irclog2html for #asterisk on 20041124

00:00.30DrukenHmedo ya see an rj45 jack ya can't figure out? :)
00:00.48sivanaya.. there's an RJ45
00:00.59DrukenHmehow many?
00:00.59intrinis everyone ignoring me?
00:01.00intrin:P
00:01.03file[laptop]D-ohhh!
00:01.12sivananot sure.. I'll have to check.. it's not here
00:01.25DrukenHmewell, doesn't really matter anyways
00:01.31DrukenHmeit'll take a t1 directly
00:01.47DrukenHmeif it's 56k, it has to be digital...
00:02.00eKo1isdn?
00:02.00goatmilkhttp://www.crn.com/sections/special/top25/top25_03.jhtml?ArticleID=52601053  <-- this is an awesome read
00:02.17sivanaso theres no way to redirect traffic from *?
00:02.24*** join/#asterisk cp5 (cp5@dsl093-032-201.snd1.dsl.speakeasy.net)
00:02.27DrukenHmeof course there is
00:02.37cp5is it possible to include a context on a sort of if-statement basis?
00:02.59intrinwww.intritech.com/zap.html = my error, if anyone could be of serivce
00:03.14sivanabased on the DNIS, I can tell which number the caller called and know if it's a dial up request
00:03.25DrukenHmeexactly
00:03.29sivanajust not sure how to send it out to the Portmaster for answering
00:04.07DrukenHmeconnect the portmaster to a t1 port, (like a pri) then dial using those ports
00:04.17SipChatAnyone know of any popular / busy conference / chat / party lines / numbers / rooms / etc. ?
00:04.18DrukenHmethe port master should pick the calls up like normal
00:04.25sivanaso I would need a second PRI then
00:04.33sudoerdo i have any options if i have a linksys router, but i need to make surevoip packets get prioritized out
00:04.35DrukenHmewell, a second port
00:04.48sivanaI have a couple of pair gains
00:04.56ariel_it's pizza night.
00:05.29sivanahow would i create a T1 port between * and the portmaster?
00:05.33*** join/#asterisk sudoer2 (~toy@pool-151-203-91-218.bos.east.verizon.net)
00:05.42*** join/#asterisk jks (~jks@0x503e4c12.arcnxx4.adsl-dhcp.tele.dk)
00:06.05SipChatsudoer - how many users share the router?
00:06.10jksHi Guys... anyone using SIPPhone.com with Asterisk and got Virtual Numbers working?
00:06.34sudoeronly about 3 users, but thats enought to saturate the crappy dls connection
00:06.39arrghfunny, I'm just fiddling with that now.
00:06.48jksarrgh: sipphone?
00:06.52SipChatsudoer - you can run local software bandwidth management software
00:06.52arrghyea
00:06.56arrghor
00:07.02arrghyou can go in with a bat and solve your bandwidth problems.
00:07.05jksarrgh: it's a bit weird... I'm simple "stuck"
00:07.06ariel_intrin, what system are you trying to install it on.
00:07.10arrgh*smack* *smack* *smack*
00:07.12Nethabanyone here connect to sipphone.com from *
00:07.14intrindebain
00:07.15SipChatsudoer - or run everything through a software router (arghh) ;) lol
00:07.20SipChatarrgh - ;)
00:07.25jksNethab: me
00:07.34Nethabdo you run CVS?
00:07.34ariel_intrin, 2.4 or 2.6
00:07.38sudoerSipChat, what kind of software? you mean sitting in the windows clients?
00:07.41jksNethab: nope
00:07.48Nethabhmm
00:07.48SipChatsudoer - yep
00:07.50intrinhmm
00:07.55intrini forget
00:07.56intrinhow do i chk
00:07.59sudoerSipChat, do you know of any good software?
00:08.12arrghI've got outbound through sipphone.com to work, it seems.
00:08.23Nethabjks: i keep getting an immediate hangup and a 488 Not Acceptable Here when connecting to sipphone.com now
00:08.25ariel_intrin, cat /proc/version
00:08.28arrghyea
00:08.30arrghI get that too
00:08.30jksNethab: it works fine for me
00:08.41SipChatsudoer - if you have to use a software solution best to use it on a gateway / router style dedicated pc
00:08.46jksI purchased a virtual number, but I can't figure out how to use it
00:08.46intrinLinux version 2.4.27-1-386
00:08.47SipChati.e.
00:09.02jksIf I call it up I get "Unknown user"... I tried using sip register with the virtual number, but that doesn't work
00:09.03Nethabjks: use it how?
00:09.06SipChatthat way if no one is using sip the spare bandwidth can be allocated to other uses
00:09.11jksNethab: for dialing in to asterisk
00:09.16ariel_intrin, did you get the files for cvs -r 1-0 or apt-get?
00:09.28SipChatsuddoer - I wouldn't recommend it but .... good software if you do want to go this way is...
00:09.29sudoerSipChat, i used to run software NAT/router on bsd/linux, but its just not as  stable as dedicated hardware
00:09.35Nethabeven if your not registered sipphone.com should say "User is not online leave voicemail after tone"
00:09.53jksNethab: my regular sipphone number works fine
00:09.53Nethaband then send the wav to you email address
00:10.04jksNethab: it's just the virtual number that I can't get working
00:10.04arrghhmm, thats where I'm at.
00:10.12jksNethab: is there anything special to do with a virtual number?
00:10.19SipChatsuddoer - netlimiter (net limiter) or netprioritizer (net prioritizer)
00:10.20arrghcalling through sipphone.com to 7474745000 gets me the "unacceptable" message
00:10.41ariel_intrin, I have to go and pickup the pizza for dinner I will be back maybe in about 1 hour.
00:10.41jksarrgh: what does sip show registry say?
00:10.42Nethabcalling any number including 12220000000 gives me unacceptable
00:10.51arrghI can
00:10.53arrghtho
00:10.59arrghcall my FWD number through sipphone.com
00:11.09intrinariel_ cvs
00:11.10SipChatsuddoer - :) software routers are not quite so stable ;)
00:11.14intrini didnt use any flags tho
00:11.37ariel_intrin, your using head then move down to stable
00:11.50sudoerwell, is there a way in case the software router/nat goes down, the comps can use an alternative gateway
00:12.00intrinhow do i do that? ;/ im new to all this
00:12.05ariel_rm -rf directory before getting the stable.
00:12.13sudoerif that is possible, i would be comfortable to use the linux box as nat box also
00:12.16intrinreinstall?
00:12.19intrineverything? :/
00:12.20*** part/#asterisk evgenyt (~Miranda@81.24.128.90)
00:13.15SipChatsudoer - that should be possible but may be geting a bit complicated.
00:13.40SipChatsudoer - you could try looking at some windows based solutions they are quite flexible and stable
00:13.54cursorstable?  Windows?  Same sentence?
00:13.59SipChatsudoer - however - they rarely come close to hardware (although some software solutions can easily be remotely restarted)
00:14.06SipChatI know it amused me ! ;)
00:15.00Nethabare "Not Acceptable Here" messages only generated for codec mismatches?
00:15.01tessier_hmmm
00:15.20SipChatThe other option is either a remote script to run on the other stations to termporarily restrict their bandwidth when a SIP call is being made or...
00:15.20tessier_It would seem that if you are listening to someone elses hold music and transfer them to another line you lose the hold music
00:15.30arrghhmm
00:15.37arrghNethab: that's a good question
00:15.43SipChatUse a domain controller type arrangement :).
00:16.10SipChatNethab - I think I have seen them for other problems.
00:16.36jksarrgh: do you have a virtual number?
00:16.41Nethabi get them for any sipphone number i dial
00:16.50cursorWindows router admin: "I have to change a route, so I'll throw everyone off while I reboot"
00:17.03SipChatNethab - what proxy and dialing format are you using etc?
00:17.11redder86hi
00:17.11SipChatcursor ; - )
00:18.09arrghjks: no, still fiddling with normal number for in/out functionality
00:18.15Nethabproxy01.sipphone.com and dtmf = rfc whatever
00:18.16jksokay :-(
00:18.39SipChatnethab - what have you tried to dial at proxy01.sipphone.com ?
00:18.39cursor"I have to change a route - who stole my mouse?"
00:18.59SipChatcursor - ;) lol (use that command prompt)
00:19.02SipChatlol
00:19.09SipChatyou deserve it ;)
00:19.52eKo1why oh why did these people decide to use windows for e-mail!? @#!%!$%!%$
00:19.58SipChatlolo
00:20.21jksI wonder what the Number Assignment feature is for, if it's not for custom/virtual numbers?
00:20.37NethabI've tried 12220000000, my own number, my friends number, and the number that guy said above
00:22.19SipChatHmm. ok :)
00:22.44MerTech_Wes!! anyone know how VoIP providers like Vonage/Primus address these systems (i.e. sip:xxxx@primustel.com)??
