affecting sound on one of the ends of a recorded call... meaning that the receiver hears good and he speaks good, but the caller never gets sound
18:55.40[TK]D-Fenderobnauticus: Cisco makes you pay for them, nobody else that I know.
18:55.49obnauticusThat's so fucking gay.
18:55.49[TK]D-Fenderobnauticus: And on the quality scale, Polycom > All
18:55.52rantshnow I'm using asterisk 1.4.5
18:55.53obnauticusk
18:55.56obnauticusthanks [TK]D-Fender
18:56.19anonymouz666[TK]D-Fender: I wonder how I can loop that 2 rows returned by the MySQL() command. Do you have a suggestion for doing that? I don't have the facilities of arrays
18:56.22obnauticus[TK]D-Fender which model do you suggest (polycom)
18:56.49rantshOh and I forgot to tell you this only happens when asterisk is NOT transcoding, meaning that both parties are using the same codec, whatever codec it is, one of the parties will not receive any sound
18:57.12kombiWriting callerids of people leaving voicemail into mysql and later adding them to a conference using a web interface, am I completely crazy or will this work? (even for the ordinary oop php mysql hacker..)
18:57.25[TK]D-Fenderobnauticus: Depends on your needs.  General use : IP 320.  Need a passthrough?  IP 330.  Want a NICE home desk-phone without PoE?  IP 501.  Receptionist?  IP 650
18:57.40Qwell[][TK]D-Fender: You recommend the 650 now?
18:57.48kombiI thought 501 was PoE?
18:57.50Qwell[]about time :p
18:57.54obnauticusthanks [TK]D-Fender :)
18:58.08[TK]D-Fenderanonymouz666: You never explained what you were doing with the data retreived, and I showed you the instructions that tell you how many values are returned.  You can look it from there.
18:58.40[TK]D-FenderQwell[]: For receptionist only and because it is only a bit more than the IP 601, IS nicer and will be supported longer.
18:58.46Qwell[]ahh
18:58.48ManxPower501 is PoE with special polycom cable
18:58.53Qwell[][TK]D-Fender: you used one yet?
18:58.59[TK]D-Fenderkombi: Not natively.  It needs a pricy adapter cable.
18:59.04[TK]D-FenderQwell[]: Nope.
18:59.08Qwell[]You need to
18:59.17[TK]D-FenderQwell[]: No I don't :)
18:59.21Qwell[]Yes you do ;)
18:59.29Qwell[]ulaw has never sounded so good
18:59.32kombiwhich polycom with PoE that gives you all the bells and whistles?
18:59.38Qwell[]it makes lpc10 sound good :P
18:59.39[TK]D-FenderQwell[]: I probably SHOULD, just like the 550 & 320/330, but not NECESSARY
18:59.49[TK]D-Fenderkombi: IP 650
18:59.49rantshanyone has a clue on what could be going on with my recording calls?
19:00.02[TK]D-Fenderkombi: Overkill for anyone but a receptionist.
19:00.12Qwell[]or a hax0r that wants the very best
19:00.19ManxPowerkombi: THE MOST EXPENSIVE ONE
19:00.30[TK]D-Fenderrantsh: You provided no CLI out, no debug info, no recorded files, NOTHING.  No, we CANNOT help you.
19:00.40kombiok.. so the sensible choice with PoE..;)?
19:00.54[TK]D-Fenderkombi: Typical low usage : IP 320
19:00.58rantshmmm...
19:01.05anonymouz666[TK]D-Fender: Yes, I understand the fetchid mechanism. The problem is the second row. I only can get the first one.
19:01.30rantshalright let me get some CLI out
19:01.32[TK]D-Fenderanonymouz666: no, you only DO get the first one, because you aren't checking the result to do the loop.  GotoIf is your FRIEND.
19:01.39[TK]D-Fenderrantsh: PASTEBIN OR ELSE.
19:01.41[TK]D-Fender~pb
19:01.42jbotwell, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
19:01.43[TK]D-Fender^^^^^^^^^^^^^^^^^
19:02.01kombi320 goes for some 100 Euro around here..
19:02.14rantshbut how do i make it visible to you?
19:02.15obnauticus[TK]D-Fender any good hardphones that support IAX/IAX2?
19:02.21[TK]D-Fenderobnauticus: Nope.
19:02.24ManxPowerobnauticus: None
19:02.26obnauticuswhy?
19:02.38[TK]D-Fenderobnauticus: Why are there no flying walrus'?!?!!?
19:02.45kombimay I repeat my enourmously complicated yet unbeleavably interesting question from above?
19:02.46obnauticusOhh
19:02.48[TK]D-Fenderobnauticus: Because GOD DIDN"T MAKE ANY.
19:02.50obnauticus[TK]D-Fender let me guess...
19:02.51ManxPowerobnauticus: vendors are REMOVING protocols from their phones (MGCP, H323, etc), not adding more protocold.
19:02.55obnauticusDAMNIT
19:02.57obnauticus>: |
19:03.06obnauticusI was gonna say because they want them compadible with everything.
19:03.06obnauticuslol.
19:03.09ManxPowerobnauticus: SIP works just fine
19:03.12[TK]D-Fenderobnauticus: Who needs it anyways...
19:03.14obnauticusya
19:03.16obnauticusthat's what i was thinking
19:03.38kombiwrite to mysql from IVR, later Page() from mysql? good or stupid?
19:03.49lymeca[TK]D-Fender: Hi again.  One quick question about ports that should be opened on my * box's firewall.  5060 for SIP, both TCP and UDP should be opened correct?  Anything else?
19:03.59ManxPowerkombi: Yes
19:04.12[TK]D-Fenderkombi: We'd need to see the big picture and why you think MySQL should be involved.
19:04.17Supaplexwho else pays you like trxtel to receive incomming calls, or outbound tollfree?
19:04.30[TK]D-Fenderlymeca: 5060,10000-20000
19:04.31kombiFender: just some way to store data..
19:04.35ManxPowerlymeca: NO!  SIP signalling uses UDP/5060.  Audio uses UDP ports 10000-20000 by default
19:04.42Qwell[]Supaplex: The state of IA, or ID, or whatever it is
19:04.54kombiFender: first thing coming to mind of the average php nerd
19:05.13[TK]D-Fenderobnauticus: No model number on that.  AVOID.
19:05.14lymecaPorts 10000-20000?? I don't need that many ports open do I??  What's the minimum number I can open?
19:05.27obnauticus[TK]D-Fenderk ya i thought so
19:05.31obnauticusmight be a fake picture
19:05.32obnauticuslol.
19:05.35ManxPowerlymeca: it is set in rtp.conf
19:05.43[TK]D-Fenderlymeca: You can restrict it.  This is the default port range for RTP
19:05.53kombilymeca: don't worry about the high ports there.. i believe udp is enough though
19:06.28[TK]D-Fenderobnauticus: Looks like a 500/501.  The price isn't set and will jump, so forget it
19:06.29rantshhttp://pastebin.ca/590723
19:06.29lymecaAh okay so SIP doesn't use 5060 TCP at all?  Just 5060 UDP for control data, and the voice data is transferred over the RTP ports?
19:06.31*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
19:06.41[TK]D-Fenderlymeca: Correct
19:06.45kombilymeca: I believe yes
19:06.48kombi...
19:06.53obnauticus[TK]D-Fender it's a buy it now lol.
19:06.56rantshso, do I need to repeat my situation's description?
19:06.59obnauticusfor 85$s
19:07.27lymecaSo in my rtp.conf can I specify a very small port range of, say, 10 or less ports?
19:07.39[TK]D-Fenderobnauticus: No statement of condition, supplying power cales, etc..... this is incrementally unhealthy....
19:07.46obnauticusya
19:07.50obnauticusi wouldn't buy it anywa
19:07.54obnauticusi'll leave you alone now
19:07.54obnauticuslol.
19:07.56[TK]D-Fenderobnauticus: caveat Emptor
19:08.13obnauticuswtf is that [TK]D-Fender
19:08.20kombiIP320 looks kewl to me, what can it not do that is important? (I mean like, really)
19:08.24obnauticusnevermind [TK]D-Fender
19:08.30ManxPowerlymeca: 2 ports per call
19:08.40[TK]D-Fenderobnauticus: LATIN
19:08.44obnauticusyes
19:08.45obnauticusit's a law
19:08.48obnauticusof some sorts lol.
19:08.53kombialaw..
19:09.00obnauticusULAW!!!
19:09.06kombimelaw
19:09.08obnauticusthat's owned.
19:09.12obnauticusourlaw honey.
19:09.20kombipoint taken
19:09.22*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
19:09.27rantshhello, anyone?
19:09.29lymecaManxPower: Right, and I read that RTP chooses an odd port and the next highest even port for a given session... so opening up 4 ports for RTP to use would mean * could route two calls at once?
19:09.38obnauticusrantsh anyone here.
19:09.47ManxPowerlymeca: WRONG!
19:09.54lymecaOh
19:09.55ManxPowerThe odd port thing is specific to CISCO
19:10.04[TK]D-Fenderrantsh: That doesn't tell us anything, nor do we see what happens upon the call terminating, nor does this let us see the FILE itself.
19:10.19ManxPowerand rtp.conf only sets the LOCAL ports.  Remote ports are set by the remote device.
19:10.30rantshthat was the whole output during a whole call
19:10.40[TK]D-Fenderrantsh: Why don't we see the hangup?
19:10.47obnauticus[TK]D-Fender cisco makes good hardware and all but they are BITCHEs
19:10.55[TK]D-Fenderrantsh: increase your verbose & debug and do it again, and host the recording.
19:10.58lymecaFrom Wikipedia: "RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications."
19:11.10[TK]D-Fenderobnauticus: And their firmware BLOWS, as does their licensing.
19:11.10obnauticusBut noo...i can't say that because we are probably using a cisco device to communicate.
