IRC log for #asterisk on 20090501

00:03.24*** join/#asterisk joobie (n=joobie@mx01.anric.com.au)
00:12.26exothermctrying to compile asterisk 1.4.24.1 and I get: http://ja.pastebin.ca/980216   any idea how to solve this?
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00:39.54*** part/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com)
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01:07.46KyleKaskerisk is NOT a sip proxy right?
01:07.52*** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924)
01:08.04Kate-oHey everyone
01:08.17Kate-oI have a question about different possibilities in FXSs and FXOs
01:08.34KyleKfire away?
01:08.54Kate-oI'm trying to configure a Linksys PAP2 with an X100P
01:09.16Kate-oand I haven't gotten any progress.. Do they make phone adapters that will serve as an FXO?
01:09.59*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com)
01:11.38KyleKlike a SPA3102?
01:11.49KyleKits for FXO+FXS
01:12.13Kate-oyeah!
01:14.48theharrussellb: did you see mirrors.xmission.com is load balancing off the link on asterisknow downloads now? :D
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01:57.43XLA187On a fresh install of Asterisk and Asterisk-GUI, I am getting a perpetual Invalid Username Password at the login no matter what I do to the config files, has anyone ran into this?
01:58.39jayteeXLA187, nope but you might have better luck asking in #asteriskgui
01:58.56jayteeoops, that's #asterisk-gui
01:59.07XLA187good idea
01:59.33jayteethis channel is mostly just for straight non-gui asterisk installs and issues
02:00.33XLA187which is about what I am going to have to do, as I have been dicking with this thing for two days
02:00.44*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
02:00.52XLA187I have not used asterisk in awhile, so I figured I'd try this new fangled gui thing
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02:20.25kc8pxyi was just looking, somewhat serious, but somewhat just to look, and i saw a  listing for a ds3 card from digium, but i don't see it on the list of products.   is it gone, or is there something else i should be looking for?
02:20.59Qwellkc8pxy: I don't know what you're talking about
02:21.59coppiceQwell: it will still be on a page at some internet archive site :-)
02:22.21Qwellall lies
02:23.00jayteeDigium does not make a card for DS3 circuits
02:23.24coppiceIt had one on its site for a while
02:23.31jayteereally?
02:23.58jayteea DS3 is essentially 28 DS1's muxed
02:24.34coppiceand all potentially running at slightly different clock rates :-)
02:24.44kc8pxy<PROTECTED>
02:24.46jayteetalk about a nightmare
02:25.21jayteeon another topic, is Germany ready for russellb? will the European Union survive this?
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02:25.57*** mode/#asterisk [+o leifmadsen] by ChanServ
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02:52.37securevoipIAX2 is vulnerable to remote DoS attacks. Digium hasn't fixed it in 8 months time http://tinyurl.com/csmadz???
02:53.28jaytee404 not found dude
02:54.36securevoipahh, the author posted it on linkedin.  apparently you have to have an account to see his exploit.
02:54.56jayteeso much for hysterical claims then
02:55.24securevoipjust wondering what it is all about.  he even has the exploit code posted...
02:57.56jayteethat subject matter is over 2 years old and the vulnerability has been patched IIRC over a year ago.
02:59.18securevoipIIRC???
02:59.27jayteeIf I Recall Correctly
02:59.52securevoipsorry, didn't know that one.
03:00.04securevoipthanks!
03:00.17jayteeyw
03:00.34jayteeum, that's short for you're welcome :-)
03:00.44securevoipfigured that one out.  ;-)
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03:41.45littleballhello, who can help me setup gtalk on asterisk?
03:42.02littleballi have configured but i cannot not make call from gmail to asterisk
03:42.26littleballi have configured two files: one is jabber.conf and the other is gtalk.conf
03:42.44littleballasterisk-1.4.21.2
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04:12.59Juggiethis is a stupid problem, any reason why on a tdm400p when i dial into an context and use backgrounddetect() to play a file it looks for the slin format but not the wav?
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04:15.25littleballhi, i am trying to have call from gmail to asterisk. from CLI, i can see the google session initiate iq package come in, but asterisk does not send any response to gtalk. so, gtalk keeps ring.
04:15.32littleballwho can help ?
04:18.04*** part/#asterisk XLA187 (n=jchase@189.154.32.251)
04:23.40KyleKgoogle supports sip?
04:26.37hardwiredoes echocancel need to be on if you have an octastic module?
04:27.37KyleKlittleball: wheres the option for a sip call?
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04:59.07hardwirewhere the crap did I ever download the octastic firmware?
04:59.10hardwirecause like..
04:59.13hardwirewtf
05:06.08hardwireok.. apparently the VPM450 onborad my card is failing.
05:06.28hardwirewith echocancel enabled in zapata.conf the load on my system skyrockets for inbound calls.
05:06.41hardwiredisabled.. works fine..
05:06.56hardwireis there a new firmware I should be using?  I can't seem to find any known issues on this.
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05:28.07AJayMNI found a patch that adds pass-through for h264 on * 1.2 but not sure how to use it.   http://web.archive.org/web/20070429103830/http://asterisk-backports.org/wiki/index.php/Passthrough-h264
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05:40.14ambush276hey guys does anyone know the program skype to SIP
05:40.18ambush276ny nhc
05:40.20ambush276nch
05:40.22ambush276or w/e
05:40.39ambush276anyway when i try and make an outgoing call teh bridge gives me a busy signal? anyone knw why?
05:41.38hardwiredid you get a sip for skype invite?
05:41.41hardwire<- jlus
05:41.44hardwireonly because I don't have one
05:41.47hardwirenot sure if I'd use it :)
05:45.42ambush276wjat
05:45.49ambush276hardwire
05:45.56ambush276invite?
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05:48.04hardwirewhat?
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07:12.32Ziggy3000has anyone used Fring (SIP) with Asterisk?
07:16.07MaliutaLapwhat does google say?
07:17.41Ziggy3000Probably yes. Just wondering how the quality is?
07:43.23*** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) [NETSPLIT VICTIM]
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08:05.51goupilsalut
08:07.14Ziggy3000hello
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09:14.57n3glvanyone awake?
09:33.33EmleyMoorn3glv: I certainly am
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09:37.23*** join/#asterisk defiancenl (n=jefrecha@m0n0.hix.nl)
09:37.31defiancenlhi all
09:37.35n3glvhi EmleyMoor
09:37.41defiancenlanyone here got res_jabber.so to function ?
09:37.42n3glvand defiancenl
09:37.58defiancenlim running into an error , and cant seem to google my way out this one
09:38.29defiancenlall i get are links to developers logs / source code etc, but no actual usefull stuff
09:39.50*** join/#asterisk shinao1 (n=shinao1@78.138.29.146)
09:39.55EmleyMoordefiancenl: What version of asterisk?
09:40.09defiancenl1.4.24.1
09:40.34EmleyMoorI have it working in a 1.4... what problem are you finding?
09:40.54defiancenlasterisk cant seem to connect/authenticate to my xmpp server
09:41.04defiancenllet me trow it on pastebin
09:41.07defiancenlhold on :)
09:41.27plantseekerhmmm
09:42.20defiancenlf$#@$
09:42.25defiancenlpastebin is unreachable :(
09:42.36n3glvdot ca or dot com?
09:42.45EmleyMoorUse paste.debian.net perhaps
09:42.56defiancenlnvm , it was my dns , i kicked it >:)
09:43.57defiancenlhttp://pastebin.com/m555eb747
09:44.25EmleyMoorhopes his web browser starts
09:44.30defiancenlheh
09:44.54defiancenlso anyone in here heard about the "hit" on the royal dutch family yesterday ? :)
09:45.32EmleyMoordefiancenl: Yes
09:45.43defiancenlsick wasnt it :X
09:45.44EmleyMoorGot a friend in Apeldoorn
09:46.01defiancenlthe guy plowed trough the crowd like they where bowling pins :X
09:46.10EmleyMoorAt least they don't now have to decide how long to lock him up for
09:46.34defiancenlyeah to bad he gave it ....
09:46.34EmleyMoorIs your Jabber server otherwise working?
09:46.37defiancenlye
09:46.55defiancenlwe use it @ office to communicate
09:46.57EmleyMoorIs Asterisk's account registered?
