IRC log for #asterisk on 20080911

00:03.54*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
00:04.19*** join/#asterisk PASCA (n=peter@abkz236.neoplus.adsl.tpnet.pl)
00:04.40PASCAHello all.
00:05.15PASCAQuick question... Is it possible to assign two public static IPs to one asterisk box?
00:06.47PASCAHello? Is there anyone here?
00:06.49ManxPowerPASCA: There is nothing special about two public IPs.  Packets will be routed according to the linux kernel routing table.
00:07.33ManxPowerof course, it could very well be a living hell to make the box route packets the way you want when using two public IPs, but that is a linux/networking issue, not an Asterisk issue.
00:07.57ManxPower(look up "source routing")
00:08.41PASCAManxPower: Ok thanks. The issue I am experiencing with Asterisk 1.4 is the following. I have two public IPs on one ethernet card. eth0 is 51, eth1 is 48.
00:08.50[TK]D-FenderPASCA: You cannot tell * to bind to 2 IP, its either one, or all.
00:08.57PASCAClients are able to register on 51, but not on 48.
00:09.20ManxPowerPASCA: put in a route to the clients to go out via 48
00:09.20[TK]D-FenderPASCA: And multi-homed * boxes cause real problems.
00:09.50ManxPowerPASCA: on the asterisk linux box
00:10.02ManxPowerthen they should be able to register to 51
00:10.06ManxPowersorry, to 48
00:10.25PASCAManxPower: So this is a linux kernel routing table issue correct?
00:10.59ManxPowerPASCA: correct.  The "multi-homed * boxes cause real problems." [TK]D-Fender is talking about is, I strongly suspect, all linux routing issues.
00:11.23*** join/#asterisk ScarEye (n=scareye@12.27.87.66)
00:11.28ManxPowerbut I do agree you should not use bindip= option
00:11.40[hC]<PROTECTED>
00:11.55PASCAI have bindip set to 0.0.0.0.
00:12.09ManxPowerPASCA: just comment it out.
00:12.54PASCAManxPower: Ok thanks I will comment it out and add a route in the routing table. This should allow registrations on 48 correct?
00:13.24[TK]D-FenderManxPower: Picture that UDP is stateless, a call comes in on A but * doesn't have a "state" (UDP!!) to know which interface to send it out.  Hilarity ensues
00:13.51[TK]D-FenderManxPower: Maybe with some conntrack zaneyness it might work... but boy, what a lot of trouble.
00:14.12ManxPower[TK]D-Fender: Asterisk DOESN'T SEND IT OUT AN INTERFACE.  Asterisk sends it out, the kernel handles the source ip because asterisk did not bind to a specific interface.
00:14.16[TK]D-FenderPASCA: Whats the point of running multiple IP's both public on the same box for this?
00:14.24lanningactually the issue is that when bound to the global socket (0.0.0.0), the source IP is the interface it routes out of, if you have two IP addresses on one card, it picks the one assigned to the physical interface (as opposed to the eth0:1 interface)
00:14.42[TK]D-FenderManxPower: Sorry, correct, not "interface", just that it'll sounce by one arbitrary interface or another
00:14.59[TK]D-FenderIP -> to go and "interface"
00:15.05[TK]D-Fenderbleh, you know what I mean
00:15.15[TK]D-FenderAnyway you go about it... trouble.
00:15.40[TK]D-FenderPASCA: If you want to do something like this I suggest you run a proxy in front
00:16.03PASCA[TK]D-Fender: I have some extensions registered on 51 and have about fifty on 48. Instead of reprogramming the ATAs with the 51 IP, I want to group the all on one server.
00:16.05ManxPowerlanning: what if you put a route in the routing table to send packets destined for 209.61.27.185 goes out interface/IP 206.87.43.12
00:16.15ManxPower(just random example addresses)
00:16.19*** join/#asterisk __jeff_O (n=silas@ool-18bab2c5.dyn.optonline.net)
00:16.41ManxPowerPASCA: it is better to change them all to use a hostname or to the same IP.
00:17.16__jeff_Oif my asterisk server is behind a firewall, what are the necessary ports to forward so that computers outside the firewall can make sip calls?
00:17.22ManxPowerotherwise sometime in the future when you totally forgot about the route line you put in /etc/rc.d/rc.local and bad things happen
00:17.42__jeff_OPorts 5060 to 5070 and 10001 to 20000 ?
00:17.43ManxPower__jeff_O: Read the sipnat document *CAREFULLY*
00:17.46ManxPower~sipnat
00:17.46jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
00:17.58ManxPowerthe wiki one is confusing, use the first lin,
00:18.00ManxPowerlink
00:18.19PASCAManxPower: That's the thing. It may be difficult to change all devices. So I would rather have 2 ips on 1 * box.
00:18.53PASCADo you really think this will cause many more problems than it is worth?
00:19.05ManxPowerPASCA: yes.
00:19.32ManxPowerI would do it as a temporary measure to make the clients registering to the old address still work -- as a transition thing.
00:19.51ManxPowerPASCA: are the clients PCs or phones?
00:20.16__jeff_OManxPower: ok, so are the 10000-20000 ports for audio? That seems to be what I've read elsewhere and the doc you gave me said they should be open.
00:20.25PASCAManxPower: Phones... Combination of ATAs and softphones.
00:21.16ManxPowerPASCA: I would try this.  Have one of the softphone pc's ping the 48 address.  See what the response is.  I suspect it will be from the 51 address.
00:21.31ManxPowerif so, then you can test the route changes on the linux box by just running a ping from one of the clients
00:22.22ManxPower__jeff_O: configurable in /etc/asterisk/rtp.conf
00:22.51PASCAManxPower: Thanks a lot for the info. You have been a great help! :)
00:22.56PASCA[TK]D-Fender: Thanks for your help as well.
00:23.03ManxPowerPASCA: let me know how it goes.
00:23.22ManxPowerrouting is bitch to get right, even on dedicated routers.
00:24.13[TK]D-FenderPASCA: You're welcome, and ManxPower easily outclasses me on the networking side of this.  Profit from whatever he can offer you.
00:24.20__jeff_OManxPower: cool, thanks. Hopefully opening those ports will fix my probs
00:24.36PASCAManxPower: I won't be able to give it a shot until tomorrow, but I will come back and let you know for sure.
00:24.40[TK]D-FenderPASCA: Mine was more warning than guidance in that respect
00:25.27PASCA[TK]D-Fender: It was some useful information indeed. Your warning and his guidance. I will give it a go, hopefully as a temp fix.
00:25.42PASCAWill let both of you know how it goes.
00:25.52[TK]D-FenderPASCA: You've walked away ahead of the game then.  Best of luck to you.
00:26.59PASCAThanks! Have a good one guys.
00:30.00*** join/#asterisk phl4kx (i=phl4kx@190.40.104.189)
00:30.02phl4kxhi for all
00:30.18phl4kxwhat is the IAX or SIP phone for call to DIGIUM SUPPORT????
00:32.57phl4kxwhat is the IAX or SIP phone for call to DIGIUM SUPPORT????
00:35.10phl4kxexten => 500,1,Playback(demo-abouttotry); Let them know what's going on
00:35.10phl4kxexten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default)     ; Call the Asterisk demo
00:35.10phl4kxexten => 500,n,Playback(demo-nogo)      ; Couldn't connect to the demo site
00:35.10phl4kxexten => 500,n,Goto(s,6)                ; Return to the start over message.
00:35.11phl4kx:D
00:36.30*** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d325993de75d3c1d)
00:38.14rue_mohrlook its asterisk related
00:39.34*** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net)
00:40.30*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
00:48.47*** part/#asterisk PASCA (n=peter@abkz236.neoplus.adsl.tpnet.pl)
00:58.33*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
01:00.10*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com)
01:05.45*** part/#asterisk jsmith-teaching (n=njsmith@72.21.36.138)
01:32.36*** join/#asterisk wiscados (n=mint@81.25.184.155)
01:36.38*** join/#asterisk synchris (n=synchris@athedsl-218704.home.otenet.gr)
01:39.58*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
01:41.21DaminPlease DIGG and pass on: http://digg.com/linux_unix/Ohio_LinuxFest_2008
01:42.13jayteehmmm, Ohio
01:42.35jayteewonder if any of the Ubuntu Loco team from here in Indiana are going
01:42.59DaminYep.. they party like rockstars..
01:43.06DaminWe have Jono Bacon keynoting..
01:43.14DaminI suspect there will be a large Ubuntu presence..
01:44.02jayteeDon't know if I'll be able to go the way my schedule is
01:45.30jayteethey do party hard. When they first got team approval they partied so hard they almost got kicked out of the Claddagh Irish Pub
01:49.23*** join/#asterisk moy (n=moy@189.169.91.147)
01:50.31*** join/#asterisk dwayne (i=dwayne@76.29.245.9)
02:11.06*** part/#asterisk Steve_J-obs (n=Chris123@209.58.251.50)
02:11.18lmadsenwell hello there!
02:11.25Qwellohai
02:11.30lmadsenomghi2u
02:11.46*** join/#asterisk sapere (n=sapere@c-71-206-107-228.hsd1.mi.comcast.net)
02:12.30sapereevening all -- question about the asterisk service provider, ironvoice ... maybe i don't understand how the process works, but where do i get to pick a phone number?
02:14.32lmadsenanyone have suggestions for Asterisk Cookbook recipes they would like to see in a book?
02:18.52*** join/#asterisk oilinki (n=oil@ppp-124-120-4-102.revip2.asianet.co.th)
02:19.33QwellMr. Madsen
02:26.51*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
02:28.19lmadsenMr. North
02:28.24Qwelllooks around
02:28.28Qwellhe's dead
02:28.31lmadsenoh really
02:28.34Qwellmmhmm
02:28.37Qwellshh
02:28.41lmadsenoic
02:28.49lmadsenMr. Parker.
02:29.03Qwellmeh, I don't like that guy
02:29.14Qwellhe's a jerk
02:29.21lmadsenMexico just scored on Canada in the World Cup qualifying, so I will now find something else to watch since that game is over
02:29.26lmadsenQwell: I agree
02:29.33lmadsenQwell: give me recipe ideas! :)
02:29.35mchouIs there a way in Asterisk to send DTMF after caller and callee have been bridged?  Kind of like 'Dial(blah,timeout,D:12345)' except after both sides have been bridged
02:29.37lmadsenI have 91 so far
02:29.47QwellI gave you one!
02:29.54lmadsenthat's why I have 91 :)
02:29.55lmadsenand not 90
02:29.56Qwell:D
02:30.04filelmadsen: new TV channels!
02:30.05Qwellhow about...
02:30.11Qwella youtube "viewer"
02:30.12lmadsenfile: oh?
02:30.24Qwellcan you convert flv to ulaw? :D
02:30.40filelmadsen: Joy!
02:30.44*** join/#asterisk EI5GTB (n=paul@78.16.158.28)
02:31.14lmadsenhas no idea what file is talking about
02:31.27lmadsenhrmmm... someone just told me he thinks dialplan is what is holding asterisk back
02:31.31lmadsenscoffs
02:32.11mchouI think so too.
02:32.32lmadsenfunny, because I've created some pretty good dialplan
02:32.41mchoumost keywords dont make any sense.....
02:33.06lmadsendefine: keywords
02:33.20mchoulike what's the diff betw. 'username' and 'fromuser'
02:33.27mchoulol
02:33.31Qwellblame sip, not Asterisk
02:33.34filethat's not dialplan
02:33.37lmadsenexactly
02:33.45mchouand the classic: 'user vs. peer'
02:33.49mchoulol
02:33.51Qwellblame sip, not Asterisk
02:34.01lmadsenalso not dialplan
02:34.04mchouuser vs. peer is sip?
02:34.21Qwellit was easier to hit up+enter
02:34.23lmadsenuser vs. peer is not really THAT hard
02:34.39Qwellbut, yeah, basically.  SIP has multiple types of servers
02:34.47mchouno, but nobody is clear on the concept
02:35.07lmadsenincoming calls try to match on users from bottom to top first on the [username], and if that doesn't match, it'll search peers for IP address from bottom to top
02:35.24mchouwhere ppl at digium basically says 'just use peer!'
02:35.26mchoulol
02:35.29mchoulame
02:35.37Qwelluser = gone
02:35.37lmadsenthat's a bit simplified, but is pretty much the jist of it
02:35.41lmadsenalso true
02:36.39lmadsenI wish I had chocolate
02:36.56mchounot to mention 'include' and extension matching priority madness
02:37.04mchoumore lameness
02:37.17lmadsenI believe that has more to do with your lack of understanding
02:37.25mchouno it doesn't
02:38.04lmadsenI've never had a problem understanding matching order in my dialplans
02:38.05mchouan include stanza gets matched last
02:38.10lmadsenright
02:38.13lmadsenthat makes sense
02:38.17lmadsenmatch local first, then follow includes
02:38.23mchoumake no sense whatsoever
02:38.29lmadsenI disagree
02:38.49lmadsenyou think it should search the includes before the local context?
02:38.54mchoucause there are plenty of conflicting 'namepaces', if you will
02:38.56lmadsenthat makes no sense to me
02:39.53mchouno, a rule like 'match specific prior to general (regardless of includes)' makes MOST sense
02:40.13lmadsenuhhhh.... it does do that
02:40.23lmadsen_123XXX is matched before _XXXXXX
02:40.38mchoumeaning include is more like a c preprocessor #include
02:41.03mchoulmadsen: it doesnt if includes ALWAYS gets matched last
02:41.24lmadsenright, it should match locally first
02:41.40mchoucause an include can clearly have more 'specific' matches than the local
02:41.51Qwellmchou: There's this book that some totally cool guys (who owe me money) wrote, that you should check out
02:41.53Qwell~book
02:41.54jbotsomebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
02:42.07mchouQwell: stop lecturing me on the book
02:42.26mchounothing I said contradicts anythingf mentioned in the book
02:42.35Qwelllecturing?  I brought it up once
02:42.48lmadsenthis conversation is going no where, so I'm going to find something to eat
02:43.42mchouI just think the way asterisk parses includes and pattern matching is lame when includes are ALWAYS matched last.
02:44.15Qwellyou can use it to your advantage fairly easily
02:44.23mchouAsterist could be a bit smarter about pattern matching rather than relegating includes last
02:46.44lmadsenI still disagree on your position. I've been writing dialplans for 5 years, and never found it to be a problem; quite the opposite
02:46.53mchoulol
02:47.18mchouit's not a problem if you don't use includes, just like 'user'
02:47.28lmadsenI use plenty of includes
02:47.33Qwellit isn't a problem if you know how to use them
02:48.46mchouyeah, now I dont know how to write a dial plan since I point out the sortcoming with include processing in asterisk
02:48.56mchoushortcomings*
02:49.43*** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net)
02:49.45mchouDoes Digium or asterisk have anything that you guys find 'irregular?'
02:50.03mchouI mean they can do no wrong, right?
02:50.15lmadsenI have found lots of things
02:50.24lmadsenI just disagree on your position about includes
02:50.50lmadsenand you can disagree with me
02:50.51lmadsenNEXT!!!
02:51.12mchoulmadsen: fine.  I can respect that.  But saying "I've never had a problem' is hardly a rebuttal
02:51.36mchouwhen I said an include can have more specific patterns than local.
02:51.49lmadsenI don't understand what you want me to say. I've written plenty of 1000+ line dialplans that utilize includes heavily, and never found a problem
02:52.11lmadsenperhaps I've not run into the same problems as you
02:52.16lmadsenbut the rules are quite clear
02:52.41mchoumaybe they are a 1000+ lines since asterisk didnt do smarter pattern matching :)
02:52.51lmadsensure thing
02:52.56lmadsenmust be it
02:53.08*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
02:54.41*** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com)
02:55.26dwayneI'm computering...
02:56.51[TK]D-Fendermchou: If you want to talk about "irregular" I'll let you know when the next bowel movement is on its way...
02:57.18dwaynelol
02:57.30mchou[TK]D-Fender: You clearly need more prunes :)
02:57.52mchou[TK]D-Fender: Constipation is a killer!
02:58.06[TK]D-Fendermchou: "constant" isn't an enviable measurement of "success"
02:58.36dwayneespecially when it is combined  with high frequency
02:58.45[TK]D-Fendermchou: As for dialplan issues, there is little I can see to complain about.  Takes very little effort to acheive 99% of peoples needs.
02:59.16[TK]D-Fendermchou: Do you have an actual issue you'd like advise with?
02:59.25[TK]D-Fendermchou: Or are you just trying to stir the pot?
02:59.47mchou[TK]D-Fender: [19:29] <mchou> Is there a way in Asterisk to send DTMF after caller and callee have been bridged?  Kind of like 'Dial(blah,timeout,D:12345)' except after both sides have been bridged
03:00.02mchou[TK]D-Fender: and no, I'm not trolling
03:00.43[TK]D-Fendermchou: After they've been bridged huh?  That'd be tough indeed.  Whats the purpose?
03:00.48mchou[TK]D-Fender:
03:00.48mchou[19:31] <lmadsen> hrmmm... someone just told me he thinks dialplan is what is holding asterisk back
03:00.48mchou[19:31] * lmadsen scoffs
03:01.11mchou[TK]D-Fender: that was all I was responding to
03:01.51mchou[TK]D-Fender: Lame ass grandcentral is the purpose
03:02.08Qwellonce the channels have been bridged, you are no longer in the dialplan
03:02.10*** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net)
03:02.11[TK]D-Fendermchou: what was intended by use of "dialplan"?  Is it a pattern matching aspect?  Priority logic?  Lack of great SIP signalling control via it?  That is an open ended wide-angle attack.
03:02.12Qwellhence the "bridge"
03:02.30mchou[TK]D-Fender: I dunno what's going on w/ Grand Central as far ast DTMF
03:02.32[TK]D-Fendermchou: Why do both sides need the dtmf?
03:02.34mchouas*
03:02.56[TK]D-Fendermchou: When you dial with "Dial" why do YOU need DTMF?
03:03.28mchou[TK]D-Fender: I dont need DTMF.  GC does.
03:03.31[TK]D-Fendermchou: I can see why you'd want to pass THEM digits.  D() does already do that.  What are you trying to have * ask itself?
03:03.52[TK]D-Fendermchou: but you said AFTER bridge.  Whats wrong with sending it to them BEFORE the bridge?
03:04.13[TK]D-Fendermchou: To them the call is set up, so whats the difference?
03:04.16mchou[TK]D-Fender: the programmtic way doesnt work successfully with GC.  On manual works
03:04.28mchous/On/Only
03:04.42[TK]D-Fendermchou: What kind of failure?  Perhaps * sends the digits too fast?