00:22.45*** join/#asterisk Zaw (~zaw@cc.cirqular.com)
00:23.52SipChatNethab - I recall GSM was working ok for you.
00:23.54SipChat?
00:24.03Nethabyeah and you direct dialed my * box
00:24.12Nethabbut i can't dial out to sipphone
00:25.57*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
00:26.20NethabMy CLI says -> invite 12220000000@proxy01.sipphone.com
00:26.28Nethabthen <- 100 trying
00:26.44bkw_sounds like someone needs to learn how to setup sip.conf
00:26.53Nethabthen <- 200 ok contact conference+12220000000@198.65.166.130
00:27.05*** join/#asterisk ctooley (~ctooley@015-835-274.area5.spcsdns.net)
00:27.22Nethabit worked before no changed to sip.cond
00:27.35ctooleyAnyone know what to put in the Polycom ftp files to be able to register an IP500 directly to Broadvoice?
00:27.36bkw_here
00:27.38bkw_[sipphone]
00:27.38bkw_type=peer
00:27.38bkw_username=NUMBER
00:27.38bkw_fromuser=NUMBER
00:27.38bkw_fromdomain=proxy01.sipphone.com
00:27.39bkw_secret=xxxxxxx
00:27.41bkw_host=proxy01.sipphone.com
00:27.43bkw_qualify=yes
00:27.46bkw_then
00:27.52bkw_dial,SIP/number@sipphone)
00:27.57bkw_dial(SIP/number@sipphone)
00:28.15Nethab[sipphone-out]
00:28.19Nethabtype = peer
00:28.23Nethabhost = proxy01.sipphone.com
00:28.33Nethabusername = 1747667XXXX
00:28.40Nethabsecret = XXX
00:28.51Nethabfromdomain = proxy01.sipphone.com
00:28.58Nethabfromuser 1747667XXXX
00:29.13cursorhttp://pastebin.ca/
00:29.15Nethabexten => _91747NXXXXXX,1,Dial(SIP/sipphone-out/${EXTEN:1})
00:29.22bkw_you're dialing WRONG
00:29.29Nethabit worked before
00:29.35Nethabthen upgraded CVS
00:29.37bkw_that whole SIP/peer/exten stuff
00:29.40bkw_I don't like it
00:29.47bkw_I do SIP/exten@peer
00:30.49arrghI discovered that the only way to show up as "online" appears to do a sip register for sipphone.com
00:30.59*** join/#asterisk Darwin35 (~darwin35@24.3.204.71)
00:31.06bkw_ya think?
00:31.08*** join/#asterisk calvis (calvis@h-66-167-52-131.sttnwaho.covad.net)
00:31.11Nethabi register
00:31.12bkw_:P
00:31.14arrgh;)
00:31.19Nethabbut that doesn't affect dialing out
00:31.22arrghno
00:31.47arrghmy exten lines are similar to bkw's
00:32.12Nethabno change still Not Acceptable Here
00:32.13*** join/#asterisk netvulture (1000@63.174.172.245)
00:32.25netvulturewho
00:32.32arrghI can 1393612 on my sipphone.com extension and get the fwd time server
00:32.33netvultureoops - wuz up guys
00:32.40calvisI am trying to cvs checkout zaptel on the CVS server, but I keep getting cvs server: Updating zaptel   It has been doing that for the past 30 min.
00:32.49cursor(netvulture) ceiling
00:32.54ZX81~seen sokol
00:32.56jboti haven't seen 'sokol', ZX81
00:33.01ZX81~seen ssokol
00:33.02jbotssokol <~ssokol@64-151-42-28-dhcp-kc.everestkc.net> was last seen on IRC in channel #asterisk, 42d 23h 50m 59s ago, saying: 'they were closed when I looked 6 months ago'.
00:33.12cursor~seen WMD in Iraq
00:33.13jbotcursor: i haven't seen 'wmd in iraq'
00:33.18Nethabbkw_: I changed my extensions.conf and still get "Not Acceptable Here"
00:33.26cursorGeorge Dubbya should have had a jbot
00:33.27netvultureHere's a dumb questions - what is the difference between Goto and Transfer?
00:33.38arrghI think W is bluescreened
00:33.46arrghever seen that blank look he gets when they fire up the teleprompter?
00:33.55eKo1netvulture: they are spelled differently\
00:34.02Atacomm~seen mark actually talk on IRC
00:34.03jbotAtacomm: i haven't seen 'mark actually talk on irc'
00:34.03tessier_netvulture: Only difference I can see is that Goto can go to different contexts whereas transfer just jumps to another extension within the same context
00:34.09NethabGoto is a dialplan command and transfer usually wants the other side to Answer?
00:34.15ZX81~seen kram
00:34.16jbotkram is currently on #asterisk
00:34.21ZX81kram == mark
00:34.27cursorreally?
00:34.34Atacommzx81: i know, that doesnt mean he talks on here
00:34.34eKo1oh no run for cover
00:34.43ZX81yah he does
00:34.50Atacommzx81: for 10 seconds yes
00:34.53arrghnethab
00:34.55ZX81ask him a question that only he can answer and he'll answer
00:34.56ZX81:-)
00:34.59Nethabarrgh: yes?
00:35.03arrghtry and dial out to fwd, bet it works.
00:35.16cursorWhat is your shoe size
00:35.20ZX81i.e. re sysmaster
00:35.21ZX81:-0
00:35.21Nethabarrgh: which number?
00:35.22ZX81lol
00:35.25bkw_haha
00:35.26bkw_sysmaster
00:35.31ZX81hmmmm
00:35.34arrgh1393612
00:35.43Atacommi need to call sysmaster one of these days, they were asking us to sell their systems
00:35.51Atacommamong many reasons to call sysmaster
00:35.55ZX81I wouldn't
00:36.05ZX81I'd rather sell windows
00:36.26Nethabarrgh: through sipphone, or through fwd
00:36.33arrghthrough sipphone
00:36.34bkw_Atacomm sysmaster is asterisk
00:36.36bkw_:P
00:36.38arrghof course it'll work through fwd
00:36.38Atacommlol, i'm not gpl religious
00:36.53Atacommi know bkw, lol
00:36.54ZX81Atacomm: your company name?
00:36.55eKo1man, i wish there was a command to reset all the sip/zap/iax/h323 channels
00:36.56arrghIAX YOUR DATA . NET PROTOCOL need flat roof
00:37.00bkw_Atacomm my job is to make sure we get everything sysmaster can do in the GPL version
00:37.03bkw_they done pissed me off
00:37.05eKo1without having to restart asterisk
00:37.05cursorChurch of GPL
00:37.40tessier_bkw_: You get 'em!
00:37.42Atacommbkw: lol, good, then they wont violate the gpl and mark wont be able to hold them hostage on the "use our gear or pay big fees" theme
00:37.55bkw_Atacomm bingo
00:38.09Atacommbkw: lol, uh, that plays more into our camp than it does digiums
00:38.17bkw_yes
00:38.24Nethabarrgh: it did
00:38.28bkw_anthm and I give that bug tracker hell EVERY DAY
00:38.31Nethabarrgh: got the date
00:39.25Nethabbkw_: do you think markster will accept the patch to fix it quickly
00:39.30bkw_ya
00:39.36bkw_I already talked to him on the phone
00:39.41Nethabnice
00:39.46bkw_he was in his car
00:39.47bkw_haha
00:40.12Nethab*car crash* there goes the * development schedule
00:40.26bkw_can more people test this http://bugs.digium.com/bug_view_page.php?bug_id=0002927
00:40.39ctooleybkw_, gone?  really, I still have all kinds of issues with our tones being detected by other phone systems
00:40.44Nethabi can try what should i test for?
00:40.56brc_~seen tclark
00:40.57jbottclark is currently on #asterisk
00:40.58bkw_ctooley using inband eh?
00:41.27*** join/#asterisk czero (~h@CPE0090f800c5b0-CM00e06f166c34.cpe.net.cable.rogers.com)
00:42.03bkw_oh that was fixed too
00:42.06bkw_like three days ago
00:42.16ctooleybkw_, yeah,  Broadvoice
00:42.21bkw_the seq numbers were rolling over
00:42.28ZX81anyone heard of implex systems?
00:42.34bkw_ctooley hehe
00:42.38Atacommnope, i've heard of implex.net (they are our isp)
00:42.52ZX81what do they do?
00:42.57ZX81just isp?
00:43.14ZX81maybe is not the same one
00:43.17Atacommimplex.net is an isp... we colocate there, they do fiber, wireless, t1, dialup and dsl in the minneapolis area
00:43.25ZX81ok
00:43.26ZX81maybe
00:43.30ZX81ta
00:43.45ctooleybkw_, I need the flexibility of the IP based carriers... we've been moving the phone system frequently for the last 3 weeks.