19:11.13rantsh[TK]D-Fender:ok
19:11.21obnauticus[TK]D-Fender i do like their catalysts though
19:11.21kombiIP320 <- I'll buy 4 or 5 now
19:11.23obnauticusi use one
19:11.25lymecaI got it backwards...
19:11.30obnauticusand i have a nortel
19:11.33*** join/#asterisk ccesario (n=ccesario@201-0-124-46.dsl.telesp.net.br)
19:11.42[TK]D-Fenderlymeca: thats what we call an RFC (request for comment).  Asterisk operates in a "NO COMMENT" capacity :)
19:11.54obnauticusYESSSS
19:12.12obnauticusbut i bet soon people will develop protocols for asterisk and write RFC's for them
19:12.22lymecaI know what an RFC is... so * ignores the RTP RFC in terms of which ports to use?
19:12.32[TK]D-Fenderobnauticus: Like IAX2!  Which... NOBODY CARES ABOUT!
19:12.43bkruse_homeoh gosh, another rfc war......lame
19:12.43Qwell[]lymeca: it doesn't ignore anything - it very clearly states that there is no standard
19:12.56*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
19:12.56obnauticus[TK]D-Fender WORD.
19:13.36ManxPowerTHERE is an IAX2 RFC
19:13.43Qwell[]There's a draft RFC :)
19:13.45obnauticusIS there?
19:13.48obnauticusOh god.
19:13.54ManxPowersorry, DRAFT RFC
19:14.05rantsh[TK]D-Fender: dude, increased verbosity from 5 to 20, got the same stupid output.... EXACTLY
19:14.11ManxPowerobnauticus: a 7 second google search would have told you that.
19:14.18lymecaOkay... but it's still true that the actual voice data of an SIP call is transferred over a single RTP port with a next-higher RTCP port for control, in addition to SIP control data on 5060 UDP?
19:14.21obnauticusManxPower i don't care that much
19:14.32ManxPowerlymeca: no it is not true
19:14.53ManxPowerlymeca: not with asterisk, at least, as Asterisk does not support RTCP
19:15.04Qwell[]yes it does O.o
19:15.21[TK]D-Fenderrantsh: and DEBUG
19:15.35ManxPowerQwell: When I say "Asterisk" I mean a released version of asterisk, not some developement version of asterisk that may or may not work and not a 3rd party patch
19:15.42Qwell[]ManxPower: like 1.4?
19:15.42lymecaSo...is my last question correct Qwell[]?
19:15.57ManxPowerQwell: 1.4 supports RTCP?  Cite your source
19:15.58rantshI don't intend to be a pain, and I'd love not to bother anyone with this n00b stuff, but I'm not very proficient at asterisk [yet] and forums are just to slow, apart from that english is not my first language
19:16.04Qwell[]umm, rtp.c?
19:16.16lymecaAn SIP call uses 5060 UDP, and a single RTP and single RTCP port?
19:16.19[TK]D-Fenderlymeca: Seriously... just let go.  You're going to give yourself an aneurysm
19:16.21*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
19:16.26lymecaNevar!
19:16.26Qwell[]<PROTECTED>
19:16.29rantshI'm not making up excuses, I'm just asking for a little patience... thanks though
19:16.48lymecaI need to plan which ports I need open in my firewall.
19:16.51ManxPowerQwell: *shrug*  Cite someone that actually has gotten it to work and uses it then
19:16.57[TK]D-Fenderrantsh: Thats fine, I'm not wailing on your ass yet so you're fine...
19:16.59Qwell[]jsmith
19:17.04ManxPowerlymeca: you will not be using RTCP
19:17.29lymecaHow do you know that/why not ManxPower?
19:17.55ManxPowerlymeca: because until 1.4 was released asterisk did not support RTCP and so I'll bet you won't need to use it
19:18.04ManxPowerdo you even know what RTCP IS?
19:18.14rantsh[TK]D-fender: hehehe
19:18.26*** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net)
19:18.29lymecaYes, but admittedly only a very conceptual understanding.
19:18.33rantshbrb with a good output
19:20.18obnauticus[TK]D-Fender i have two cards with http://www.conexant.com/products/entry.jsp?id=122 that chipset...can those work as FXO/FXS cards?
19:20.29Qwell[]obnauticus: no
19:20.32obnauticusdamnit
19:21.57[TK]D-Fenderobnauticus: No, your craptastic little winmodem really IS worthless.  Seek therapy now.
19:22.07ManxPowerobnauticus: that should have been "I'm a cheap bastard, damnit."
19:22.14[TK]D-Fender~ygwypf
19:22.15jboti heard 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.
19:22.17[TK]D-Fender^^^^^^^^^^^^^
19:22.40obnauticusi didn't pay for them
19:22.48obnauticusthey came with some POS oem comptuers i have running as serverrs now
19:22.49obnauticuslol.
19:22.53ManxPowerso they are worth nothing
19:22.57obnauticusword.
19:23.07*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
19:24.20*** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net)
19:24.29[TK]D-Fenderobnauticus: Or every penny you payed for them
19:24.39obnauticus1?
19:24.46[TK]D-Fenderobnauticus: Double your money back garauntee!
19:24.51obnauticusrofl
19:24.53obnauticus2!!!
19:25.17obnauticusI just need to stop looking for crap i don't need
19:25.18obnauticusbut looks cool
19:25.18obnauticuslol.
19:25.29[TK]D-Fenderobnauticus: If I offer someone a penny for their thoughts, and they give me their 2 cents worth... what happens to the other penny?
19:25.41ManxPowerobnauticus: you need to start USING asterisk
19:25.48[TK]D-Fenderobnauticus: INDEED
19:25.49ManxPowerBut first read The Book
19:26.06[TK]D-Fenderobnauticus: THIS is what you need.....
19:26.11[TK]D-Fender~osmosis
19:26.11jboti heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
19:26.13obnauticusManxPower im using asterisk...
19:26.17obnauticusand configuring menus and stuff
19:26.22obnauticuswhile i ask retarded questions over irc.
19:26.29obnauticusand reading the O'Reilly book.
19:27.02ManxPowerI would suggest reading the book before asking questions
19:27.25*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
19:28.38[TK]D-Fenderobnauticus: I suggest.... OSMOSIS
19:31.30*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
19:33.05Slingkydoes * support mgcp as client ???
19:34.14[TK]D-FenderSlingky: not AFAIK
19:35.21Slingkythanks man
19:35.32Slingkythat means * can'T be connected to primus tbb
19:35.39Slingkycause they use mgcp
19:35.53Slingkynow sip but they restrict access via mac address
19:35.53*** join/#asterisk nibbler_de (n=nibbler@as250.net)
19:36.01ManxPowerwhy can't the Primus be configured as an MGCP client?
19:36.18thevokeheer nibbler_de :)
19:36.30nibbler_demr. vookie! ;)
19:36.49Slingkyprimus tbb is a voip provider which use mgcp
19:36.50ManxPowerI know Asterisk can be an MGCP client, I seem to vaguely recall MGCP server support being added before 1.4 was released, but I don't know if it made it into the release or not
19:36.58nibbler_dedoes anybody of you have a hint for me how to terminate calls automagically after 2 hours?
19:37.09[TK]D-FenderSlingky: Time to start Spoofing....
19:37.23thevokenibbler_de: set the duration to 2hrs?
19:37.32ManxPowerSlingky: They can't restrict on MAC address as the MAC can only be read on the LOCAL NETWORK
19:37.33Slingkymy linksys spa2102 can do mac clone, but i don't know sip settings for primus
19:37.47nibbler_dethevoke: where?
19:37.48[TK]D-Fendernibbler_de: "show function TIMEOUT"
19:37.53thevokeone sec
19:37.53nibbler_dethanks
19:37.54Slingkyi think they restrict on firmware
19:37.56ManxPowerMaybe you mean that the MAC is used as the USERID????
19:38.05[TK]D-FenderSlingky: Change your useragent, etc
19:38.25ManxPowerThe source MAC of your packets will have the MAC of the internal interface of the Primus router at the Primus NOC
19:38.31Slingkyanyway, i'm looking for a good sip provider that support faxing (acanac is bad !)
19:38.50NuggetSlingky: asterlink.com does a shedload of fax stuff these days
19:38.51thevokeSlingky: mainly depends on the jitter towards your sip provider
19:39.01ManxPowerSlingky:  You think you'll get FaxOverVoiceOverIPOverInternet to work?
19:39.17ManxPowerAh, to be young and optimistic again.
19:39.32thevoke:>
19:39.32*** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-248.bwic.chi.wayport.net)
19:39.37*** join/#asterisk zotz (n=zotz@24.244.163.157)
19:39.49[TK]D-FenderManxPower: this is SO you : http://www.despair.com/pessimistsmug.html
19:40.49ManxPowerYup!
19:41.08[TK]D-FenderFailure : When your best jsut isn't good enough.
19:41.32ManxPowerExactly!
19:43.01rantsh[TK]D-Fender: alright this is the debug (complete) output http://pastebin.ca/590788
19:44.23rob0:-( [tm]
19:47.48*** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net)
19:49.42rantshI don't know what else to try...
19:49.58rantsh:'(
19:52.31rantshis there anyone here?
19:53.23rantsh[TK]D-Fender: can you read me?
19:53.48*** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29)
19:54.01*** join/#asterisk Eradan (n=eradan@66.218.245.192)
19:54.56*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
19:55.03*** join/#asterisk johann8384 (n=johann83@gateway.myogre.com)
19:56.00*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
20:00.03rantsh[tk]D-Fender: http://pastebin.ca/590813
20:02.18NovceGuruwow extensions.conf hurts my head :(
20:02.34[TK]D-FenderNovceGuru: ...
20:02.36[TK]D-Fender~osmosis
20:02.37jbotsomebody said osmosis was the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
20:02.49NovceGuru:)
20:03.23NovceGuruit can still hurt!