09:47.11defiancenlsure :)
09:47.38EmleyMoorA traceroute betwoon the two machines may help
09:47.42EmleyMoorbetween
09:51.23defiancenlhummm it wont even connect to talk.google.com :X
09:52.13defiancenlthing that worries me : res_jabber.c:1573 aji_recv_loop: JABBER: socket read error
09:52.40EmleyMoorHmmm - this sounds like firewalling
09:52.58defiancenli tried using an external php script to do jabber, but the execution takes to much time
09:53.05defiancenlfirewall is empteh
09:53.17defiancenl[root@asterisk /]# iptables -L
09:53.17defiancenlChain INPUT (policy ACCEPT)
09:53.17defiancenltarget     prot opt source               destination
09:53.18defiancenlChain FORWARD (policy ACCEPT)
09:53.18defiancenltarget     prot opt source               destination
09:53.20defiancenlChain OUTPUT (policy ACCEPT)
09:53.22defiancenltarget     prot opt source               destination
09:53.30EmleyMoorNo idea in that case -
09:53.47defiancenlsame as me then :)
09:54.02defiancenldamned it tho, i got everything working as it should
09:54.12defiancenlexcept this litte thing, and that annoys the shit out of me
09:56.42defiancenl<stream:error><not-authorized xmlns
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09:59.55defiancenlis what i get back from gtalk
10:00.07defiancenlemley waht are u using as a jabberserver ?
10:00.09defiancenlxmpp ?
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10:08.11defiancenlnoone else :(
10:12.27EmleyMoorejabberd
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10:45.25SparFuxHello all. I get this error for my ekiga after update, once it worked. I just change the ekiga config probably, because it was a new version and I lost the old config: chan_sip.c:14035 handle_request_invite: Call from '' to extension '266663' rejected because extension not found.
10:45.44SparFuxBut the extension is there. I get into the same context with other phones and the extension works.
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10:53.07SparFuxAha! [May  1 12:52:39] DEBUG[19145]: chan_sip.c:15531 sipsock_read: Invalid SIP message - rejected , no callid, len 351
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11:23.54virtualme123I'm using the manager API to originate a call, however I want to collect the Originate response and just can't see how to get the reason status. I have tried setting 'async' to true and setting an ActionID, but don't understand how to get this reason status ...
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11:40.58defiancenlemley --> ejabberd ?
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12:11.13defiancenlhummm @ emley
12:11.18defiancenlsuddenly it worked :
12:11.20defiancenl:X
12:11.29defiancenlahh then my project is finished :D)
12:20.52viraptorwhen using a remote attended transfer, is there any way to make asterisk call the uri from refer-to, instead of trying to resolve the destination through the local dialplan?
12:25.09[TK]D-Fenderviraptor: No.  * is not a proxy, it is a B2BUA
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12:29.24viraptor[TK]D-Fender: yeah, but if something sends back "Refer-to: XXX@ip", isn't it wrong to assume that asterisk knows anything about XXX?
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12:38.14viraptorI'm tempted to report it as a bug, but just wanted to get more info...
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12:44.35*** join/#asterisk Raffles666 (n=bgates@datavox.plus.com)
12:45.12Raffles666hello.  has anyone encountered (and fixed) the Queue app not sending ring tone to the caller?
12:48.34Raffles666v1.4.24.1 btw
12:49.12Raffles666wonders if there is anybody there....
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12:52.47[TK]D-Fenderviraptor: Again, * is not a proxy, and since * will be the one doing the call-out, imagine they try to poit * to the IP of a gateway that only auth's by IP.  * will then rather blindly call out and terminate a call it might not otherwise allow for that peer.
12:53.02[TK]D-Fenderviraptor: In this case it could be considered a security issue
12:53.14[TK]D-Fenderviraptor: Either way B2BUA <-------
12:53.56[TK]D-FenderRaffles666: pastebin a call you send to the queue.
12:53.58[TK]D-Fender~pb
12:53.59infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
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13:16.26Raffles666i don't really need to pb the log - unless you really want it.  If I use "Dial (SIP/200&SIP/201&SIP/202,,)" everything is rosy (ringing is played to the caller) - if I use "Queue (main,r)" then the caller hears silence (instead of moh of course).
13:16.31Raffles666if i remove the 'r' - then the caller does get moh
13:16.51Raffles666calls get queued properly either way
13:17.39Raffles666it's a problem with indicationd.c (the "can't play that much data!" error)
13:18.20[TK]D-FenderRaffles666: You want an autopsy, show us the body.
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13:18.30Raffles666hehe - ok - 1 sec
13:19.10defiancenlqueues suck anyways
13:19.12defiancenl^^
13:19.19defiancenlsimply make the queue in the dialplan itself
13:19.21defiancenl:)
13:19.53defiancenlExten = x,n,Dial(SIP/200,15)
13:20.10defiancenlExten = x,n,Dial(SIP/201,15)
13:20.11defiancenletc
13:20.22defiancenlafter 15 sec ring it will go to next
13:20.53defiancenli couldnt get the queues to dial my isdn tunk for instance... :X
13:20.57defiancenl*dial out
13:21.10defiancenlbut manually it worked fine
13:21.17defiancenlso i created the queue in teh dialplan
13:22.13[TK]D-Fenderdefiancenl: that is not a Queue.
13:22.27defiancenlahh it works the same ;)
13:22.34[TK]D-Fenderdefiancenl: Not at all.
13:22.41defiancenlwell it does what i wanted :P
13:22.56defiancenlfirst dial pers A, if he doesnt pick up , dial person B
13:23.06Raffles666but queues have agents ;)
13:23.14defiancenlwe dont use agents :P
13:23.17[TK]D-Fenderdefiancenl: No logging in of agents, no call stats, multiple callers will ring BUSY PHONES, no prioritization, no backgrounded announcments.
13:23.21Raffles666u don't use queues
13:23.57[TK]D-Fenderdefiancenl: a queue implies first come, first served (with optional weighting".  Yours is anything but.
13:24.10Raffles666anyway, i an't get this damn fault to re-appear now (typical eh?).  I'll post up a link as soon as it happens again
13:24.11defiancenlahh ok ,then i misunderstood Queue on itself :)
13:26.03[TK]D-FenderRaffles666: Yes, problems always magically disappear when i ask for evidence.  Story of my life...
13:26.39defiancenlwell
13:26.45defiancenli had this with asterisk over and over again
13:26.50defiancenlthings didn work
13:26.56defiancenli went and got a coffee
13:27.00defiancenland then they did :
13:27.11defiancenlwithout me changing it :X
13:27.52Raffles666<PROTECTED>
13:29.39*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
13:29.49defiancenli must admit th
13:29.53defiancenli love phpagi :)
13:29.57Raffles666on a slightly different note - did agents.conf ever get 'realtime' support?
13:30.05*** join/#asterisk areay (n=areay@93-97-161-123.zone5.bethere.co.uk)
13:31.21*** join/#asterisk puzzled (n=foobar@94.157.195.130)
13:32.25*** part/#asterisk dany303 (n=dany@mip001.dnetx.net)
13:35.58areayif i call through my asterisk from an outside line my voice echoes
13:36.07*** join/#asterisk davevg (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)
13:36.24areayi can hear it fine the other end, with no distortion or anything... but the echo's kinda annoying... any way to fix it?
13:37.45eppigyHELLO
13:39.12[TK]D-Fendereppigy: YOU ARE DAVE
13:39.39[TK]D-Fenderareay: what is on each end of the call?
13:40.04areay[TK]D-Fender, a landline and x-lite
13:40.32*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
13:40.41[TK]D-Fenderareay: please be a whole lot more specific than just "land line".  For X-Lite, what are you listening and talking on?
13:40.54ZeeekI feel right at home :)
13:41.43areay[TK]D-Fender, it's a cordless home phone... i've tried from a cell too (nokia n96) and it's the same... i'm using a headset with mic/headphone jacks through my sound card
13:41.43*** join/#asterisk laggo (n=user@80.175.142.209)
13:42.32[TK]D-Fenderareay: Lets try this again.  You said "land line".  What boody interface are you using?
13:42.42laggoin sip.conf, how does the server definition know which "register =>" line it's applying to?
13:43.00[TK]D-Fenderlaggo: they have nothing to do with each other
13:43.09[TK]D-Fenderlaggo: No implied relationship whatsoever
13:43.48areay[TK]D-Fender, sorry i didn't explain myself... the home phone is connected to the PSTN, and i'm dialling my 05-number
13:43.53ZeeekIf a client registers, and there's no server there, does it make a sound?
13:44.00[TK]D-Fenderare!?
13:44.01laggo[TK]D-Fender: ok. im trying to figure out how my current setup works at the moment then, because the register line has the username and password i use to login, but the server definition has the context to redirect those calls to
13:44.25*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:44.27[TK]D-Fenderareay: AGAIN, what INTERFACE is letting you talking to the PSTN?