00:43.46_Vileheh heh heh heh
00:43.52_Vileluff muh lnp full # check hack
00:44.09czeroeveing all
00:44.19Atacommbkw: hows your scalability stuff coming along
00:44.21bkw_someone buy me a broadvoice account
00:44.27bkw_i'll fix that too
00:44.35ctooleybkw_, besides both the TDM400P
00:44.53Atacommwhat does P stand for anyways ?
00:45.02*** join/#asterisk implicit (~implic1t@roam-15.roam.intelenet.net)
00:45.03ZX81~p
00:45.05jbotsomebody said p was pee
00:45.10ZX81:-)
00:45.11bkw_power hungry?
00:45.13Atacommlol
00:45.19implicitheh
00:45.24implicithi everyone
00:45.24ctooleybkw_, and the single channel card had horrible quality
00:45.26modulus_jbot pee?
00:45.27jbotOk.
00:45.32ZX81lol
00:45.32NethabP = persnickity
00:45.34Atacommhey implicit, how goes
00:45.54implicitAtacomm: not bad
00:45.58Atacommgood good
00:47.46redder86PCI
00:47.48*** part/#asterisk eKo1 (~abc@63.245.57.70)
00:47.52Atacommimplicit: the train is getting close to the end of the tunnel :)
00:48.01redder86yeah, as if there were ever going to be an ISA version
00:48.14Atacommcould be a VESA Local or EISA
00:48.27arrghnethab
00:48.43arrghtry canreinvite=no
00:48.50implicitimplicit: oh ya?
00:48.51arrghin you sip.conf entry for sipphone
00:48.56implicit:-p
00:49.08Nethabarrgh: yes the fwd number worked, and canreinvite is no and notransfer is yes already
00:49.13redder86okay, so we need TDM400I, TDM400V, and TDM400E versions in addition to TDM400P
00:49.25Atacommimplicit: talking to yourself?
00:49.37impliciti am pretty out of it right now
00:49.41Atacommrofl
00:49.48implicitlol
00:49.50implicithahahaha
00:49.54Atacommatacomm: i've never addressed myself before now
00:49.55NethabNethab: have you seen my pants, I know i left them somewhere around here
00:49.58*** join/#asterisk bdeb5 (~bdeb4@alb-24-195-238-207.nycap.rr.com)
00:50.09funraps-asguys
00:50.26Atacommimplicit: yeah, converting between 4 to 6 layers right now, meeting with some important people next week in our supply chain
00:50.29funraps-asis there a cheap FXS card out there? just need 1 port
00:50.35implicitAtacomm: nice
00:50.43redder86funraps-as: nope
00:50.51bkw_someone paypal me some cash to get a broadvoice account :P
00:50.59redder86funraps-as: the least-expensive thing would be an ATA
00:51.11Atacommimplicit: my vendor got mad that one of the suppliers was falling behind, so they are flying one of their reps in-state so we can bitch together at him
00:51.18funraps-asisn't ata fxs to usb or ethernet?
00:51.50Nethabyes ata's are usually inline from rj11 to rj45
00:51.53redder86funraps-as: okay, sure, it's still an FXS port, though
00:52.13Nethabor usb
00:52.58funraps-asgotcha, any cheapie you recommend to try out?
00:53.14arrghnethab check your codecs?
00:53.35Nethabdisallow = all
00:53.35Nethaballow = ulaw
00:53.35Nethaballow = alaw
00:53.35Nethaballow = gsm
00:53.36arrghI'm getting the recording at 5000 now, so, sip.conf was malconfiged
00:53.58*** join/#asterisk _Simon (irc@i216-58-40-193.avalonworks.net)
00:54.14_Simonhey guys
00:54.40*** join/#asterisk SipChat (jirc@m71-mp1.cvx2-b.lng.dial.ntli.net)
00:55.00bkw_the dns resolver in asterisk is fucking lame
00:55.09HellHoundanyone know the url for the SixNet IAX itsp ?
00:55.12bkw_OH lets block everything for X seconds just for shits and giggles
00:55.39_Simonwas curious if anyone got iaxclient to work in .NET? I'm getting catestrophic errors trying to call methods from the object, it won't register with regsvr32 either
00:57.08tessier_.NET? Catastrophic errors?
00:57.23_SimonI think its because the COM object won't register
00:58.59*** join/#asterisk HitTop (~Miranda@Toronto-HSE-ppp3732702.sympatico.ca)
00:59.38ta[i]ntedcan someone send me their broadvoice sip.conf?
01:01.01*** join/#asterisk juice (~juice@mo-205-240-40-86.dyn.sprint-hsd.net)
01:01.10redder86can someone send me their broadvoice authentication information, their credit card information, and a signed form authorizing me to use them without limits
01:01.23tessier_ok, I am confused.
01:01.38ta[i]ntedredder86: i think any reasonable person would obscure their authentication information
01:01.52cursor(tessier) I'm cursor
01:01.53redder86hopefully
01:02.15ta[i]ntedglad to hear you're so caring of strangers
01:02.15Nethabarrgh: well it's connected now, but no sound so that's some kind of progress
01:02.17ta[i]ntednow stfu
01:02.26implicitta[i]nted: i don't like reasonable people for that reason
01:02.49ta[i]ntedthat's reasonable
01:03.14cursorreasomable people are always so unreasonable, for some reason
01:03.41redder86ta[i]nted: I don't use Broadvoice anyway, but I stopped handing out my confs like candy because sometime, someday, I'm gonna forget to obsure my authentication information.
01:03.45Nethabit's unreasonable to reason that reasonable people would have a reason at all, no?
01:03.53tessier_Ok, I have a PRI...
01:03.58ta[i]ntedthey wouldn't be reasonable people if they were unreasonable, even for a reasonable amount of time
01:04.00tessier_And I have 100 DID's on this PRI
01:04.14tessier_Is it possible to have several calls all coming in at the same time to the same DID?
01:04.33redder86tessier_: yes, or of what value would your expensive PRI be?
01:04.44cursorWhat reason could any reasonable person have to suspect that anyone should be reasonable to an unreasonable person?
01:04.48tessier_redder86: I thought I had to have a separate DID for each line I wanted coming in.
01:05.05redder86tessier_: it's only one line, 24 channels
01:05.11ta[i]ntedredder86: why don't you contribute to voip-info wiki then? and rid the hassle of routine sharing with one fell swoop!
01:05.13*** join/#asterisk glm2k (~glm2k@rrcs-24-199-11-45.west.biz.rr.com)
01:05.21redder86tessier_: 100 DIDs... all independent
01:05.40redder86ta[i]nted: too busy, sorry
01:06.10ta[i]nteddid u just say that in an IRC channel?
01:06.21ta[i]ntedlol
01:06.31ta[i]ntedthe irony
01:07.22bkw_http://codeen.cs.princeton.edu/codns/
01:07.28_Simonanyone know how to get the COM object to work for IAXClient?
01:07.47redder86wikis are an abomination to all that is holy about documentation: usefulness
01:08.19*** join/#asterisk drbyte (~byte@byte.fedora)
01:09.26bkw_redder86 if people would only verify what they write before putting it on the wiki
01:09.44bkw_and by verify.. I mean read the code
01:10.11Nethabwiki's are ok, if you carry around your own grains of salt
01:10.16redder86it's the root problem of a wiki, though, the design is to allow the lay-person the ability to write documentation for something for which they are not really "authorized" to do
01:10.25file[laptop]let's all go to the lobby
01:11.02libpcphi all
01:11.05czerofile[laptop] your not goign to start disco dancing agian are you
01:11.17ta[i]ntedi propose an electoral wiki..
01:11.40libpcpin latest cvs of zapata, does wcfxs replace by wctdm ?
01:12.02libpcpi couldnt find the wcfxs in the Makefile
01:12.07Nethabbut then the red wiki states would have just as much say as the blue wiki states
01:12.56EquinoxIf I have *#1 registering as a peer to *#2.. *#2 needs to have a user entry, right?(IAX2)
01:13.23libpcpah okay, the wcfxs was alias to wctdm
01:13.44*** join/#asterisk drazvan (~drazvan@romsat011.fx.ro)
01:13.59Nethabequinox: yes otherwise #2 would show up as guest
01:14.14EquinoxNethab- Well, I wanted it unidirectional
01:14.20drazvanhi everyone. could anyone help me with some ANI + Asterisk questions?
01:14.21Equinox#1 -> #2
01:14.28Nethabyou should try trunking
01:14.38EquinoxWhat advantage does trunking have?
01:14.40redder86it would be nice if the people who wrote the code actually documented it outside of the code (in addition to inside it)
01:15.04Nethabless overhead per call if you make several calls between systems
01:15.08Equinox<PROTECTED>
01:15.11Nethabotherwise none
01:15.15*** join/#asterisk Kumbang (~kumbang@167.205.22.54)
01:15.23EquinoxNethab- So without trunking the reduction in overhead isn't there?