03:04.47*** join/#asterisk nr4q (n=dave@wsip-70-183-143-230.hr.hr.cox.net)
03:04.55mchounope. plenty of wwwwww
03:04.59[TK]D-Fendermchou: as in they need you to wait a sec before litening for your answer...
03:05.06mchouw=0.5sec delay
03:05.14nr4qman... asterisk gives me a chub
03:05.20[TK]D-Fendermchou: I'm not certain "w" works inside of D() has this been confirmed?
03:05.29QwellI think w only works on zap
03:05.37[TK]D-Fendermchou: AFAIK, "w" only works in the tech part of a zap call
03:05.45mchouw works.  At lest it ways before sending DTMF to me
03:05.49[TK]D-Fendermchou: Do you see it documented somewhere?
03:06.08mchoucore shore application Dial in * 1.4
03:06.17mchoushoe*
03:06.22mchoushow**
03:06.24mchoulol
03:06.27fileoh it'll work any time Asterisk sends DTMF to a channel...
03:06.40*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
03:07.55[TK]D-Fendermchou: I don't see it in the insrtuctions.
03:08.33filemchou: are you sending a string of DTMF digits?
03:09.24mchoufile: just :wwwww1
03:09.47mchouvarious amounts of w's of course :)
03:10.08Qwell:?
03:10.25filethe internal API call would just discard that as an invalid digit
03:10.27Qwelloh, right
03:10.49mchoufile: huh??  what's an invalid digit?
03:10.51file0-9*#aAbBcCdDfF are valid
03:11.08file: is invalid
03:11.15filebut not your problem.
03:11.37[TK]D-FenderSounding like 'w" is not an option for D()
03:11.45mchoufile: ':' is not a part of the DTMF string :)
03:11.47Qwellwait, :?
03:11.55fileit's to separate calling/called
03:11.56Qwellyou're sending the DTMF to the calling party?
03:12.03[TK]D-Fendermchou: here's a though, call it yourself manually and the very instant you see it answer, try dialing.
03:12.03mchouD([called][:calling]) - Send the specified DTMF strings *after* the called
03:12.03mchou<PROTECTED>
03:12.03mchou<PROTECTED>
03:12.03mchou<PROTECTED>
03:12.03mchou<PROTECTED>
03:12.19[TK]D-Fendermchou: Nowhere does that say "w" is valid
03:12.26file[TK]D-Fender: It's valid.
03:12.31mchouQwell: yup.
03:12.38Qwelldid you answer?
03:12.42mchou[TK]D-Fender: use the source luke :)
03:12.43[TK]D-Fenderfile: I'm still looking through the code for it
03:12.51file[TK]D-Fender: ast_dtmf_stream in main/app.c
03:13.06[TK]D-Fendermchou: I am looking in the code, just haven't gotten there yet.  If you care to direct me I'll happily look there./
03:13.19mchouQwell: If by 'answer' you mean did I pick up the phone, then yes :)
03:13.26QwellThe calling chanenl
03:13.30Qwellchannel too
03:14.01QwellIf grand central doesn't allow DTMF before the call is answered...
03:14.01mchouQwell: I'm not sure the channel is explicitly answered......
03:14.18fileQwell: hrm... I wonder if Dial() doesn't do that...
03:14.24Qwellfile: pre-bridge?
03:14.27[TK]D-Fenderfile: I'm seeing it.  If that is what's referenced for all realted apps, then ok/fine/sure.
03:14.32Qwellit might
03:14.35fileQwell: this would certainly be executed pre-bridge...
03:15.02Qwellworth a try though.  put an answer/wait before the dial
03:15.02[TK]D-FenderNo, the line would have to be answered to pass DTMF no?
03:15.13file[TK]D-Fender: maybe.
03:15.14Qwell[TK]D-Fender: the question is whether it's answered at that point yet or not, implicitly
03:15.14mchouQwell: oh, Answer would work.  but it would have side effects
03:15.21Qwellmchou: such as?
03:15.30Qwellnevermind
03:15.32[TK]D-Fendermchou: a thought, try it in M() <-
03:15.52mchouif you 'Answer programmatically' GC regards that called as picked up.
03:16.20mchouI dont want to that to happen b4 I pick up......
03:16.33mchouphysically pick up, that is
03:17.02file[TK]D-Fender: that is a solid idea
03:17.15mchou[TK]D-Fender: sorry? you mean macro or ?
03:17.19[TK]D-Fenderfile: I prefer to think of it as "all thats left" :)
03:17.24[TK]D-Fendermchou: Yes.
03:17.40*** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net)
03:17.41fileis too tired to look through app_dial
03:17.47[TK]D-Fendermchou: in yoru macro do "answer", "wait(2)", SendDTMF", etc
03:18.54mchouI dont see how that avoid the side effect of having the call answered b4 I physically pick up?
03:19.02mchouavoids*
03:19.36filethe Macro only executes once a called channel is answered
03:19.59file(if done via the M dial option)
03:20.03[TK]D-Fendermchou: Your side isn't the issue is it? you are calling OUT to them, no?
03:20.18mchou[TK]D-Fender: nono.  GC calls me
03:20.43[TK]D-Fendermchou: tehn ANSWER their call, and jsut do SendDTMF before calling inside anywhere
03:21.05filedrifts off to bed
03:21.12[TK]D-Fendermchou: and D() should send back to them anyways.
03:21.15mchou[TK]D-Fender: back up.  lemme descibe how GC works
03:21.28[TK]D-Fendermchou: yeah I know you have to ACK the call
03:21.44[TK]D-Fendermchou: I ran into others trying to automate the "annoy the human" to answer
03:22.24mchou[TK]D-Fender: the stated purpose is NOT to annoy the human
03:22.26mchou:)
03:22.54[TK]D-Fendermchou: yeah, thats YOUR goal, not theirs.
03:23.03mchouit's for callee me to decibe whether I want it to go to VM and the like :)
03:23.13mchoudecide*
03:23.27[TK]D-Fendermchou: Have you tried just Answer first, then SendDTMF direct in the dialplan and THEN passing off to your inside phone, etc?
03:24.12mchou[TK]D-Fender: no, not yet, but others have and they say it seems to work (according to the intarweb)
03:25.05mchou[TK]D-Fender: I have no reason to doubt that might work
03:25.10[TK]D-Fendermchou: well m() is definately no use, so if you can't get D() to work (DO Answer the call before the dial at the bare minimum, and play a sound (even silence) BEFORE You Dial (might help to set up the audio path).
03:25.23[TK]D-Fendermchou: Aside from that you have my only other suggestion
03:25.48[TK]D-Fendermchou: and others have gotten it to work.  I HIGHLY recommend the "Answer" + "Playback" right from the start
03:26.07[TK]D-Fendermchou: Just saved someone elses sanity with that earlier today
03:26.27mchou[TK]D-Fender: yeah.  I just dont like the side-effect of programmatic answer
03:26.56mchouthat's the whole point, isnt it?  Automating this crap :)
03:28.01[TK]D-Fendermchou: GC is generally considered a toy for cheap-ass kiddies who haven't gotten around to trying to bypass CID blcok with that other kiddie schmuck provider they saw a youtube video for.
03:28.18mchouhaha
03:28.37mchouI resemble that remark :)
03:29.07[TK]D-FenderA slow painful progression towards the "I give up on VoIP, so I can use my 5$winmodem as FXS/FXO right?@!?! plaeze!!!"
03:29.14mchoubut I got it from David Pogue of NYT :)
03:29.37[TK]D-Fendermchou: Pogue reminds me of "Hate By Numbers" videos.
03:29.48Nuggethttp://macnugget.org/stuff/asterisk-irc.txt
03:30.01mchou[TK]D-Fender: Never seen those videos
03:30.23[TK]D-Fendermchou: Same kinda sarcastic tone in their teardowns of things.
03:30.33Nuggethate by numbers is a cracked.com thing
03:31.55[TK]D-FenderThat guy is kinda funny...
03:32.35mchouwell, at least david pogue is not completely clueless
03:32.46Qwellyeah, neither is Markoff
03:32.52mchouunlike 2bigtelephony guys
03:33.17mchouI heard their crap the other night any nearly died laughing
03:33.30Qwelllink?
03:33.41mchouCLEC==Common Local Exchange Carrier.
03:33.43mchoulol
03:33.56mchouand other faux pas
03:34.32Qwellmchou: who?
03:34.57mchouQwell: some jerks doing audio blog
03:35.10Qwellgot a link?
03:35.20[TK]D-Fendermchou: well I think I'll more permanently move you off of the "troll" list and into the "rabble rousing / trouble seeking / agitant" list :)
03:35.25mchouhttp://2bigtelecomguys.com/
03:35.26*** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net)
03:35.35mchouit's a real hoot
03:35.55mostyis there a manager command to take two channels and bridge them?
03:35.59Qwellthis weeks?
03:36.36mchouQwell: nah
03:36.55mchouQwell: Aug 19
03:40.33Qwelltheir voices are annoying
03:40.40Qwellrecord in > 8khz, thanks
03:41.00mchouQwell: Try not to focus on their voice but content
03:41.06Qwelltrying
03:41.35mchouI mean the voices are annoying but that's not what get's my goat
03:41.41mchougets*
03:44.53*** join/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca)
03:45.28*** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net)
03:45.48prg3I'm having a problem with asterisk-gui/http, it's giving me 403 forbidden for files that are there.. and I can't seem to get any sort of debugging output from it.  Ideas?
03:46.29mchouprg3: did you authenticate?
03:46.54prg3mchou: in the browser? I'm not given any options to do so.. is there a .htaccess that I'm missing or something?
03:47.02mchousigh
03:47.33*** join/#asterisk Levonk (n=lk@75.62.129.25)
03:47.51mchouprg3: you think asterisk will let _anyone_ mess with it over the web?
03:48.25prg3mchou: I'd expect it to prompt me for a username and password, yes.  I was expecting to have the first page I see being the authentication page.
03:49.09mchouprg3: afaik it doesnt prompt.  you need to go to the approriate page to gog on
03:49.15mchoulog*
03:50.31prg3according to this page: http://www.asteriskguru.com/tutorials/asterisk_gui.html I've done all of that, yet it still will not prompt for authentication info..
03:50.39prg3is there a debugging flag for http somewhere?
03:51.09Qwellwhat URL are you trying to hit?
03:51.38prg3asterisk/static/cfgbasic.html although all of them do the same thing, and cfgbasic.html is just a meta refresh to index.html
03:51.42*** part/#asterisk dwayne (i=dwayne@76.29.245.9)
03:53.14QwellT3=45mbit?
03:53.37Qwelloh, I guess it is
03:55.38prg3Is there something better then asterisk-gui?  I'm mostly looking for a reporting, and listening in on recorded (or live) conversations.. call center managers want to keep an eye on the call center type thing
03:56.27mchouprg3: most of us here dont support GUIs
03:56.59mchouprg3: and better is a rather subjective term
03:57.41[TK]D-Fenderprg3: Asterisk-GUI was not made for any of that
03:58.29prg3mchou: Yeah, I know :)
03:58.54prg3I'd prefer it.. but the managers won't be able to do anything CLI based..
04:00.25prg3Oh, maybe better question, is there a decent (pick your own definition of decent) web frontend for CDR data? (mysql)
04:00.26*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
04:00.31[TK]D-Fenderprg3: If you're at all competent at programming it'd be easy enough to make a web interface for them for this.
04:00.53*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
04:01.08[TK]D-Fenderprg3: For reporting there are already 3rd party web-based solutions available.  Spying would be a small custom job
04:01.17heedlyhaving people write programs for you is pretty smart though.
04:01.18prg3I'm not.. but that's where I might need to go.
04:01.27mostyis there an easy way to bridge two existing channels in asterisk 1.4?
04:01.32prg3Spying we can leave out, or they can drive that via a phone
04:01.55[TK]D-Fenderprg3: then go hit the WIKI for a list of these other reporting tools.
04:01.59prg3[TK]D-Fender: Any suggestions for the first one I should look at?
04:02.11[TK]D-Fenderprg3: Asterisk-stat, etc.
04:02.20[TK]D-FenderAreski...
04:03.43prg3[TK]D-Fender: Thanks
04:04.09*** join/#asterisk Ardi (n=NiMiTz_C@cpe-72-227-190-157.nyc.res.rr.com)
04:04.48*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
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04:20.48a1faare there any tools that can stress test a number based on SIP?
04:23.15voxterhas anyone implemented multicast paging in asterisk yet? i saw a couple posts on the list, but nothing definitive
04:23.50jameswf-homea1fa: sipp
04:24.32a1fasipp can dial a did number?
04:24.48Qwellbrownie points for anybody who can tell me the longitude of the north pole
04:25.35a1fai need to join a conference bridge and basically flood the shit out of it with number of concurent calls from my * box
04:25.56Qwell(tip, latitude is 90)
04:26.43jameswf-home0?
04:27.05Qwellyou lose
04:27.41jameswf-homeeh well
04:30.02ArdiQwell: which one geographic or magnetic north pole
04:30.25Qwelltrue north
04:30.45Qwell(geographic)
04:30.46jameswf-homeLongitude is meaningless at the north pole (and the south pole too). It has all longitudes at the same time.
04:30.55jameswf-homegoogle foo hah
04:31.02Qwell:D
04:31.26jameswf-homesadly found on kde.org wtf ever that has to do with anything
04:31.27*** join/#asterisk techman97 (n=myweiner@97-91-103-181.dhcp.roch.mn.charter.com)
04:32.20jameswf-homeJesse ventura looks like crap
04:32.25techman97evening all - my mind is blanking, hopefully you can help me.  I'm setting up a Sangoma card (US PRI) and the line I put in the zapata.conf file that allows me to send my station ANI is escaping me...help?
04:32.37techman97jameswf-home:  yes he does.  I'm from MN and he never looked *good*
04:32.38techman97hahahaha
04:33.00Qwelltechman97: ...and yet you elected him
04:33.01Qwell:p
04:33.37techman97yeah...well....It's a lot like South Park's episode of choosing between a douche and a turd sandwich..
04:33.52techman97the three people that were running - Ventura was the lesser of the three evils.
04:34.04techman97and he didn't do crap - good or bad - while in office, so it was a net wash.
04:34.19jeevQwell
04:34.26Qwelljeev
04:34.39jameswf-homehttp://boozhy.typepad.com/my_weblog/images/jesse_ventura.jpg <<< i would put change in his coffee
04:34.43techman97I think I found the command - "usecallingpres=yes"
04:34.52jeevhi.
04:35.43jameswf-homeSo is obama the douche and mccain the turd sandwitch
04:35.43techman97he made an appearance at the Ron Paul event during the RNC....if he would just shave and style his hair, he wouldn't look like a cracked out street pimp
04:35.56techman97jameswf-home:  it's kinda turning out that way
04:36.07techman97you tell me the last election where that WASN'T the choice...LOL
04:36.07jeevslaps jameswf-home, obama aint no douche! he's gonna save us f00|
04:36.15*** part/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca)
04:36.16jameswf-homeI am voting for the pig in lipstick
04:36.19techman97hahaahahah
04:36.21jeevwow
04:36.22jeevreally ?
04:36.40jameswf-homeVPILF bayyyybeeee
04:36.41jeevjameswf-home must be one of those who installed asterisk from a package!
04:36.55Qwelljeev: I hear he uses trixbox
04:37.15jeevwow
04:37.19jameswf-homeuses ubuntu-server (shuddup)
04:37.25jeevubuntu
04:37.27jeevEVEN worse
04:37.32jeevreal men use slackware
04:37.44jeevrepublicans use ubuntu
04:38.07jameswf-homehttp://dontcallmyboss.blogspot.com/2008/09/my-pet-project.html << what i use :)
04:38.33jeevhttp://www.screamingpoints.com/archives/idiot-genius.jpg
04:38.34jeevhahahahahah
04:39.18jeevyour pet project blows
04:39.38techman97ouch jeev, don't hurt the boy's feelings or nuttin'
04:39.44techman97hehehe
04:39.50jameswf-homeI am not a fush ban i mean bush fan but im not sure obama is the cure no matter how well he parts the waters and turns water to wine
04:40.03techman97but I love wine man...
04:40.32jeevhttp://tbn0.google.com/images?q=tbn:PHVlzf-bH_e41M:http://willful-ignorance.com/uploaded_images/pro-war-735076.jpg
04:40.32jameswf-homeI am more of a 151 and coke guy
04:40.32jeevhahahah
04:40.36techman97the biggest problem is that they're both the same ol' politics....big talk to get in, do nothing but status quo when in office.
04:40.55jeevjameswf-home, doesn't matter, obama talks change, mcbush talks the same, perm tax cuts, war in iraq.. this and that, then picks that hot ass milf and then says it's time for change.
04:41.06jameswf-homeMoran is minisotan for moron also used in wiscansan
04:41.22jeevhence why i dont consider minisotan's or whatever people.
04:41.22jeevhahah
04:41.32techman97oh SNAP jeev
04:41.53techman97'dem fighin' werds ya know....don't make me get out my waders and show ya' who's the best on the lake
04:42.01techman97yea fer s00re
04:42.02jameswf-homeI want a Ron Paul Ross Perot ticket
04:42.09jeevok ok o
04:42.14jeevmy 2 other colo'd servers are online
04:42.17jeevwhat the hell should i use them for
04:42.22techman97pr0n
04:42.31techman97what else is the intertubes good for?
04:42.31jameswf-homeipv6 porn
04:42.56techman97setup a pr0n site and then advertise on CraigsList for developers to help you out for free access
04:43.20jeevlol
04:43.43jameswf-homewas contacted to set up a 900 line
04:43.48jeevlol
04:44.00jeevwho pimps out 900 lines anyway
04:44.12jameswf-homeumm 800 lines
04:44.33jameswf-homethis phone seks powered by Asterisk
04:45.07jameswf-homeoh crap memory leak crash
04:45.28techman97that's what you get for hosting fone seks
04:45.33techman97dirty memory
04:45.38techman97and leaky intertubes
04:46.15jeevjameswf-home, your e-penis is too small in this channel
04:46.32jameswf-hometigerdirect has 1TB drives for 149... I hink i shal set up a 1 TB raid
04:46.56jameswf-homeI read somewhere that someone had voipsex with kerry's wife
04:47.30jeevlol
04:48.35jameswf-home167 for "voipsex". (0.25 seconds
04:48.59*** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net)
04:49.34*** join/#asterisk SpaethCo (n=SpaethCo@cache.spaethco.com)
04:50.33drmessanoSIPSEX.com
04:51.20drmessanoI w_n_ t_ _??__ _ou al_ _ve_ oh _?_
04:51.44trnzmetacan I buy an e?
04:51.53trnzmetaoops already there
04:51.59techman97I won't take your clothes off while you're drunk?