13:44.50laggo[TK]D-Fender: so somehow asterisk figured out to use the username/secret from the register line in the server def
13:45.04[TK]D-Fenderlaggo: No.
13:45.19areay[TK]D-Fender, asterisk is not directly connected to the PSTN. i'm receiving calls through my IAX2 trunk...
13:45.44laggo[TK]D-Fender: can i pm you my config (~15 lines)
13:46.01areay[TK]D-Fender, i'm just using asterisk for voip, and i have a PSTN phone number forwarding to the "incoming" context of my extensions.conf
13:46.39areay[TK]D-Fender, so to answer your question, I'm connected to the PSTN via IAX2
13:47.56*** join/#asterisk bullium (n=RAIN\wbr@216.68.250.30)
13:48.08defiancenlanyone here that can help me on a little phpagi script ?
13:48.24defiancenli need to use a drop down menu on a html/php page to set a variable wich will be used in a diaplan
13:48.44defiancenlso option1 from the dropdown will be $option1 etc
13:49.06defiancenlbut option1 also needs to be filled with a number
13:49.07bulliumin the extensions.conf file how do I make an extension dial a SIP peer if they exist and are online but if they don't exist and are offline to send the call to the same extension out a zap channel
13:49.31*** join/#asterisk puzzled (n=foobar@91.141.192.160)
13:50.18Raffles666bullium - have a look at 'followme'
13:50.39areay[TK]D-Fender, it doesn't matter if the call is initiated from the PSTN or from x-lite... The person on the PSTN hears his voice echo
13:51.14[TK]D-Fenderareay: test another headset / softphone
13:51.27[TK]D-Fenderareay: If that fails then your provider sucks and you're screwed
13:51.45pmhaddad-work[TK]D-Fender, you are always so positive
13:51.46Raffles666headsets are reknowned for causing echo problems - use a 'real' phone
13:51.48pmhaddad-work:)
13:52.07[TK]D-Fenderpmhaddad-work: Yeah... I'm pretty positive he's screwed :)
13:52.17pmhaddad-worklol
13:52.17Raffles666lol
13:52.23laggo[TK]D-Fender: ok how do i specify the context to use for an incoming sip call?
13:52.25*** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do)
13:52.41[TK]D-Fenderlaggo: Usually in a peer entry
13:53.08[TK]D-FenderlaggBarring that it falls to the context specified under [general] if it doesn' fail auth
13:54.23laggo[TK]D-Fender: ahh ok i was using the wrong word. i said server definition when i meant the peer entry
13:54.55[TK]D-Fenderlaggo: I know, I just didn't bother calling you on it
13:55.20laggo[TK]D-Fender: i figured. so my question is how the register line matches the appropriate peer entry block
13:56.36laggo[TK]D-Fender: Raffles666 says it's what follows the / in the register line, but i thought that was the extension to use for incoming calls (instead of 's')
13:57.51laggoat the moment, they match based on the hostname (the peer entry is named the same as the host from the register line) - but now i need to register two from the same host so im confused
13:59.11[TK]D-FenderRaffles666: That means that Raffles666 is not indeed consistently wrong
13:59.18[TK]D-Fenders/not/now/
13:59.22[TK]D-Fender:)
13:59.26[TK]D-Fender~sipregister
13:59.27infobot[~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register.  Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently.  Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW)
13:59.53[TK]D-Fenderlaggo: REGISTER has NOTHING to do with how an incoming call is processed <-
14:00.21[TK]D-Fenderlaggo: It only tells the server WHERE to send it.  It has no impact on the AUTH they demand when sending the call
14:00.32Raffles666i know - i was getting confused between incoming and outgoing - my bad!
14:00.55bulliumRaffles666, I have, but I know I should be able to do it with dialplan logic in the extensions.conf file....
14:01.07[TK]D-FenderRaffles666: Registration also generally has absolutely no impact on OUTBOUND calls either.
14:01.25[TK]D-FenderRaffles666: At least its used to tell the provider where you are so they know where to try to send calls to
14:01.46defiancenli need a script that will fill a variable (phpagi)
14:01.50defiancenlanyone have an idea
14:01.58defiancenlim not to good @ php coding
14:02.20[TK]D-Fenderdefiancenl: You seem to be talking about a web script.  This has nothing to do with AGI
14:02.42ZeeekThere's a good book on PHP AGI
14:02.51[TK]D-Fenderbullium: You could check the result of DIALSTATUS right after the call ans see why it failed and then dial out your Zap channel as desired
14:03.18Raffles666admits defeat then
14:03.19Zeeekdefiancenl:  check this: http://www.theopensourcerer.com/2009/04/16/asterisk-agi-programming-with-packt/
14:03.42[TK]D-Fenderapp_followme = almost worthless
14:03.47Raffles666anyway - guess who's just had the queue problem come back - link to follow
14:04.02[TK]D-Fenderhardly anything that you couldn't do more fine-grained in the dialplan yourself
14:04.26Raffles666[May  1 15:01:27] WARNING[2858]: indications.c:149 playtones_generator: Can't generate that much data!
14:04.27ZeeekTK yes, you could write your own math functions in C
14:04.50defiancenld-fender
14:04.52timeshell_atworkAnyone here getting asterisk to work correctly behind a checkpoint fw?
14:04.55defiancenlim talking about phpagio
14:04.57defiancenl-o
14:04.59timeshell_atworkwith SIP that is
14:05.14defiancenlthnx for the url
14:05.35defiancenlwe have a group of 4 ppl that are say "weekend support"
14:05.44defiancenlone of these 4 will have the weekend shoft
14:05.49defiancenl*shift
14:05.52laggo[TK]D-Fender: if register has nothing to do with incoming call processing, how is my setup working currently when the only place i mention the secret password is in the register line??
14:05.54rob0shaft!
14:06.03defiancenlinstead of manually editing the config for the coresponding number
14:06.11defiancenli want to be able to set it via a webpage
14:06.17defiancenlia phpagi
14:06.21Zeeekshort (shrift)
14:06.26[TK]D-Fenderdefiancenl: AGI is not something you would sue from a web page.
14:06.35defiancenlno but phpagi is ;)
14:06.52Raffles666laggo - i think he means that it isn't directly involved - to call processing - just getting the trunk registered
14:06.52eppigyNEIN
14:06.53[TK]D-Fenderlaggo: because they send the call to you with NO AUTHENTICATION.  Many providers do this
14:07.08laggoahh
14:07.33Raffles666anyway - D-F - what do you want in this log?
14:07.52[TK]D-Fenderdefiancenl: "set it".  What is "it"?
14:07.55laggook so my asterisk receives the incoming call with no authentication - how is it supposed to decide which peer entry to match the call to?
14:08.15defiancenlthe phone number of the corresponding employee
14:08.20[TK]D-Fenderlaggo: Looked at your "host" line of your peer lately?
14:08.46laggo[TK]D-Fender: YES exactly, but now i need to register two incoming trunks with the same host - this is why im here
14:09.10[TK]D-Fenderlaggo: So far you fail to have shown or described a problem...
14:09.34[TK]D-Fenderlaggo: Only shown a sense of confusion over why hat you have now even works.
14:09.37[TK]D-Fenderwhat*
14:09.53[TK]D-Fenderdefiancenl: WHAT are you "setting'?
14:10.04defiancenlok let me try and explain it more
14:10.13defiancenlin the dialplan i have for instance now set hard
14:10.14Raffles666please do :p
14:10.17[TK]D-Fenderdefiancenl: Try using more nouns
14:10.38defiancenlx,3,Set(User1=phonenumber)
14:10.57defiancenlx,4,Dial(${User1})
14:11.03laggo[TK]D-Fender: sorry i will try once more - the problem is i need to receive incoming calls from two trunks with the same host. the two incoming lines need to be redirected to different contexts, so i need a separate peer entry for each. my problem is how to specify which peer entry matches with which registered line.
14:11.05BobPierceI'm setting up an Asterisk cluster at our office for HA reasons. I'm trying to decide between an Active-Active setup with phones registering semi-randomly to both servers or an Active-Backup setup with a secondary server only used if the first server fails. Does a desire to use ADA influence this decision? Does ADA work well or even at all in an Active-Active cluster?
14:12.31[TK]D-Fenderlaggo: You often don't have more than 1 peer for the same provider regardless of multiple registrations, and you do not normally need a separate context for this.