01:15.50*** join/#asterisk JerJer (~mine@d2-236.rt-bras.che.centurytel.net)
01:16.17brc_~seen JerJer
01:16.19jbotjerjer is currently on #asterisk (29s)
01:16.19Nethabif one * server is the main system, and has the connection to the PSTN or the main voicemail box, and calls from the second box need to go through it to get to the PSTN then trunking is very good
01:16.20brc_er...doh
01:16.35bkw_libpcp yes wctdm replaces wcfxs
01:16.37bdeb5anyone here been able to unlock the pap2?
01:16.44ta[i]ntedit would be nice if i was fellated regularly by random attractive women on a daily basis
01:16.52_Simonanyone here use IAXClient COM object?
01:16.58redder86bdeb5: it's not locked
01:17.01ta[i]ntedbut we all have dreams, don't we
01:17.03Nethabi disbabled my COM service sorry
01:17.21bdeb5redder86: it's locked into vonage, i can't get in to set my sip hostname, etc
01:17.23redder86bdeb5: but don't count on anyone here telling you how to use it with Asterisk
01:17.31_SimonNethab: hmm?
01:17.42Nethabi was being facetious
01:17.50bdeb5redder86: how come?
01:18.15_Simonhehe ok, well I can't get the IAX ActiveX object to work
01:18.28redder86bdeb5: "locked" would indicate that something in the firmware is forcing it to Vonage, and that's not true, it pulls a configuration file and gets the Vonage config from the config file.
01:18.32ta[i]ntedbdeb5: that would be hax0ring
01:18.37redder86bdeb5: those who know aren't telling
01:18.43*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-201.d4.club-internet.fr)
01:18.55*** part/#asterisk drazvan (~drazvan@romsat011.fx.ro)
01:18.57bdeb5og
01:18.59ZX81what is the best interface to use with the Fritz cards? i.e. chan_capi, zaphfc etc
01:18.59bdeb5oh
01:19.07Nethabeverything i know i learned from the wiki
01:19.12ZX81assuming European BRI
01:19.18redder86bdeb5: the cardinal rule about hacking stuff and keeping your hack functional: don't tell anyone about it
01:19.30bdeb5i've been able to roll back the firmware and get it to download the xml file, but it wont take the changes - it needs to be salted i think
01:19.55bdeb5i see
01:20.18ta[i]ntedsalt your ATA, grasshoppa
01:20.24cursorhttp://www.cursor.biz/tmp/flame_form.txt
01:20.33redder86salt on slugs is better than salt on ATAS
01:22.12cursorNot if you believe what they say in the adverts on British TV
01:22.48redder86what, that slugs are good for the earth?
01:23.13bkw_ok the dns resolver lib in asterisk needs to be replaced with something that doesn't block
01:23.16redder86they eat decaying plant matter, help the food chain, save the whales?
01:23.18ta[i]ntedcursor: i <3 flame_form
01:23.32*** join/#asterisk david (~david@muffin.davidcoulson.net)
01:23.33cursor:-)
01:23.51redder86bkw_: the IAXy just plain needs a resolver
01:24.03ta[i]ntedsomeone needs Preparation H
01:24.09redder86hehe
01:24.14cursorWhat's that?
01:24.15ZX81bkw_: agree
01:24.24cursorhttp://www.rtfm.info/tmp/flame.txt
01:25.01bkw_http://www.gnu.org/software/adns/
01:25.28redder86bkw_: they won't sign the disclaimer
01:26.33nestArcursor: you might want to add "DIE IN A FIRE" to the FURTHERMORE: section.. :)
01:26.51EquinoxI'm trying to get IAX2 working from my home machine, behind a Linksys NAT router, to another asterisk server on a DS-3.. This isn't working and I don't seem to see any debug information saying why. Any suggestions?
01:28.00Nethabif your behing NAT you should really forward the ports to the * box behind the linksys
01:28.11redder86UDP port 4569
01:28.14bkw_you shouldn't have to
01:28.24cursordisclaimer?  what's that then?
01:28.29redder86for incoming calls you have to, unless you register, then you shouldn't have to.
01:28.33cursordon't answer that :-)
01:28.35EquinoxI'm registering.
01:28.41Nethabif you originate calls from the non nat box to the nat box you should have the ports forwarded
01:28.49EquinoxEven with registering?
01:28.59EquinoxActually I'm mostly concerned about outgoing calls from nat.
01:29.03redder86Equinox: I forward the ports even though I register.
01:29.27EquinoxCan't hurt.. 1 sec
01:29.32Nethabif you don't reinvite and leave the * box behind the nat in the media path you should be fine
01:30.21ZX81does anyone have any questions they would like asked of Mark Spencer for an interview for the Asterisk Daily News?
01:30.28brc_yes
01:30.39ZX81mail me?
01:30.39brc_what is the short term plan for 1.1
01:30.43ZX81ok
01:30.44brc_and what is the long term vision for 2.0
01:31.00ZX81:-)
01:31.15brc_there's tons of new stuff in head
01:31.17NethabEquinox: i forwarded UDP/5060 for SIP as well
01:31.33Nethabbrc_: like what new stuff?
01:31.41brc_read the cvs list
01:31.49EquinoxWhere is the cvs list?
01:31.50Nethabi did didn't mean much to me
01:32.00brc_Equinox, same place as all the mailing lists
01:32.03Nethablists.digium.com
01:32.03brc_asterisk.org
01:32.05brc_look hard
01:32.17brc_http://www.asterisk.org/index.php?menu=support
01:32.17EquinoxStill not working.
01:32.46NethabEquinox: are you in the * console (CLI)?
01:32.50cursor<brc_> read the cvs list
01:32.51cursorhaha
01:32.52cursor"oops"
01:33.10MavvieZX81: I have a lot of questions, but I don't think any of them makes sense.
01:33.17ZX81:-)
01:33.20EquinoxNethab- on both machines
01:33.22ZX81mail to editor@sineapps.com
01:33.31brc_cursor, wtf?
01:33.32*** join/#asterisk michael12345 (mick1234@202.43.239.10)
01:34.10cursorUseless commit message of the week:
01:34.11cursor---
01:34.11cursorModified Files:
01:34.11cursorpbx_realtime.c
01:34.11cursorLog Message:
01:34.11cursorRealtime improvements
01:34.12cursor---
01:34.25michael12345anyone got the new oh323 module to compile
01:34.26Nethaband you don't get any messages with 'set verbose 4' or iax2 debug?
01:34.40michael123450.70
01:34.40drumkillacursor: oh come on ... it is exactly what he said it was   :p
01:34.44cursor:-)
01:34.46brc_luke, read the code...ya know...the stuff right under that message
01:34.59EquinoxI get some IAX2 messages, but they don't tell me anything.  The set verbose thing same as -vvvv?
01:35.05Nethabyes
01:35.07cursorok - how about this one, which was last week's useless commit message of the week:
01:35.08cursor---Modified Files:
01:35.08cursorlogger.c
01:35.08cursorLog Message:
01:35.09cursorFix little logging issue
01:35.10cursor---
01:35.23drumkillawell there you go
01:35.28drumkillaobviously it was pretty little
01:35.32drumkillaso you don't really need to worry.
01:35.41michael12345please anyone
01:35.41drumkilla:)
01:35.54brc_michael12345, contact the authors...
01:35.56drumkillamichael12345: i don't use it
01:35.58brc_use cvs head
01:36.05Nethabbecause some of us are low level programmers and res = chan->pvt->setoption(chan,... means nothing to us
01:36.08EquinoxHmm
01:36.11EquinoxFixed it
01:36.18Mavviecursor: and he wants us to read these to figure out what has changed in the configuration files....
01:36.19EquinoxApparently Asterisk isn't happy being on a 2nd interface IP..
01:36.25michael12345brc_ are you using it
01:36.29brc_no
01:36.41MavvieEquinox: use bind addresses.
01:36.57EquinoxBind = 0.0.0.0 won't do it?
01:37.03EquinoxCan I have multiple bind statements?
01:37.12Mavvienot if you want the eth0:1
01:37.13Nethabbindaddr = 0.0.0.0
01:37.30EquinoxThat's default isn't it?
01:37.32cursor0.0.0.0 should still work
01:38.43cursor<Equinox> Can I have multiple bind statements?
01:38.43cursorno
01:39.14Mavvielsof -n  | grep asterisk | grep -E '(TC|UD)P' <- should all be bound on your eth0:1 IP address.