04:52.22drmessanoThat's why phone sex ops don't use SIP
04:52.45drmessanoooohhh
04:52.55drmessanoI wonder if IAXME.com would take off
04:52.56techman97ooooohhhh
04:53.15a1falol
04:54.20*** join/#asterisk wiscados (n=mint@81.25.184.155)
04:58.16techman97nite all
04:58.36*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
05:01.25a1faah midnight
05:01.31jameswf-homeThe price of gold canyon candles are going up
05:02.18jeevwtf is that
05:03.06jameswf-homecandles sold in house parties... the local sheriff Raided their warehouse and arrested 65  illegals
05:03.26jameswf-homeneed a job?
05:03.27jameswf-home:))
05:03.56trnzmetaweeet weeet
05:06.01jeevhuh
05:06.03jeevi'm lost
05:06.10jeevwe need to arrest illegals and mcbush
05:11.16jameswf-homelook you can put lipstick on a pig http://www.mondaymorningmemo.com/mmm_images/June6_2005MMM.jpg oh wait its still a pig
05:14.17jeevhahah
05:15.39*** join/#asterisk Bananaskin (n=Banana@user-5af01b01.wfd96.dsl.pol.co.uk)
05:20.24jameswf-homeHOLY CRAP http://www.bunnyranch.com/virgin/
05:35.17*** join/#asterisk costal79 (n=ivan@203.84.225.2)
05:35.33costal79hi people
05:35.41costal79I have this group
05:35.57costal79GROUPSUPPORT=SIP/620&SIP/624&SIP/621&SIP/622&SIP/623
05:36.26costal79and also I'm doing something like this
05:36.29costal79exten => o,1,Dial(${GROUPSUPPORT},20,tTwWr)
05:36.34costal79so my question here is
05:36.50costal79the order of the group define which extension will get the call first ?
05:40.31*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-893bd2a09f2fc996)
05:41.49rue_mohrno
05:41.54rue_mohrthey all ring togethor
05:42.25rue_mohrcostal79,
05:43.19rue_mohrthe only time you might see a difference is if they are all set to autoanswer
05:43.42rue_mohrbut network topology would effect is just as much
05:43.49rue_mohrI should think
05:45.31rue_mohrI also think your doing it wrong for a set of support phones, but I dont know how to do any of the rotation type stuff I think you should be doing
05:46.28rue_mohrI think it involves some lists with for loops and if busys
05:46.43costal79mmm
05:46.59costal79this is an old asterisk system and I'm trying to do some changes
05:47.10costal79they have only this group definition
05:47.18costal79and it works like a queue
05:47.24*** join/#asterisk tengulre (n=tengulre@125.71.208.16)
05:48.18*** join/#asterisk jsin (n=jsin@gentoo/developer/jsin)
05:48.59jsinCan someone please make a soft phone recommendation?
05:49.23mchoujsin: like don't use one
05:49.31jsin:)
05:49.41mchoujsin: not joking, really
05:49.56jsinhow come?
05:50.22mchoupoor audio & inconvienence
05:50.50jsinI just want to play while I wait on my hard phone to come in the mail
05:51.05rue_mohrjsin, what os?
05:51.07mchouthat's what they all say
05:51.17jsinwindows
05:51.21rue_mohrok
05:51.26mchouhaha.
05:51.26rue_mohrwhast was it called...
05:51.32mchouMS Messenger :)
05:51.33jsinI have a GXP2000 on the way
05:51.38rue_mohrjusta sec
05:51.44mchouxlite?
05:51.55*** join/#asterisk Rico29 (n=Rico@static-120-146.blueline.mg)
05:51.56rue_mohrno
05:53.43rue_mohrexpresstalk
05:54.00rue_mohrhttp://www.nch.com.au/components/talksetup.exe
05:54.20rue_mohrto configure it..
05:54.20rue_mohrclick options...
05:54.20rue_mohryou need to get to "sip setup"
05:54.20rue_mohrhttp://i80.photobucket.com/albums/j192/tpad-com/ExpressTalkSIPProvider.jpg
05:54.20rue_mohrclick the "I already have an account" (then next)
05:54.32rue_mohrthe rest is specific to your thing
05:55.12rue_mohrmy experiance is that its 'descent'
05:55.18mchoulol
05:55.27mchoudescent into hell is right!
05:55.47rue_mohr2 hours past my bedtime
05:55.48rue_mohrgnight
05:56.09rue_mohrtwinkle under linux
05:59.24jsinthanks rue_mohr, good night
05:59.40jsinthis better than x-line in your experience?
06:04.20drmessanoGotta love any software that offers you to download a whole effing PBX when registering
06:04.43drmessanoX-Lite is 10x better than ExpressTalk.. that thing gave me daymares
06:08.30*** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net)
06:10.37jsinexpresstalk installed and setup quickly... I can make outbound calls, now to figure out how to get incoming calls :)
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06:18.50trnzmetacan you have skins to x-lite?
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07:01.20r0bisThe funniest thing I discovered about my problem when inbound calls do not get hung up properly (i.e. remote side remains with open line after asterisk issues hangup()). It appears telco problem, since it is exactly reproducible when just an analog phone and nothing else is connected to the line. lol and I think so far the condition has not been detected because its so natural for the other side to hangup after speaking
07:01.20r0bis<PROTECTED>
07:04.23*** join/#asterisk johnakabean (n=none@pool-72-82-106-158.nrflva.east.verizon.net)
07:04.37johnakabeanhey room i need some help with 1.4 (latest stable)
07:09.59kaldemar~ask
07:10.00jbot[ask] Questions 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.
07:11.18kaldemareither jbot scared the living poo out of him or he didn't really need help.
07:11.23MooingLemuruh huh
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07:31.02*** join/#asterisk [gnubie] (n=gnubie@203.177.180.52)
07:31.06[gnubie]waves to all..
07:33.41tzafrir_laptophi [gnubie]
07:35.30[gnubie]i am running 1.4.21.2 here.. my ip phone is connected remotely.. if i call another extension phone located inside the lan where the asterisk is located, the callee hears me.. but if i call a telephone connected to a local pstn (pots) via my fxo, the callee cannot hear me at all.. no problem with all incoming calls regardless if it comes from one of the extension phones or from the pstn (pots)
07:35.40[gnubie]hello tzafrir_laptop
07:37.12tzafrir_laptopif you call voicemail of playback? (not direct sip<->sip)?
07:37.23[gnubie]in short, inbound calls to my remote phone is ok but outbound calls from my remote phone, the callee cannot hear a sound
07:38.11[gnubie]tzafrir_laptop: no problem if the remote phone calls to another extension number or vice versa
07:39.02[gnubie]calling the echo test is good
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07:39.55[gnubie]remote_phone => asterisk => pstn(pots) => telephone
07:39.58[gnubie]that's the problem
07:40.11[gnubie]but if the other way around, there's no problem at all..
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07:43.00[gnubie]i'm using sip here by the way
07:43.53[gnubie]outbound call from a remote phone produces a one way audio where the callee cannot hear the caller
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07:44.49[gnubie]any idea?
07:45.20mchouyeah.  it's your firewall
07:49.00[gnubie]mchou: firewall.. where? on the asterisk box or on where the remote phone is connected?
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07:51.43mchouwhereever there is a firewall needs to be checked
07:51.51[gnubie]mchou: my firewall rules => http://paste.debian.net/16965/
07:51.54costal79exit
07:51.58costal79\q
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07:52.47mchouI'm not going to debug your issue for you, sorry
07:53.24mchouone way audio is almost always a firewall traversal issue
07:54.12[gnubie]i didn't changed anything except that i upgraded my asterisk installation from 1.4.17 to 1.4.21.2
07:54.30[gnubie]the configurations are all the same
07:54.51[gnubie]now, i have this issue already..
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08:01.21drmessano~sipnat
08:01.22jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
08:01.28drmessanoIt's not a firewall problem
08:01.45drmessanoSounds like you need to fix your localnet externhost/externip settings
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08:10.34[gnubie]drmessano: you mean on my sip.conf ?
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08:13.47[gnubie]drmessano: as far as i understand, it's good.. and it worked before with the same configuration as i have for asterisk-1.4.17..
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08:15.11dandreHello,
08:15.24dominic1if I program a forward in my telephone I get a 302 move temporarily from the device. Is there any possibility to read that statement in the dialplan?
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08:15.32dandreHow can I put a call on hold using the manager interface?
08:15.53dandreI haven't found an action: hold
08:17.51[gnubie]besides, my asterisk box is facing the internet already.. eth0 is using a public ip and eth1 is the one that faces to my lan.. my remote phone is located somewhere the internet outside the lan
08:18.09[gnubie]so when i try to place a call, the route is:
08:18.43[gnubie]remote_phone => internet => asterisk => pstn(pots) => telephone/callee
08:19.14[gnubie]sorry, it's not like that.. here is the right one:
08:20.01[gnubie]remote_phone => lan => modem/router => internet => asterisk => pstn(pots) => telephone/callee
08:20.34[gnubie]if it's the other way around, it's ok.. like this:
08:21.01[gnubie]remote_phone <= lan <= modem/router <= internet <= asterisk <= pstn(pots) <= telephone/callee
08:21.24[gnubie]s/callee/caller
08:23.13dominic1hold ist not possible
08:23.19dominic1you can park and unpark somebody
08:23.24dominic1but there is no command to hold sb
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08:39.35dominic1sorry, back again
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08:59.00Rico29hi
08:59.15Rico29i'm trying to configure zaptel through /etc/zaptel.conf
08:59.46Rico29i get this error msg  : ZT_CHANCONFIG failed on channel 1: No such device or address (6)
09:00.18Rico29http://debian.pastebin.com/m7869a72b
09:01.14tzafrir_laptopRico29, what is the output of: zaptel_hardware
09:01.27tzafrir_laptopwhat device(s) do you actually have?
09:01.42Rico29oxygen:/etc# zaptel_hardware
09:01.45Rico29nothing
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09:01.48Rico29tdm411B
09:02.20Rico291FXS / 1FXO
09:03.09tzafrir_laptopdo you see that card on lspci ?
09:03.22tzafrir_laptopthat card? or a TigerJet?
09:03.41tzafrir_laptopWhat kernel? What linux distribution is it?
09:03.46Rico29mmmh
09:04.02Rico29my boss told me that pluging the card to the computer's alim was not needed...
09:04.09Rico29i think he was wrong
09:04.13Rico29debian 4
09:04.14JTalim?
09:04.20Rico292.6.18-6-686
09:04.25Rico29power
09:04.35JTalim == power, huh?
09:04.55Rico29yes there is a slot to plug a power cable on the card
09:05.00JTyou do need to hook it up if you want the fxs port to work
09:05.06Rico29like these on HD
09:05.10JTsince when has that been called "alim"?
09:05.16Rico29sorry for my bad english...
09:05.26Rico29french word sorry
09:05.28Rico29:)
09:05.30JTok
09:05.51JTit's a molex connector
09:06.20Rico29yes
09:06.23Rico29that's it
09:07.55tzafrir_laptopRico29, as a hint: dmesg | tail
09:08.13tzafrir_laptopI think you should see there an error message telling you just that :-)
09:08.51tzafrir_laptopZaptel is kernel, so looking at dmesg (or in Debian: better: /var/log/kern.log) can be useful
09:08.59Rico29ok
09:09.00Rico29thanks
09:09.18Rico29i'll see
09:09.21Rico29i'm currently rebooting
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09:31.11Rienzillahey all
09:32.06RienzillaI was wondering whether a linksys voip-phone adapter should be able to carry fax and modem signals?
09:32.25mgdmfaxes and especially modems don't work very well across VoIP
09:32.41Rienzillayeah I suspected that
09:33.16Rienzillawon't it work at all? or just with low bitrates/unreliably?
09:33.42mgdmModems I suspect won't work at all, faxes can be made to work, I believe, with a lot fo tweaking
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09:41.51gr0mitRienzilla, it is all rather flakey
09:41.55Rienzillaok. Thanks.
09:42.24Rienzillayeah the point is we switched to voip, but we forgot to take into account some old modem connection to the building's heating system :)
09:43.02Rienzillaso ill probably have to keep some pots line into the building to operate that modem
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09:43.58tzafrir_laptopRienzilla, will it be OK for the modems to work on lower baud rates?
09:44.09Rienzilla9600 8n1 is probably what they're talking
09:44.29Rienzillait's connected to the serial port of some management system
09:44.35tzafrir_laptopso this is basically like a fax
09:44.38RienzillaI bet it's 9600 baud
09:44.41Rienzillayeah
09:44.48RienzillaI don't expect 56k6 to work across voip ;)
09:45.18tzafrir_laptopthat's marginal for an ATA. You need a good network
09:45.52RienzillaI'm pretty well connected (100mbit fiber to the datacenter where the signal goes onto isdn)
09:46.14WimpManWhat about Serial-Ethernet thingies?
09:46.22tzafrir_laptopThere are some ATAs that support T.38 (fax over IP). But T.140(?) (modem over IP) is hardly supported
09:46.37Rienzillahmmkay
09:46.54RienzillaIll keep the analog line for the next few years then, until this is better supported
09:46.59Rienzillathanks for the advice
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11:44.35bugotafHi
11:46.27bugotafI'm using an digium AA50 and i have some trouble with voicemail, i' have define some user with voice mail with password equal to phone number, when i call voicemail and type phone number then pincode the autorisation is always refused
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11:46.44Specialist1hello everyone
11:47.41bugotafi have another problem with voicemail when i call internaly my voicemail sound is in french but when outdoor call fall in voicemail voice are english
11:48.27Specialist1<PROTECTED>
11:50.17gr0mithands bugotaf a dictionary for his external callers ;-)
11:50.49gr0mitbugotaf, i think you need to set the language for the SIP or IAX peer that handles your incoming calls
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11:51.49heedlySpecialist1: it always doesn't ring?
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11:52.01heedlysounds like a provisioning thing to me...
11:52.19Specialist1hi heedly, actually it rings at times but not always
11:52.51Specialist1first I saw  "auto congesting " message on the console..
11:53.28Specialist1then I changed 'autoqualify' to 'no' ! to overcome auto congesting.
11:54.35Specialist1Before I set 'autoqualify' to 'no' It showed two times ringing tone and after that it said "the number is not avaialble"
11:55.12Specialist1Now after I change it to 'yes' It shows ringing on the calling side but acutally it does not ring on the called side!!
11:57.09heedlymaybe a nat problem?
11:58.36Specialist1hmm . Do you think so.  Is there any way that I can communicate from the server with the client  . Like ping or something?
11:59.12Specialist1But ping can only tell if the client machien is responding!
12:04.57heedlythere are a number of things you can do to test.
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12:08.10Phob[0S]hi all
12:08.13Rienzillahmm
12:08.27Rienzillais there a straightforward solution for cancelling echo in phonecalls?
12:09.14gr0mityes - use isdn
12:09.27gr0mitor anything with 4-wire connections the whole way
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12:09.30Rienzillawell
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12:09.38RienzillaI don't use any analogue lines
12:09.41gr0mitok
12:09.50Rienzillaand my voip provider does use isdn bundles afaik
12:09.54gr0mitok
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12:10.16gr0mitso if you are using a voip provider then it is them who needs to cancel the echo
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12:11.08Rienzillaoh
12:11.22Phob[0S]hi can i ask support for a strange issue?
12:11.29Specialist1heedly sorry for the delayed reply... How can i test if its due to nat
12:11.31gr0mitthe echo points will be any 2-4 wire conversion point
12:11.54gr0mitor a speakerphone or handset with accoustic path from Rx to Tx
12:12.04WimpManPhob[0S]: We are unaware of your abilities to do so.
12:12.20Phob[0S]???
12:12.39WimpMan~ask
12:12.40jbotrumour has it, ask is Questions 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.
12:12.40Phob[0S]i have a strange issue with rtp stream
12:12.46Phob[0S]can we talk in pvt?
12:13.19Great_Anta_Bakawhat does silence suppression do?
12:13.41gr0mitmesses up Asterisk, Great_Anta_Baka
12:13.44[TK]D-FenderGreat_Anta_Baka: Stops sending RTP at all when the endpoint decides its too weak to consider conversation
12:13.58Great_Anta_Bakai see
12:14.06Great_Anta_Bakaso i should disable it?
12:14.16gr0mitalways disables it
12:14.25[TK]D-FenderGreat_Anta_Baka: * doesn't support it, so yes.
12:14.25Great_Anta_Bakammm kk
12:14.50kalel008how do i stop this from playing http://pastebin.com/m4009fb94
12:15.20[TK]D-Fenderkalel008: "core show application voicemail" <-
12:16.14Odd_BlokeDoes this stack trace look familiar to anyone: http://paste.pocoo.org/show/85007/ ?
12:16.28Phob[0S]Ok, i've an asterisk implementation with a Cisco voicegateway. Cisco Voice Gateway had an E1 PRI Isdn. RTP streams for internal calls are setup P2P between phones, but external calls (From Internal Phone to public Network through Cisco) RTP Stream it setup through Asterisk Server
12:16.47Odd_BlokeI'm about to start debugging an issue with a channel we're writing, but want to check we're not hitting a known problem in Asterisk itself.
12:17.02Phob[0S]anyone can help me on how to resolve this issue?
12:18.12Phob[0S]no one?
12:19.14Rienzillagr0mit: thanks so far
12:19.35gr0mitso Rienzilla under what conditions do you get echo?
12:19.42RienzillaIm not sure
12:19.49Rienzillasome calls experience echo, some don't
12:20.08gr0mithave you raised this with your carrier?
12:20.17RienzillaI just have :)
12:20.25Phob[0S]how can i avoid rtp proxying?
12:20.27gr0mitmy carrier had prbs on calls to French landlines
12:20.27RienzillaI was just wondering whether this could be my own ault
12:20.43gr0mitso i juest send calls via a different carrier and they are fine
12:21.23gr0mitare these calls within .nl or different countries?
12:21.50Rienzillaalmost exclusively .nl
12:22.01Rienzillait's customers of a dental practice, so hardly any international calls
12:22.10gr0mitah ok
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12:22.34gr0mitcan you try another carrier?
12:22.55Rienzillanot for incoming calls
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12:23.10RienzillaI could set something up for outgoing calls, maybe
12:23.18gr0mityup
12:23.36Rienzillabut I'll first wait to see what th ecarrier says
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12:23.53RienzillaI know they just received new incoming lines from another phone company
12:24.03Rienzillaso maybe it's a new problem which they dont know yet
12:24.06gr0mityup
12:24.12gr0mitwhich carrier is this?
12:24.16Rienzillaspeakup.nl
12:24.41gr0mitjust looking
12:27.29gr0mitnot the cheapest, Rienzilla
12:27.55gr0mitso not a bargain basement company!