14:12.41pmhaddad-workBobPierce, ADA?
14:12.53[TK]D-Fenderdefiancenl: that is a channel variable.  that is not something you can set from the outside.
14:13.08defiancenlyes you can
14:13.13defiancenlim able to do so
14:13.16BobPierceADA - Asterisk Desktop Assistant
14:13.18defiancenlwith a callerid script
14:13.19[TK]D-Fenderdefiancenl: You need to set a PERsistent valu of some kind like a DB entry you lookup in the dialplan, or AstDB for example
14:13.20defiancenlphpagi
14:13.55laggo[TK]D-Fender: is see, so i must specify different extensions for them to call in each register line and then redirect one of them to another context in the dialplan?
14:14.00laggoi see*
14:14.19[TK]D-Fenderdefiancenl: AGI is for use IN a call.  Not for some external interface.
14:14.35pmhaddad-workBobPierce, oh, I wouldn't know then. Sorry. My HA stuff is all Active-Backup though. Works out well unless you truly *need* load balancing between * nodes
14:14.42[TK]D-Fenderlaggo: Wow... didn't even take you 3 tries!
14:14.45*** join/#asterisk davevg (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)
14:14.49[TK]D-Fenderlaggo: Hope for you yet
14:15.33laggo[TK]D-Fender: i dont understand why this question isnt asked more often - it's not just context that you specify on a peer basis, i'll also need to make sacrifices on the codec and language front
14:16.32*** join/#asterisk ingenius (n=alektro@190.137.17.188)
14:18.08[TK]D-Fenderlaggo: Different codecs for differnt #'s at the same provider?
14:18.28laggoit could happen :P
14:18.31Raffles666http://www.dv-ip.com/temp/queue_problem.txt
14:18.33laggocertainly different languages
14:18.36[TK]D-Fenderlaggo: ...
14:19.03defiancenlD-Fender
14:19.04*** join/#asterisk bartimaeusson (n=bartimae@c-76-23-46-62.hsd1.ut.comcast.net)
14:19.04defiancenlexten => _0492520095,8,agi(/var/spool/asterisk/callid.php)
14:19.05defiancenlexten => _0492520095,9,NoOp(${CALLERID(all)})
14:19.05defiancenlexten => _0492520095,10,Playback(gesloten)
14:19.15defiancenlin this script i set the callerID(all)
14:19.21[TK]D-Fenderlaggo: have each reg target a different exten and all processing is in the dialplan... you CAN separate them, jsut because they start in the same context does not stop you from splitting them off and dealing with them separately
14:19.36BobPiercepmhaddad-work: Yeah, I'm leaning more to Active Backup, because we are talking about approx. 150 phones. I don't expect any problems handling that with one server day to day. Do you have any tip or pointers on HA Active-Backup scenarios?
14:19.43laggoyes i understand this now
14:19.49laggojust found it weirdly restrictive
14:19.50defiancenlall i want is a script that can set say for instance $variable1 to 4 names
14:19.52[TK]D-Fenderdefiancenl: And before you're asking about using a PHP drop-down box which implies web-use
14:20.05defiancenlyes thats a second step
14:20.11pmhaddad-workBobPierce, sure! use heartbeat+DRBD and replicate * on the DRBD partition :)
14:20.25[TK]D-Fenderdefiancenl: that second step is what I'm telling you is inappropriatel.
14:20.29pmhaddad-workyeah and BobPierce for 150 phones, you really don't need Active-Active
14:20.30defiancenlahh k :)
14:20.35defiancenlthen i misunderstood you
14:20.36defiancenlsorry
14:20.40defiancenlmy english is poor
14:20.55[TK]D-Fenderdefiancenl: for your in-call processing PHP-AGI's function for setting a channel variable is well documented
14:23.05defiancenlcan you give me a pointer to a howto/doc/whatever
14:23.16Raffles666Asterisk Gateway Interface 1.4 and 1.6 Programming
14:23.23Raffles666packt
14:23.55Raffles666that nook will help
14:23.57Raffles666book
14:24.24Raffles666anyway - anyone got a resolution to the queue/ring problem?
14:24.57[TK]D-Fenderdefiancenl:  $agi->agi_exec("SET VARIABLE MYVARIABLE $asamplephpvar");
14:25.16defiancenlyes
14:25.18defiancenlthank you
14:25.20defiancenl:)
14:25.29defiancenlthat was what i was looking for :)
14:25.58laggo[TK]D-Fender: fun fact - if you specify context as 'default' in the peer entry, extensions specified after the / in the register lines are ignored, all calls sent to 's'
14:26.37*** join/#asterisk Xanathar_ (n=Xanathar@pool-173-49-123-246.phlapa.fios.verizon.net)
14:26.56[TK]D-Fenderdefiancenl: http://www.google.ca/search?hl=en&q=php-agi+set+variable&btnG=Google+Search&meta=&aq=f&oq=
14:26.58Raffles666then don't use 'default' :p
14:27.05laggoyeah :
14:27.07laggo)
14:27.16laggobut it's all working fine now, which is cool
14:27.21[TK]D-Fenderdefiancenl: I got that out of the first link in a searcht aht took me under 10 SECONDS, and I've never actually USED PHP-AGI before.
14:27.30[TK]D-Fenderdefiancenl: Try a little harder next time
14:27.37Raffles666but that's because you're a god :p
14:27.49ZeeekRaffles666: is correct
14:27.56Raffles666:D
14:28.04Raffles666for the first time today
14:28.08*** join/#asterisk J0llyR0ger (n=J0llyRog@216.243.105.18)
14:28.26ZeeekRaffles666: even a broken clock shows the right time twice a day
14:28.32J0llyR0gerLOL
14:28.34Raffles666thanks
14:28.55[TK]D-Fenderlaggo: no, the "?" to specify is not ignored based on dialplan context.  that is a provider decision.  And some ignore it REgARDLESS and send another SIP header to specify the DID or account the call is realated to and you ahve to parse it yourself in the dialplan.
14:29.00*** part/#asterisk J0llyR0ger (n=J0llyRog@216.243.105.18)
14:29.12*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
14:29.13[TK]D-Fender"/"
14:29.44Raffles666anyway - now all that is over - i really need to cure this queue/ringing problem - anyone know of a patch or workaround?
14:29.45BobPiercethanks pmhaddad-work: we are currently running about 90 phones and will only be adding 50 more phones. The desire from my managers was really about HA not load balancing. Our Asterisk system has been very stable, but the last time it had problems came with poor timing. So, I've been tasked with building a redundant system to provide as close as we can get to 99.999% service. I'm also looking at moving to Asterisk 1.6.0.0 during this rebuild. We are c
14:29.45laggo[TK]D-Fender: i disagree, i JUST tried it - changed context from 'default' to 'incoming' in the peer entry, no other changes. with default -> routed to 's' regardless
14:30.41Raffles666laggo - what happens if you take 's' out of [default]?
14:30.43*** join/#asterisk aenaus (n=hdgfghf@79.107.169.106)
14:30.48*** join/#asterisk aenaus_ (n=hdgfghf@79.107.169.106)
14:31.14laggoRaffles666: i don't know, and seeing as it's working i'm not going to try it ;)
14:31.24Raffles666jumps - just spotted a ghost :p
14:31.32Raffles666laggo - ok - i was just curious
14:31.47[TK]D-Fenderlaggo: proper debug would show you why I'm sure...
14:31.48laggomaybe you can try it and we can have some independent verification hehe
14:33.03Raffles666i would - but sipgate seem to have 'banned' asterisk servers (it doesn't want to work for me either way anymore)
14:33.17pmhaddad-workBobPierce, I'm using 1.6.0.9 atm
14:33.19pmhaddad-workworks fine
14:33.24laggoi'm using sipgate :)))
14:33.44laggoasterisk is a temperamental one
14:33.58pmhaddad-workand really with that many phones i honestly don't think you need HA, but I suppose for disaster recovery its a viable option
14:34.04Raffles666hang on then - i'll retry
14:34.15[TK]D-FenderRaffles666: Well your situation is looking like a bug...
14:34.19pmhaddad-workBobPierce, just maybe more work than its worth
14:34.27defiancenlhummm fender
14:34.34defiancenlyour example doesnt work :X
14:34.45defiancenl<PROTECTED>
14:34.46defiancenl$agi->agi_exec("SET VARIABLE 0634793552 $Jef");
14:34.53[TK]D-Fenderdefiancenl: It was a sample and I don't know if YOU implemented it wrong or not.
14:34.57defiancenlthis would set $Jef to the number right ?