01:39.36cursorlsof: command not found
01:39.37cursor:-)
01:39.39JerJer0.0.0.0 will work in IAX and SIP, maybe MGCP and most certianly NOT chan_h323
01:39.43drbytehello all. i have a Winbond 6692, and i'd like to configure asterisk to make use of that
01:39.44Equinoxudp        0      0 0.0.0.0:4569            0.0.0.0:*
01:39.48EquinoxNetstat -n -l
01:40.08timecopheh
01:40.18timecopasterisk behind nat = waste of time.
01:40.20timecopgive up now.
01:40.25MavvieEquinox: doesn't five you the process name.
01:40.41Equinoxtimecop- My main asterisk is on a static
01:40.44Equinoxtimecop- This is my home
01:40.48Nethabfuser 4569/udp?
01:40.49MavvieI'm only interested in the asterisk side of the story.
01:40.59timecopequi: its still a waste of time
01:41.03cursor*  sys-apps/lsof
01:41.03cursor<PROTECTED>
01:41.03cursor<PROTECTED>
01:41.03cursor<PROTECTED>
01:41.10EquinoxWhy?  Outgoing calls to my other box?
01:41.14timecopuse IAX.
01:41.23drbytecursor: what rpm are you using? i.e. which repository is it from? i'm rebuilding from fedora.us to make it generally work
01:41.26timecopor better yet
01:41.31timecopgive your "home machine" a external IP.
01:41.37Nethabmy * box is behind a Linksys NAT and i can make calls and receieve them all day long
01:41.46JerJerLFS
01:41.47Optic2hey
01:41.49Equinoxtimecop- I wanted to experiment with NAT.
01:41.51bkw_yo yo yo
01:41.57JerJertar zxf blah.tar.gz
01:41.58Optic2what would cause like 2 seconds of lag on an iax conversation? :)
01:41.58EquinoxActually the problem isn't NAT at all..
01:42.02EquinoxIt's asterisk not working on eth0:1!
01:42.03Mavviecursor: only on linux machines :-)
01:42.03Optic2audio is perfect, but very delayed
01:42.03JerJer./configure && make install
01:42.05Nethabexternip = <your ip>
01:42.11EquinoxIt works fine if I point the other config file to eth0...
01:42.12MavvieEquinox: works fine here.
01:42.12timecopequin: ugh
01:42.14EquinoxIs this a bug?
01:42.18bkw_localnet=localnet/localmask
01:42.22bkw_oh
01:42.24bkw_FYI
01:42.27EquinoxMavvie- You have a default gw on eth0:1?
01:42.35bkw_you can't do auth on a virtual interface using iax2
01:42.36Nethabwhat's your route -n say
01:42.43bkw_if say you have eth0 and eth0:1
01:42.52bkw_and you try to register to the ip on eth0:1
01:42.52MavvieEquinox: no, default gateway is pointing to eth0 IP address.
01:42.54bkw_YOU CANT
01:42.57EquinoxOooh
01:43.00_Simonanyone here use the ActiveX IAX client?
01:43.03*** join/#asterisk czero (~h@CPE0090f800c5b0-CM00e06f166c34.cpe.net.cable.rogers.com)
01:43.04EquinoxWell that explains it.
01:43.07bkw_its a bug
01:43.11czerokelw XP Pro bluescreens on MeTaBSD
01:43.11bkw_I have been screaming about it for months
01:43.19bkw_but nobody seems to think its really a bug
01:43.19czerofirst aht that has happended in a LONG time
01:43.29bkw_Equinox are you getting hit by that?
01:43.31znoGshiiit, 37000 messages in my asterisk folder
01:43.34Mavviebkw_: can you elaborate on that one?
01:43.36znoGtime to delete 85% of them
01:43.53bkw_eth0 = 192.168.1.1 and eth0:1 = 192.168.1.2
01:43.57cursorfolder - ugh
01:44.02bkw_chan_iax2 is on 0.0.0.0
01:44.05bkw_ie all right?
01:44.20bkw_if you try to register or do anything that requires auth on eth0:1 IT WILL FAIL
01:44.40Mavviebkw_: is your subnet mask on eth0:1 /32 ?
01:44.48bkw_doesn't matter
01:44.49EquinoxI tried /32 and /30
01:44.53EquinoxJust for the record ;)
01:44.57bkw_its a flaw in chan_iax2
01:45.18bkw_but I know it does give us hell
01:45.19bkw_:P
01:45.20EquinoxGlad it's a bug
01:45.23EquinoxWas driving me insane.
01:45.25tessier_redder86:   I just talked to a PRI guru and he suspects they had each DID set up as an individual call route and not as a trunk and then changed it when I called up wanting a hunt group.
01:45.35bkw_Equinox try a real ip on a real interface
01:45.44EquinoxThey are both real IPs.
01:45.49bkw_ok try the other one
01:45.50bkw_haha
01:45.51EquinoxBut I'm switching to the non aliased eth :)
01:45.53bkw_the one thats not an alias
01:46.17Mavviebkw_: it seems to work fine here, can you give some more information on how I can prove it is not working?
01:46.18Equinoxbkw_ Yes.. You're right.  I figured it out about 2 sec before you said it's a bug
01:46.24Equinoxbkw_ Of course it's been torturing me for days . . .
01:46.44bkw_Mavvie can you make calls and register to an aliased ip?
01:47.09sudoer2is it possible to get a business address that isnt a po box and you dont live there?
01:47.22bkw_sudoer mailboxes etc
01:47.37cursorWhy not get a business address that points to your home/business?
01:47.51Mavviebkw_: I think so.
01:48.11bkw_try it to be sure
01:48.33Mavviebkw_: I want to know where you think it is going wrong, since I can't see it here.
01:49.15Optic23 second lag at the end of a 5 minute call
01:49.24Optic2almsot no lag at the beginning
01:49.26MavvieOptic2: same here.
01:49.27Optic2jitterbuffer off
01:49.40Optic2iaxComm <-> asterisk <-> sip
01:49.48EquinoxIf there is no auth=whatever statement are plaintext pws exhcnaged?
01:50.16tessier_oh jeezus freakin' christ
01:50.24tessier_The email said DO NOT discuss it on the mailing list.
01:50.35EquinoxMavvie- Um.. It isn't working.
01:50.39EquinoxMavvie- Trust me :)
01:50.45EquinoxMavvie- At least, in bkw and my case.
01:50.49sudoer2cursor, i dont want people to have my home address
01:50.50*** join/#asterisk Nethab (~Nethab@adsl-67-113-141-170.dsl.sntc01.pacbell.net)
01:50.54EquinoxMavvie- The one thing I didn't try was putting a route out eth0:1.
01:51.03MavvieEquinox: yes, and then I ask for more information so I can emulate it here.
01:51.13EquinoxMavvie- Do you have a route out of your eth0?
01:51.16EquinoxMavvie- Do you have a route out of your eth0:1?
01:51.29cursorDon't you have a legal obligation to reveal your business address?
01:51.33MavvieEquinox: eth0
01:51.36NethabThe problem is when registering to the external * box the src IP will always be eth0 not 0:1
01:51.36cursorso people can send you official letters etc.
01:51.39Equinoxcursor- Your registered agent.
01:51.50cursorok
01:51.56cursorIt'd obviously different over here
01:51.56Equinoxcursor- A few states have a minimum of a RA.
01:52.02Equinoxcursor- Where ru?
01:52.08cursorEngland
01:52.11EquinoxBut in Florida my business address is public record.
01:52.20EquinoxAnd most places in the US is the same.
01:52.27cursorok
01:52.47Nethabthe default route which is what is used to find the other * server says to go through eth0 not the IP alias on eth0:1
01:52.49cursorIn England, you have to reveal your "registered address"
01:52.57cursoreven if that's different from your office address
01:53.03Equinoxsudoer2- Just rent an office?
01:53.10Equinoxsudoer2- It's 1. A business address, and 2. You don't live there.
01:53.14EquinoxYour qualifications :)
01:53.14yxaanyone knows of a online site that sells digium products and ships internationally as well?
01:53.15cursorso one company might have several offices, but only one "registered address"
01:53.21Equinoxcursor- Sure.
01:53.25MavvieNethab: yes, but if it bound to an IP address it will take that one anyway.
01:53.28Optic2reading iax2 protocol doc
01:53.32Equinoxcursor- The registered agent is there so you can sue them.
01:53.42cursorok
01:53.48sudoer2Equinox, its not a bad idea, unfortunately thats too exspensive for me
01:53.51cursorAgain, slightly different over here :-)
01:53.57Equinoxsudoer2- Then use your home addy.. I do ;)
01:54.02Equinoxcursor- How so?
01:54.22cursorWe don't tend to run around sueing one another
01:54.27EquinoxI think it's a crime if your registerd agent isn't valid.
01:54.28MavvieEquinox: can you show me the output of that lsof command I just pasted?
01:54.31EquinoxYes, I've heard that.