12:28.12RienzillaI know the company owners personally
12:28.36Rienzillabut they have a quite ok service
12:28.38gr0mitok, well what I did was to run a tcpdump on my asterisk box
12:28.45gr0mitlog the traffic
12:28.52gr0mitand send it to my wholesaler
12:28.59Great_Anta_Bakais there a tool i can use to analyze the route to another pbx to check if placing a sip call over it is viable?
12:29.13gr0mitwhen they heard the echo they said ' oh yes, that is bad'
12:29.40RienzillaWell I'll have my users write down the phone numbers they have problems with
12:29.51gr0mityup - date time and destination
12:30.01gr0mitthen they can look through the logs
12:30.04Rienzillayep
12:30.07Rienzillak
12:30.17gr0mitI have had issues with calling DECT phones
12:30.34gr0mitcoz they introduce latency and echo
12:30.40jblackThey work fine for me.
12:30.57jblackIt's probably the particular model of phone you're using.
12:30.59gr0mitjust some dect phones on some lines
12:31.12gr0mitmost as you say are perfect
12:31.16RienzillaI have voip phones exclusively
12:31.31Rienzilla(and all the same models, too)
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12:32.12darkskiezhow can i find out what Pickup() is expecting..
12:32.15Rienzillaand the weird thing is, I hav used the same voip provider at home, with the same model phone, and not experienced any problem
12:32.25Rienzillaso it might just be their new outgoing lines
12:32.40darkskiezi'm trying with the text displayed under  location col in show channels
12:32.46gr0mitRienzilla, could be anything really
12:33.11jblackIt could be anything. Perhaps a bad ground in one phone. Maybe a little space alien has set up camp in one phone.
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12:33.42gr0mitRienzilla, best get them to look into it
12:34.43gr0mitRienzilla, if you really don;t get any joy let me know - might be able to help.
12:37.51Rienzillaok, thanks so far
12:39.10gr0mitgraag gedaan.
12:40.52[TK]D-Fender[08:28]<Great_Anta_Baka>is there a tool i can use to analyze the route to another pbx to check if placing a sip call over it is viable? <- Dial
12:42.12Phob[0S]anyone can help my on my rtp trouble?
12:42.28Phob[0S]i need to know if i can solve my problem
12:43.06Phob[0S]??
12:43.56Phob[0S]c'mon guys i only would to avoid rtp stream pass through my asterisk PBX
12:44.21Phob[0S]this happens only in particular cases
12:44.52Rico29with canreinvite=yes ?
12:45.07Phob[0S]hi rico
12:45.10Phob[0S]thnx for your reply
12:45.25Rico29you're welcome
12:45.29Phob[0S]i've already set it to yes
12:45.39Phob[0S]i can esply to you from beginning
12:45.52Rico29sorry i've a lot of work
12:45.56Phob[0S]ok
12:45.59Phob[0S]tnx
12:46.00Rico29i'm at the office
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12:46.06Phob[0S]np
12:46.51lmadsenI can't seem to remember where this is enabled, but there is an option I thought that would display 'hint' information to the manager interface?
12:46.58lmadsenI can't seem to find it in sip.conf though
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12:47.28WimpManlmadsen: That's 1.6 only, isn't it?
12:47.43lmadsenWimpMan: that still doesn't answer my question :)
12:47.49lmadsenuses all versions of asterisk
12:47.49Phob[0S]i've resolved my issue
12:48.12Phob[0S]this happens cause i passed 'tr' option in Dial application in my dialout macros
12:48.19Phob[0S]tnx to all for the supports
12:48.25Phob[0S]Google it's your friend
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12:58.10Great_Anta_Bakahow can i calculate how big my jitterbuffer should be?
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13:01.46[TK]D-Fenderlmadsen: "core show hints"
13:01.56lmadsenthanks
13:01.58*** part/#asterisk lowlevel (n=Stuart@lowlevel.ca)
13:02.10*** join/#asterisk dominic1 (n=dob@213.221.82.242)
13:02.23WimpManDidn't I read manager?
13:02.46lmadsenWimpMan: I talked to a developer, and what I just requested does not exist in asterisk, but I'm gonna attempt to fix that...
13:02.52[TK]D-FenderWimpMan: Yes, but you didn't use your imagination :)
13:02.54dominic1can anybody tell me, why I am not able to set the callerid of a number from my isdn pstn and not always show the network number?
13:03.05[TK]D-Fenderlmadsen: *I* do it this way
13:03.25lmadsenI'm not gonna parse through the output of the CLI if I can help it
13:03.28lmadsenhacks
13:03.33lmadsencoughs
13:03.41[TK]D-Fenderlmadsen: And have for years.  my call center Polycom Idle app mointors 2 queues, 2 VM boxes, and 4 presence entries
13:03.53[TK]D-Fenderlmadsen: Parse it!
13:04.22[TK]D-Fenderlmadsen: If you don't like brute force, you clearly haven't found a big enough stick!
13:04.24*** join/#asterisk postel (n=jp@wikimedia/Postel)
13:04.41[TK]D-Fenderhas an even BIGGER stick, and I use ti too! *WHAM*
13:05.42*** join/#asterisk Levonk (n=lk@adsl-75-62-142-149.dsl.lsan03.sbcglobal.net)
13:05.53Alton2Postel, a famous name in computing.
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13:12.44dominic1I want to transfer the ddi of my isdn pri. The main msn is 15450. I set the callerid of my outgoing call to 1545123 but the called party only sees 15450. Is there any option, which I will have to adjust?
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13:14.40nacerhi
13:14.50naceri need a information about digium card :)
13:14.55WimpMandominic1: Check the type of number.
13:15.04lmadsendominic1: sounds like no, as the provider is restricting your number
13:15.08naceri need to know the exact size of the eax 2400 card
13:15.17lmadsen42!
13:15.24nacer:)
13:15.27Great_Anta_Bakacheck out the specifications page
13:15.30Great_Anta_Bakaon the digium site
13:15.31lmadsennacer: your best bet is to just call digium sales -- they'll tell you
13:15.33WimpMandominic1: Might be you have to set the area code as well, or set the correct TON.
13:15.34lmadsenor that :)
13:15.52nacerthereis no informtion about this in the documentation
13:16.12Great_Anta_Bakanormally the specifications pag eincludes all of that crap
13:16.20naceryeahh normally
13:16.22Great_Anta_Bakaotherwise call digium
13:16.33nacerdam my english speaking is bad
13:16.56lmadsenyour english seemed fine -- just the spelling, but even native english speakers seem to have that problem :)
13:17.48lmadsenand the only word I actually see wrong is s/dam/damn :)
13:18.18dominic1WimpMan: TON?
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13:22.59WimpMantype of number
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13:29.50WimpMandominic1: -> msg
13:30.14dominic1ah okay, I understand
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13:39.27sapereHey guys, anyone familiar with the Asterisk service provider, IronVoice.net?
13:39.48Rienzillahmm is it possible with an asterisk queue to put "important" clients in the front of the queue?
13:40.04saperemake that IronVoice.com
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13:42.31lmadsenRienzilla: that is a very good question. I had someone ask me that recently, let me see if I can find any configuration option or variable that allows that -- but I have a feeling it may not be possible
13:42.42lmadsenunless there is a weighting system I can find that I don't know about
13:43.07RienzillaI cant find it in the Queue command documentation
13:43.30lmadsenya, I think it'd have to be a channel variable or something since you have to apply it to the channel before it enters the queue
13:44.56*** join/#asterisk moy (n=moy@nat/ibm/x-94b8c591a6d565b4)
13:45.46lmadsenRienzilla: ${QUEUE_PRIO}           Queue priority
13:45.56lmadsenRienzilla: looking for more docs on it -- but that might be the ticket
13:45.59[TK]D-Fenderlol.... pre 1.0 stuff only discovered now!
13:46.07*** join/#asterisk jtodd (i=pbgpw24b@ns2.loligo.com)
13:46.26lmadsenoh my god.... JTODD!
13:47.16nacerlmadsen: the hotline give the information
13:47.17Rienzillaahh
13:47.42lmadsennacer: the rooster flies at midnight
13:48.01nacerlmadsen: i dont understand :)
13:48.09lmadsennacer: me either :)
13:48.15[TK]D-Fendergives lmadsen the secret handshake
13:48.30lmadsenI don't want to know your name!
13:48.31*** join/#asterisk l2trace99 (n=jr@75.112.133.235)
13:48.39Rienzillahttp://lists.digium.com/pipermail/asterisk-users/2006-May/150398.html
13:48.39[TK]D-Fenderlmadsen: I jsut want...
13:48.41Rienzillalooks like it
13:48.42lmadsen! ! !
13:48.46[TK]D-Fenderlmadsen: zomg!
13:48.54lmadsenlawlz
13:49.39lmadsenRienzilla: I will have to test. It may be possible that variable doesn't work and needs to be fixed. If that is the case, then I will file a bug -- unless you get to it before me :)
13:50.09*** join/#asterisk korihor (n=korihor@190.78.32.60)
13:50.17Rienzillai'll try now
13:51.47Rienzillahmm
13:51.56*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:52.01Rienzillamy queue is not large enough to see where they are queued at the moment :)
13:52.47jeevaaaaaaaaaa
13:52.56Rienzillahow do you enable debug output for queue?
13:56.20jeevleef
13:58.10l2trace99leaf | life | loof whats the difference
13:58.11l2trace99?
13:58.42jeevdunno
13:58.43jeevi'm tired
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14:00.11*** join/#asterisk Blackvel (n=blackvel@dslb-088-065-118-239.pools.arcor-ip.net)
14:01.05Blackvelhi all. found GotoIf cmd. I am in the need to manually change to callerid for outgoing call depending on various callerid strings. Is there an simple If/else statement too? where could I look at?
14:02.50KobazBlackvel: nope, you're stuck with BASIC-style syntax unless you go with ael
14:03.13*** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it)
14:03.20*** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it)
14:03.49l2trace99or use agi
14:04.28Kobazi like the freeswitch method better for language integration
14:05.04[TK]D-FenderBlackvel: How many different values?  What are the calls coming in from?
14:05.41Blackvelprobably 4 (so far I can think for)
14:05.42Blackvelhttp://radio-downtown.de/voip-info/wiki/view/Asterisk+func+if.html
14:06.04Blackvelhmm no if/else :)
14:06.50[TK]D-FenderBlackvel: go read IF's **instructions**
14:07.00[TK]D-FenderBlackvel: You've clearly missed the big print
14:09.32Blackvel[TK]D-Fender: What are the calls coming in from? how do you mean that? the calls come in as sip from pstn. but the the callerid can have at least 4 different numbers (when I set clip no screening on my mobile phone I have to completely change the incoming callerid otherwise clip is not working with the p
14:09.40Blackvelwith the sip provider...
14:10.49[TK]D-FenderBlackvel: Ok, so in from a non "local phone", and then right out another channel.  Then 4 IF's are in order
14:12.54*** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk)
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14:15.22Blackveltoo bad func_logic.so is in 1.4, not 1.2 :) and asterisk-1.4.21.2 segfaults with patton bri gateway.
14:15.43heedlyjust copy iy over!
14:15.47heedlyand hope it works..
14:16.47*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
14:17.00Blackveli dont need func_logic.so in 1.2?
14:17.13[TK]D-Fendertotal waste.  4 little "IF"s
14:17.13Blackveli mean necessarily need func_logic.so?
14:17.20BlackvelI see
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14:18.09BlackvelI feel lost so many times. I always miss the entry point to find the stuff with * I need (like if)
14:19.12[TK]D-FenderBlackvel: "core show applications" , "core show application [appwithoutbraces]" , "core show functions" , "core show function [functionwithoutbraces]" <- all you need.
14:19.50[TK]D-FenderBlackvel: So before you ask "how do I XXXXX", go look at the list first.
14:20.22*** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net)
14:21.57Blackvelvery good tip with show functions. looks like I was missing that :(
14:25.34Rienzillahmm, anyone ever used a linksys pap2 ata thingy? My analogue device insists the line is off hook if I plug it into the ata
14:26.09*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
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14:31.39gr0mitRienzilla, yup.
14:31.49gr0miti have instaled one at a customer site
14:34.07*** join/#asterisk gabegundy (n=gabe@57.251.sfcn.org)
14:35.22[TK]D-FenderRienzilla: It'll do that if it isn't registered
14:35.29*** join/#asterisk seanmh (i=seanmh@216.31.101.77)
14:36.53gabegundyI have a client that wants CentOS/Asterisk/FreePBX and wants RPMs for everything.  I'd love to build it for him, but from what I understand, AsteriskNOW will provide all of that.  Does anyone know if that is true?
14:37.10Rienzilla[TK]D-Fender: the registration stat eis online though
14:37.16Rienzillaaccording to the web interface
14:37.29gabegundyThe next version of AsteriskNOW that is.
14:38.47[TK]D-Fendergabegundy: I've heard as much myself
14:38.57*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)
14:39.26gr0mitRienzilla, you have checked the pins are correct?
14:39.51gr0mitif you plug a phone in, do you get dialtone?
14:40.18gabegundyWhat are the chances someone in this channel can confirm that and save this guys some money (even if it means I get less billable hours)?
14:40.56Rienzillayes, a phone will give a dialtone
14:41.00Rienzillaand the wiring must be correct
14:41.14Rienzillait's just standard rj11 and rj45 cabling
14:43.59*** join/#asterisk spokra (n=spokra@host093-179-144.sea0.speakeasy.net)
14:44.49Rienzilla(if I plug a phone to the fax wall outlet which is connected to the ata I also get a dialtone)
14:45.22gr0mitok
14:45.46gr0mitI wonder if you can increase the line voltage in the pap2
14:46.18RienzillaI can increase the ring voltage
14:46.30*** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com)
14:47.10Rienzillaand I read something similar (same problem on a different device) where a solution was to increase a CPC value, but I don't know what that is :)
14:47.20*** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com)
14:48.07rwaitehi all. i am scanning my server and port 1720 is open ... i looked for a module i could disable to close this port but i dont know what could be using it?
14:48.21Rienzillarwaite: are you using a linux system?
14:48.53lmadsenBlackvel: you could use the IF() function too
14:49.08lmadsenmay be way out of date now though as he didn't read the entire scrollback
14:49.23Rienzillarwaite: (if so, run netstat -tapn  | grep LISTEN | grep 1720 to find the process which is listening on port 1720
14:49.24gr0mitCPC is calliing party clear.  should not affect line voltage detection
14:49.41Rienzillarwaite: or lsof -i -n | grep 1720
14:50.13rwaiteahah
14:50.15rwaitethanks
14:50.15[TK]D-Fenderlmadsen: ClueBat action has been missed...
14:50.22Rienzillaso probabnly my device is thinking the line is open because the voltage on the wires is different than it would expect in an on-hook situation?
14:50.30*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
14:51.00jameswf-homeso asterisknow is debian?
14:51.26rwaiteRienzilla: hmm. i get nothing for that, but nmap still lists it as open. maybe i'm missing something
14:51.49rwaiteit says "filtered"
14:51.52Rienzillaah :)
14:52.00Rienzillafiltered means a firewall is blocking that port
14:52.21Rienzilla(so a connect to the port does not yield "connection refused", but it yields nothing)
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14:53.25rwaitewhat would cause this?
14:53.34rwaite(no firewall on this machine, yet)
14:54.21[TK]D-Fenderjameswf-home: rPath currently
14:54.40jameswf-homerpath uses apt?
14:54.40Rienzillarwaite: some firewall between the machine you scanned from, and the machine you scanned
14:54.51[TK]D-Fenderjameswf-home: conary..
14:54.52rwaiteok
14:54.56*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-8734327f753ebbd9)
14:54.56*** mode/#asterisk [+o putnopvut] by ChanServ
14:55.46jameswf-homeah must be old instructions
14:57.29*** join/#asterisk weinerk (n=irc@82.80.129.212)
14:59.05weinerkHello, please help - why do I have a line repeating over and over in the log
14:59.18weinerk[Sep 11 02:12:47] VERBOSE[26681] logger.c:     -- <SIP/ZDt92ruS96-b7800468> Playing 'vm-extension' (language 'en')
15:01.28*** join/#asterisk ManxPower (n=manxpowe@90.sub-75-248-241.myvzw.com)
15:01.37*** join/#asterisk pjz (n=pj@zachs.place.org)
15:02.00pjzanyone have recommendations on firmware version to use with Polycom IP330s ?
15:02.51pjz3.0.3B (since it's the latest) ?
15:02.56Rienzillagr0mit: I don't think the pap2t can change the line voltage
15:03.17gr0mithmmm oh dear
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15:04.11x86anyone have any thoughts on the Linksys SPA921 phone?
15:05.54[TK]D-Fenderx86: FFS WHY!?! :p
15:06.56[TK]D-Fenderx86: Polycom IP 320 kicks its ass for the same price.
15:08.27gr0mit[TK]D-Fender, so should i get hold of one of these Polyjuice phones?
15:09.00[TK]D-Fendergr0mit: If you are looking for a good SIP phone, Polycom is it.
15:09.52gr0mitthey look better than the snom 300
15:11.05rwaitethe the asterisk .X releases fixes for severe bugs that couldnt wait until the next .XX release to fix?
15:11.11[TK]D-Fendergr0mit: Very little point to Snom in my books.  To pricey for normal/high models, too lacking in features on lower, too unstable.  Linksys is my #2 choice.
15:11.26[TK]D-Fenderrwaite: 4th position, yes
15:11.36ManxPowerThe Polycom 300s are pretty ugly, but the other models (320, etc) are quite nice.
15:11.43*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:11.54gr0mitwell i like the busy lamps on the snom phones.
15:11.58rwaitethat's what i thought. thanks.
15:12.00[TK]D-FenderManxPower: 30x = never buy > $50
15:12.04gr0mithave not seen this on an others
15:12.20ManxPower[TK]D-Fender: my customer has many many 301s
15:12.23gr0mitdo polycom do any thing like that?
15:12.25ManxPower..er.. 300s, that is
15:12.39[TK]D-Fendergr0mit: Thats 1 thing Snosm is good for on the middle models... but its a very specific segment that profits from the # it has based on model
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15:12.56[TK]D-FenderManxPower: I just got my first 2 IP320's here.  They look nice
15:13.00ManxPowergr0mit: Polycom allows you to set any line appearance as a BLF.  Polycom 60x has a side car available, 14 line appearances per sidecar, up to 3 side cars
15:13.20ManxPower[TK]D-Fender: my customer moved to the 320s for new phones
15:13.30[TK]D-FenderManxPower: cool
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15:16.58*** join/#asterisk wacky__ (n=root@nat/digium/x-ae82c3d51cf8a4f2)
15:17.42wacky__with Asterisk 1.6, when it run "core show translation", I get some 1-digit values, and some 5-digits values..