14:35.10[TK]D-Fenderdefiancenl: WRONG
14:35.13Zeeekso by the way, Nufone is totally dead this time
14:35.17defiancenlok :X
14:35.31[TK]D-Fenderdefiancenl:  $agi->agi_exec("SET VARIABLE MYVARIABLE $asamplephpvar");  <--- you tried putting a NUMBEr where you need to put the VARIABLE NAME
14:35.56Raffles666<[TK]D-Fender> : thought so.  Time for a bug report?
14:36.13[TK]D-FenderRaffles666: Well its clearly crapping out on you.  What ver exactly?
14:37.05Raffles6661.4.24.1 (source from 1.4-current.tar.gz) - not sure of rev.
14:37.19[TK]D-FenderRaffles666: Ok, definitely report it
14:37.20BobPiercepmhaddad-work: I agree, I've seen the system bee very stable for periods of longer than six months at a time. But for those times (if and when) a hardware failure or outage may occur (even potentially with the Ethernet switch it is connected too) my manager would like to have a backup box that phones could fail over to and keep providing service to our users.
14:37.29Raffles666will do - ta
14:37.56defiancenlok then it would be agi->agi_exec("SET Jef 0634793552 (what the fuck goes here then?)");
14:38.22pmhaddad-workBobPierce, yep Active-Backup with heartbeat+DRBD is what you want
14:38.46BobPiercepmhaddad-work: Thanks again. Have a great day.
14:39.07virtualme123has anyone used asterisks HTTP protocol to collect originate response information from a call? I am trying to use GetVar as an action to return the HANGUPCAUSE result of a channel, is this a good way to get the hangup information?
14:39.24[TK]D-Fenderdefiancenl: $agi->agi_exec("SET VARIABLE MYVARIABLE $asamplephpvar");  <- MYVARIABLE = the name of your fucking variable
14:39.39[TK]D-Fenderdefiancenl: You removed a keyword entirely
14:39.59defiancenlfuck , i dont understand :X
14:40.04[TK]D-Fenderdefiancenl: $agi->agi_exec("SET VARIABLE Jef $asamplephpvar"); <---
14:40.17bullium[TK]D-Fender, could you give me an example of how to use DIALSTATUS?
14:40.34defiancenlbut where goes the value of the variable then ?
14:40.42[TK]D-Fenderbullium: its a VARIABLE.  "core show application dial"
14:41.09pmhaddad-workBobPierce, anytime, let me know if I can help you with anything else :)
14:41.17[TK]D-Fenderdefiancenl: My sample shows you setting the * var to the value of a PHP var.  Please wake up.
14:41.30defiancenlahhhhhhhhhh
14:41.32defiancenlok damn
14:41.38defiancenlsorry
14:41.42defiancenlwoke up now
14:41.55[TK]D-Fenderdefiancenl: Holy shit, didn't I fucking nameit obviously enough for you?  "$asamplephpvar"
14:42.00defiancenllol
14:42.03defiancenlsshht
14:42.04defiancenl:(
14:42.09Raffles666bullium : http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
14:42.13defiancenlsometimes im oblivious man
14:42.13[TK]D-Fendergrabs his ClueBat (tm)
14:42.16defiancenlhaha
14:44.40virtualme123I currently just send an originate request and get a response packet back that either says 'success' or 'failed' but what I want is a status code, any ideas on which way to go?
14:45.28*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
14:45.38[TK]D-Fendervirtualme123: What kind of "status"?
14:45.55virtualme123I'm looking for the originateResponse status
14:46.09virtualme123So if it fails I know why
14:46.58[TK]D-Fendervirtualme123: I'd recommend dialing a local channel, and using a UserEvent if you're looking to poll for the response via AMI
14:47.13[TK]D-Fendervirrather than calling another device.
14:48.15*** join/#asterisk propellerhead (n=yogurt2u@host177.190-30-180.telecom.net.ar)
14:49.37virtualme123Well I was going to make an action of GetVar and get the variable HANGUPCAUSE status
14:49.47virtualme123After say 30 seconds
14:53.08[TK]D-Fendervirtualme123: Well logically your channel will not exist at that point.
14:53.30[TK]D-Fendervirtualme123: It should have long since died.
14:55.03virtualme123I suppose that if I polled for it though I would know when it had died and take the last know cause? Or is that too flakey ...
14:55.58*** join/#asterisk moy (n=moy@74.12.124.89)
14:56.29virtualme123I'm not too sure how I'd use the UserEvent ...
14:59.26*** part/#asterisk defiancenl (n=jefrecha@m0n0.hix.nl)
15:00.38[TK]D-Fendervirtualme123: In your local channel you could jsut spit out an event with the dialstatus, hangupcause, etc
15:01.23[TK]D-Fendervirtualme123: You could prefix it with a unique identifier set as a channel variable in your initial Originate so as to be able to track it to the original call
15:03.37*** join/#asterisk bl4 (n=kim@64.0.29.254.ptr.us.xo.net)
15:04.24*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
15:04.40*** join/#asterisk jelly-bean (n=mikesmul@c-76-23-46-62.hsd1.ut.comcast.net)
15:05.00jelly-beanwhere can i find a good web front-end for asterisk with beautiful reporting?
15:05.26pmhaddad-workjelly-bean, a2billing
15:06.10[TK]D-Fenderjelly-bean: front end to do what?  Reporting of what?
15:06.24jelly-beani want to see reports on like call volume etc.
15:06.40[TK]D-Fenderpmhaddad-work: Shouldn't jsut fire off an answer to an unqualified request like that :)
15:06.53[TK]D-Fenderjelly-bean: Why kind of call voluem?  General CRD?  Queues?
15:07.08jelly-beaneverything? i can't have it all-in-one?
15:07.43ajohnsonHeh, define "everything"
15:07.48pmhaddad-work[TK]D-Fender, a2billing will do call volume and CDRs and such from my understanding
15:08.21jelly-beani am looking for something like what trixbox has with pie charts and bar charts
15:08.24[TK]D-Fenderpmhaddad-work: What is "call voluem" apart from CRD?  You are wording them as 2 entries and in and of itself is not a "thing"
15:08.46[TK]D-Fenderjelly-bean: Are youplannin g on running queues?
15:08.54jelly-beanyes
15:09.23jelly-beansee "ACD Reporting": http://www.trixbox.com/products/trixbox-pro/screenshots
15:09.28[TK]D-Fenderjelly-bean: don't know whats decent for queues.  check out QueueMetrics , and Asterisk-stat
15:12.26Raffles666jelly-bean : http://www.voip-info.org/wiki/view/OrderlyStats
15:12.33*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
15:13.33pmhaddad-workjelly-bean, yeah trixbox has a nice call reporting panel
15:14.55*** join/#asterisk n3glv_lap (n=n3glv@76-220-96-162.lightspeed.fyvlar.sbcglobal.net)
15:14.58jameswf~trixbox
15:14.59infobotextra, extra, read all about it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
15:15.15jameswf~fonality
15:15.16infobotfonality is, like, "no Asterisk knowledge needed" or, "I just installed asterisk now what?", and will be the first guys against the wall when the revolution comes!
15:15.36[TK]D-Fenderpreps the firing squad
15:15.37n3glv_laplol at infobot
15:16.17Raffles666cocks his rifle ;)
15:16.19*** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net)
15:16.30[TK]D-Fenderjameswf: Careful, we have fonality lurkers... I mean "guests" here often.. wouldn't want to hurt their feelings :)
15:16.33n3glv_lapanyone have any info about TDD?
15:16.54[TK]D-FenderRaffles666: Sex with firearms is illegal in several states...
15:16.56n3glv_lapRaffles666: watch where you point that thing
15:17.00n3glv_lap;-)
15:17.02Raffles666:)
15:18.02n3glv_lapso, * is supposed to support TDD
15:18.23n3glv_lapI'm guessing like the fax softmodem thing?
15:18.37n3glv_lapanyone know is it still there?
15:18.40ddickensonare there any problems using older digium cards with dahdi? Yesterday I set up server with a sangoma card and dahdi would load fine.  I decided to use one of my digium t1 cards instead and then dahdi wouldn't load.  I finally started with a fresh install of centos and reccompiled libpri, dahdi, and asterisk to start completely fresh and I still can't get dahdi to load.  Any ideas?
15:19.19n3glv_lapddickenson: I have a tdm410p, had to hand edit
15:19.23n3glv_lapbut it works then
15:19.24Qwellddickenson: well, how old?