20:04.29[TK]D-FenderNovceGuru: You can't feel pain while unconscious, and we'll feel a lot better too....
20:06.45Eradanquick question ... recently installed PRI w/ Digium 120p ... outbound calls i get the "Your call cannot be completed as dialed". i've tried the 'ww' trick and can't seem to figure it out.
20:06.53[TK]D-Fenderrantsh: I still don't trust your mix.  Don't do it.  Also you provided SIP debug, I wanted CORE debug, not SIP./
20:07.29[TK]D-FenderEradan: You don't use "w" with digital channels and had better NOT.
20:07.46rantshthat'll be done with "core debug" then?
20:07.51Eradan[TK]D-Fender: i removed it but did test using it.
20:07.54[TK]D-FenderEradan: Make sure your PRIDIALPLAN, and PRILOCALDIALPLAN values are set right in zapata as well
20:07.55rantshhow can I stop debug after?
20:08.08[TK]D-Fenderrantsh: "sip no debug"
20:08.57rantsh[TK]D-Fender: ok, and to start core debug? "core debug"?
20:09.50[TK]D-Fenderrantsh: Sounds about right
20:10.02[TK]D-Fenderrantsh: or jsut "set debug 10)
20:11.08DragoraNwhat softphone to use on windows?
20:11.12DragoraNwhich has everything
20:11.25anonymouz666anyone in here can give a good reason to not use a gateway fxo
20:13.01[TK]D-FenderDragoraN: eyeBeam
20:13.15DragoraN[TK]D-Fender: free...
20:13.21[TK]D-Fenderanonymouz666: Try using a more specific MODEL.....
20:13.29[TK]D-FenderDragoraN: NONE
20:13.36DragoraN:)
20:14.10anonymouz666[TK]D-Fender why would you use a asterisk with an tdm8 if compared to some gateway 8 fxo
20:14.21[TK]D-Fenderanonymouz666: LINK it please
20:14.35anonymouz666it could be ANY gateway
20:15.51[TK]D-Fenderanonymouz666: And what is a TDM8.
20:15.58[TK]D-Fenderanonymouz666: Please be SPECIFIC.
20:16.25anonymouz666TDM808P
20:16.32anonymouz6668 fxo digium card
20:17.39anonymouz666tdm is more expensive and you need to convice your boss to buy the card and not the gateway. He wants to know the quality impact in pratics terms
20:17.45[TK]D-Fenderanonymouz666: Well TDM card is cheaper, removes 1 voip layer for better direct access for the basis of faxing, etc.
20:17.45anonymouz666what would you see?
20:17.57[TK]D-Fenderanonymouz666: I wouldn't use EITHER
20:18.14[TK]D-Fenderanonymouz666: And you did not specifi what unit you were considering in place of that card.
20:18.30[TK]D-Fenderanonymouz666: Sangoma A200d
20:18.41[TK]D-Fenderanonymouz666: is what I would use personally.
20:18.48anonymouz666could you explain the part "removes 1 voip layer for better direct access for the basis of faxing"?
20:20.20Sweeperbeen flaky since the weekend, so I stripped out the old crappy demark and put in a new one, and cleaned out the rats nest of wiring
20:20.45mercestesHey Qwell.  Ping.
20:20.49Qwell[]?
20:20.53SweeperPONG
20:20.55[TK]D-Fenderanonymouz666: You don't want to introduce latency to SpanDSP or any extra conversion on the line if faxes are involved.  Mind you you probably don't want * involved AT ALL in that case.
20:21.02*** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com)
20:21.07mercestesYou remember that internal RTCP NTP clock skew deal?
20:21.10Qwell[]yeah
20:21.12mercestesMake that 3 systems.
20:21.14*** part/#asterisk catch23 (n=catch23@69.60.124.109)
20:21.19vltHello. I want to connect my legacy PBX "behind" my bristuffed Asterisk as shown in the "BRI-PBX-integration" sample from qozap. I got the following config: http://pastebin.ca/590839 -- The 2 TE spans work perfectly but I don't get the NT ports running. Any idea what I missed here?
20:21.21mercestesstandard box, TDMP400p, Debian
20:21.25mercesteserr.
20:21.25Qwell[]Why has nobody opened a bug report yet? ;/
20:21.28mercestess/debian/ubuntu/
20:21.30techman97_andyhowdy doo all - anyone know of a good interface I could use to manage my recordings?
20:21.38Qwell[]techman97_andy: bash?
20:21.39mercestesHeh, not a problem.
20:21.40bkruse_homeQwell[]: and they expect you to know and fix it already :/
20:21.58*** join/#asterisk thedingaling (n=jjones@216.70.38.210)
20:21.59techman97_andyI would prefer a webpage for my call center manager\
20:22.15*** part/#asterisk diskfree (n=bve@www3.datarack.nl)
20:22.20[TK]D-Fendertechman97_andy: Go wrinte one.
20:22.25[TK]D-Fenderwrite*
20:22.41anonymouz666[TK]D-Fender: faxes are involved. And it is very important.
20:22.56[TK]D-Fenderanonymouz666: Keep them the hell away from * then.
20:23.14[TK]D-Fenderanonymouz666: At which point it doesn't matter what you use as a gateway.
20:23.22anonymouz666but the IP phones will be using g711 in pass-through mode
20:23.39anonymouz666that way works. I used here in the office
20:24.23*** join/#asterisk andrewc (n=andrewc@dsl254-017-249.sea1.dsl.speakeasy.net)
20:25.16mercestesWriting a bug report now.
20:25.45[TK]D-Fenderanonymouz666: This has nothing to do with faxing however so again, you can use whatever card or gateway you want.
20:27.52*** join/#asterisk andrewc (n=andrewc@dsl254-017-249.sea1.dsl.speakeasy.net)
20:27.56*** join/#asterisk waverly360 (n=waverly@209.12.249.243)
20:28.27*** join/#asterisk zaggynl (n=zaggynl@dsl-083-247-110-137.solcon.nl)
20:31.24ManxPoweranonymouz666: If you want reliable fax you will not route them thru Asterisk
20:32.01[TK]D-FenderOk, I'm off, back in many hours time...
20:32.09zaggynlHi, I've got this asterisk server set up, with all ports forwarded and such, and we can connect to each other, but there is no audio
20:32.29ManxPowerzaggynl: did you set localnet and externip options?
20:32.39zaggynlyes sir
20:32.41*** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar)
20:32.55zaggynlmy sip.conf: http://pastebin.ca/590865
20:33.17ManxPowerSince ports are forwarded I assume that you are running the Asterisk server behind nat and what are "all ports"  ports 1-65535 TCP and UDP?
20:33.59zaggynlnah, just the service ports required by asterisk
20:34.08ManxPowerset direct rtp and srv lookups to NO
20:34.17ManxPowerzaggynl: What ports did you forward?
20:34.34zaggynljust 5060
20:35.07ManxPowerSIP signalling uses port 5060/UDP.  AUDIO us sent using the RTP protocol, which defaults to ports 10,000-20,000, you can set the port range in rtp.conf
20:35.13zaggynlahah
20:35.29ManxPowerIn the furture do not use the term "all ports"  Say what you mean
20:35.37zaggynlokay, sorry
20:36.09zaggynldo I have to forward the whole range? I can't forward a few of them?
20:36.24zaggynlI can't enter full ranges in my router
20:36.37ManxPowerif you want to forward fewer then set the range in /etc/asterisk/rtp.conf
20:36.48ManxPowereach call will require two ports
20:36.53ManxPowerfor audio
20:36.58zaggynlokay
20:37.35DragoraNwhich is the most complex web gui (free)
20:37.49Qwell[]why on earth would you want a complex web gui?
20:38.12__DAWto look cool man...
20:38.26Qwell[]install phpshell and use that.  It's technically a web gui
20:39.21zaggynlheheh
20:39.27zaggynlphp linux shell
20:40.16Nuggetew, that's disgusting.
20:40.18ManxPower__DAW: Trixbox/AMP
20:40.30__DAWhuh?
20:40.40ManxPowerbut if you ask questions about it on THIS channel we will hunt you down, tie you to a tree and force you to use Cisco Call Manager
20:40.53ManxPower__DAW: a complex gui for Asterisk is called AMP or Trixbox
20:41.02ManxPower~amp
20:41.02jbotrumour has it, amp is NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
20:41.04__DAWahhh
20:41.04ManxPower~trixbox
20:41.04jbot[trixbox] a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
20:41.07anonymouz666I am just curious... how much is a cisco call manager solution ?
20:41.31Qwell[]anonymouz666: anywhere from $20000 on up to several million
20:41.33Nuggetanonymouz666: the answer to that question is "how much can you afford?"
20:41.35ManxPoweranonymouz666: The quote we got was about 1/4 of a million USA dollars for one 50 person office
20:41.37Nuggetseriously.
20:41.38ManxPowerthis was a while ago.
20:41.48Qwell[]ManxPower: good lord
20:42.00Qwell[]why does that surprise me though?  It really shouldn't
20:42.02Nuggetcisco is like ibm global services.  trying to get a simple price out of them is like nailing jello to the wall.
20:42.05ManxPowerthat is about on par with solutions from Nortel, etc
20:42.25Nuggetyou always have to come at the price backwards, starting at your wallet.
20:43.14Qwell[]stare at it, hell.  just give it to them, they'll give you what you can afford
20:43.19ManxPowerQwell:  You need 4 PCs w/4 copies of licensed Windows NT to start off with.
20:43.27*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
20:43.27Qwell[]for FIFTY users?
20:43.28ManxPowerprimary and backup PBX, primary and backup voicemail
20:43.29mockerI love jbot.
20:43.42Qwell[]mocker: jbot loves you too, I'm sure
20:43.47Qwell[]~love
20:43.47jbotLove suffereth long, and is kind; love envieth not; love vaunteth not itself, is not puffed up, doth not behave itself unseemly, seeketh not her own, is not easily provoked, thinketh no evil; rejoiceth not in iniquity, but rejoiceth in the truth; beareth all things, believeth all things, hopeth all things, endureth all things....  And now abideth faith, hope, ...