15:18.01wacky__is there a cache somewhere to clear ? anything to update to get all 4 to 5-digits micro-seconds numbers ?
15:18.59ManxPowerwacky__: show translation has some options to help
15:19.16ManxPowerI believe it's a "recalc" option
15:20.16wacky__oh wow..
15:20.24wacky__that's neat! thanks
15:21.01ManxPowerBTW, thanks to the person that sent me a paypal donation!
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15:23.07x86[TK]D-Fender: I've had nothing but problems with the IP330's
15:23.20x86[TK]D-Fender: 330's and 430's both have serious flaws
15:23.23ManxPowerx86: then you set them up wrong. 8-)
15:23.29x86[TK]D-Fender: where the IP301 and IP501 work fine
15:23.37x86ManxPower: not according to Polycom's engineer
15:24.01ManxPowerWe had massive problems with the 320s.  Until we updated the sip.cfg and phone1.cfg to add the required options for those phones.
15:24.19ManxPowervolume problems, the icons next to the line appearance buttons not working.
15:25.00x86the problem we had with the 330's and 430 is that they would start to boot, and then just lock up
15:25.16x86some times they would sit there locked up and never do anything, sometimes they would go into an infinite reboot
15:25.30x86Polycom couldn't figure the issue out
15:25.41ManxPowerx86: we never had those problems and we have a fair number of 320s and 330s
15:25.43x86not had a single problem with 301's, 501's, nor 601's
15:26.01x86ManxPower: *shrugs*
15:26.19x86ManxPower: the network cabling here is cat4 (*ugh*), so that might have something to do with it
15:26.30x86ManxPower: but the 301, 501, and 601 have no problems with that at all
15:29.28ManxPowerx86: are you *sure* your sip.cfg and phone1.cfg files support the new options for the 320/330?
15:30.41[TK]D-Fenderx86: Neither myself nor any of my clients ever have problems with any of them.
15:31.42pjzanyone have recommendations on firmware version to use with Polycom IP330s ? 3.0.3revB ?
15:32.45pjzx86: I'm having the infinite reboot problem with my phones right now, but I thought it was due to upgrading my asterisk server
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15:35.40ManxPowerI'll look at what version we use.
15:35.52*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
15:38.04[TK]D-Fenderpjz: I would switch to the latest 3.X release and if that causes issues, the latest 2.2.X release
15:38.18Rienzillagr0mit: would changing settings for FXS Port impedance, FXS input gain and FXS output gain help anything?
15:38.22Rienzillacould*
15:38.23ManxPowerPolycom SIP firmware 2.1.1.0052
15:38.34ManxPower(according to the phone menu)
15:38.38pjzManxPower: on a 330?
15:38.50ManxPowerpjz: we only have one firmware version.
15:39.04ManxPowerIIRC 2.1 was what version supported 330/320/430
15:40.18ManxPowerYou would have to check the release notes to be *sure*.
15:40.53ManxPowerWe tend not to change things unless they are broken
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15:46.39Rienzillaaargh this is driving me nuts :)
15:47.58Specialist1I want to send all my outgoing traffic to a H323 server . do i need to create an H323 trunk for that on my server ?
15:48.57gr0mitRienzilla, nope
15:50.09*** join/#asterisk weinerk (n=irc@82.80.129.212)
15:50.53ManxPowerSpecialist1: not many people here use H323.  Which H323 channel are you using in Asterisk?
15:51.53*** join/#asterisk ManxPower (n=manxpowe@40.sub-70-221-156.myvzw.com)
15:52.11ManxPowerstupid verizon
15:52.49*** join/#asterisk theHub (n=theHub@69.177.93.21)
15:52.53Specialist1actually i am using a sip server but i want to forward my traffic to the h323 sever
15:53.35*** join/#asterisk scurb (i=scurb@94.191.130.55.bredband.tre.se)
15:57.51ManxPowerSpecialist1: Best of luck.  As I said, virtually nobody here uses H323
16:00.00Blackveloh my dear, I wish I had choosen external Java AGI for this ivr thing. how can I avoid that 4x IF...SET gets executed line after line (even on 1st match)? is there any way to combine the IF with GOTOIF? like "if .. else if .. else if .. else if ..."
16:00.39aliverI'm using an ITSP (bandwidth.com). In my sip.conf they want me to have "fromuser=+1234567890" where the number there is one of my DIDs they handle for me. If I comment out the line, I think the "callerid=Some Young Guy <1234567890>" is sent to them instead and they reject the call because they only want the number in it's 11 digit form.
16:01.00aliverDoes that sound unlikely. Mainly I want to know, does fromuser == callerid more or less?
16:01.15aliverie.. is callerid sent as "fromuser" when there is no "fromuser" ?
16:02.45[TK]D-FenderBlackvel: FFS its jsut 4 lines of dialplan, stop complicating it
16:03.16lmadsenBlackvel: sounds like you want to use a case statement in AEL (which is just a set of GotoIf()'s
16:04.55lmadsenGotoIf($["${foo}" = "${bar}"]?true:${IF($["${jim}" = "${beam}"]?true:${IF($["${jack}" = "${daniels}"]?true:false)})})})
16:05.04lmadsenif you really wanna make it hard and put it all on one line
16:06.04lmadsenso yes, there IS a way -- but even *I* don't do that, and I love nesting functions
16:06.36Blackvelso you guys dont care in normal extensions.conf when the four ifs get executed one by one
16:06.42lmadsenand actually I have one too many sets of closing I believe
16:06.59lmadsenBlackvel: explain again simply what you're trying to solve
16:07.10lmadsenyou're trying to execute 4 GotoIf()'s at the same time? That's what it sounds like...
16:07.20lmadsenbecause obviously we aren't getting it
16:07.22Knightfalcan I limit the amount of calls I recieve to an inbound route on a per day basis
16:07.27[TK]D-Fenderlmadsen: Even that is more complex than it deserves
16:07.34KnightfalI have a client that has 100 calls per day in thier contract and they are sending me 300+
16:07.35lmadsenKnightfal: yes you can
16:07.52KnightfalCan you point me in the right direction
16:07.57lmadsenKnightfal: combination of GROUP(), GROUP_COUNT() and STRFTIME() I would think
16:08.14lmadsenor STRFTIME and DB() perhaps
16:08.33Blackvellmadsen: sorry I dont get it to describe in English :)
16:08.45x86lmadsen: group and group count only count the number of calls currently in use in that group, right? they don't do cumulative totals over a period of time, I thought
16:09.02lmadsenx86: yes, that's why I realized using a DB() call is more appropriate :)
16:09.09x86:)
16:09.18lmadsenthx though for making that point obvious
16:09.20sapereHey guys -- Asterisk/VOIP newb question here: Asterisk-based appliances provide what benefits that a simple analog VoIP adapter doesn't?
16:09.29lmadsenBlackvel: then good luck! :)
16:09.29x86or MYSQL() that ties right in with your billing system ;)
16:09.45lmadsenyep, something like that :)
16:09.48Blackvel4 lines with SET cmd combined with prior IF. when I have 4 lines, the if gets called 4 times (which you normally try to avoid in programming languages)
16:10.28lmadsenif you normally try to avoid it in programming languages, why would you do it in asterisk?
16:10.44Blackvellmadsen: [TK]D-Fender tries to tell me that I should program those 4 if/set lines and shut up :)
16:11.00lmadsenI still don't get what you're trying to do
16:11.05Blackvelexactly
16:11.35Blackvelforget it. maybe I can explain it in 10 years when my English progresses :)
16:11.53lmadsenI don't think its an english problem -- I think it is a logic problem
16:12.22sapereDo I even need an analog adapter if I receive service over ethernet?
16:12.48sapereah, wait ... nvm
16:12.51lmadsenIF($[expression]?true:else)
16:12.54sapereanalog phones, not analog incoming
16:14.18lmadsen[10:00]  <Blackvel> hi all. found GotoIf cmd. I am in the need to manually change to callerid for outgoing call depending on various callerid strings. Is there an simple If/else statement too? where could I look at?
16:14.35lmadsenBlackvel: do you mean based on the incoming callerID string, you need to change the callerID string going outbound?
16:15.26lmadsenexten => _NXXNXXXXXX/4165551212,1,Set(CALLERID(num)=8002223334)
16:15.46lmadsenexten => _NXXNXXXXXX/5195915119,1,Set(CALLERID(num)=8665551212)
16:16.11ManxPowerBlackvel: See "ex-girlfriend option" in the Asterisk docs
16:16.14lmadsenexten => _NXXNXXXXXX/5195915119,n,Goto(standard_stuff,1)
16:16.49lmadsenexten => standard_stuff,1,NoOp(This is where all major logic happens, regardless of who called in -- you've already set the CID for the outgoing call)
16:16.58*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:18.10*** join/#asterisk enno (n=enno@lightsource.verbrennung.org)
16:19.40enno"Extension can never match" <-- i tried to find out where in the externsion.conf to place an i or _X rule for testing first isdn dialin. Any hints? Google didnt help
16:20.21ManxPowerenno: "i" is only for IVRs
16:20.32ManxPoweryou would place it in the context your ISDN dialing ports are located
16:20.44ManxPowerwhere "it" is _X.
16:20.57*** join/#asterisk wiscados (n=mint@81.25.184.155)
16:22.39ennoManxPower: thanks
16:25.32*** join/#asterisk jsin_ (n=jsin@gentoo/developer/jsin)
16:26.58*** join/#asterisk StormD (n=stormd@64.3.54.171.ptr.us.xo.net)
16:27.29StormDhow do I determine what version I'm running?
16:27.55[TK]D-FenderStormD: "show version"
16:29.00lmadsen"core show version"
16:29.01StormDso I think I did something stupid, running version 1.2.23
16:29.15lmadsenheh
16:29.18[TK]D-FenderStormD: Thats a self-fulfilling statement :)
16:30.18StormDThank you for the vote of confidence in my self-appraisal.
16:31.31StormDanyway, what happened is, a couple months ago, one of my clients demanded I enable voice mail on their system which had apparently been set up and configured without any voice mail.
16:32.31StormDFiguring that was a completely retarded way to set up a voip system, I went ahead, read a few docs, and set up basic voice mail, which worked pretty well until two weeks ago, when it became apparent why voice mail was never set up in the first place, when the 10gig hard drive that the system was running on completely overflowed.
16:34.15StormDSo I went through /var/spool/asterisk/voicemail/<group>/<boxes>/INBOX/ and deleted a bunch of messages to clear up a little space, and I also created some room by deleting /var/log/asterisk/messages.0
16:34.23StormDI think that last part was retarded of me, correct?
16:35.06[TK]D-FenderStormD: no, thats fine
16:35.12l2trace99it should be a log file by the path of it
16:35.39StormDnow, several users can't log into their mailboxes...they dial into their mailbox, enter their password, and then the system just disconnects them
16:36.40l2trace99do the users show up in  'show voicemail users' ?
16:37.17StormDyes.
16:37.23*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:38.06StormDoh hmmm...
16:38.56StormDBeing connected to the console when logging into the maibox is a valuble troubleshooting tool, innit?
16:39.21l2trace99does the same user that owns the asterisk process owns the /var/spool/asterisk  dir
16:39.22l2trace99yes
16:39.47*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
16:40.09StormDapp.c:1232 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/<group>/<user>/Old': File exists
16:40.54StormDwell, root owns that directory
16:41.04[TK]D-FenderStormD: Did you delete the txt files AND all of the recordings to match?
16:41.05l2trace99and asterisk is running as ?
16:41.17StormDroot
16:41.47StormDFender: yeah I pretty much scoured their boxes clean
16:41.51*** join/#asterisk arpu (n=arpu@chello080109017114.12.14.vie.surfer.at)
16:43.31Kobazwow, our main asterisk hub just crashed
16:43.44[TK]D-FenderStormD: copy their recordings, delete the entire vm folder, leave a new VM to reinit the box, then copy the recordings back
16:44.19StormDbut when I was cleaning the "Old" directory, I think I just ran ls, and didn't 'ls -al' and missed that .lock file which I now see in there and just cleared out.
16:44.25StormDtesting again
16:44.29Kobaz[Sep 11 12:37:59] WARNING[20140] res_features.c: Bridge failed on channels IAX2/AlbanyIn-5 and Zap/1-1, res = -1
16:44.42Kobazthat's the last message in the log before asterisk crashed
16:44.43Kobazweird
16:45.04StormDoh that totally fixed it.
16:45.17*** join/#asterisk bmg505 (n=leon@196-209-8-66-ndn-esr-2.dynamic.isadsl.co.za)
16:46.17*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
16:46.24*** join/#asterisk newvn (n=newvn@70.252.57.249)
16:47.03*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
16:47.23StormDNow, the main thing I need to do is install a bigger hard drive on this damn computer to mount on /var so that this doesn't happen again in another month.
16:52.13*** join/#asterisk jtodd (i=fn8t2q0n@ns2.loligo.com)
16:52.34StormDso /var/log/asterisk/messages, messages.0, messages.1 just cascading records of voice-mails gone by....not necessary for functioning of system?
16:53.28[TK]D-FenderStormD: Feel free to trash them
16:53.29StormDI notice those files get huge, being voice-recordings.
16:53.45StormDand I only have a 10G drive for the whole system here.
16:58.35CrashSysUse mp3
17:02.22[TK]D-Fender...
17:02.30[TK]D-Fender~cluebat CrashSys
17:02.31jbotACTION pulls out a ClueBat (tm) and thwaps CrashSys.
17:02.39CrashSysThank you sir may I have another
17:02.41CrashSyserr, sup?
17:03.34StormDI'll look into that, but the thing is, installing a hard drive and mounting a partition is something I know how to do in less than an hour and only costs my client $100 or so. Tracking down documentation on converting voice-mails to utilize some compression I'm not sure how to use in a system I'm only barely familiar with, and teaching myself how to implement that is... ??? on my schedule and billing sheet.
17:04.10CrashSyslol :)
17:04.21mort_gibStormD: you are better off doing both anyway, if you have enough CPU power
17:04.33mort_gibAnd it's not really difficult
17:05.00[TK]D-Fender~cluebat CrashSys
17:05.00jbotACTION pulls out a ClueBat (tm) and thwaps CrashSys.
17:05.02CrashSysHard-drive is the easier option
17:05.09[TK]D-FenderStormD: the answer you're looking for is "more" :)
17:05.12CrashSysplus 10-gig's is pretty miniscule
17:05.15*** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk)
17:05.23CrashSysby today's standards anywyas
17:05.35bkw_10 gig of voicemail?
17:05.39mort_gibYes, how did you manage to find a 10GB drive??
17:05.43[TK]D-FenderMy cell phone has 4, and thats because I got it free :)
17:05.50bkw_aha
17:06.14[TK]D-Fender\o/ WinMo6.1 upgrade
17:06.20[TK]D-Fender(pending)
17:06.45StormDI have no idea...this system came with the client. One of their IT-guys who no longer works for them since before they hired me set it up.
17:07.29StormDThe weird thing is, it's a Dell that looks like it's 2 years old at the max....I have no idea how it shipped with such a tiny drive.
17:07.42StormDI suspect some hanky-panky might have gone on.
17:08.27StormDin fact, I think I'm going to look up the Service Tag on Dell's site and see what size drive it actually shipped with.
17:09.04Kobazmaybe the drive failed and a crappy old spare was quickly found
17:09.17Kobazor it came with a 500gig drive and someone jacked it
17:09.18StormDbkw: 10 gig of total hard disk...that's OS AND voice mail.
17:09.46StormDall mounted on a single partition, naturaly
17:09.52KobazStormD: 10 gig isn't too shabby for a small linux system, i used to run linux on like 100 meg drives
17:11.43*** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net)
17:12.07*** join/#asterisk Levonk (n=lk@adsl-75-62-140-156.dsl.lsan03.sbcglobal.net)
17:12.24StormDKobaz: oh, I know...I used to run Red Hat on a 800MB drive when I was in college. But I wasn't trying to record and serve voice-mail for an office, either.
17:13.00Kobazheh
17:13.19*** join/#asterisk sapere (n=sapere@209-249-12-72.ip.openhosting.com)
17:13.27coppiceI remember doing voicemail on a 10M drive
17:13.53StormDyou get 1 30 second message
17:14.05mvanbaakit all depends on the ammount of users and the count of messages
17:14.26mvanbaakoh, and the codec ;)
17:14.31*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
17:15.00StormDanyway, thanks for the help, I gotta jet.
17:16.18sapereafternoon all, quick question about minimum hardware required for an IP phone small business asterisk setup: I'd need a computer to run my asterisk server, a switch, and an IP phone. That's it correct? (other than the ITSP service)
17:16.46saperewell, and cat5 cables between the PC & switch, and switch & phone(s)
17:17.21[TK]D-Fendersapere: Yup, thats about it.
17:17.46mort_gibsapere: and some time on your hands
17:17.47saperethanks, [TK]D-Fender
17:17.55[TK]D-Fendersapere: You could technically do without the switch by adding more NIC's to your server, but who wants to do all that? :)
17:17.59[TK]D-Fender(except me)
17:18.39[TK]D-Fenderloves his everything-and-the-kitchen-sink-home-server
17:18.42sapereIf I wanted to run something like Elastix or Trixbox, instead of Asterisk directly, am I really gaining anything?
17:18.52[TK]D-Fendersapere: Yes, our wrath :)
17:19.01mort_gibsapere: flaming from TK
17:19.03[TK]D-Fendersapere: Free with every install!
17:19.45[TK]D-Fendersapere: GUI's don't do much of anything you can't do yourself and force you to do things based on their method of thinking.
17:19.53errrIm having some issues with  an ivr.. http://paste2.org/p/72794  I have pasted the error Im getting here
17:19.57saperehah -- i can't find any real value they provide, other than claims that make me skeptical (trixbox claims to "lockdown" security in asterisk)
17:20.06mvanbaak[TK]D-Fender: you forget that vim is also a gui
17:20.18mort_gibsapere: If you do have issues, doing it ground up will prepare you better...
17:20.42[TK]D-Fendermvanbaak: Real programmers use SOLDER
17:20.51mvanbaakor ed
17:20.53sapere[TK]D-Fender : are they strictly a GUI? If they're only a GUI, then FreePBX is also in that group
17:21.03mort_gibOr rather, WHEN you have issues, HAVING done it ground up prepares you better
17:21.20mvanbaaksapere: freepbx is not a GUI, it's a virus
17:22.04sapereand I only need FXS/FXO to connect to analog phones or analog phone lines, correct
17:22.09[TK]D-Fendersapere: Those distros you mentioned all USE FreePBX and is the core problem.