15:19.30Qwell(but no, in theory)
15:19.47n3glv_lapQwell: is  your man there ddickenson
15:20.01n3glv_lapo/ hi qwell
15:20.05[TK]D-FenderQwell: Well.. you know I have this old Tormenta 1 PCI caud that was just laying around an...
15:20.15ddickensonit's a TE410P
15:20.41ddickensonI didn't know if there would be some type of firmware updates or something
15:20.57bullium[TK]D-Fender, thanks
15:20.59bulliumRaffles666, thanks
15:21.19ddickensonI'm not sure how old it is, my boss acquired it at some point...
15:22.19n3glv_lapmeh I hate rain... on payday no less
15:25.31eppigyDear Katty, What is for lunch? Love, Dave
15:25.52Zeeekh0m3 Impr0v3m3nt today on VUC: http://tr.im/voip
15:26.06ZeeekSee you at AMOOCON in two days
15:26.38Zeeekby the way, join us: #voip-users-conference
15:26.40n3glv_lapgod I hate l33t sp33k even more than rain
15:26.57ZeeekRain rocks
15:27.13Zeeekbut we digress
15:28.29n3glv_lapyes, so, is fskmodem.c still in * ?
15:28.31n3glv_laphttp://www.voip-info.org/wiki/view/tdd+mode
15:29.01bartimaeussonDoes anyone know if trixbox can run concurrently on an asterisk install? Or is trixbox a full install that replaces the OS?
15:29.21smash-centos
15:29.26smash-download trixbox is an os
15:29.41n3glv_lapsmash-: he's a troll
15:30.45bartimaeussonDoes anyone know of any good live reporting front-ends for asterisk then?
15:31.07n3glv_lapgo ask your mom
15:31.29smash-hey question how hard is it to setup sip trunk through trixbox gui
15:31.33smash-or should i use cli
15:31.39*** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com)
15:31.48n3glv_lapsmash-: usually seconds
15:31.49smash-im having trouble getting a damn provider to respond i have my telco guy on the phone right now
15:31.52n3glv_lapvia gui
15:31.58smash-but he says that, he doesnt sell sole sip trunks
15:32.10n3glv_lapthen move on to another
15:32.15smash-and none of the places i have sent emails too have replied with anything but auotmated response
15:32.34n3glv_lapwhat country?
15:33.03ariel_smash-: not really hard
15:33.03ariel_bartimaeusson: I would not use trixbox, but http://pbxinaflash.net/  would give you the same things but not the over head that trixbox has.
15:33.24n3glv_lapor asterisknow
15:33.32n3glv_laphi ariel_ o/
15:33.45ZeeekWhere is my Katty ?
15:33.53bartimaeussonariel_: Thanks!
15:34.31*** part/#asterisk n3glv_lap (n=n3glv@76-220-96-162.lightspeed.fyvlar.sbcglobal.net)
15:35.18Aiatekor if you want just a reporting cdr
15:35.35Aiatekareski
15:35.44Aiatekits a very good choice
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15:39.41*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
15:40.05smash-the over on trixbox is nice
15:40.09smash-gui is fucking dope
15:40.14smash-vs. this box i admin for fonality
15:40.17smash-that u pay for webui
15:40.19smash-pbxtra
15:40.22smash-ha
15:40.32smash-i wanna make my own asterisk distro
15:40.42Raffles666ties smash- to the steak
15:40.58smash-fuck i wish i knew someone who ran a sip supplier
15:41.03smash-i want to do some freakin test's
15:41.23smash-do most sip trunks still allow you to setcallid to anything u want
15:41.44smash-or does it ahv eto be relative to the did u are allowed
15:41.56Raffles666depends on supplier
15:42.13Raffles666and if it's going pstn - telco provider (not allowed at all in the UK)
15:42.46*** join/#asterisk b14ck (n=comradeb@72.37.252.50)
15:42.46b14cksup
15:43.41[TK]D-FenderAh, our resident Fonality shill has arrived!
15:43.49*** join/#asterisk jelly-bean (n=mikesmul@c-76-23-46-62.hsd1.ut.comcast.net)
15:43.50Raffles666:)
15:43.55b14ckw0rd
15:44.16b14ckim not here for fonality, im here for the fun ^^
15:44.17jelly-beani want a real-time call viewer type of report that i can leave up on the screen to always see things like: calls in queue, longest waiting on hold, etc.
15:44.19Raffles6662 ya muvva - sorry - went all 'streetz' then ;)
15:44.24jelly-beanis there any interface that provides that?
15:44.37Raffles666write one (AMI)
15:45.07[TK]D-Fenderjelly-bean: Plenty of 3rd party apps, or feel free to write one yourself.
15:45.41jelly-bean[TK]D-Fender: can u link me to a few? or mention the names? i need some keywords for google
15:46.12Raffles666sounds like you need a 'wallboard' app - I wrote ours in VB (yes, I know) using the AMI output
15:46.15*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
15:46.44mockerjelly-bean: Flash Operator Panel is pertty popular.
15:46.49jameswfHe said VB I need to go wash my eyes out they feel dirty
15:46.50mocker+spelling!
15:47.15*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
15:47.26ddickensonD-Fender:  So after messing with my dahdi stuff yesterday I decided that I had muddied the water enough and started from scratch.  Still can't get dahdi modules to load though.  Fresh centos install (this time with all dev tools) fresh libpri compile, dahdi compile, and asterisk compile.  It worked fine when I had sangoma card in there but now that I put TE410P in there it won't load, could...
15:47.28ddickenson...it be card?
15:47.28Raffles666sorry james - but how do you think I feel using it ?
15:48.02b14ckwrite it in x86_64 at&t unix asm
15:48.04b14ckthats the only way to go
15:48.09b14ckheh
15:48.26Raffles666careful - or I'll re-write it in COBOL!
15:49.00b14cklol
15:49.13b14cki was experimenting last week with the quivr c library for AGI
15:49.14Raffles666yes - I am that old
15:49.16b14ckits really nice, actually
15:49.21b14ckif u like c, id recommend it
15:49.43Raffles666I'll take a look when I get chance - maddddddddddd busy at the moment
15:49.49b14cksame here
15:49.52b14ckheh
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15:58.00Raffles666off home - laters guru's ;P
15:58.43*** join/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com)
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16:05.59_brent_does anyone know how asterisk 1.6 determines if a forwarded call is to itself and should be Local/foo@context?
16:07.06_brent_what i see it doing is attempting to send the call to itself, detecting a loop, and then forwarding it to Local/foo@context
16:07.40_brent_is this by design now (i don't see it doing this in 1.4) or do i have a misconfiguration?
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16:11.57pmhaddad-workis there any easy way i can limit what options are shown in the a2billing main panel?
16:12.10pmhaddad-workor do i have to muck through all the php and html and delete the ones i dont want
16:12.39pmhaddad-workspecifically, i dont want the asterisk config option under administrator
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16:45.06dbuHi, I have a sdp / rtp question not related to asterisk, which channel should I ask my question on?
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17:06.36EUSEricDCAPhey guys
17:08.05pmhaddad-workhi EUSEricDCAP
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17:12.08_brent_to answer my own question about how * determines if a forward should be local or not, it appears to respect promiscredir absolutely
17:12.29_brent_promiscredir=yes      ->      no forwards treated as local
17:12.36_brent_and vice versa
17:15.01hardwireextenspy == chanspy?
17:16.06hardwireapps/app_chanspy.c:res |= ast_register_application(app_ext, extenspy_exec, tdesc, desc_ext);
17:16.08hardwireTrue
17:19.05pmhaddad-workhey anyone ever had an issue with meetme() where lets say you have 1000,1,meetme(1000,i,1234) it doesn't prompt for authenticiation?
17:33.16nealixNew user here, please tell me if this is the wrong place to ask.  I have installed an asterisk distro in a VMware machine for learning.  Where can I find a
17:33.50nealixquick start tutorial for what steps need to be configured to get the first basic comms going.  Something like;
17:34.22nealix-  Config a SIP softphone extention.    -  config an outbound trunk.  -  config a dial plan.   - test a call ?
17:34.40nealixI have a 300 page doc, but want to learn the first steps to play/learn.
17:34.55[TK]D-Fendernealix: the BOOK
17:34.57[TK]D-Fender~book
17:34.58infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
17:35.05[TK]D-Fendernealix: And for inspiration go read :
17:35.09pmhaddad-workis reading the BOOK right now
17:35.09[TK]D-Fender~jerjerguide
17:35.10infobot[~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/
17:35.15pmhaddad-workfreakin MeetMe()... :/
17:35.28[TK]D-Fendernealix: that last guide only for setup, not installation.