20:43.59mocker:)
20:44.01ManxPowerQwell: it was $12,000 just for the phones.
20:44.06Qwell[]umm
20:44.16*** join/#asterisk waverly360 (n=waverly@209.12.249.243)
20:44.24Qwell[]that's like, what, 500% markup from what they sell them for usually?
20:44.32ManxPowerEveryone required speakerphone and display
20:44.43ManxPowerQwell: What does a 7960 cost new w/service contract?
20:44.46Qwell[]oh, maybe not, my math wasn't right
20:44.47Nugget$12k / 50 is "only" $240 per
20:44.50Qwell[]that's about right
20:44.52Nuggetthat's not too awful
20:45.02DaminHmmm.. anyone having 800 termination problems today?
20:45.25ManxPowerDamin: I think China is.
20:45.38DaminManxPower: Well, that is normal...
20:45.50DaminManxPower: Specifically, X/O seems unable to terminate an 800 call today..
20:46.17mockerohwtf.
20:46.34mockermy fax problems may actually be the damn fax machine.
20:46.49mockerbecause someone decided to stack 400 pages of paper on top of it.
20:46.57ManxPowerRight around the time we were looking at the CCM solution we switched to a provider that gave us unlimited LD in the two states where there are offices.
20:47.13ManxPowerSo toll bypass
20:47.17ManxPowerSo toll bypass was not a selling point
20:47.19anonymouz666FAX sucks.
20:47.25anonymouz666must be banned from earth
20:47.38ManxPowerI never have fax problems with my customers
20:48.16zaggynl:D
20:48.57*** join/#asterisk sradu (n=sradu@82.79.243.82)
20:49.20sraduhi
20:49.56sraduI am trying to setup openser+asterisk following the voip-info.org howto. I am encountering a problem when trying to start asterisk after I enable odbc storage
20:50.00sraduJun 26 20:56:59 VERBOSE[31012] logger.c:  [app_voicemail.so]Jun 26 20:56:59 WARNING[31012] loader.c: /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: vm_delete
20:50.00sraduJun 26 20:56:59 WARNING[31012] loader.c: Loading module app_voicemail.so failed!
20:50.03mockerI'm going to petition my elected officials to end of life fax machines like they are doing w/ broadcast TV.
20:50.20sraduif I don't enable odbc storage (at compile time) it works OK
20:50.30mockerManxPower: How do you do your faxes?
20:50.37mockerdedicated line? iaxmodem?
20:50.38sraduam using 1.2.19
20:51.12*** join/#asterisk dalfry (n=dalfry@70.89.177.109)
20:51.40ManxPowermocker: I plug the fax machine into a POTS line from the telco
20:51.49mockerManxPower: cheater. :)
20:52.05*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
20:52.06ManxPowermocker: No.  This is the way to run fax.
20:52.26NuggetListen to ManxPower.  ManxPower is wise.
20:52.28ManxPowerThere were no end of problems when I tried to run fax via Asterisk
20:52.43Nuggetfax + voip = endless headaches culminating in your termination.
20:52.55dalfryanyone knows a good / scalable fastagi server?
20:53.08ManxPowerI SUSPECT it had to do with the fact that I had to really screw up the gains to eliminate echo.  Someone using hardware EC or the HPEC might have fewer problems, but even then I would not run it over IP
20:53.23ManxPowerdalfry: Your question makes no sense.
20:53.43dalfryManxPower: well, I was looking at Orderlycalls and it seems it hasen't had a release since 2005
20:54.03dalfryI have an app currently coded using phpagi and the server dies after 90 - 100 calls
20:54.12ManxPowerdalfry: That is not a fast agi server.  That is a specific application that uses fastagi
20:54.26dalfryokay...
20:54.27ManxPowerdalfry: then fix your application
20:54.47zaggynlOkay, I forwarded the ports, but still no audio, and after a few seconds, I get this from the asterisk shell:
20:54.51zaggynl[Jun 26 18:11:29] WARNING[6752]: chan_sip.c:1900 retrans_pkt: Maximum retries exceeded on transmission OTBkNGZiMzgxMzVmNzBmNGQzNTkzOWZhYTI5NGJiNzI. for seqno 2 (Critical Response)
20:54.54ManxPowerAll FastAGI does is send/get it's input via a socket, rather than a local file handle
20:54.57dalfryManxPower: there is a limit to which I can tune phpagi. need something that can use the resources of the machine better
20:55.12ManxPowerdalfry: You have the source code to PHPAGI
20:55.13dalfrythreading, etc
20:55.27ManxPowerdalfry: so what you are REALLY looking is a language specific wrapper for the AGI functions.
20:55.39dalfryI understand
20:56.04ManxPowerI am not aware of any other PHP AGI libraries.  It should be trivial to write one.
20:56.41dalfrywhat I am looking for is something that uses the fastagi protocol but does not spawn a new process for every call that it gets
20:57.33ManxPowerdalfry: All asterisk does is connect to the specified socket and blast the info at the application.  What your application does with it is up to you
20:57.34KDandalfry: what you're saying is you want a kind of mod_php for asterisk
20:57.37KDanright?
20:57.59dalfryyeah kinda
20:58.02KDanso that it doesn't spawn off a new php proces for each call
20:58.03ManxPowerKDan: no.
20:58.14dalfrybut not exactly that
20:58.26KDanwhy not exactly?
20:58.28ManxPowerYOUR FAST AGI PROGRAM IS A SEPERATE PERSISTANT PROCESS
20:58.31ManxPowerIT IS NOT PART OF ASTERISK
20:58.37dalfryManxPower: yes I know
20:58.44KDanah, well, then, what's the problem?
20:58.46ManxPowerKDan: doesn't know.
20:58.58dalfry:)
20:59.02ManxPowerKDan: He is trying to work around a bug in his application
20:59.23KDani don't think you'll find many PHP implementations of anything that don't assign a process for each request
20:59.32ManxPowerFast AGI uses a TCP socket to communicate with your FastAGI application.  The application can run on the same machine as asterisk or on a remote machine
20:59.36kombiwhat is a good tool go audio edit voice prompts on mac?
20:59.39KDansounds like you need a non-php implementation
20:59.44kombis/go/good/
20:59.52kombishoot...
21:00.00KDankombi: SoundStudio Pro
21:00.08KDaner
21:00.10KDanSoundStudio
21:00.13KDan3.5.2
21:00.21KDanavailable at your local mac shop or bit-torrent client
21:00.34kombiKDan: that was my next question..;)
21:00.42kombithanks!
21:01.20KDanpleasure
21:01.31KDanprobably can be bought online from their site, too
21:02.22kombiKDan: err.. what is the BT-client of choice (mac noob here)
21:02.33kombitomato something?
21:02.37KDanAzureus seems to still be the best, unfortunately
21:02.44KDandespite being a bloated monster
21:02.54KDanit's the most stable and feature-ful so far... :-(
21:03.02*** join/#asterisk saftsack (n=saftsack@pD9E06927.dip.t-dialin.net)
21:03.02kombilol, I like bloated monsters..
21:03.08KDanthen install Azureus
21:03.11KDanyou won't be disappointed!
21:03.23KDanMakes Jabba the Hut look like a stick-man on a diet.
21:03.36KDanHmm. nice analogy. Maybe I should become a stand-up comedian.
21:04.32kombiKDan: allways a possibility.. beat digging ditches!
21:04.35mercestesThat was pretty graphic there, KDan.  I wouldn't quit your day job tho.
21:04.40kombis/beat/beats/
21:04.51kombiagain.. scary bot!
21:05.01mercestes~botsnack
21:05.01jbotmercestes: :)
21:05.25mercesteskombi, He's better than abot - ibot.
21:05.37KDanmercestes: well, i'm about to quit the day job, but i have other plans. Hopefully they involve going to comedy shows as a spectator
21:05.39kombiKDan: 3.0 or 2.5? The former looks even more bloated..
21:05.57KDani've got the latest one. It's overbloated, but hell, i have a mbp
21:06.03KDan:-)
21:06.36KDanit is full of useless crap though
21:06.47KDantry 2.5 maybe
21:06.52KDanlet me know how it goes
21:06.55kombinice icon (yeek)
21:07.40kombijeez...
21:07.47iruleis there a CLI command that will tell me about missing sound files?
21:10.08*** join/#asterisk sweeper (i=sweeper@softcheese.net)
21:10.49ManxPowerirule: no.  Asterisk thinks you are smart enough to know of the sound file exists or not
21:11.14iruleok lol thanks
21:11.15ManxPowerYou can do a Playback and then check the value of PLAYBACKSTATUS or whatever it is called.  "show application playback" for more info
21:11.27ManxPowerbut not from the CLI
21:18.20kombiKDan: "You already registered an account on this machine" <- how do you get rid of that? (screwed up the first time)
21:18.50KDankombi: ??
21:18.59KDanno idea
21:19.05kombiKDan: I registered..
21:19.08KDanwhere?
21:19.11ManxPowerkombi: we don't even know what the hell you are talking about
21:19.33KDantry #macosx perhaps... or #mac .. not sure which one exists
21:19.35kombiManxPower: that happens to me often..;)
21:19.44KDani've only had my mac for about a month
21:20.01KDanbefore that i was a good old fashioned ubuntu boy
21:20.11kombiKDan: I had it for a bit longer but all I ever use is Terminal
21:20.16KDanblech
21:20.19KDanterminal..
21:20.21KDanat least use iTerm
21:20.32KDanfree. go google for it
21:21.07kombiTerminal is ok really, transparent and all
21:21.15kombienough mac talk, sorry everyone
21:21.17KDantrust me - iTerm is best.
21:21.27kombi^^^^
21:22.17KDanmust-have apps: iTerm, Quicksilver, Firefox, MacTheRipper
21:22.18Supaplexit's tEh slow
21:22.24KDanwhat is?