01:54.41Equinoxmavvie- lemme find it again
01:54.48EquinoxRepaste it? :)
01:55.01Mavvielsof -n | grep asterisk | grep -E '(TC|UD)P'
01:55.07timecopso how do I get asterisk to setup for sccp
01:55.11Nethablsof -n | grep UDP
01:55.18cursorPerhaps your accountant will allow you to use its address as your registered office
01:55.47EquinoxOr your lawyer.
01:55.58jksAnyone here got a SIPPhone Virtual Number working? (with Asterisk)
01:56.00cursorThe registered address (in England) is used for tax reasons - it's the address the tax man uses to send stuff
01:56.18cursorand the address anyone should be able to use to send stuff
01:56.40cursorso your accountant could be a good choice
01:56.48Nethabjks: nope, i can't even get regular calls to sipphone working right
01:56.49cursor(for a fee - probably)
01:58.07*** join/#asterisk Equinox (~secret@star.l93.com)
01:58.16cursorExcess flood?
01:58.17EquinoxIf u want more just give me an email ;)
01:58.21cursorAre you in Florida?
01:58.25EquinoxYeah I pasted that thing mavvie wanted
01:58.27EquinoxAnd owned myself
01:58.32MavvieEquinox: port 2727 and port 4569 aren't bound to an IP address.
01:58.33cursorpastebin
01:58.45Equinoxasterisk  22968 asterisk   15u  IPv4   32679767                   UDP *:2727
01:58.45Equinoxasterisk  22968 asterisk   16u  IPv4   32679768                   UDP *:4569
01:58.46Nethabasterisk  18394   root    8u  IPv4     242563                 UDP *:4569
01:58.51EquinoxMavvie- They are
01:58.57Nethabthat means all ips
01:58.58EquinoxMavvie- I just died before you got those two lines
01:59.00Nethabnot none
01:59.02*** join/#asterisk syslod (~sysglod@65.114.0.198)
01:59.07Mavvieif they are they wouldn't be a * there
01:59.11jksnoone using sipphone.com here?
01:59.14Equinox* means all?
01:59.16Nethabi am
01:59.21Nethabbut not virtual bumbers
01:59.21MavvieEquinox: yes
01:59.22Gand_DJI think I have my asterisk server setup to link to fwd.
01:59.29cursorsomewhere
01:59.31Gand_DJoutgoing calls work so far :)
01:59.49Equinoxtcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN
01:59.53EquinoxSsh works just fine . . .
01:59.54Equinox(on all Ips)
01:59.55jkshmm, I think I wasted 69.99$ on my virtual number then :-(
02:00.07Equinoxjks- What happened?
02:00.07Mavvieyes, ssh listens on every port.
02:00.14jksEquinox: well, it doesn't work
02:00.15Gand_DJI use sipphone.com.... but don't pay for anything on it
02:00.18Nethabthat's netstat not lsof silly
02:00.24Equinoxjks- what's a virtual number?
02:00.26jksEquinox: my regular sipphone number works fine with asterisk
02:00.28EquinoxWell yeah
02:00.28Gand_DJI like fwd better
02:00.36cursorIt's almost a number, but not
02:00.36EquinoxBut won't netstat show the listening ports?
02:00.38jksEquinox: virtual  number = a real US number that can be dialed from PSTn
02:00.50cursorDDI
02:01.03jkscursor: sorry?
02:01.06Nethabthe virtual number should simply forward to your sipphone number no difference, if the virtualnumber says User unknown then something else is wrong
02:01.10Nethabon sipphone.com's end
02:01.11cursorDirect Dialin number
02:01.14Equinoxsshd        196     root    3u  IPv4        350                   TCP *:ssh (LISTEN)
02:01.16EquinoxFine
02:01.17EquinoxThere :)
02:01.17jksNethab: how do you know that?
02:01.19EquinoxHehe
02:01.20jkscursor: thanks
02:01.42Nethabbecause either way, the connection should come through proxy01.sipphone.com
02:01.50Nethabor your local proxy
02:01.59MavvieEquinox: for your iax.conf:
02:02.03jksNethab: how do you know that?
02:02.05Mavvie[general]
02:02.05Mavviebindport=4569
02:02.05Mavviebindaddr=218.1.2.3
02:02.12jksNethab: i.e. are you just guessing or do you know this/tried this?
02:02.14cursorare you having trouble with sipphone in or out?
02:02.16cursoror both?
02:02.19EquinoxMavvie- Actually I tried binding it directly to that IP, didn't work.
02:02.25jkscursor: regular sip phone works fine for me.
02:02.26EquinoxMavvie- I've tried lots of things ;)
02:02.30MavvieEquinox: what didn't work about it?
02:02.35Mavviethe binding ?
02:02.37EquinoxIt was just as broken as it is now
02:02.38jkscursor: it's just this new number, I can't dial into it from PSTN.. it just says user unknown
02:02.41Nethabbecause the PSTN incoming termination is happeneing before it gets to sipphone that's how they know it's your number
02:02.43EquinoxIt wouldn't register
02:02.44EquinoxOr pass traffic
02:02.52jksNethab: okay, so you're just guessing
02:03.06Nethabwell it's not terminating at your house
02:03.07cursorWhat does your register line look like in sip.conf?
02:03.08Mavviecan you, before you try again, please make sure they're *all* bound to an IP address?
02:03.10jksNethab: i.e. I'm looking for a "missing link" here... something which must be done, which I didn't do
02:03.23jksNethab: like for example I have to do number assignment from a web control panel, or I need to register with a new username or something
02:03.30jkscursor: like regular?
02:03.36jkscursor: do I need to change it?
02:03.41cursorShould be: register => phoneno:password@proxy01.sipphone.com/phoneno
02:03.45Equinoxjks- Using sip or IAX?  You using asterisk or a sip phone?
02:04.04jksEquinox: SIP, Asterisk
02:04.29Nethabeven if you register using x-lite, the PSTN number should ring you through sipphone
02:04.34cursorYou'll also need a type=user section
02:04.39cursorwith username = phoneno
02:04.53jkscursor: yes, I have already have all that working
02:04.54EquinoxMavvie- The same setup works great if I go to the eth0 IP
02:04.56Nethabdid you use the my.sipphone.com to get the virtual number?
02:04.57jkscursor: with my regular sip phone number
02:05.01jkscursor: which works fine
02:05.08jkscursor: now I bought a DDI number as an add-on
02:05.13jkscursor: do I need to change anything then?
02:05.23timecophm
02:05.26cursorI don't know, but I suspect so
02:05.28MavvieEquinox: then use the eth0 IP.
02:05.31jkscursor: I tried adding the DDI number the same way as the regular sip number, but Sipphone.com won't let me register then
02:05.31timecopcant get this cisco communucator thing to do anythign with asterisk
02:05.34cursordid you get a second account with the second number?
02:05.38jksNethab: yes?
02:05.40EquinoxMavvie- I am.. I switched the DNS for the 2nd IP to the eth0 IP
02:05.41cursorThe number I have is used as the username
02:05.46jkscursor: no, it's an addon to the same account somehow
02:05.49EquinoxMavvie- Just kind of.. Strange.  Glad it's a known bug tho
02:05.49cursorso perhaps you'd need two accounts for two usernames
02:05.51jkscursor: same here
02:06.03cursorset up a second type=user block
02:06.06jkscursor: I tried using my new number as the account name, but that didn't work
02:06.27Nethaband under the "Premium features to your account" it shows your Virtual number
02:06.29jkscursor: I can't authenticate with the new number as account name, so it won't owrk.
02:06.36cursorok
02:06.43cursoryou'll need to call sipphone support then
02:06.44jksNethab: There's no Premium features menu?
02:06.45EquinoxI wonder if they just bind the 2nd # to your first account?
02:06.48cursorthey'll probably know
02:06.50EquinoxSo you have 2 "lines" on your sip phone?
02:06.51jkscursor: I've written a ticket... they don't reply
02:06.55Nethabno use your sipphone user as the registry user name
02:07.05cursor2:06am - give them time :-)
02:07.08*** join/#asterisk tris (tristan@camel.ethereal.net)
02:07.10jksNethab: sorry?
02:07.18jkscursor: well, I wrote in two days ago
02:07.22jkscursor: still no reply
02:07.23timecopthis shit isnt working at all
02:07.24timecopugh
02:07.27Nethabyou only need to register to sipphone as 1747 whatever
02:07.36cursorphone them
02:07.49jksNethab: done that
02:07.53jksNethab: what more do I need then?
02:08.06jkscursor: I'll try that then... I just couldn't find their number the other day
02:08.07cursorWhat does your register line look like in sip.conf?