17:22.17[TK]D-Fendersapere: Correct.
17:22.31[TK]D-Fendersapere: for that I'd suggest a Linksys SPA-3102 if for home use
17:22.54sapereThe cheapest ITSP i've found is Broadvoice's BYOD service at $6/mo for 100 minutes, any opinions on them?
17:23.06mvanbaakthose SPA thingies are that good ?
17:23.50jayteeI just got a 3102 in today. haven't set it up yet
17:24.10*** join/#asterisk carpenike (n=ircap@82.ecb7d1.client.atlantech.net)
17:24.16sapere[TK]D-Fender : but I wouldn't need a Linksys SPA-3102 if I'm doing IP phones and no FXO/FXS interface
17:24.29*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
17:24.29[TK]D-Fendersapere: thats $.06/min with SUCKS
17:24.39[TK]D-Fendersapere: Yes, but you ASKED
17:24.39*** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat)
17:24.56[TK]D-Fendermvanbaak: Not awesom for FXO, but good enough for home.
17:24.57sapereoh, sorry I misunderstood your response
17:25.09carpenikeHey anybody know much about faxing inside 1.6?
17:25.13mvanbaak[TK]D-Fender: ah. good to know
17:25.15[TK]D-Fendermvanbaak: and the FXS is Linksys regular.  Excellent value overall
17:25.28CrashSysI pay $0.0089/min
17:25.43CrashSysonly have to spend about $2000/mo to get it too :)
17:25.44sapere[TK]D-Fender : that price is outgoing, I should mention. Free incoming
17:25.45[TK]D-FenderI pay $0.000/min:)
17:25.53CrashSysDAMN
17:25.57[TK]D-Fendersapere: bleh
17:26.08mvanbaakI just want a device that will reroute calls going from $outside to local analog line to sip/iax/skinny
17:26.19[TK]D-Fendersapere: could still do better probably.  You need to reall calculate your usage when shopping around.
17:26.25mvanbaakso ppl can keep their loved/hated analog line and still experiment with voip
17:26.31[TK]D-Fendermvanbaak: It'd do the job OK
17:26.46[TK]D-Fendermvanbaak: its ESPECAILLY good because of the power failover
17:27.07mvanbaakyeah, I read that
17:27.17carpenikeHey anybody know much about faxing inside 1.6?
17:27.59*** join/#asterisk moy (n=moy@nat/ibm/x-cbcb7d536591e364)
17:28.46carpenikeSpecifically, I'm tring to figure out if it's possible to do a Email to Fax gateway using the new termination/origination built into Asterisk 1.6 all locally installed... And if so, what I actually need installed to do so...
17:29.00carpenikeI'd like to get a T.38 provider and pass the faxing SIP/IAX2.
17:32.48*** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk)
17:33.08sapereDo you guys generally recommend AsteriskNOW over Asterisk as a separate server? In my case I have no need to reuse my server for any other purpose
17:33.58heedlysapere: most people in here will recommend compiling asterisk from source.
17:34.13heedlyand using a distro you are comfortable with.
17:34.21saperefair enough, thanks
17:34.50carpenikeGentoo. :-D
17:35.28*** join/#asterisk makkksimal (n=makkksim@f054120109.adsl.alicedsl.de)
17:35.48errrisnt asterisknow going to start to come with freepbx?
17:36.07heedlymost modern linux distributions include ways to build from source.
17:36.16heedlygentoo is the only one that requires it ;)
17:36.33heedlyslap an "R" on it goes fffffaaaast
17:36.56carpenikeI just got Gentoo + OpenVZ + Asterisk 1.6 Source + Freepbx to all play together. :)
17:37.08carpenikeInside of VMWare.
17:39.22heedlya VM inside a VM huh
17:39.32heedlymy reality is shriking man
17:39.54carpenikeHeh... Goal is to put it into ESX.
17:40.42carpenikeAnd use the same VM to run additional isolated services like mail.
17:43.54*** join/#asterisk ohayden (i=ohayden@adhd.irule.net)
17:44.40CrashSyscarpenike: How did you get a reliable hardware timer through openVZ?
17:45.39carpenikeinstalled Zaptel inside of the HN and gave the client access via vzctl set --devnodes
17:45.56CrashSyswhat's zttest report?
17:47.16Blackvelhow can I call functions on a dialplan? within {} or something?
17:47.32Blackvellike IF(expr? Set(....))
17:47.34*** join/#asterisk makkksimal (n=makkksim@92.224.50.208)
17:47.45Blackvelit interpretes my if as cmd
17:47.47carpenikeopenvz ~ # zttest
17:47.48carpenikeOpened pseudo zap interface, measuring accuracy...
17:47.48carpenike99.963379% 99.926758% 99.987793% 99.926758% 99.987793% 99.902344% 99.926758%
17:47.48carpenike99.987793% 99.938965% 99.987793% 99.902344% 99.963379% 99.914551% 99.914551% 99.975586%
17:48.07CrashSysimpressive... at idle load?
17:48.27carpenikeyeah.
17:48.47CrashSysSo you are running VMWare inside OpenVZ on top of Gentoo?
17:49.06[TK]D-FenderBlackvel: Go read the basics on functions and variables on the WIKI....
17:49.15carpenikeGentoo Host --> VMWare Server --> Gentoo VM --> OpenVZ --> Asterisk
17:49.16jeevas fender always says: reading is fundamental
17:49.23carpenikeErr
17:49.26carpenikeGentoo VM is OpenVZ.
17:49.46CrashSysis the Gentoo/VMWare just for testing or actual production set-up?
17:50.07CrashSysIn other words, is Gentoo/VZ the actual platform as what you would do in production?
17:50.16carpenikeFor my house yeah.
17:50.21carpenikeAt work...
17:50.36carpenikeI'd do ESX -> GentooOpenVZ --> Asterisk
17:50.52CrashSysHmmm, ok...
17:51.13carpenikeAnd use the same Gentoo OpenVZ VM to run a few other VPSes.
17:52.04carpenike--- Results after 313 passes ---
17:52.04carpenikeBest: 100.000000 -- Worst: 92.480469 -- Average: 99.764670
17:52.25CrashSysehhh... 92 = hell for IAX/MeetMe
17:52.43carpenikeYeah.
17:52.58CrashSysI have IAX/MeetMe dependencies...
17:53.52carpenikeHmm.. What's the main draw in hosting your own conference room and not going with a hosted provider IE: freeconferencecall.com? Aside from branding and I can see it if everybody is internal to the office or dialing in through VPNs...
17:54.01*** join/#asterisk bbryant (n=brett@68.208.65.50)
17:54.39x86anyone know of a good classic rock icecast stream that I could stream via an Asterisk extension?
17:54.57Blackvelhave a good evening
17:57.04*** join/#asterisk lanning (n=lanning@66.151.128.195)
17:58.23*** join/#asterisk ManxPower (n=manxpowe@231.sub-70-222-214.myvzw.com)
17:58.47*** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
17:58.47*** mode/#asterisk [+o russellb] by ChanServ
17:59.13Kobazx86: you could stream shoutcast/icecast via mplayer, and pipe the output to music on hold
17:59.58*** join/#asterisk gr0mit (n=tim@81.187.32.146)
18:01.50*** join/#asterisk soulfreshner (n=root@dsl-241-171-94.telkomadsl.co.za)
18:02.31soulfreshnerI'm struggling to get my tdm400 card working
18:02.40soulfreshnerI compiled the drivers
18:03.42soulfreshnerbut when I run ztcfg -vv I get "failed on channel 1: No such device or address (6)
18:03.56Qwelldid you load the drivers?
18:04.25soulfreshneryep - modprobe zaptel; modprobe wctdm
18:05.16*** join/#asterisk skr1p7 (n=toor@line111-36.adsl.kirov.ru)
18:05.55Yourname`Hi, is there a way to 1) Login an agent via AMI 2) Transfer a call via AMI?
18:05.57*** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com)
18:06.12[TK]D-FenderYourname`: Yes, Yes, and BOOK <-
18:06.46skr1p7hi all. how to fix that: "system('asterisk -rx "dialplan reload"');" return "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" ?
18:06.54skr1p7srwx---r-x 1 asterisk asterisk 0 Sep 7 00:43 /var/run/asterisk/asterisk.ctl
18:07.01skr1p7^ file is exist
18:07.31ManxPowerskr1p7: remove the '
18:07.35ManxPowerboth of the ' actually
18:07.43skr1p7where?
18:07.48ManxPowerI doubt that's the problem, but it's a good idea.
18:07.50Yourname`[TK]D-Fender: Which book?
18:08.01ManxPowersystem('asterisk -rx "dialplan reload"')  <-- remove the 's
18:08.02soulfreshnerI get another error at startup as well: wcopenpci:[00] Firmware version 0 not supported by this driver contact Voicetronix to have it updated
18:08.03[TK]D-Fender~book
18:08.04jbotmethinks book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
18:08.29skr1p7run sudo -u www asterisk -rx "dialplan reload" return same error in shell
18:08.41ManxPowersoulfreshner: edit /etc/sysconfig/zaptel  remove the drivers you don't need
18:08.51ManxPowerskr1p7: but that was not your question
18:09.12ManxPoweras you can see the file is owned by ASTERISK but you are trying to access it as WWW
18:09.44*** join/#asterisk UtopiahGHML (n=rrrec@lns-bzn-33-82-252-24-232.adsl.proxad.net)
18:09.50skr1p7yeah. need own by www? i think if chmod o+rx is good :)
18:10.02soulfreshnerManxPower, I don't have /etc/sysconfig/zaptel (using ubuntu)
18:10.27*** join/#asterisk makkksimal (n=makkksim@92.224.50.208)
18:11.19skr1p7ManxPower: thx. chown to www fix that problem
18:11.56ManxPowerskr1p7: not going to help in the long run.
18:12.25ManxPowerThat file is re-created every time Asterisk starts.
18:12.32*** join/#asterisk Bananaskin (n=Banana@user-5af01b01.wfd96.dsl.pol.co.uk)
18:12.44*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
18:13.23skr1p7ok. i will add chown command to rc.d script :)
18:14.24ManxPowerskr1p7: that is the wrong way to do it.
18:14.30ManxPowerWhy don't you fix the actual problem?
18:14.59ManxPowerThe problem being you want to run Asterisk as user "www", but Asterisk is configured to run as user "asterisk"
18:15.25skr1p7run asterisk as www? that not good fix too
18:15.40ManxPowerskr1p7: no, that is the CORRECT fix.
18:18.30Yourname`[TK]D-Fender: The QueueAdd, does that matter if it's AgentLogin()?
18:18.36Yourname`..in the dialplan I mean
18:19.50[TK]D-FenderYourname`: And I've answered this many times, agentlogin has NOTHING to do with dynamic members
18:20.23Yourname`Ok, cool.
18:20.26Yourname`God bless you.
18:20.45Kobazwe don't want blessings, we just want food and beer
18:20.45ManxPowergod has nothing to do with it
18:21.38Yourname`Well, [TK]D-Fender needs 'em, ok?
18:21.56ManxPowerI doubt that.
18:22.02Yourname`With the blessings of your employer, your requisite for food and beer will be fulfilled.
18:22.28Kattyherroes.
18:22.32ManxPowermost tech people don't seem to believe in those superstitions.
18:23.08Yourname`Of an employer fulfilling your food/beer needs?
18:23.44ManxPowerno.  of some super powerful being that watches over people.
18:23.59ManxPowerkind of like Santa Claus or the Easter Bunny
18:25.10*** join/#asterisk imcdona (i=imcdona@2001:5c0:8fff:fffe:0:0:0:b917)
18:25.10Yourname`Ah, I don't know tech world you live in Mr. Power, but a couple engineers here who happen to be the geekiest of techs have little Hindu shrines on their desks.
18:25.11scooby2damn this issue happened again today. 5 calls waiting to be answered, 3 people logged in but idle, asterisk thinks everyone is busy.
18:25.26Yourname`scooby2: autofill=yes?
18:25.28ManxPowerscooby2: tell your agents to stop tranfering people
18:25.54ManxPowerIIRC an agent that transfers a call is still considered in use.
18:26.18scooby2Yourname`: no autofill
18:26.41scooby2not sure about transfering but I doubt all three agents transferred someone
18:27.45scooby2just moved to 1.4.21.2 looks like something to try
18:28.53scooby2Yourname`: thanks, I will try it now
18:28.56carpenikeHey does anybody know when the HP Proliant dl380 G4s were introduced?
18:29.04ManxPowerscooby2: moved to 1.4.12.2 FROM what version
18:30.05scooby21.2.15
18:30.27ManxPowerscooby2: Did you read upgrade.txt and upgrade-1.2.txt?
18:30.52scooby2yeah
18:30.58MindTheGap_hello all, i get this:
18:31.02MindTheGap_<PROTECTED>
18:31.03MindTheGap_<PROTECTED>
18:31.03MindTheGap_Segmentation fault
18:31.20carpenikeCould try moving the modules directory to a backup and re-installing
18:31.44MindTheGap_when registering a peer w md5secret
18:31.51MindTheGap_asterisk 1.6.beta9
18:32.23russellbupdate to a 1.6.0-rc
18:32.31russellbdownloads.digium.com/pub/telephony/asterisk
18:32.35russellbup to rc6 now
18:33.01russellband also make sure you empty out /usr/lib/asterisk/modules when upgrading
18:34.21Kobazi always build with a prefix
18:34.37Kobaz./configure --prefix=/opt/asterisk-1.6.0rc6
18:34.48Kobazand then you just make symlinks
18:35.30*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
18:35.30*** mode/#asterisk [+o lmadsen] by ChanServ
18:37.06Kobazso if i wanna switch asterisk versions, i just change one symlink
18:39.21*** join/#asterisk makkksimal (n=makkksim@92.224.50.208)
18:39.29thedonvaughnKobaz: doh, so simple i feel dumb for not doing that myself.  I run vicidialer with a few asterisk servers here.  I've gotten in trouble with vici with an upgrade.  Changing with symlinks woulda been the smooth.
18:41.04Kobazheh
18:41.28*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)
18:46.27*** join/#asterisk Thorn (n=thorn@unaffiliated/thorn)
18:46.41x86Kobaz: or you can use an icecast stream directly as MoH ;)
18:46.49x86Kobaz: I was looking for suggestions on a stream
18:47.28*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk)
18:48.54Kobazx86: oh hmm, dunno
18:48.58*** join/#asterisk korihor (n=korihor@201.211.168.130)
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19:02.04jameswf-homemy MOH is leekspin
19:12.53*** join/#asterisk zerohalo (n=zeroHalo@75.150.77.161)
19:16.26MindTheGap_russelb, thanks... just upgraded to rc6 and it wont crash anymore but the password wont match. Realmed password should match peer@realm:password (from memory may not be it) yes? but i need to match it to just a password, no realm no peer, just password. i should use MD5Secret? im using MD5Secret but it fails authentication.
19:17.27MindTheGap_btw, i'm authenticating against an ldap server
19:17.39*** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137)
19:18.01*** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk)
19:18.12MindTheGap_s/russelb/russellb/
19:18.32russellbi do not recall
19:19.25*** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk)
19:20.05rgsteele||workSo, I'm constructing a dialplan in which 011 gets stripped from the number, but only if that 011 is present.  My original attempt stripped it all the time, so I worked up this to circumvent that problem, but was hoping to have some eyes more experienced with AEL give the yay or nay before I test it early tomorrow morning before anyone gets to the office: exten => s,n,Set(${NUMTODIAL}=${IF($["${MAC
19:20.07rgsteele||workRO_EXTEN}" : "(....)"] = "011")?${MACRO_EXTEN:3}:${MACRO_EXTEN})
19:20.24rgsteele||workBah, formatting.  Let me try that again:  exten => s,n,Set(${NUMTODIAL}=${IF($["${MACRO_EXTEN}" : "(....)"] = "8011")?${MACRO_EXTEN:4}:${MACRO_EXTEN:1})
19:20.28MindTheGap_russellb, i asked about asterisk crashing using MD5Secret, you told me to upgrade do rc6. i did upgrade and it wont crash anymore but the password wont match. Realmed password should match peer@realm:password (from memory may not be it) yes? but i need to match it to just a password, no realm no peer, just password. i should use MD5Secret? im using MD5Secret but it fails authentication.
19:21.06heedlyrgsteele||work: why not just write two seperate extens that match the different formats?
19:21.23heedlythat just looks confusing...
19:22.22rgsteele||workheedly: I suppose I could, I just thought about doing it in one go initially.
19:22.49[TK]D-Fenderrgsteele||work: Close, but you are not setting a varibale properly there...
19:23.09rgsteele||work[TK]D-Fender: That's the part I was worried about :)
19:23.33rgsteele||workI was pretty sure the ternary statment was okay (though certainly I could be wrong about that too!)
19:24.00[TK]D-Fenderexten => s,n,Set(${NUMTODIAL}= <-- already bad
19:24.35rgsteele||workErgh, yeah, that should be ...Set(NUMTODIAL=....)
19:25.14rgsteele||workWhich leaves me with: exten => s,n,Set(NUMTODIAL=${IF($["${MACRO_EXTEN}" : "(....)"] = "8011")?${MACRO_EXTEN:4}:${MACRO_EXTEN:1})
19:26.55*** join/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com)
19:26.57rgsteele||work[TK]D-Fender: Thanks for the once-over - I hate chasing down silly things like that really early before those first two cups of coffee have kicked in :)
19:27.35*** join/#asterisk nr4q (n=dave@24-183-225-98.dhcp.kgpt.tn.charter.com)
19:28.51nr4qscratching my head over this. I think it's replated to the dialplan on the phone. with a polycom phone, handset on phone type a 3 digit extension (103), press end and it dials. handset off hook dial 103 and it sends 10 to asterisk. default dialplan on the phone and in sip.cfg
19:29.50nr4qonly thing i can think of is that the phone sees the 0 in "103" and things i'm trying to call the operator "0" and automatically sends "10"
19:29.51[hC]So I got an email from polycom about a polycom/digium-created wifi voip phone. is this anything new, or is this the same phone weve known about for ages that polycom had out?
19:30.07nr4qhc: oh that sounds cool
19:30.44*** join/#asterisk Levonk (n=lk@adsl-76-238-250-64.dsl.lsan03.sbcglobal.net)
19:30.52nr4qhc: do you have a link?
19:31.13[hC]not yet
19:31.37[TK]D-Fender[hC]: Ages ago Polycom bought out Spectralink
19:32.04nr4qare there any decent voip 802.11g phones these days? i know a few years ago i heard that wasn't the case
19:32.12rgsteele||workHaven't the VoIP over WiFi phones been out for a bit, though?