21:22.29Supaplexmac terminal
21:22.39KDaniterm is fine. and pretty.
21:23.46sweepera eterm is fine too
21:24.06KDanon os x?
21:26.57watchyi'm having problems dialing out with asterisk 1.2.9.1. Sometimes it works but sometimes I get that it needs more digits
21:27.08watchybut its sending the complete # to the zap
21:29.26ManxPowerwatchy: add a "w in front of your dialed digits
21:29.35watchyhrm
21:29.37*** join/#asterisk ruied (n=ruied@bl7-218-152.dsl.telepac.pt)
21:29.46ManxPowerthis is a question asked all the times.  a 5 min google search should have turned up helpful information
21:30.22watchyin front of every digit or just in front of the #?
21:31.01irulethis is weird, I just installed 1.2.19 on a base installed debian, everything seems perfect, BUT I get an error that there is no vm-youhaveno sound file? this is part of standard voicemail sounds! so I just find it and there it is! how may I fis this problem? here is the evidence http://pastebin.ca/590939
21:31.14ManxPowerjust before the first digit
21:32.07ManxPower[root@pbx-1 sounds]# ls vm-youhaveno*
21:32.07ManxPowerls: vm-youhaveno*: No such file or directory
21:32.23ManxPowerthat file is not part of asterisk
21:32.49*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)
21:33.21ManxPowerlooks like that file might be part of the sounds-extra package
21:33.24obnauticussergee
21:33.26obnauticusare you there?
21:33.50iruleManxPower that file is the basic voicemail message telling you that there are no messages when you log into an empty mailbox :s
21:34.47kombiKDan: got that thing wrangled, now where do you put the search term?
21:35.01ManxPowerirule: it certinally isn't going to find it in the es.anterior directory
21:35.08irule<PROTECTED>
21:35.10ManxPowerand no, that file is NOT part of the standard asterisk
21:35.44ManxPower[root@pbx-1 sounds]# find /var/lib/asterisk/sounds -name "vm-youhaveno*" -print
21:35.45ManxPower[root@pbx-1 sounds]#
21:35.49KDankombi: search term?
21:35.57ManxPowerIt might be part of 1.4 or sounds-extra
21:36.08KDanyou put it in isohunt.com, i guess
21:36.08ManxPowermind you my searches are on a production box
21:37.47ManxPowerirule: on 1.2 it plays vm-youhave vm-no and vm-messages
21:38.01ManxPowerI just tested it
21:38.09iruleoh I see, thanks!
21:39.03*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
21:39.07*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
21:39.49ManxPowerirule: http://pastebin.ca/590951
21:40.09ManxPowerirule: I've been at this much longer than you have been.
21:41.29iruleManxPower I bet, well I still thank you for your response :)
21:43.50*** join/#asterisk CyBeR_GeeK (n=CyBeR_Ge@201.89.88.30)
21:44.06*** join/#asterisk asterisknerds (n=asterisk@66.7.122.93)
21:44.06asterisknerds<PROTECTED>
21:44.24*** join/#asterisk anthm (n=anthm@dhcp64-134-34-248.bwic.chi.wayport.net)
21:44.24*** mode/#asterisk [+o anthm] by ChanServ
21:44.32*** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net)
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21:44.56*** mode/#asterisk [+o anthm] by ChanServ
21:45.20*** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-248.bwic.chi.wayport.net)
21:46.19watchyexten => _9NXXXXXX,1,Dial,Zap/g1/www${EXTEN:1}
21:46.23watchywould that be correct?
21:46.55ManxPowerwatchy: one or two "w" should be enough.
21:47.26ManxPowerAsterisk does NOT wait for dialtone  before dialing.  If you telco is even a little slow, it will miss the first digit.  "w" on FXO ports puts in a .5 second delay
21:47.26dasuberdavidwatchy: each w is 500ms to wait before sending digits
21:49.38watchyso 2 ws is enough?
21:50.01ManxPowerwatchy: How would I know how long of a delay your telco requires?
21:50.17watchymagic?
21:50.25Juggieits also worth noteing the delay can vary.
21:50.51ManxPowerI may be a fairy, but I'm not magical
21:51.07Juggiehah
21:52.37*** join/#asterisk ZX81 (n=matt@202.20.97.211)
22:01.20iruleI made a few dialplans with 1.4, so Set(CHANNEL(language)=es) is all over, then I put them on 1.2 but that uses Set(LANGUAGE()=es), and my dialplans are broken :s  ...is there a variable in both that will let me find out where I am and act accordingly? thanks
22:01.39*** join/#asterisk moeSizlak (n=0mar@static-69-95-250-34.buf.choiceone.net)
22:02.04moeSizlakhey guys i need a way to defeat a howard stern war-dialing prank-bot.  any ideas?
22:02.48Qwell[]eh?
22:03.49irulehttp://www.softwink.com/iwar/
22:03.56*** join/#asterisk ManxPower (n=manxpowe@74.sub-70-221-89.myvzw.com)
22:05.52*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) [NETSPLIT VICTIM]
22:05.52*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)
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22:07.48*** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net)
22:08.35Aces1Uphey all...  if i want to create a billing system, and also have multiple DID's that have to be organized in a database, is an AGI the best thing for this sort of thing?  or can i code it all in the dialplan?
22:10.46Maxxedhttp://www.sun.com/emrkt/blackbox/index.jsp
22:10.49Maxxedthats some cool shit
22:10.50vltHello. I have some problems getting NT mode running on my HFC-4S based BRI card (using qozap). The 2 ports configured as TE work perfectly but I get "DEACTIVATED" and "No D-channels available!" on the NT ports. Can anyone help?
22:11.03vltThe setup: http://pastebin.ca/590839
22:11.24Maxxedfuckin datacenter in a shipping container
22:11.55vltThe ports are shown as [TE TE NT NT] in dmesg and /proc/zaptel/[1-4]
22:13.28vltAny BRIstuff experts around?
22:15.05*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
22:19.14*** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl)
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22:20.02nohopis anyone here using voipbuster.com? and... is it correct that it's sip server is down ? (otherwise i have to make a call to my ISP...)
22:20.57nephflhello
22:22.38vltThe TE ports are connected directly to the NT ports of the telco's NTBA and the legacy pbx's "internal S0" and work fine. The NT ports on their part are connected directly to the TE ports ("external S0") of the PBX. Do I have to plug something between them?
22:23.42vltOr use crossed cables? Or some power supply? Or ...? Any help?
22:25.14moeSizlakcan i fake caller-ID info w/ asterisk ?
22:25.43vltmoeSizlak: Set(CALLERID(num=0123456789)) should work.
22:25.51moeSizlakcan i change the text too?
22:26.06vltmoeSizlak: sorry, Set(CALLERID(num)=0123456789)
22:26.10vltmoeSizlak: yes
22:26.15moeSizlakniiiiiiice
22:26.30moeSizlakdoes this work for any type of phone line?
22:26.37moeSizlakie residential
22:26.55moeSizlakwell actually its a fiber-optic phone service: Verizon FiOS
22:27.25vltmoeSizlak: I _think_ you can't send names on non VoIP lines ...
22:29.03moeSizlakit MAY be a voip line, im not sure
22:29.09moeSizlaki highly suspect it is
22:29.13vltmoeSizlak: ... and maybe even your (VoIP) provider checks if you are allowed to do calls FROM this number.
22:29.22vltmoeSizlak: You're not sure!?!
22:29.30moeSizlakwell theresno copper wires
22:29.35vltmoeSizlak: How is it connected to *?
22:29.44moeSizlakto the verizon fiber termination system
22:29.49moeSizlakin my closet
22:30.16moeSizlakthey run copper wires from the termination system to the phone jack, but no copper to my house
22:30.32dasuberdavidi cant wait till fios comes to our area
22:30.39Qwell[]dasuberdavid: I have fios
22:30.41moeSizlakim loving the 20Mbit downloads
22:30.58Qwell[]dasuberdavid: we're never gonna get fios here - Verizon won't come near us :D
22:31.58*** join/#asterisk javb (n=ws10@tdev213-77.codetel.net.do)
22:32.28vltmoeSizlak: But if the whatever line is converted to copper before it reaches your *box you won't be able to send names. (BTW what channel driver are you using? zaptel?)
22:33.15moeSizlaki havent installed asterisk yet, i just wanna know if it can do what i want it to do.  I assume it runs on any flavor of linux?
22:34.08javbHi, have this issue, want to know if someone here can help me (too much googling): When i call an inside exten, i can pick it up from another exten, with the cmd "pickup()", no prob. But, when the call comes from outside (TDM400P in my case,) from a Zap Chan, i cant pick it up. On my GrandStreams, i get busy tone with 603 error...
22:34.12vltmoeSizlak: yes.
22:34.26moeSizlaksince it does  get converted to copper, ill be able to send #'s but not names?
22:34.59javbAny ideas guys ?
22:35.18Qwell[]moeSizlak: call Verizon and ask them whether you can or not
22:35.39moeSizlakyeah, the average IQ of a verizon cust service rep is 12
22:35.42*** join/#asterisk sandorp (n=sandor@firewall2.wsi.net)
22:35.59Qwell[]and?
22:36.27Qwell[]If I had access to your account, I'd look at tell you.  But I don't.  They do.  Ask them.
22:36.32moeSizlakyeah ill just call em up and tell em i want to fake my caller id info, and ask how....
22:36.35vltmoeSizlak: If the signal over the copper line is a classic analog phone line you even won't be able to change the caller id there. If it's BRI (ISDN) you propably can.
22:37.10javb:s
22:37.40sandorpI have a 2.gG celeron with 512M of RAM and will have, at most, 6 analog lines active at the same time;  what's a sage setting for the echocancel value in zapata.conf ?