02:08.19Nethabthe virtual number should get translated by their PSTN provider and sent through sipphone to your box
02:08.32jksNethab: ofcourse
02:08.45*** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net)
02:08.46jkscursor: register => 1747667xxxx:9634@proxy01.sipphone.com/1000
02:08.54cursorok
02:09.02*** join/#asterisk w0w0 (~apardo@80.26.167.46)
02:09.11Nethabthe only thing i would try is don't register with asterisk, trying registering with a softpohone on your PC and see if that rings from the virtual number
02:09.29Nethabif not then it's on sipphones side
02:10.07jksNethab: well, either that - or there's something I don't know
02:10.22jksI was wondering if there was any kind of "extra clue" you could send along in the sip registration to sign in for more than one number
02:10.26cursorThis message probably won't change while you're staring at it.
02:10.36Nethabnot really, X-lite and other softphones should receive the call
02:10.37jksor something I needed to do in their my.sipphone.com to assign the new virtual number to a specific sip phone number
02:10.50cursorThis message is slightly different than the one that was here a minute ago.
02:10.51Nethabdo you have more than one account?
02:10.57Nethabon sipphone?
02:11.07jksNethab: yes, but that doesn't have anything to do with it
02:11.12*** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net)
02:11.13Nethabyou can have multiple register commands
02:11.19jksNethab: I know.
02:11.54drbytei've setup asterisk like this: http://www.voip-info.org/wiki-Asterisk+quickstart, and am using kphone. but the registration repeatedly fails :(
02:12.16Nethabjks: if a softphone can't receive the call then it's not your fault
02:12.31jksNethab: It could easily be for the reasons, I've just lined out.
02:12.33cursorI'm going to take a break for a while
02:12.35Darwin35ok having some problems with trunking
02:12.35cursorback later
02:12.37cursorprobably
02:12.50Darwin35I add the needed info to iax.conf
02:13.09Darwin35and the 2 boxes reject each other
02:13.10*** join/#asterisk oncemore (~oncemore@222.33.36.198)
02:13.30Nethabjks: if the virtual number appears under the "Premium features on Your account" and it says status is on then you've done your part
02:13.39timecopso i'm still running 08/05/03 CVS of asterisk because anything past that (not sure of exact date) fails registering with my SIP providers.
02:13.50timecopand i'm now trying to use chan_skinny and its failing hard
02:13.50jksNethab: as I said before, there's nothing in my.sipphone.com that says "Premium Features"
02:14.01timecopany suggestions on what to od?
02:14.14jksNethab: okay, I found what you said now... it wasn't in the menu
02:14.23jksNethab: it lists the number there fine... but as a Custom Number?
02:14.33timecopso much for opensores help
02:14.37jksNethab: the status show "on line"
02:14.40Nethabjks: on my my.sipphone.com page i see "Basic Features on Your account"Basic Voicemail on
02:14.44jksNethab: do you know the phone numebr for their setup ?
02:14.46oncemorehi
02:15.07oncemorehow to make busydetect work?
02:15.31Nethabjks: if you dial ** into sipphone.com you should get your number back?
02:15.38Nethabis that what you mean
02:15.46*** join/#asterisk funraps-as (~funraps@adsl-64-164-82-84.dsl.lsan03.pacbell.net)
02:15.49oncemorei set busydetect=yes in zapata.conf ,but it does not wok
02:16.06jksNethab: I get my number back.. the sip phone number (1-747-etc)
02:16.14jksNethab: sorry, not setp
02:16.17Nethabthat means your connected
02:16.19jksNethab: I meant, the number for their _support_
02:17.48Nethabjks: not offhand sorry
02:19.05oncemorewho can help me?
02:19.19funraps-ashi guys
02:19.21oncemoremake busydetect work.
02:19.22funraps-asI got dissed
02:19.39Nethabi've never used busydetect sorry
02:19.41*** join/#asterisk routerheads_atho (~root@user-12l2p2p.cable.mindspring.com)
02:19.47funraps-asis there a document online that shows ALL of the asterisk extensions.conf commands?
02:20.04*** part/#asterisk routerheads_atho (~root@user-12l2p2p.cable.mindspring.com)
02:20.12oncemoreO ~~ x100p can not detect pstn phone hungup
02:20.21oncemoreo
02:20.23*** join/#asterisk NormAst (NormAst@Toronto-HSE-ppp3677946.sympatico.ca)
02:20.29oncemorethanks
02:20.30Mavviefunraps-as: because they can be generated when you're using app_xxxx, no.
02:20.43Optic2funraps: voip-info.org is the first place I go for asterisk doco! :)
02:21.24*** join/#asterisk nassy (~mark@24-193-231-136.nyc.rr.com)
02:21.46*** join/#asterisk n0where (~ken@o152244.ppp.asahi-net.or.jp)
02:22.24*** join/#asterisk UdontKnow (udontknow@udontknow.staff.freenode)
02:22.27UdontKnowhello
02:22.54oncemorehow understand dsp.c chan_zap.c channel.c ?
02:22.57UdontKnowuh, can someone recommend a console sip client besides cornfedsipua? cornfed doesnt support tcp...
02:23.01*** join/#asterisk Johaan (~johaan@ip24-56-24-181.ph.ph.cox.net)
02:23.14drbytei can't even get kphone to dial in properly :(
02:23.15oncemoreI want to modify busydetect Alg.
02:24.10Nethabheh asterisk can dial out from console, does that count?
02:24.35ariel_hello all
02:24.43drumkillaNethab: probably overkill, but it will do the trick  :)
02:25.09sudoer2can i set macros per client in sip/iax conf, like CID, and then set the CID to whatever is set for the cid in their client setup in sip/iax conf?
02:25.49drbytei followed exactly everything http://www.voip-info.org/wiki-Asterisk+phone+Kphone and i can't get it to work. i'm trying the "simple" config
02:26.01*** join/#asterisk routerheads_atho (~root@user-12l2p2p.cable.mindspring.com)
02:27.17Mavviedrbyte: use sip-debugging on asterisk to figure out what is going wrong.
02:28.51UdontKnowNethab: heh. do you have "full phone" on console with it?
02:29.07UdontKnowNethab: like receiving, calling, waiting and so on
02:29.32_Vilehrmm
02:29.36*** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca)
02:29.40_Vilewonder how I can make lynx post from the command line
02:29.44Nethabon linux is /dev/dsp is configured to your sound card asterisk will use the microphone and speaker to call with
02:30.04NethabCLI> call 912220000000
02:30.28Nethabwill call and use the local mic and speakers
02:30.43UdontKnowNethab: can I register an asterisk with fwd? or need to use e164 stuff?
02:30.54drbyteMavvie: okay, i'm running that in the console. i just turned on sip debug. nothing. kphone is trying, but not working
02:31.04Nethabyes you can register asterisk to fwd and many other providers
02:31.28Mavviedrbyte: if it is trying, does that mean asterisk is receiving the packets?
02:31.51drbyteMavvie: by the looks of it no. asterisk console has nothing responding to it.
02:32.04UdontKnowNethab: nice
02:32.26drbyteusing kphone 4.0.5, compiled off tarball
02:32.27Nethabbut admittedly that's like carving a steak with a chainsaw
02:32.53Mavviedrbyte: does or doesn't asterisk receive the packets? do you see them being outputted on the asterisk console?
02:33.09Chujithat's good
02:33.26Nethabif your phone sees trying that came from the proxy or asterisk
02:33.37Nethabotherwise it would say connecting
02:34.08*** join/#asterisk santiago (~santiago@63.245.86.75)
02:36.05drbyteMavvie: it doesnt. nothing is being output on the console
02:36.20Nethabset verbose 4
02:36.32Nethabwhoops
02:36.40Nethabbad xwindows bad
02:36.52drbyteand i started it up with -vvvvgc
02:38.14*** join/#asterisk soulz2 (~Soulz-@host-137-132-45-204.imcb.nus.edu.sg)
02:38.20soulz2hello all
02:39.24ZX81hello
02:40.59UdontKnowNethab: some friend tells me that it chews 100% cpu when calling from asterisk... is that true?
02:41.54Nuggethow would we know if your friend is lying?
02:41.58Nethabwell not 100% but asterisk has to convert your raw audio to a voip codec instead of your pc so it's a little heavy yeah
02:42.16zigmanbut never ever 100%
02:42.18*** join/#asterisk calvis (calvis@h-66-167-52-131.sttnwaho.covad.net)
02:42.20Nethabthat and chan_oss sucks
02:42.23zigmanunless its a 386 something
02:42.57*** join/#asterisk Legend (~legend@24.244.142.134)
02:43.26*** part/#asterisk sigtom (~sigtom@rrcs-67-78-17-122.se.biz.rr.com)
02:43.31*** join/#asterisk sigtom (~sigtom@rrcs-67-78-17-122.se.biz.rr.com)
02:43.51Nethabok those of you helping me get my sipphone outbound working again, I was able to get the 1222 number to work but only intermittently
02:44.29Mavviedrbyte: then asterisk isn't receiving anything.