19:32.20rgsteele||work(Quality aside)
19:33.41[hC][TK]D-Fender: yeah. I know, I just got this email from a dude at polycom, so it seemed like maybe something new happened. turned out its just a spectralink 8002
19:33.56*** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net)
19:33.57[hC]http://www.polycom.com/usa/en/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html
19:34.45*** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk)
19:35.58jameswf-home[hC]: that doesnt mention digium as a partner just that it is ABE Certified
19:38.14[hC]jameswf-home: my email said "Polycom and Digium join forces to bring an enterprise grade VoWifi wireless solution to the SMB market segment!!!!!!!!!"
19:38.41MindTheGap_nr4q, we got some nokias e65 and e51. quality is decent but they cant handle re-registration (when phone looses wifi coverage) properly sometimes.
19:39.34nr4qi have a FXS card for a tdm400. i thought the best solution may be to just use an analog phone on that FXS port
19:39.38sapereAsterisk The Future of Telephony speaks highly of the Linksys SPA942, has anyone had any negative experiences with one?
19:39.58jameswf-home[hC]: i see nothing newer than 07 in the googlesphere so maybe hasnt hit the blogs
19:40.50[hC]jameswf-home: its probably just announcing ABE certification..
19:40.59jameswf-homenothing on russellb's blog either
19:41.05[TK]D-Fendersapere: Forget Linksys, go Polycom.  www.telephonydepot.com
19:41.22nr4qi'm impressed with my polycom phones
19:42.08nr4qI just wish I weren't clueless
19:42.15russellbwhat what what?
19:42.30WimpMansapere: They try to use the in the company I used to work for but ar very dissatisfied with speech quality. I don;t knlow however what else might be wrong there nowadays.
19:42.52russellb[hC]: I got that email, too.  I lol'd at its sillyness level.
19:43.02WimpManI just disliked the UI, but then I haven't seen a voip phone with a UI I liked, so far.
19:43.06jameswf-homeOpenR2 in Asterisk - MFC/R2 Free of Headaches or Your Money Back << funny :)
19:43.17[hC]russellb: I actually had to check wether it was from a "real person"
19:43.20russellbVoWifi wireless solution from the department of redundancy department!!!!!111111111oneone
19:43.34voxterdamnit i need to change my xchat config.
19:43.41russellbvoxter: was it?
19:43.51bkrusevoxter: I remember those phones coming into Digium
19:43.54voxterrussellb: @polycom.com address.
19:44.03russellbvoxter: didn't look very ... um ... well, it didn't look like something that polycom would do
19:44.22voxterrussellb: no kidding. i bet that guy got in shit at lunch. :)
19:44.27jameswf-homei must only be on the asia spammer list :(
19:44.28russellbi hope he did
19:44.30voxter"Bill Newman"
19:44.40russellbheaders say it came from polycom in atlanta ...
19:44.48[TK]D-FenderWimpMan: For user interface I haven't a complaint about Polycom....
19:44.49voxternever heard of him, but i dont exactly pay attention to polycom's employees either.
19:44.59russellbpolycom is a big company ..
19:45.54jameswf-homeBill Newman - Manager, Dect Pre-Sales, Americas
19:47.01*** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net)
19:47.25voxterhas multicast/rtp paging made its way into asterisk yet?
19:47.32voxteri would love to stop using unicast Page()...
19:49.26Kattyanyone use exim4 and mutt a lot? i'm having some email problems i can't seem to figure out.
19:50.36tzafrir_laptopmutt and postfix here...
19:50.48voxtercrazy. people still use mutt.
19:50.55Yourname`I do.
19:51.02*** join/#asterisk soulfreshner (n=D@dsl-241-171-94.telkomadsl.co.za)
19:51.08Kattyit's easy to test with.
19:51.29Kattyi'm just not sure why emails aren't sending, and i'm not sure which logs to check
19:52.16soulfreshnerat last I got the tdm400 card working - I had to uncomment #define TDM_REVH_MATCHALL in zconfig.h ...
19:52.17voxterno idea about your system, but if you're sending from it, id start with /var/log/mail.log, as a suggestion
19:52.17voxterthat may not exist on your system.
19:52.22*** join/#asterisk wacky__ (n=root@nat/digium/x-4763810caeefbe6b)
19:52.42soulfreshnersomething to do with the PCI subvendor ID being out of date
19:53.09*** join/#asterisk n3hxs (n=HAMming@151.196.87.132)
19:53.26Kattywell that's a good place to check i guess.
19:53.26*** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net)
19:53.30soulfreshnernow what - how do I set up hunting lines for outgoing calls?
19:54.10[TK]D-Fendersoulfreshner: "group=" in zapata.conf
19:54.13*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
19:54.45[TK]D-Fendersoulfreshner: Dial(Zap/g1/1234567890) , etc for example
19:54.59soulfreshnerthanks, [TK]D-Fender
19:55.04Kattyvoxter: mailling to remote domains is not support, claims eximlog
19:55.14*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
19:55.25*** join/#asterisk phl4kx (n=supervis@200.48.200.66)
19:55.31phl4kxhi for all
19:55.35phl4kxI call digium asterisk support
19:55.45phl4kxbut the system say I dont have registered products
19:55.50voxterKatty: so your exim is misconfigured. there's a howto on the google.
19:55.57phl4kxIm register the product in digium
19:56.26Kattyvoxter: mmhmm, yeah. i'm rerunning dpkg reconfigure
19:58.08russellbphl4kx: hmm?  what product do you have?
19:58.17phl4kxAEX 800P
19:58.39russellbwhere did you get it from?
19:59.07sapereIf I have only one phone line and I want to add fax functionality, I'm kinda screwed right? I mean, since fax machines don't know how to enter an extension...
19:59.08*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
19:59.12*** join/#asterisk CamargoBP (n=BP@jive.fttp.xmission.com)
19:59.47jaytee[TK]D-Fender, do you think it might be possible with Asterisk and an SPA3102 to have the line for the FXS port always off-hook and in an on-hold condition with MOH playing?
19:59.57CamargoBPAnyone know a good tool to monitor Jitter?
20:00.05CamargoBPHopefully that is opensource.
20:01.04[TK]D-Fenderjaytee: Umm.... why?
20:01.15voxterCamargoBP: using just icmp, you can use mtr.
20:01.23Kattyoh
20:01.27CamargoBPThat's what we do right now
20:01.34[TK]D-FenderKatty: Mew.
20:01.37Kattysomething just came to me about our stupid exchange reverse dns lookup server
20:01.44jayteeI'm trying to jury rig something with the paging PA system I told you about using Valcom's equipment. it has a separate audio source connection for music.
20:01.45Kattyit's floating about my head, like dust particles
20:01.47CamargoBPvoxter: I'm trying to make a nice web interface or maybe a plugin for Zenos
20:01.48CamargoBPs
20:01.50Katty[TK]D-Fender: ohai
20:01.53Katty[TK]D-Fender: howrechu
20:02.25jameswf-homehttp://www.hanlongtek.com/pro_2_4.html <<looks strangely like a GS Handytone
20:03.24[TK]D-FenderKatty: gesundheit :)
20:03.43Kattyvoxter: ahahahahahahaha
20:03.44voxterCamargoBP: well jitter is really just large deviation between round trip measurements... so however youd want to capture that. but its something that you can only determine by monitoring over time
20:03.48Kattyvoxter: my exhcnage server was dumping it
20:03.53Kattyvoxter: cause reverse DNS FAILED
20:04.18jayteejameswf-home, best way to tell if it's a Handytone on the inside with a different case is just make a call with it and say hello and see if you hear yourself saying hello back. :-)
20:04.29CamargoBPvoxter: When you talk time does that mean secons, minutes, hours or days?
20:04.32Katty[TK]D-Fender: well come on then
20:04.34Katty[TK]D-Fender: how're you?
20:04.38Katty[TK]D-Fender: mister gesundheit
20:04.42CamargoBPI've seen some solutions where you can see real time jitter
20:04.58CamargoBPMaybe they aren't really real time
20:05.08jayteejitter should be in milliseconds
20:05.23phl4kxrussellb:
20:05.25phl4kxIm from peru
20:05.46*** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin)
20:06.19n3hxsjitter is sometimes caused by too much coffee.
20:06.51CamargoBPn3hxs: lol
20:07.12CamargoBPI'll get all my customers to stop drinking coffee
20:07.34n3hxsEverclear will help clear things up.
20:07.45n3hxsAt least they won't care as much.
20:08.07[TK]D-FenderKatty: Getting by, a piece at a time.
20:08.22CamargoBPYou bet send them all a 40 and hope they don't complain
20:08.40*** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
20:10.11CamargoBPWould you say encapsulating all the VOIP traffic in TCP rather than UDP will help with jitter?
20:10.25*** join/#asterisk SteveTotaro (n=Administ@pool-70-17-230-174.balt.east.verizon.net)
20:11.45SteveTotarohey guys, i want to do a dial on the h exten using asterisk 1.2 has anybody done this, with a regular dial, it tries and "Cancels" I believe because there is only one channel at that point
20:12.20*** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
20:12.20SteveTotaroI am trying to use local channels but I get caught in a loop and I lose the variables that were set in a macro
20:14.29[TK]D-FenderSteveTotaro: No, there is no channel at all.  The call is DEAD
20:15.08*** join/#asterisk xacatecas (n=jkroon@196.46.173.152)
20:15.46xacatecashi all, hope all is well.  i'm new to the isdn arena and was hoping someone can point me at some good resources.
20:15.59*** join/#asterisk bbryant (n=Brett_Br@adsl-153-53-11.chs.bellsouth.net)
20:16.08soulfreshneris there a simple sample dialplan out there?
20:16.24*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
20:16.33soulfreshnerthe one included with the asterisk installation is a bit over the top for me
20:16.53[TK]D-Fendersoulfreshner:...
20:16.56[TK]D-Fender~jerjerguide
20:16.56jbot[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * :  http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/
20:16.57xacatecasi'd like to learn a little bit about the technology, specifically, i've got a potential client that claims he's got 10 BRI's along with 5 BRI based premis plugged into 3 x 4-port Digium BRI cards ... but 15 > 12 ... so I can't figure how that works.
20:16.59[TK]D-Fender^^^
20:17.19afinkHey guys is there any specific linux distro that seems to work best with asterisk?  I was using opensuse but I wasn't able to use digium HPEC software b/c suse doesn't use incremental init scripts, then I tried Debian etch which gave me kernel panics and now I am using debian etch n half that just stops working almost every 24 hrs.  Can someone point me in the right direction please?
20:17.44soulfreshnerthanks [TK]D-Fender :)
20:17.45[TK]D-Fendersoulfreshner: the sample one includes tons of crap that is not necessarily sanely assembled.  This above guide has a very minimalistic structure for you to get an idea from
20:18.13[TK]D-Fenderafink: Debian is normally OK if you know what you're doing, as is CentOS (my recommendation)
20:18.30afinkThanks [TK]D-Fender
20:19.43SteveTotarothe call is not dead
20:20.02SteveTotaroi can do a dial from the h exten using a local chan
20:22.46gr0mitafink, i use Debian on all my boxes
20:22.56gr0mitit Mostly Just Works
20:23.12afinkgr0mit: What version and are you using any HPEC software?
20:23.29gr0mitam using etch on two production asterisk boxes
20:23.36*** join/#asterisk jtodd (i=bqs9e3jl@ns2.loligo.com)
20:23.55gr0mitHPEC??
20:24.13afinkDigium High Performance Echo Cancellation for TDM cards
20:24.34gr0mitah nope
20:24.49sapereIf I have only one phone line and I want to add fax functionality, I'm kinda screwed right? I mean, since fax machines don't know how to enter an extension... right?
20:24.56gr0mitno reason to believe they would not work though!
20:24.56bochwhere can i read main diff between * 1.4 and 1.6  ?
20:24.57afinkthat seems to be the source of all my debian problems.  It worked fine before I installed HPEC
20:25.44[TK]D-Fendersapere: "go read up on
20:25.54[TK]D-Fendersapere: "go read up on "Asterisk Standard Extensions" on the WIKI
20:25.56[TK]D-Fender~wikis
20:25.56jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
20:26.05[TK]D-Fenderboch: changelog.txt
20:26.11[TK]D-Fenderboch: upgrade.txt
20:26.14sapere[TK]D-Fender : thanks
20:27.47sapere[TK]D-Fender : another question for ya: if my ITSP gives me 4 channels of service, and each channel allows me a separate incoming or outgoing call, what does this mean for bandwidth? In other words, what does asterisk generally use per channel?
20:28.09saperei've read some conflicting information, including some saying as little as 16k per channel
20:28.55[TK]D-Fendersapere: go read up on "bandwidth consumption on the WIKI.  this is protocol, and codec-sensitive
20:29.01[TK]D-Fenderout for now, back later
20:29.20*** join/#asterisk amiral_ (n=amiral@gob75-2-81-56-65-10.fbx.proxad.net)
20:29.46amiral_hello ;) how can i have email => fax  and fax => email with asteris
20:30.12*** part/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com)
20:34.23*** join/#asterisk kraptv (n=ryan@magic.skylab.org)
20:34.30x86amiral_: take out asterisk and insert hylafax
20:34.56kraptvDoes anyone know if there's any ztdummy workaround for some of the stranger operating systems like OSX (Darwin) ?
20:36.40kraptvOr, another way of putting it, is there an alternative to MeetMe that doesn't rely on zaptel timings?
20:37.33Kattyi had some zaptel problems on boot. it said to check dmesg.
20:37.40Kattybut dmesg doesn't show the zaptel errors.
20:38.36Kattychan_zap doesn't appear to be loaded. it was complaining about lib modules
20:38.49Kattyany idea where to start on this?
20:41.09amiral_x86 hylafax can send fax over sip ?
20:41.23tzafrir_laptopKatty, what libs?
20:41.28Kattyhttp://pastebin.ca/1200408
20:41.50Kattytzafrir_laptop: still looking for the lib thing i saw.
20:42.54tzafrir_laptopKatty, is that pastebin related?
20:42.56Kattytzafrir_laptop: does that pastebin mean anything to you?
20:42.57Kattyaye
20:43.03*** part/#asterisk wacky__ (n=root@nat/digium/x-4763810caeefbe6b)
20:43.09Kattyi'm not sure what i'm looking at really
20:43.27tzafrir_laptopit means that the kernel module "wanpipe" failed to load
20:44.20tzafrir_laptopbecause of missing exports such as wanpipe_ec_register
20:44.25Kattyhmm.
20:44.29*** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net)
20:44.51Kattydo you think i should try to recompile wanpipe?
20:44.51jameswf-homeI dont know why pastebin.ca hates me.
20:44.54x86amiral_: not over sip, but using iaxmodem you can send faxes via IAX2
20:45.54sapereWould using IAX over SIP be preferable with an ITSP, if it were available as an option?
20:46.00tzafrir_laptopKatty, no idea
20:46.18Kattyk
20:47.45tzafrir_laptopgrep in the sources for EXPORT_SYMBOL(wanpipe_ec_register)
20:48.02x86sapere: it's less bandwidth if you're running multiple concurrent channels, but in my experiences, sip seems more reliable
20:48.03*** join/#asterisk zydoon (n=zydoon@41.225.153.182)
20:48.22x86amiral_: I wouldn't recommend fax over IP at all... it's never really reliable
20:48.25*** part/#asterisk zydoon (n=zydoon@41.225.153.182)
20:48.44x86amiral_: just setup a fax server with some regular old external faxmodems, and hylafax
20:49.48amiral_x86: problem, my company don't have pstn
20:50.05amiral_x86: big problem...
20:51.55x86very big problem if they want reliable faxing
20:52.07x86perhaps you can solve that problem by installing some POTS lines?
20:53.03*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
20:53.23Kattytzafrir_laptop: i'm just recompiling wanpipe
20:53.33Kattytzafrir_laptop: i have no idea what's wrong
20:53.40Kattytzafrir_laptop: but i recall i updated grub.
20:53.48MindTheGap_hello all, how do i take advantage os an md5 password already in a database to authenticate users? our md5 password contain only the pasword as there is samba, ldap, etc authenticating against it, but asteris expects a hashed "peer:realm:secret" in the db not a hashed "secret".
20:53.49Kattytzafrir_laptop: maybe that did something funky.
20:54.07MindTheGap_s/ldap/imap/
20:54.26tzafrir_laptopKatty, strange stuff in /proc/cmdline ? yeah, a very longshot
20:54.51jameswf-homeMindTheGap_: AGI
20:56.51*** part/#asterisk kraptv (n=ryan@magic.skylab.org)
20:57.17MindTheGap_jameswf-home, sorry but i dont get it. how will an agi help me?
20:57.17*** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net)
20:57.40tvirusIf someone calls in on a land line (zap module) is it possible to transfer that over to SIP to free up the land line?
20:57.59Kattytzafrir_laptop: i really don't know :/
20:58.12Kattytzafrir_laptop: i was tinkering with the grub splash image and it suggested to apt-get upgrade it...
20:58.18Kattytzafrir_laptop: so i updated grub.
20:58.26Kattytzafrir_laptop: reboot, and wanpipe spewed.
20:58.55Kattytzafrir_laptop: only thing i can think of.
20:59.08Kattytzafrir_laptop: i'm fscking the disk right now, just to make sure
20:59.58*** join/#asterisk Levonk (n=lk@adsl-76-230-108-247.dsl.lsan03.sbcglobal.net)
21:02.24Kattytzafrir_laptop: well the recompile worked.
21:02.29Kattytzafrir_laptop: i dunno know really happened
21:02.37*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:02.42tzafrir_laptopdifferent kernel?
21:03.01jayteequittin time, be back later
21:11.06ManxPowertvirus: generally No
21:13.54legisHow do I send a call via SIP without a username?
21:14.20lmadsen...
21:14.29lmadseneh?
21:14.45legisheh
21:16.01legisIs that i'm dialing this VSP and the call goes through fine through console, which I suppose it doesn't send a username?
21:17.09mchouwhich VSP is this?
21:17.11legisIf i call from the softphone it gets rejected
21:17.43legisdialing through this VSP I mean
21:17.59legismchou: one that you probaly haven't heard of
21:18.09mchoulegis: try me anyways
21:18.21WimpManI think the console sends "asterisk" just as an originate will.
21:18.42mchouyup, agree with WimpMan
21:19.00legismchou: UNE, in Colombia
21:19.53legisWimpMan: nice, the softphone is using a number instead of letters, so maybe thats it, let me try
21:20.37Kattytzafrir_laptop: hrmm, no. i didn't do a kernel update.
21:20.53Kattynot that i can remember, anyway ^_-
21:20.57WimpManCould it be the choice of codec?