22:37.58sandorpoops, that's 2.6GHz celeron
22:39.58*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
22:40.27vltAnyone here who can help me with my HFC-4S (module qozap ports=12) and NT mode problem?
22:41.09JTvlt: what card?
22:42.55vltJT: Hello JT. It's the "CologneChip HFC-4S evaluation board" that's like a Junghanns QuadBRI you helped me configure several months ago. Now I want to change my setup to the legacy pbx _behind_ asterisk. That's what I need NT mode for.
22:43.11JTvlt: does it have jumpers?
22:43.44vltJT: No, it seems to be all done by the "ports=" option of qozap module.
22:44.11JTvlt: so there absolutely no jumpers visible on the board?
22:44.48vltJT: Yes, no jumpers visible.
22:45.14JTyou will need to make an S-bus crossover cable then
22:45.42vltJT: Which lines are crossed over then?
22:45.57vlt... I'll google it ...
22:46.43*** join/#asterisk harlequin516 (n=sham@styk.net)
22:47.07JThttp://www.zytrax.com/tech/layer_1/cables/telephone.html
22:47.08harlequin516Is there a startup file that will execute command on the CLI at startup?
22:48.12snuff-workyou can make a script which starts * then just does 'asterisk -rx 'cli command here'
22:50.22*** join/#asterisk anthm][ (n=anthm@dhcp64-134-34-248.bwic.chi.wayport.net)
22:51.27*** join/#asterisk anthm (n=anthm@dhcp64-134-34-248.bwic.chi.wayport.net)
22:51.28*** mode/#asterisk [+o anthm] by ChanServ
22:52.50*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
22:53.22*** join/#asterisk J4k3 (n=jsuter@openwrt.us)
22:53.28harlequin516snuff-work: Hmm Thanks
22:53.43harlequin516I guess that'll work, though not what I was hoping for.
22:53.52*** join/#asterisk remmo (n=junk@203.62.147.3)
22:54.46ManxPowerharlequin516: What are you trying to ACCOMPLISH?
22:56.20J4k3anyone know of a reasonably affordable DECT VOIP cordless phone?
22:57.13ManxPowerHAHAHAHAHA!
22:57.21ManxPowerJ4k3: I assume you are in Europe
22:57.58ManxPowerDECT was approved only recently in the USA
22:58.14*** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl)
22:58.24ManxPowerUses the 1.9 Ghz band
22:58.31*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es)
22:58.34kink0hello
22:59.02kink0any way to get ( And probably setuserfield to add to CDR ) the information about RTP remote IP:port ?
22:59.52JTDECT can have different frequency bands
22:59.52J4k3ManxPower: I'm in the USA, and as a responsible radio operator I would not be interfering with any licensed communications in place (I've checked the FCC's site, it can't find anything within 50 miles in that range)
22:59.56J4k3DECT 1.9
23:00.01J4k3I don't want a 6.0 phone
23:00.01kink0that is the c=IN IP4 1.1.1.1 SIP message body while INVITE
23:00.08J4k3it won't work at all in this house, 1.9 is sketchy enough
23:00.35ManxPowerJ4k3: I suspect you would have to order a DECT VOIP device from europe and hope the FCC doesn't toss you in jail
23:00.55ManxPowerJ4k3: the DECT 6.0 phone I recommended to a friend REALLY REALLY sucks.
23:00.58J4k3well, I find them on ebay for shipment in the USA.
23:01.00ManxPowerI wish I could find a 900Mhz one
23:01.21J4k3yeah, I wish I Could use a 900 mhz cordless here but I'm using the entire 900 band for other things.
23:01.22ManxPowerDECT 6.0 uses 1.9 Ghz IIRC
23:01.39J4k3ahh, I thought it was actually "6.0 ghz"
23:01.42ManxPowerJ4k3: I would LOVE to have some DSS 900Mhz.  That band is wide open around me
23:02.00ManxPowerJ4k3: I would have to look it up to be totally sure.
23:02.05J4k3ManxPower: ofdm 900 is more fun ;)
23:02.19ManxPowerJ4k3: as long as it is digital I don't care.
23:02.27javbHi, have this issue, want to know if someone here can help me (too much googling): When i call an inside exten, i can pick it up from another exten, with the cmd "pickup()", no prob. But, when the call comes from outside (TDM400P in my case,) from a Zap Chan, i cant pick it up. On my GrandStreams, i get busy tone with 603 error...
23:02.37ManxPowerI've not been able to find more than 1 model of Digital 900Mhz phone.
23:02.56J4k3I want a vowifi phone that doesn't suck, but that appears near impossible ;)
23:03.02ManxPowerJ4k3: 50 acre campground, needs 900Mhz because of trees
23:03.11J4k3ahh yeah
23:03.43ManxPowerI was hoping 1.9 Ghz would be low enough to have it work, but it doesn't.
23:04.33J4k3I need a 49 mhz cordless
23:04.39ManxPowerI've been looking for an afordable wireless phone.
23:04.45J4k3and a 6 meter amp and some cans
23:04.49ManxPowerJT: as long as it is digital HF I'm happy
23:04.57J4k3:D
23:05.24ManxPowerThe campground owners are in a bind because their phones don't work over 90% of the campground
23:05.54J4k3ManxPower: could a hotspot buildout pay for itself there?
23:06.09*** join/#asterisk galeras (n=root@200.31.204.42)
23:06.10ManxPowerJ4k3: The owners are comitted to keeping the WiFi service free.
23:06.15J4k3there are some vowifi phones that are slightly better than being completely disconnected
23:06.17ManxPowerand I will NOT use a VoIP phone
23:06.18J4k3oh snap, forgot
23:06.22J4k3how about a walkie talkie setup?
23:06.38J4k3like a dingotel
23:06.46J4k3its better than total disconnection.
23:07.00ManxPowerJ4k3: As I understand it you are not permitted to link the PSTN to FRS or GMRS walkie talkies
23:07.07J4k3ManxPower: MURS.
23:07.10harlequin516javb: I wonder if you did answer the call that you wanted to pickup?
23:07.19ManxPowerAny affordable MURS devices?
23:07.21J4k3I believe you can autopatch with a GMRS license
23:07.24JTthere are digital modes for hf, but they're not cheap
23:07.33harlequin516javb: I was inm a funny situation where Answering the call made the difference.
23:07.39J4k3cough up your $80-per-organization "The Everyone-on-earth Group!" ;)
23:07.52ManxPowerharlequin516: you never answered my question
23:07.53J4k3JT: bah, psk31 is cheap.. just don't expect to speak in realtime
23:07.55J4k3iparty over psk31.
23:08.00javb<harlequin516> ... could u elaborate?
23:08.13JTManxPower: you're in the middle of nowhere, as if the fcc knows or cares
23:08.53ManxPowerJT: I'd rather not take the chance if I can help it.
23:08.54javbi mean, imagine me calling my phone number using my cellphone, an exten ringing inside, i cant pick up that call.. but i can pick it up if call is made from another exten inside
23:09.04harlequin516ManxPower : Sorry I must have missed the question
23:09.05JTJ4k3: i meant G4GUO mode
23:09.13ManxPowerand yes, there are farmers around here that would just LOVE to send some govt agency against us
23:09.28JTwhy?
23:09.28ManxPowerharlequin516: I asked What are you trying to ACCOMPLISH
23:09.40harlequin516Oh I see it.
23:09.42galerasPlease: How can i prevent ARP broadcast traffic reaching the SIP phones when those phones are power off?
23:09.43ManxPowerJT: a gay campground in the middle of rural farm land Alabama.
23:09.51JTah right
23:09.57JTputs things into perspective
23:10.14ManxPowergaleras: your question makes no sense
23:10.16harlequin516ManxPower : I was just trying to automate my setup for development such that the debug was specified for only specific channels users etc...
23:10.24ManxPowerah
23:10.31ManxPowerharlequin516:  good luck with that
23:10.34harlequin516ManxPower : upon startup I mean.
23:10.54ManxPowerasterisk -rx "command" is the way to go
23:11.18ManxPowerJT: I guess nobody knows of any decent 900 Mhz phones?
23:11.24harlequin516Well I just want to run about 4 CLI commands.  Do you know if I can put \n between the differnt commands?
23:11.35JTManxPower: shrug, they're easy to get in Australia
23:11.49JT900MHz, 2.4GHz, 5GHz cordless
23:11.55ManxPowerJT: I meant something approved for USA
23:12.04JTi don't know
23:12.13ManxPowerYou can get 900Mhz here, but there are all analog
23:12.28ManxPowerand their range is about that of two dixie cups and a string
23:12.31ai-aharlequin516: what about ; as command seperator ?
23:12.51x86ManxPower: you live in a gay campground in the middle of BFE eh?
23:12.53JT2.4GHz phones seem to go around 1km in the open
23:12.58sandorplol
23:13.08javbnothing?
23:13.25ManxPowerWe had a refurb phone, did EXACTLY what we wanted.  The only issue is that Iit dropped calls even if you were next to the base station.
23:13.29ManxPowerx86: yes
23:13.35x86ManxPower: neat
23:13.35galerasmy * is generating a lot of arp broadcast, trying to reach sip phones when those phones are turned off
23:13.46ManxPowergaleras: ignore that
23:14.03x86ManxPower: so you're a flamer eh?
23:14.10JTgaleras: wow, normal tcp/ip operation
23:14.11ManxPowerAnd NO ASTERISK IS NOT GENERATING ARP REQUESTS.;  The OS and networking stack of the OS is.
23:14.12harlequin516ai-a : Does that work?
23:14.15J4k3ManxPower: I had this AWESOME uniden two line POTS phone...  I loved it for many years
23:14.34ManxPowerJ4k3: And it was discontinued, right?
23:14.37J4k3battery pack died, built a new one for it.
23:14.49J4k3it was discontinued when I got it.. $40 off the frys shelf "refurbished"
23:14.55ManxPowerx86: something like that
23:15.01x86ManxPower: cool
23:15.07kink0any way to get ( And probably setuserfield to add to CDR ) the information about RTP remote IP:port ?