02:44.50*** join/#asterisk rustyb (~rustyb@68.235.250.116)
02:45.45drbyteMavvie: any idea why that might be the case? is kphone an appropriate tool to use?
02:46.15Mavviedrbyte: I don't know why it happens. First thing I would do is to run tcpdump in the local and remote machine to see where the packets are actually going to.
02:48.53drbyteMavvie: i think i solved it. what happened was i was running asterisk via ssh. now i run it locally on the console and it works! now to get the isdn card working (the winbond)
02:49.23Mavviegood luck :-)
02:49.39drbyteMavvie: thanks. have you had experience in setting it up?
02:51.54*** part/#asterisk KHague (~khague@72.39.cm.sunflower.com)
02:52.51oncemorewhy busydetect alg. can not detect busy tone?
02:53.45implicitlol
02:53.47implicitbecause it sucks
02:57.05oncemore?
02:57.08oncemorewhy???
02:57.18oncemorewhat means?
02:57.23fraglethi all
02:57.28fragletquick question if i may
02:57.44Mavviedrbyte: I did everything via ISDN4Linux
02:57.46fragletwe are looking to use pri card on asterisk, does it handle it well?
02:58.04Mavviefraglet: no problems here using the zaptel driver
02:58.22fragletmavvie: thats using one of the digium cards?
02:58.34Mavviethat's using the TE410P cards
02:58.36sigtomsucks=bad
02:59.20fragletthats the one i was looking at.
02:59.29oncemore:(
02:59.37oncemorepool busydetect alg.
02:59.39fragletyou know if it scales well.. ie multiple te410`s in one box?
02:59.52Mavviefraglet: I have two TE410Ps in one box.
02:59.56oncemorecallprocess can not work outside US
03:00.05oncemorehow to du?
03:00.10oncemorehow to do?
03:00.25fragletmavvie: thanks for you help, makes us feel more comfortable ordering up some hardware and pri to give it a go
03:00.38Mavvieno probs.
03:01.04oncemorewho can help me to detect busytone outside US?
03:01.19Nethabyou mean ringing tones hehe
03:01.59oncemoreBUSY tone 450Mhz 0.35s<On Time(ms)> 0.35<Off Time(ms)
03:02.08ta[i]ntedanyone here use broadvoice
03:02.10jksanyknow knows the exact meaning of this sipphone error message: "the user you're trying to reach is unknown"
03:02.20Nethabi was commenting on how european ring tones sound similar to US busy tones
03:02.38oncemorehow to detect this tone? now busydetect alg. not work
03:02.46oncemoreit is China tone plan
03:04.37UdontKnow;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup)
03:04.42UdontKnowhow does one use that?
03:04.43NethabI get "The user you are trying to reach is unavailable" all the time
03:04.52drbyteMavvie: well, thats what i'll be using. got to read up on isdn4linux now then. it seems it gets automatically detected on Fedora Core 3 (the card), but other than that, i guess i'm in for some fun...
03:05.08oncemoreI also played with BUSYDETECT_MARTIN and/or BUSYDETECT_TONEONLY and it
03:05.09oncemoremakes no difference. I also tried editing dsp.c and adjusting
03:05.09oncemoreBUSY_MIN and BUSY_MAX, but nothing fixes these problems.
03:06.13Mavviedrbyte: make sure you have the patches from http://bugs.digium.com/bug_view_page.php?bug_id=0002704 in your code.
03:06.30Mavviehttp://bugs.digium.com/bug_view_page.php?bug_id=0002667 that is.
03:06.40*** join/#asterisk goatmilk (~goatmilk@cae168-249-184.sc.rr.com)
03:07.25*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || DUNDi (http://www.dundi.com) #dundi || Find a bug... help squash it out! http://bugs.digium.com
03:07.27drbyteMavvie: thanks, though i'm using stable? 1.0.2 iirc
03:07.43Mavviedrbyte: just check if that patch is in it.
03:08.06drbyte1.0 is when its fixed. i will double check in a bit
03:08.38goatmilkanyone get the error about radfw.h file not found during compile of zaptel w/ make linux26 ?
03:09.00NormAstAnyone know why this won't work ???     Zap/G1/9056198844,,D(90565554000)
03:10.12bkw_NormAst no clue
03:10.16bkw_that should be fine
03:10.22bkw_show your full dial line
03:10.41bkw_goatmilk nope
03:10.43file[laptop]nowhere else I'd rathe rbe
03:11.06*** join/#asterisk t3t (~t3t@207.67.0.20)
03:11.15NormAstbwk: it's getting called in C    ret = pbx_exec(chan, app, data, 1);
03:11.54goatmilkbkw_: apparently you need to run make radfw.h before make linux26  ..
03:12.05bkw_goatmilk no you don't
03:12.07bkw_make clean
03:12.09bkw_then do it again
03:12.22goatmilkwell i did that, and it didn't work
03:12.28goatmilkso then i did my way, and it worked
03:12.52bkw_just do make
03:12.56bkw_it works the same
03:13.00bdeb5how could i configure asterisk to call out to a prepaid calling card, dial my pin, and then have it dial the number that i want to dial?
03:13.16bkw_the D option on app dial
03:13.39NormAst<PROTECTED>
03:13.39NormAst<PROTECTED>
03:13.53puzzledhey guys, what's the "List propostion" threat about?
03:14.00bkw_try it other another way
03:14.08bkw_NormAst try it witohut pbx_exec
03:14.18NormAstdbeb5: Create a macro..
03:14.25bkw_dont need a macro
03:14.29bkw_the D flag can do it
03:15.22file[laptop]you don't need a reason, let the rain go on and on
03:15.46NormAstbkw: So how do I dial out without pbx_exec?
03:15.56file[laptop]bkw_: how are you oh Brian-like one?
03:16.06bkw_exten => 555,1,Dial(Zap/g1/blah||D(1234))
03:16.17bkw_NormAst you know C and you had to ask that?
03:16.27bkw_:P
03:16.44NormAstIt's been 10 years since I played in C and linux :)
03:16.49bkw_;)
03:17.23file[laptop]Enya is nice.
03:17.37bkw_yactopitc
03:18.42bkw_file[laptop] yactopitc
03:20.19NormAstbkw: needs the || and not the ,,
03:20.27bkw_both work
03:20.54NormAstI tried the ,, and pbx_exec and it did not work!
03:21.18NormAstbugs?
03:21.23bkw_doubt it
03:21.41NormAstI bet it is..
03:21.51bkw_bet its not:P
03:22.12bkw_file[laptop] do you wish me to smack you.. YACTOPITC
03:22.12NormAstbkw:  PRI issue?
03:22.24bkw_NormAst who knows let me try it
03:22.30file[laptop]whatttttttttt? I was reading code change reports
03:23.13*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnjl.dialup.mindspring.com)
03:23.23NormAstbkw:  Okay... What would you like to do?
03:23.35file[laptop]bkw_: you've lost me
03:25.02bdeb5how could i configure asterisk to call out to a prepaid calling card, dial my pin, and then have it dial the number that i want to dial?
03:25.39bkw_file
03:25.42bkw_666
03:25.45file[laptop]one sec
03:26.20brc_YACTOPITC!
03:26.35brc_wtf
03:26.36brc_bkw_,
03:27.10NormAstdbeb5: it was answered!
03:27.34NormAstexten => 555,1,Dial(Zap/g1/blah||D(1234))
03:28.10NormAstJust change the 555 to the area code, and the blah to ${EXTEN}
03:28.27NormAst555 to Area code and number ie: 7165551212
03:29.51UdontKnowI have some doubt regarding fwd setup as I found on http://lists.digium.com/pipermail/asterisk-users/2004-January/034050.html
03:30.05UdontKnowthe sample says
03:30.06UdontKnowexten => s,1,Hangup
03:30.06UdontKnowexten => 74928,1,Answer
03:30.23UdontKnowwhere can I find how to do it right for my setup?
03:30.45UdontKnow(the sample uses multiple fwd lines...)
03:33.00oncemorehi
03:33.15brc_bye
03:33.18oncemorewho understand dsp.c channel.c chan_zap.c
03:33.30brc_nobody
03:33.39brc_it's voodo magic
03:33.48brc_it wrote it self really
03:34.00czero:)
03:34.04czeroits selfaware
03:34.18*** join/#asterisk alphaque (~Alphaque@218.208.238.245)
03:34.18puzzledunless "it" receives money off course. then it explains by itself
03:34.34czeroif you try to modify it, it will appear in you office and steal your soul, only to grow more strong and ruthless
03:34.43brc_genetic evolutionary code using highly advanced heuristic models
03:35.01puzzledsome whisper it is more evil than bill himself
03:36.00puzzledwhich would explain the money part