17:35.50nealixvery nice.  Really appreciate it.  I'll go read both of those.  Thanks a ton.
17:37.37KyleKI guess for an outgoing sip request Contact: <sip:line1@192.168.0.10> should be my public IP?
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17:45.28pfnamong other things
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17:53.25[TK]D-FenderKyleK: Clearly you have still not configured your system properly to wrok from behind NAT
17:53.28[TK]D-Fender~sipnat
17:53.29infobot[~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
17:53.33[TK]D-Fender^^^^^^^^^^6
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18:01.03shozaib[May  2 03:56:51] WARNING[9072]: chan_sip.c:12334 handle_response_invite: Received response: "Forbidden" from '"unknown" <sip:6131731@192.168.2.3>;tag=as548b281c'
18:01.24shozaibreceving error asterisk any can help plz
18:01.54*** join/#asterisk EUS-Taz (n=chatzill@ip67-152-18-226.z18-152-67.customer.algx.net)
18:04.29*** join/#asterisk EUS-Taz (n=chatzill@ip67-152-18-226.z18-152-67.customer.algx.net)
18:05.03EUS-Eric-DCaPthat's interesting.
18:05.20EUS-Eric-DCaPwhat is the IP of your PBX?
18:07.49l0st-soulhi there
18:08.29l0st-soulam i the only one to experience config files getting emptied by asterisk ?
18:09.38*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
18:09.39l0st-soullike, fresh install on debian .. i create 1 extension with voicemail, i go on the voicemail and change my pin code ...  voicemail.conf is emptied.
18:10.10rob0belches
18:10.16rob0I ate it, sorry.
18:10.23shozaib192.168.1.60 ip address of pbx
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18:31.13shozaibcan any one help me
18:31.33EUS-Eric-DCaPdo you have something with IP 192.168.2.3 on your network?
18:32.15shozaibyes
18:34.43telnettechdo you have something yhat can route between networks
18:35.05shozaib192.168.2.1 with this ip my sip device is connected
18:35.19shozaiband at my eth0 i have 192.168.1.60
18:35.47shozaibnope
18:36.40telnettechnevermind what i was thinking.....it is friday afternoon and i am fried from the week
18:36.52EUS-Eric-DCaPwhat's happening is your sip device doesn't have the sip account properly configured on it.
18:37.02EUS-Eric-DCaPalso you may need to tell asterisk about the multiple networks.
18:37.25EUS-Eric-DCaPare the networks natted to each other, or just routing between them
18:37.51eppigywow
18:38.11telnettechthank you Eric....thats what i was meaning......device didnt register cause the username and password is not correct
18:38.36telnettechverify that the sip.conf has a host and type
18:38.40telnettechfor the user
18:39.42shozaibhos is 0.0.0.0 and type
18:40.19EUS-Eric-DCaPset "host=dynamic" in the SIP.conf entry for that client
18:40.24EUS-Eric-DCaPtype=friend
18:40.31EUS-Eric-DCaPdo that and reboot the phone
18:40.52shozaib[6131731]
18:40.53shozaibtype=peer
18:40.53shozaibusername=6131731
18:40.53shozaibfromuser=6131731
18:40.53shozaibsecret=xxxxxxx
18:40.53shozaibhost=0.0.0.0
18:40.55shozaibdomain=10.215.11.250
18:40.57shozaiboutboundproxy=10.215.11.250
18:40.59shozaiboutboundproxyport=5060
18:41.01shozaibcanreinvite=yes
18:41.03shozaibcontext=default
18:41.05shozaibcallerid="" <6131731>
18:41.07shozaibqualify=yes
18:41.09shozaibport=5060
18:41.11shozaibdisallow=all
18:41.13shozaiballow=ulaw
18:41.15shozaiballow=alaw
18:41.17shozaibnat=yes
18:41.23_brent_pastebin
18:41.23EUS-Eric-DCaPuh...you have some problems there.
18:41.32EUS-Eric-DCaPthat's just for a regular old sip phone,right?
18:41.38EUS-Eric-DCaPwhat brand/model
18:43.03telnettechis this user a remote location?
18:43.43shozaibsiemens/giga set SX682
18:44.03shozaibno
18:44.16shozaibuser is on my local lan
18:51.47[TK]D-Fender*b00m*
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18:52.45[TK]D-Fender*b00m*
18:54.57jplankyay I never live through netsplits
18:56.15pmhaddad-worklol
18:57.06Qwelljplank: only newbs are on brown.freenode.net
18:57.41Qwellhmm, didn't that servers namesake die recently?
18:58.20Qwellnm
19:00.05nealixAnyone have pointers for small, low power HW to use for home asterisk (1 line, 5 extention).
19:00.26nealixWas thinking an ITX motherboard with Intel Atom, but open to ideas for a small system
19:00.52nealixWill use the Linksys SPA-3102 for the analog adapter, so I don't need space for PCI card at this time.
19:01.26jplankI have a ATX board with an atom and 2gb ram in a slim pc case
19:01.29jplankworks fine
19:01.37jplankbut we only use it for demo's and trade shows
19:01.46jplankwhat about a pika box?
19:02.28nealixwhat is that?
19:02.36jplankI've had great success with Pika's asterisk box
19:02.40jplankpikatechnologies.com
19:03.12jplankits basically an appliance so to speak
19:03.45jplankits just a box the size of a linksys router, running their brew of linux (which I would recommend changing, some guy ported debian to it) and asterisk
19:03.57jplankhttp://pikatechnologies.com/english/View.asp?x=652
19:07.21nealixIt's cute, looks fantastic, but I don't have the $750.  I'll take a look at the $69 Intel mobo, as I already have disks, power supplies, etc.  So I should have asked if people have suggestions for DIY hardware for low power setup?
19:07.49[TK]D-Fendernealix: Just about anything will do
19:08.08_brent_http://www.voip-info.org/wiki/view/Asterisk+Linksys+WRT54G
19:09.45jplankdoesn't really trust asterisk on a linksys
19:10.25jplankno memory, no disk space, slow processor = shotty performance IMO
19:11.15_brent_i've never tried it, but it fits the low power requirement
19:11.29nealixperhaps a small single board x86 computer, so I can use a linux based asterisk distro like AsteriskNow or Elastix.org
19:11.46_brent_fg
19:11.51_brent_sorry, wrong window
19:11.53_brent_doh
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19:15.00*** join/#asterisk crevetor (n=crevetor@bureau.ubity.com)
19:15.48crevetorI have a problem : I have sip peers with nat=yes and qualify=5000 and asterisk sends it'soptions packet to the non natted IP (e.g. 192.168.1.x) Why would it do that ?
19:16.41crevetorHi by the way
19:18.59[TK]D-Fendercrevetor: pastebin the complete SIP debug and your peer entry masking only the password
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19:28.22EUS-Eric-DCaPin the general second of your SIP.CONF do you have externip defined?
19:28.24EUS-Eric-DCaPand localnet?
19:28.51EUS-Eric-DCaPwhen you changed these settings did you do a reload chan_sip?
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19:32.05crevetorEUS-Eric-DCaP: yes, yes and yes
19:32.22EUS-Eric-DCaPwhat kind of firewall do you have?
19:32.34EUS-Eric-DCaPwell, are they set correctly?
19:34.24crevetorEUS-Eric-DCaP: the weird thing is that my peers are up and have nornmal qiualify response (around 40ms)
19:35.08crevetorand I am doing a sip debug on a peer and asterisk tells me it's sending the options packet to the right IP and I do get an answer to that packet
19:35.19crevetorbut it seems like it also sends packet to the natted IP
19:35.53EUS-Eric-DCaPodd
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19:36.33crevetorAaahhaaa
19:36.47crevetorWhen I don't specify a peer I get interesting debug
19:36.55crevetorwhere can I paste it ?
19:37.25pmhaddad-work~pb
19:37.26infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
19:37.36pmhaddad-work^^ crevetor
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19:38.06pmhaddad-workhi leifmadsen
19:38.09crevetorthx
19:38.13pmhaddad-worknp
19:39.20leifmadsenpmhaddad-work: yo yo
19:39.37crevetorhttp://pastebin.com/d1174c627
19:41.15crevetorhi leifmadsen
19:41.31pmhaddad-workleifmadsen, what's the happenings in the land to the north?