23:15.15J4k3it'd work out to my mailbox til we put the SR9's (900 mhz ofdm radios) up.
23:15.19ManxPowerkink0: what version of asterisk
23:15.35ManxPowerJ4k3: what are those radios used for?
23:15.38kink0ManxPower, I am ussing both 1.2.x and 1.4
23:15.47*** join/#asterisk {Sean} (n=Sean@206-171-8-130.ded.pacbell.net)
23:15.53kink0mainly I still ussing 1.2
23:15.53J4k3ManxPower: wireless internet access (distribution
23:15.54J4k3)
23:15.55ManxPowerkink0: in 1.4 I believe that information is available.  check README.variables
23:16.08{Sean}Is there anyone here that does contract asterisk development?
23:16.20*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
23:16.23ManxPowerJ4k3: I could get cheap access to one of the cell towers on the mountian I'd love to do internet backhaul using it
23:16.49J4k3ManxPower: cell towers are usually owned by folks that want WAY too much money.
23:16.51kink0ManxPower, let me check it, I tried ussing getsiphead , but that is unable to get info from the body of SIP message
23:16.59ManxPowerIt's about 5 miles to the tower and about 10 miles to the nearest DSL or cablemodem
23:17.09J4k3they're basically "bought off" by the cellcos to keep the competition (ie - anything and everything else) away
23:17.15J4k3ahh
23:17.17ManxPowerJ4k3: That's whyt I've not even bothered to check it out
23:17.20x86J4k3: around here they want anywhere from $2-$5/foot/month
23:17.20J4k3yeah that'd be do-able
23:17.34ManxPowerx86: define "foot"
23:17.41x86ManxPower: 12 inches
23:17.46J4k3x86: yep, it usually works out cheaper to build your own if thats an available option
23:17.52x86J4k3: *nod*
23:18.08x86J4k3: but then you have to get zoning variances usually, which are a major PITA
23:18.17ManxPowerif your box is 100 ft off the ground you pay $200-$500/month or if your box is 2 sq feet you pay $4-$10/month
23:18.39J4k3x86: depends on where you're at... I'm in rural texas... I put up 150' and didn't have to do *anything* first.
23:18.47ManxPowerMy "comms tower" is a 28 ft 2nd hand telephone pole
23:18.51harlequin516Is there a service where I can start small and buy more phone lines as I need them from a telco, building up to a full T1?
23:19.07JTharlequin516: you meant Fractional T1?
23:19.25ManxPowerSo exactly is the cost computed?
23:19.27harlequin516I guess so.
23:19.32ManxPowerfor renting someone else's tower
23:19.36J4k3ManxPower: my first "mast" was two pieces of 3/4" conduit with a short piece of 1" conduit and lots of duct tape holding it together, with a 108" stainless whip on top
23:19.54ManxPower(6:17:01 PM) x86: J4k3: around here they want anywhere from $2-$5/foot/month
23:19.56harlequin516Is there an asterisk card that will do fractional T1?
23:20.01J4k3ManxPower: its computed on how much they think they can ask without getting punched in the face.
23:20.08ManxPowerharlequin516: all Asterisk T-1 cards do frac T-1
23:20.22ManxPowerA Fractional T-1 is just a full T-1 with some of the channels turned off
23:20.35x86and just as costly on the local loop ;)
23:20.42J4k3they wanted $2/ft for a 350 ft mount on a 500' tower southwest of here.
23:20.53*** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell)
23:20.53*** mode/#asterisk [+o Qwell[]] by ChanServ
23:20.53harlequin516Oh I see, so the hardware is the same, its just you will never be able to utilize more lines than you paid for?
23:20.58ManxPowerok, so it is cost per foot of height off the ground
23:21.03x86ManxPower: right
23:21.10ManxPowerharlequin516: correct
23:21.10J4k3I got a land use agreement in a hayfield for 10 years, in exchange for $30/month worth of internet and $10/month prepaid yearly.
23:21.15x86ManxPower: sorry, didn't understand what you were asking me ;)
23:21.16ManxPowerx86: that really sucks
23:21.21x86ManxPower: yeah heh
23:21.30*** join/#asterisk sabakas1 (n=solapus@66.90.121.129)
23:21.34J4k3on the other side of the road on a taller hill
23:21.42J4k3I can put 170' up there and be perfectly happy :P
23:21.43harlequin516Is there a way to get internet access data and phone on the same line?
23:21.45x86ManxPower: by FCC rules, if you put up a tower, you have to allow others to share it (although you can certainly charge them for access)
23:21.49ManxPowerJ4k3: If I can anchor it, I could build a tall tower.
23:21.51Qwell[]harlequin516: like DSL?
23:21.54J4k3harlequin516: ADSL?
23:22.02Qwell[]or T1?
23:22.11J4k3ManxPower: depending on soil type, you can use some cheap anchoring devices
23:22.14harlequin516Well I mean using a Asterisk T1 hardware device.
23:22.19J4k3if you're on rock, you can drill/screw
23:22.22ManxPowerx86: any idea how the FCC defines a tower?
23:22.28J4k3if you're on the rock, you should go to rehab!
23:22.42J4k3we're on fairly solid soil, we used 8" screw-in ground anchors
23:22.43ManxPowerJ4k3: granit(sp!) and sandstone
23:22.51x86ManxPower: best anchors I've used are J-bolts into some rebar'd concrete about 6 feet into the ground ;)
23:22.56ManxPowerusually with 2" to 4' of dirt over it
23:22.56JTgranite is rather strong
23:23.00J4k3yeah
23:23.02J4k3if the chunks are large
23:23.05J4k3thats a great anchor
23:23.10J4k3if nothing else
23:23.13J4k3get some big chunks moved in
23:23.16J4k3drop 'em in the hole
23:23.17x86ManxPower: no idea
23:23.20J4k3screw into them
23:23.20x86ManxPower: check /msg
23:23.22ManxPowerJ4k3: the chunk is 5 miles long by about 2 miles wide
23:23.23J4k3and use them for an anchor :)
23:23.35J4k3ManxPower: thats not gonna fit in a trailer ;)
23:23.43harlequin516So I bascially desire an asterisk hardware interface to a T1 line that can provide fractional T1 FXO and fractional Data to my ISP?
23:23.48harlequin516Is this possible>
23:23.48ManxPowerx86: my /msg says someone did a CTCP version
23:23.51harlequin516?
23:24.07JTharlequin516: PRI voice is far better
23:24.11x86ManxPower: ah, your nick isn't registered with nickserv eh?
23:24.14J4k3ManxPower: bx?
23:24.15sweeperharlequin516: look into wanpipe
23:24.24sweeperbut yea, pri plz
23:24.25harlequin516hmm
23:24.31harlequin516okay lemme read about it
23:24.44x86PRI is only really for local access
23:24.46lesouvageAt the risk of nagging the channel I have a question about installing a b410p card. I updated Asterisk and zaptel with make b410p and followed all the other instructions on the digium site. Asterisk still doesn't recognize the card. btw, it's just there to have a hardware timer avalable (for now). see http://www.pastebin.ca/591082
23:24.49ManxPowerharlequin516: yes, zaptel can do data,
23:24.55ManxPowersweeper: Digium cards do data too
23:24.56x86for long distance, you want a CAS T1 (which is cheaper anyway)
23:25.09ManxPowerJ4k3: Pidgin aka GAIM
23:25.10x86otherwise you'll be stuck paying switched rates on the PRI
23:25.21x86where on a CAS T1, you pay dedicated rates
23:25.28sweeperManxPower: yea, but sangoma does it so much better :)
23:25.30ManxPowerJ4k3: my point is that to anchor it I would prolly have to drill into the bedrock
23:25.39harlequin516I am going to host a local phone service.  So it will be all inbound local calling.
23:26.04sweeperManxPower: totally dependent on the windload of the tower
23:26.15ManxPowerThe telephone pole is anhored in 800 lbs of high strength concrete
23:26.20harlequin516PRI CAS?  What's that?
23:26.41ManxPowersweeper: the tower would have multiple 3ft dishes on it
23:27.06x86harlequin516: PRI == 23 B channels and 1 D channel for dedicated signalling, supports passing ANI from the PBX to the LEC
23:27.21JTx86: that's the T1 version or PRI, anyway
23:27.28ManxPowerThere is no such thing as PRI CAS
23:27.37JTs/or/of/
23:27.40sweeperManxPower: really, it's just a matter of having a big enough slab :D
23:27.52x86harlequin516: CAS == 24 DS0 channels with no dedicated signalling, not supporting passing of ANI from the PBX to the LEC
23:27.54ManxPowerCAS is "channel associated signaling"  The advantage of PRI is that it is NOT CAS
23:28.09x86right
23:28.15x86but PRI has disadvantages too
23:28.22x86like having to pay switched rates ;)
23:28.42x86and, IIRC, you can't PIC it to the LD carrier of your choice
23:28.49JTyeah, it's moot most places outside of the US
23:28.55ManxPowerx86: we do
23:29.02x86ah, ok
23:29.04JTwe don't have these crazy CAS long distance trunks
23:29.08ManxPowerwell, our CLEC does for us, at least
23:29.32sweeperManxPower: if you can't anchor, you go with a rigid tower that you can stick in a slab
23:29.49sweeperhow high do you need?
23:29.53ManxPowersweeper: A tower is not really on the horizon.
23:29.59ManxPowerMy point is that a tower would be complicated.
23:30.05sweeperoh
23:30.25ManxPowersweeper: Well at 24 ft of height I can't see much of anything, so I imagine something higher than that
23:30.42ManxPowerthe dishes can be low on the tower to reduce wind load (I do that with the existing "tower")
23:31.08ManxPowerdishes == satellite dishes
23:31.32sweeperoh, you're just trying to get LOS to the bird
23:31.33harlequin516I read : PRI requires a switched rate per line.