21:22.11legisno, cause if I use a DID of the them as the user, the call works too.
21:23.57*** join/#asterisk PakiPenguin_ (n=junaid@linuxpakistan/admin/pakipenguin)
21:23.59PakiPenguin_http://www.youtube.com/watch?v=NLRS1b5-Kg8
21:25.31legisWimpMan: thanks!, looks like they only accept letters or they own DID
21:26.39*** join/#asterisk hfb (n=hfb@96.247.52.72)
21:26.54*** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583722.dsl.bell.ca)
21:28.34sapereexit
21:28.44*** join/#asterisk n3hxs (n=HAMming@151.196.87.132)
21:28.57legisnow I have to test up fax with them :/
21:30.32*** join/#asterisk great_Anta_Baka (i=c636caf6@gateway/web/ajax/mibbit.com/x-e0ded14cd1bb3220)
21:30.37*** join/#asterisk hmodes (i=hmodes@B1-66ER.matrix.gs)
21:30.52legisis there an app to send faxes?
21:30.53great_Anta_Bakahow do i get rid of this how do i get rid of this linux/compiler.h: No such file or directory?
21:31.03great_Anta_Bakatrying to install h323 support for asterisk
21:31.53legisgreat_Anta_Baka: looks like you are missing dependacies
21:32.18great_Anta_Bakathat file doesnt exist in the kernel any more
21:32.35legisgreat_Anta_Baka: what distro?
21:32.46great_Anta_Bakathis is a fedora box
21:33.13legisdoes it have the kernel headers installed?
21:33.13great_Anta_Bakaits when i am compiling open h323
21:33.33great_Anta_Bakai can install them again but linux/compiler.h doesnt exist in the headers
21:33.42great_Anta_Bakait was removed in an earlier kernel version
21:34.13legisinstall the kernel headers
21:34.18great_Anta_Bakai have done this before just cant remember what i did :(
21:34.22ManxPowergreat_Anta_Baka: What kernel version are you running?  What version of Asterisk?  What version of Asterisk-Addons?
21:34.40*** join/#asterisk StooJ (n=stooj@stooj.plus.com)
21:34.41great_Anta_Bakalatest asterisk on the digium page
21:34.59ManxPowergreat_Anta_Baka: I am not going to go to Digium's page to look that up.
21:35.07ManxPowerAnyway, best of luck.
21:35.18great_Anta_Baka1.4.21.2 or something like that
21:35.23ManxPowerI was not aware that Digium had linux kernels for download.
21:35.40great_Anta_Baka2.6.18-53.1.4.el5
21:35.44great_Anta_Bakakernel version
21:36.02great_Anta_Bakai mean asterisk version
21:36.22great_Anta_Bakabut this hasnt got to do with asterisk
21:36.29great_Anta_Bakaasterisk isnt compiled yet
21:36.37great_Anta_Bakaneed to install openh323 first
21:36.48ManxPowerwell nevermind then
21:37.04ManxPowerSorry, I thought you were having Asterisk problems.
21:37.16*** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net)
21:37.23great_Anta_Bakalegis kernel headers installed
21:37.35ManxPowergreat_Anta_Baka: which of the 3 or 4 H323 drivers do you want to install?
21:38.00great_Anta_BakaManxPower: installing from this guide http://astrecipes.net/index.php?from=40&q=AstRecipes/Compiling%20Asterisk%201.4%20with%20TDM400%20and%20H323
21:38.06scooby2Another strange one. Agent logs out. Directly as logging out, a call goes to him. It cancels the logout thus leaving him logged in.
21:39.09ManxPowergreat_Anta_Baka: I suggest you use the one in Asterisk-addons
21:39.47great_Anta_Bakai see... do i compile asterisk then the addons and then asterisk again?
21:39.58ManxPowergreat_Anta_Baka: you follow the instructions for it.
21:40.26great_Anta_Bakakk
21:40.27ManxPowerthe one in asterisk-addons is the one DIGIUM PAID FOR.
21:41.33great_Anta_Bakaaccording to to voip-wiki asterisk addons on has 4 packages and none of them has h323 in it o_0
21:41.37*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
21:41.56ManxPowergreat_Anta_Baka: perhaps you should look at the asterisk-addons package rather then the old and outdated information on the Wiki.
21:43.21great_Anta_Bakammm looking
21:43.25ManxPowerYou should report it as a bug if the H323 channel driver is not in asterisk-addons
21:45.12scooby2would hitting dnd before trying to logout help?
21:47.55*** join/#asterisk nr4q (n=dave@12.160.78.184)
21:52.35great_Anta_BakaManxPower: after installing the asterisk-addons package, do i have to ./configure and make asterisk again?
21:52.54ManxPowergreat_Anta_Baka: I have no idea what you have to do.  I assumed there would be instructions.
21:53.00great_Anta_Bakacos in the make menuselect of asterisk chan_h323 still has [XXX] next to it
21:53.07great_Anta_Bakano instructions
21:53.44ManxPowerThen I guess you need to report a bug
21:53.55ManxPowerand it is NOT chan_h323
21:53.59great_Anta_Bakawell restarted asterisk and did module show like h323
21:54.03ManxPoweras you can see in the source code in the asterisk-addons.
21:54.04great_Anta_Bakaand it seems like its loaded
21:54.07ManxPowerit is chan_ooh323
21:54.08great_Anta_Bakaso thanks :)
21:54.20great_Anta_Bakayeah
21:55.13great_Anta_Bakaoh and how do i stop my iax trunk from reregistering every 10-20 seconds?
21:56.05[TK]D-Fendergreat_Anta_Baka: "noload => chan_iax2.so" in modules.conf
21:56.25*** join/#asterisk Paige (n=Paige@208.89.241.31)
21:56.40[TK]D-Fender;)
21:56.44Paigehi, i need some help getting mysql realtime to work
21:56.46great_Anta_Bakalol
21:56.56*** join/#asterisk Levonk (n=lk@adsl-76-238-248-235.dsl.lsan03.sbcglobal.net)
21:57.15[TK]D-Fendergreat_Anta_Baka: Could be your provider is insisting on that timeout.  Otherwise hit up the WIKI page for iax.conf and I'm sure you'll find it...
21:58.28Paigemy error is: ERROR[8206]: res_config_mysql.c:845 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on /var/lib/mysql/mysql.sock (err 1045).
21:58.39great_Anta_Bakammm well i set the minexpire time to 60 on the side where the trunk is registering to
21:58.44great_Anta_Bakawill see if that changes anything
21:58.46*** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
22:00.09Paigecan someone help me trouble shoot this error?
22:01.59scooby2does /var/lib/mysql/mysql.sock exist?
22:02.06Paigeyes
22:02.20great_Anta_Bakaok so now that chan_ooh323 is installed... in my sip.conf file do i just put allow=h323 for the user that?
22:02.32scooby2username/password exists on the mysql db you are trying to use?
22:02.45great_Anta_Bakahttp://www.google.co.za/url?sa=t&source=web&ct=res&cd=1&url=http%3A%2F%2Fforums.mysql.com%2Fread.php%3F11%2C19531%2C19531&ei=dpXJSI27B6TIsgKrpvXUBg&usg=AFQjCNEhCbVzgWUDqpEmpjYFBSD0t1jPEg&sig2=cLvq0NGEsB_EtLR97OolVg
22:02.57Paigematches in res_mysql.conf and works by command line
22:03.06great_Anta_Bakaaccess denied
22:03.07great_Anta_Bakaerror
22:03.37great_Anta_Bakacheck your privileges Paige
22:03.49Paigei did
22:04.23scooby2set the logs to include verbose and debug? What does messages say?
22:04.32*** part/#asterisk zerohalo (n=zeroHalo@75.150.77.161)
22:05.20great_Anta_Bakadid you set up a custom user to connect the mysql database or you using the root user?
22:05.40Paigecustom user
22:06.02great_Anta_Bakaand what grant command did you use?
22:06.19Paigeeverything is yes
22:07.17jameswf-homeAs mom said in mysql dont forget to flush
22:07.24great_Anta_Bakalol
22:07.28great_Anta_Bakamy thoughts exactly
22:07.44great_Anta_Bakaalso just see if it works connecting with the root user to the database
22:07.46scooby2Paige said that he/she could login manually
22:07.50Paigei lushed
22:07.57Paigeflushed
22:08.09jameswf-homedid you jiggle the handle?
22:09.03great_Anta_Bakaare you connecting with the -h option?
22:09.32Paige?
22:09.47great_Anta_Bakamysql -h 127.0.0.1 -u root -p
22:09.57great_Anta_Bakaor mysql -u root -p
22:09.59Paigeyes
22:10.06great_Anta_Bakayes?
22:10.13great_Anta_Baka<PROTECTED>
22:10.35Paigemysql -u astuser -p
22:11.10great_Anta_Bakaany white spaces in your cofig file? not that i think that makes a difference
22:11.22Paigeno whitespace
22:11.22great_Anta_Bakajust check if you can log in with your root mysql user
22:11.28great_Anta_Bakain the asterisk conf file
22:11.48great_Anta_Bakai mean the mysql
22:11.51Paigethe database config is in res_mysql.conf
22:11.57great_Anta_Bakayes
22:11.59great_Anta_Bakathat one
22:12.12*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
22:14.56Paigei found it. typo
22:15.34scooby2lol
22:15.43scooby2always something simple
22:17.29great_Anta_Bakaanyone done any video through asterisk.. i dont know where to start other than having installed chan_ooh323
22:20.32Paigethanks for the help guys!
22:20.39*** part/#asterisk Paige (n=Paige@208.89.241.31)
22:26.29j0does anyone have a nice pdf of the voicemail options?
22:26.42j0i guess every system is different, but at least one for the generic comedian mail
22:30.34*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
22:36.29hohumhey guys
22:36.43hohumwhat happened to the "odbc show" command int he latest version of asterisk 1.4
22:37.45hohumhelp please
22:40.35hohumnever mind I don't have res_odbc even compiled
22:46.12*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:49.33jayteeanyone know of a cheap analog line audio coupler that can plug into a line jack, seize the line in an off-hook condition and has an audio output like a handset without a mike?
22:50.26*** join/#asterisk sapere (n=sapere@c-71-206-107-228.hsd1.mi.comcast.net)
22:50.33*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
22:51.27pjzlike a cheap phone?
22:51.31WimpManjaytee: Any old modem?
22:52.04*** part/#asterisk pjz (n=pj@zachs.place.org)
22:52.14jayteeyeah, but I only need to recieve audio, don't need mic output on the analog line, don't need a dtmf keypad either.
22:55.14jayteethis would be used in place of a phone so I can use it with an SPA3102 FXS port setup as a hotline to an extension on Asterisk that answers and immediately plays musiconhold so I can take the audio output and hook it into the music channel of a PA system. The PA side works with an FXO port but I can't use the MOH on that channel because the Page application doesn't support MOH.
23:00.52*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
23:04.01*** join/#asterisk imcdona (i=imcdona@2001:5c0:8fff:fffe:0:0:0:b917)
23:05.01*** join/#asterisk tmccrary (n=tmccrary@d14-69-192-41.try.wideopenwest.com)
23:05.18tmccraryCan anyone recommend a somewhat decent sip termination service provider?
23:06.35bminishtmccrary, where are you ?
23:06.44tmccraryoh sorry, midwest usa
23:07.31bminishok then blueface.ie will not be that great for you (they also do IAX trunking  if you ask them.)
23:07.40mchoutmccrary: what's you definition of decent?
23:07.58mchoutmccrary: and what's your call volume min/mo?
23:08.04bminish!crap I would guess ;-)
23:08.04mchouyour*
23:08.19mchoubminish: lol
23:09.25mchoubminish: would that include vonage? :)
23:10.03tmccrarydecent, something that is fairly reliable
23:10.10bminishdunno, never used them, happy with blueface.ie though
23:10.14tmccrarycall volume very low, like maybe a few calls a weak
23:10.18tmccrary*week
23:10.39mchoutmccrary: if you call volume is low go with these guys: diamondcard.us
23:10.59mchoutmccrary: no affliation except a reasonably happy customer
23:11.11tmccrarythanks I'll check it out
23:11.43mchouDIDs in all US and termination a-z
23:12.10mchoutmccrary: dont be discourage by their mickey mouse kindergarten web site
23:12.18mchoudiscouraged*
23:12.59*** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au)
23:13.10phixhey
23:13.40*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net)
23:16.55*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
23:16.55*** mode/#asterisk [+o mog] by ChanServ
23:17.46*** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
23:17.46*** mode/#asterisk [+o russellb] by ChanServ
23:18.26jayteemy boss just got approval for me to go to Advanced Asterisk training, YAY!!!!!
23:18.34russellbjaytee: nice!!
23:18.44*** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.158.28)
23:18.45russellbjaytee: it sounds like a pretty cool course
23:18.50jayteeit does
23:19.06russellbwants to take it
23:19.06russellb:)
23:19.21jayteerussellb, are you in Huntsville?
23:19.25russellbyes, i am
23:19.36jayteecool, maybe we can do lunch!
23:19.40russellbsure!
23:19.49fileit's a trap!
23:19.52russellbjust let me know when you're here, and I can come eat lunch with the whole group
23:20.38jameswf-homejaytee: has a big club...
23:20.44russellbo.O
23:20.46mchouThere's no such thing as a free lunch :)
23:20.58jayteeI actually kind of had the choice between Huntsville and Vegas but Vegas just isn't anything that thrills me. not a gambling man and I don't care for hookers.
23:21.20russellbHuntsville has ... um ... cotton fields
23:21.22jayteebut I figured I'd stay an extra day and see if I can get into a tour of the Marshall Space Flight Center
23:21.26russellband a space center
23:21.27mchoujaytee: what attractions does Huntsville have?
23:21.37mchouok, besides the space ctr
23:21.41jayteeI'm too old for Space Camp :-)
23:21.47*** join/#asterisk SteveTotaro (n=Administ@pool-70-17-230-174.balt.east.verizon.net)
23:22.01*** join/#asterisk Levonk (n=lk@adsl-76-227-116-162.dsl.lsan03.sbcglobal.net)
23:22.05jameswf-homeI wanna go to the flea market in MG
23:22.09russellbyes!
23:22.51jameswf-homei hear its just like a minimall
23:23.36jayteeMG?
23:23.41russellbMontgomery
23:23.45russellb~nowwhat
23:23.45jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
23:23.49russellbsee the video
23:23.59jameswf-homecant spell Montgomery
23:24.22russellbhuh ... what's with the music on that one
23:24.23russellblame
23:24.38jayteeok, that's almost scary!
23:24.46russellbanyway, search for "flea market montgomery"
23:24.48jameswf-homehas leekspin as his MOH it is awesome
23:26.50mchouI dont even get the concept
23:27.08mchouthe whole flea market is owned by one guy??
23:27.39mchouhe must have lots of crap if he can fill 73000 sq ft :)
23:27.54russellbit's just like a mini-mall, obviously
23:28.23mchourussellb:  Is it mostly new or used stuff?
23:28.29russellbi have never been there
23:29.16mchouahh, too bad.  I was hoping you'd enlighten me
23:31.02jayteeok, I'm looking for something similar to this little gizmo but I need it to act like a phone that's off-hook, not a passive tap into an extension line. http://www.omnicronelectronics.com/analog/lic-390.htm
23:31.17jayteeanyone know of anything "cheap"
23:32.53*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:33.16*** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net)
23:34.22mchouanyone here ever installed HW for a telemarketing call center?
23:35.48russellbtelemarketing is eeeeeevil
23:36.13mchourussellb: I know it's evil.  that's why I'm seeking info
23:36.29russellbedits the Asterisk license the exclude telemarketer's from being allowed to use it
23:36.39mchoulol
23:36.44Rienzillahehe
23:37.00Rienzillamaybe asterisk could be patched to include an equivalent of the evil bit
23:37.11Rienzillathe m extension, for telemarketeers
23:37.23mchourussellb: I'm sure the telemarketers will find an escape clause somewhere.
23:37.50russellbwell, we can't retroactively change the license of every version previously released
23:38.24jameswf-homeyou can if you hire chuck norris
23:38.32mchouhaha
23:39.48mchouThis is what REALLY irritates me:  http://800notes.com/articles/News.aspx/0PX8Dr_Y_AAcJgjKvtDs0g
23:40.25mchouyou break the law, get a wrist slap
23:40.40*** join/#asterisk onescomplement (n=dave@mail.davefuller.net)
23:41.16mchou"Based on the companies' ability to pay, the amounts were reduced to $20,000 and $75,000 respectively."
23:41.20mchouwtf??
23:42.49mchouAnd I dont understand why the govt didmt go after dish network instead of their contractors
23:42.57mchoudidnt*
23:45.38jameswf-homehttp://dontcallmyboss.blogspot.com/2008/09/epic-battle.html << hitler beat Chuck norris 5 to 1
23:48.36*** part/#asterisk tmccrary (n=tmccrary@d14-69-192-41.try.wideopenwest.com)
23:50.24mchouso here is a question.  suppose at telemarketer calls. I use asterisk to defeat AMD and get a live person on the line.  Then I play this: "The number you have called is on the Federal "Do not call" registry.  This call may be recorded.  You will risk legal action if you call again."
23:50.58mchouDoes this have teeth or am I whistling in the wind?
23:53.29jayteemchou, whistling into the wind
23:53.40mchoujaytee: why do you say that??
23:54.13mchoujaytee: even if I get a live telemarketer?
23:54.31Rienzillawhat is AMD?
23:54.52mchouAnswering Machine Detection (on their end)
23:55.09Rienzillaah
23:55.25jayteeDo Not Call has not real teeth. Legal action? yeah, so you sue, the lawyers settle, your lawyer keeps their cash and bills you an additional amount. feelin better yet?
23:55.49mchousure I fell better.  They wont bother me again
23:55.54mchoufeel*
23:56.06jayteemchou, wanna know my solution?
23:56.14mchouthat's the whole point
23:56.21mchoujaytee: sure
23:56.57mchoujaytee: plus there's no lawyer involved.  It's all small claims
23:57.18mchoucause each fine I think is <$5K
23:57.47jayteeI filter by callerid, if it's blocked they get a message that says, "I don't know who you are so go away". If it's any number with an 800 NPA I route it into a queue with music on hold and announcement every 30 seconds that says, "Your call is unimportant to us, please continue to hold and someone will be with you when they are bored senseless"
23:58.10mchoujaytee: I do something similar already
23:58.33jayteeyou could also do Zapateller and then dump them in a queue
23:58.42mchoubut that's not enough.  I dont want them to call "on my dime," as it were
23:59.26mchoujaytee: I do Zapateller already.  You know that doesnt work cause telemarketers get PRI signalling

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