23:31.38JTwatch ebay for a 20-30m high second hand ham radio truss/wind up/etc tower :)
23:31.40ManxPowerthe wireless antenna would have to be at the top
23:31.46J4k3ManxPower: you can get 'push up masts' up to 70'
23:31.50harlequin516What's the other optrion for rate?
23:31.53J4k3I've done 50' with a helper
23:31.56J4k3and it was fairly easy
23:32.02J4k310 mph breeze too, that was the scariest part
23:32.03sweeperJ4k3: those need anchoring, I'm pretty sure
23:32.12J4k3sweeper: yeah, but pretty darned laughable anchoring
23:32.27ManxPowerI'd rather rent space on an existing tower, if I can
23:32.38J4k3you need stronger anchoring to keep a pissed off pit bull in place.
23:32.38*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:32.48ManxPowerI just need line of sight to Ashville, AL or Rainbow City AL
23:32.59sweeperManxPower: how many miles?
23:33.16J4k3ManxPower: if the 900 is empty, you may not need LOS
23:33.30J4k3I've done rock solid 'dirty' 900 shots with SR9's and yagis
23:33.40ManxPowerWe are 15 miles from the CO, but that CO does not have DSL, so I would need 10 - 15 miles or range
23:33.44J4k3but not colocated with, say, an 800/850 cell carrier ;)
23:34.10sweeperANYTHING on a decent sized cell tower is gonna be shitty, really
23:34.28sweeperI drove up to one post-katrina, to see about doing a wifi shot to some shelters
23:34.37J4k3sweeper: yeah, once you get about 100' above the trees 900 and 2.4 go totally insane here.
23:34.49sweepermy facking car kept locking and unlocking itself
23:35.00J4k3haha
23:35.09sweeperthat's how much noise there was
23:35.15harlequin516Are there many different providers of phone services (T1's) or is it all through my local carrier?
23:35.18*** join/#asterisk ajohnson (n=ajohnson@corpex.pivotal.televerde.com)
23:35.27J4k3my grandparents lived in missouri city (houston, tx's tower farm town)... my explorer's alarm would NOT quit going off
23:35.30sweeperthe cell eng that was there was like "yea, we'd let you use the tower, but you're not gonna get shit"
23:35.33*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
23:35.34ManxPowerharlequin516: That TOTALLY depends on exactly where the lines will be
23:35.34J4k3I had to pull fuses to go visit my people.
23:35.52*** part/#asterisk ajohnson (n=ajohnson@corpex.pivotal.televerde.com)
23:35.55x86harlequin516: you can get a CAS T1 anywhere
23:36.01J4k3I'd have asked the cell guy to get his shit under control :P
23:36.11J4k3"ever heard of an 8 pole filter, sir?"
23:36.18ManxPowerharlequin516: for example where I am I have exactly one company other than bell that can provide me service.
23:36.18harlequin516I worked at a company where Worldcom provided everything but Atlantic Bell was the local telelphone company.  I didn't understand how they did that.
23:36.26sweepereh? nah dude, it was just the noise coming off the gear
23:36.37ManxPowerat my clients they have like 15 companies they can choose from
23:36.57sweeperyou don't put out 5000watts and not kick up some RFI
23:37.08sweeperI mean, your cellphone can screw with your ipod
23:37.09harlequin516How can I find out what companies serve my area?
23:37.11J4k3yeah but cell towers aren't pushing 5000 watts
23:37.16JT5kW, that's not cell gear
23:37.26sweeperover the whole thing?
23:37.27J4k3they're pushing 0.1-100 watts with some antenna gain.
23:37.45*** join/#asterisk kombi (n=kombi@213.160.14.18)
23:37.52sweeperthis was a big tower :)
23:38.01kombito store a callerid to mysql, what does one do?
23:38.02ManxPowerThe towers near me look more like microwave than anything else.
23:38.03JTthey're on different frequencies, so you can't just add all the powers together
23:38.05J4k3well
23:38.06harlequin516x86 : CAS T1 means 24 telephone lines ?  With 24 telephone numbers? Or all the same?
23:38.09JTdirty transmissions cause rfi
23:38.13JTespecially gsm
23:38.14J4k3I see insane stuff with 2.4 at night here
23:38.19ManxPowerbig "drum" looking things as well as the "horn of plenty" things, all with covers
23:38.34kombijeez, where're you at right now?
23:38.38J4k3stuff skips in from 200 miles away.  Can throw the radios into 'snoop' mode and capture valid beacon packets for INSANE distances.
23:38.47JTManxPower: the drums are just shrouded parabolics
23:38.50kombiI hear you..
23:38.53J4k3the farthest one I could find with wigle was 250 miles.
23:38.53JTwith waveguide feeds
23:39.03J4k3200 miles is pretty regular.
23:39.26J4k3ManxPower: sounds like a microwave relay site.
23:39.29ManxPowerJT: I assumed that, but what about the horn of plenty looking ones
23:39.34kombican I ask my incredibly complicated, yet overwhelmingly interesting question again?
23:39.36ManxPowerJ4k3: chances are it is.
23:39.42J4k3ManxPower: reflector waveguide.
23:39.46JTManxPower: i'm not sure what a horn of plenty looks like
23:39.49ManxPowerJ4k3: Ah, OJ
23:39.53ManxPower..er  OK
23:39.58kombibut it sounds terrific
23:40.02ManxPower<-- is not an RF geek, but has learned a lot about RF
23:40.05Jinglesif the waveguide fits, you must acquit
23:40.06J4k3ManxPower: at the bottom of those theres a little antenna feed
23:40.16J4k3prolly no more than 1.5 cm long
23:40.17J4k3hehe
23:40.21J4k3cantenna.
23:40.29J4k3(I've been told you can light them with a cantenna)
23:40.30harlequin516kombi : Overwhelmingly interesting, I doubt it.But I think it would fall within the norms to ask again.
23:40.39kombithanks!
23:40.42ManxPowerkombi: the answer to your question is "it will work.  your question is WHY you would want to go thru all that extra work"
23:41.02kombito make it available in a web app
23:41.08Jinglesisn't callerID already shoved into the CDR?
23:41.12x86harlequin516: can be provisioned however you want it
23:41.18JinglesI mean, in cdr, there's a field called 'clid'
23:41.32x86harlequin516: you can have 24 outbound lines with NO inbound phone numbers at all
23:41.44anonymouz666lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h -> what package provides that header file ?
23:41.51x86harlequin516: or you can have bi-directional lines, each with their own number
23:41.55sweeperanonymouz666: HA
23:41.57kombiJingles: really? but one would have to connect that to other data from that call such as key punches..
23:41.58sweepersame problem I had
23:42.07sweeperjust a day ago
23:42.15sweeperyou need the kernel source, mang
23:42.16anonymouz666sweeper: how you solve that ?
23:42.19sweeperand the headers
23:42.19x86harlequin516: or you can setup hunt groups, where a certain number of channels are all bound to the same inbound number (useful for 800 numbers / etc)
23:42.25anonymouz666I got the headers
23:42.29kombiJingles: or are those in there as well..?
23:42.35harlequin516Do they charge differently for bi-directional lines?  I am sure they charge for each phone number.
23:42.36anonymouz666whats the name of the source package?
23:42.36sweeperand the source?
23:42.42sweeperuhh
23:42.47anonymouz666kernel-source
23:42.47Jingleskombi: I'm working on how that ends up in the cdr - I suspect that where you're navigating the dialplan gets put into the database as well - it doesn't track *all* your steps through the tree.
23:42.48anonymouz666?
23:43.29kombiJingles: If I understand you right, yes
23:43.49sweeperanonymouz666: kernel-devel
23:44.09anonymouz666sweeper thanks
23:44.13Jinglesfar as I know, the database does retain caller ID, though.
23:44.54sweeperanonymouz666: btw your kernel is out of date
23:45.00anonymouz666it's ok
23:45.11kombiwhat I actually try to do is: caller hears charming female voice, enters four digits and leaves a message, all of which gets shown in a web app
23:45.16anonymouz666i don't need a up to date kernel
23:45.23anonymouz666unless has security issue
23:45.37ai-akombi: parse the logs.
23:45.53ai-aoh ;) wrong way hehe.
23:46.00sweeperkombi: you can do odbc queries in the dialplan
23:46.29kombisweeper: like put real sql in there?
23:46.33sweeperyep
23:46.41kombigreat!!
23:46.52kombiwhat's the * command though?
23:46.56*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
23:47.15kombianyway, I look it up..
23:48.15vltJTJT: Ok, I soldered a crossover cable (3->4, 4->3, 5->6, 6->5) and plugged it between the card's NT and the PBX's TE port. Now I see "ACTIVATED" in /proc/zaptel/3 and calls _to_ the PBX are signalled (mostly) but I can't establish a connection (answering isn't recognized). And the asterisk warnings have become less frequent but don't disappear. I found a (German) description on how to setup an HFC card (http://www.pro-linux.de/work/asterisk/a
23:48.15vlteen NT and TE port. Do I need this?
23:48.28sweeperkombi: http://www.loligo.com/asterisk/misc/apps/odbc/app_odbcexec/
23:48.37kombithanks!
23:49.02*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
23:49.55*** part/#asterisk galeras (n=root@200.31.204.42)
23:50.27kombijust what I need..
23:51.00JTvlt: i assume you've done a pri intense debug already
23:56.59*** join/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net)
23:57.27vltJT: :-)
23:58.27[hC]has anyone hacked shared line appearance functionality into 1.2, specifically using polycoms?
23:58.44kombisweeper: just saw app_spl_odbc.so in the Makefile, wouldn't that maybe do also?
23:59.09vltJT: Ok, I fetched an old spare NTBA and plugged it in (without power supply). Now it works. Hmmm, if it works now w/o external power ... what is it necessary for?
23:59.30JTpower supply is for isdn phones that need power feeding

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