19:45.35crevetorOooooohh I think I've got it
19:45.43crevetorstupid stupid stupid
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19:55.07ddickensonanyone know config to run an adtran tsu600 from a t1 card on asterisk box?              [* box]----->TSU-600---->analog phone sets.   I've only ever configured t1's to go to PSTN
19:58.15[TK]D-Fenderddickenson: "same span def, no "bchan" / "dchan", signalling "like you would for analog "fxols, fxsks", etc
19:59.06ddickensonso I should set it up as fxo since the cb is fxs?
19:59.44ddickensonI was thinking it needed to be pri_net?
20:01.02watchyi wanna make some rissoto
20:01.13ddickenson[TK]D-fender: this is in the /etc/dahdi/system.conf you're talkinga bout
20:01.41[TK]D-Fenderddickenson: and dahdi_channels.conf  Only part that looks like PRI is the span line
20:01.56[TK]D-Fenderddickenson: its a question of how its channelized
20:03.10ddickensonok.  What about as far as extensions.conf goes?  does it need to be somehow a 1 to 1 deal for DID's that come into asterisk on my other t1 span and should be going out to these certain channels on the channel bank
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20:13.50[TK]D-Fenderddickenson: Channel groups are the same throughout *
20:14.50watchymake me some risotto tk, i know you can
20:15.04watchyyour probably a great cook
20:17.44ddickensonTK, can you look at my files and tell me where I'm screwing up?
20:18.05[TK]D-Fenderopens his copy of "Hannibal Lecter's Dinner of two" and eyes watchy
20:19.15Corydon76-digwith a nice Chianti?
20:20.33[TK]D-FenderCorydon76-dig: Well met.
20:22.34[TK]D-Fenderok, checkout time, BBIAB
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20:27.34ddickensonanyone willing to take a look at my config files for setting up this channel bank and tell me where I'm messing up?
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20:52.34lesouvageare there some example questions somewhere on the Internet to get an idea about what the150 dcap exam questions are about?
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21:04.23joako_What exactly does "cdr_mysql: cannot connect to database server localhost." mean? That the host can not be connected to at all or there could be a database issue?
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21:24.34joesuffcerenMy asterisk process (running 1.4.22 on an 8-core Xeon box) is consuming enormous amounts of CPU with only a few active phone calls. This box has been in production for months with no issues, and no changes have been made recently
21:25.12joesuffcerenany thoughts on what I could do to isolate the problem? I realize I could restart asterisk and it would probably go away, but I have active phone calls, and it's consuming enough CPU that incoming calls are getting a busy signal
21:26.22ariel_joesuffceren: are you using any system calls? via perl scripts? or other items like that?
21:26.40ariel_top's could tell you what is running and what is taking more of the processes
21:27.05joesuffcerentop shows me just my asterisk process consuming 390+% CPU
21:27.10joesuffcerennothing else is consuming cpu
21:27.31joako_What exactly does "cdr_mysql: cannot connect to database server localhost." mean? That the host can not be connected to at all or there could be an authentication issue?
21:28.42ariel_joako: it means there is a problem with the connection to mysql, Is it running? did you configure it correctly?
21:32.53ariel_joesuffceren: when I used to run 22 boxes of asterisk in the call center I worked for, we used to get our queue boxes take those hits.  But it was mostly due to a perl script being called via a systems call. Only way around it was to let it unlocked or just restart the asterisk. We would loose the connections on it.
21:34.13joesuffcerenariel_: It looks like there's only one active call left. I'll IM the user and get him to hangup so I canrestart asterisk
21:36.43rOfLzhello there
21:37.03ariel_if it's one user only then it might not be a big issue.  Just imagine when we had about 100 calls stuck this way.
21:37.22joesuffcerenyeah. yuck
21:37.46ariel_it's time to head home.  gnite folks
21:37.50rOfLzI am having a problem with outgoing calls there is no voice on both ends from my sip chan
21:38.03mocker~sipnap
21:38.05mocker~sipnat
21:38.06infobot[~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
21:38.15mockerrOfLz: Check that.
21:38.30ariel_wow I was about to say that most of the no voice is due to a nat issue.
21:38.42rOfLzI do not have any router or any firewall within asterisk box and client
21:39.05mockerrOfLz: iptables?
21:39.14rOfLznops
21:39.28ariel_two free systems out there. wow...
21:39.35joako_ariel_: Yes it is running, other apps use it. But how do I get more verbose error for asterisk? E.g. what PHP gives?
21:40.48Pan3DrOfLz: is the call actually dropped or is there silence?
21:41.51jameswfwow If I am reading this right there is pretty much NO paid support options for trixbox CE http://trixbox.org/fonality-trixbox-ce-scope-service
21:41.59rOfLzsilence
21:43.09*** join/#asterisk heit0050 (n=andy@75-149-156-145-Minnesota.hfc.comcastbusiness.net)
21:43.15rOfLzPan3D: call get connect with complete silence
21:43.44*** join/#asterisk manxpower (n=Administ@router.asteriasgi.com)
21:53.49mykhyggzSo, are there any situations that * can handle with SIP device behind NAT w/o SIP proxy? Or am I wasting my time thinking there is?
21:56.39manxpowermykhyggz: almost all situations can handle sip devices behind NAT
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21:56.58manxpoweryou won't get reinvites, but you don't need some "sip proxy"
21:57.52mykhyggzRight. Classic problem, I guess. Dropping the call after about a minute. No response to critical packet.
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22:00.26manxpowercanreinvite=no.  NO port forwarding on the NAT router the SIP device is behind.  I assume Asterisk is on a public IP?
22:00.56manxpowerIf asterisk is behind NAT then it's a bit more complicated
22:01.03drmessanojameswf: How so?
22:01.51mykhyggzmanxpower: yes * is on public IP
22:02.47manxpowermykhyggz: then it should be as easy as setting canreinvite=no in the sip.conf section for that device
22:03.33mykhyggzThat's what I thought.
22:05.28manxpowerjust make sure you did not enable any NAT stuff on SIP device.  You might want to add qualify=yes to that sip peer too if your router closes the nat translations too quickly
22:05.59mykhyggzokay, cool.
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22:07.40rOfLzsorry guys got dc
22:08.30rOfLzmykhyggz: I am using sip device and got account register with asterisk
22:09.12mykhyggzYes, that's not the problem. It's bridging the call between SIP device behind nat and SIP trunk. Call drops reliably.
22:09.14rOfLzbut asterisk box and user is on same subnet there is no router or firewall within both
22:09.46mykhyggzno problem using with local asterisk.
22:10.24rOfLzwhen I set canreinvite to yes then softphone hangup the call but on other end call remain establish with no voice
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22:10.51manxpoweryou should not reinvite when using nat
22:11.37rOfLzbut I am not using nat but I am not sure about sip proxy coz it is on other subnet
22:15.45*** join/#asterisk crevetor (n=crevetor@bas1-montreal07-1176436428.dsl.bell.ca)
22:16.19crevetorHow ould you explain this : Retransmitting #6 (NAT) to 192.168.1.139:5060:
22:16.32crevetorwhen 192.168.0.0/255.255.0.0 is in localnets
22:20.25mykhyggzWell, for me it looks like there's no ACK to an OK so * hangs up. Packet2Packet bridging  all good, then hangs up.
22:20.59crevetorThat's for an options packet by the way
22:22.09*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:23.16juancferrerI'm trying to run asterisk on an amazon ec2 machine.  When I call the 'real' phone number attached to the sipphone account from my cellphone I get a message saying the person i'm trying to reach is unavailable.  If I call my cellphone from one of the extensions (granstream phone), then i get a busy dialtone.   Where did I go wrong?
22:23.44juancferrerhttp://dpaste.com/40180/
22:29.02mykhyggzWell, that's that, I guess. Time to try a different SIP trunk provider.
22:29.38timeshell_atworkSIP Response 482 "Loop Detected"
22:29.45timeshell_atworkHow do I fix this?
22:33.17*** join/#asterisk carl689 (n=carl@rrcs-24-227-79-218.se.biz.rr.com)
22:34.37carl689Hello, I just got a TDM844 today. Had it running fine this morning, after a reboot however I do not get a dial tone or pickup from FXS.  Any ideas?
22:35.41jayteecarl689, DAHDI or ZAPTEL?
22:35.47carl689DAHDI
22:36.10jayteecarl689, if you type help dahdi at the CLI what do you get?
22:36.17drmessanoClowns
22:36.30jayteeKiller Clowns
22:36.46carl689In asterisk cli or terminal?
22:36.52jayteeasterisk CLI
22:37.30carl689If I type dahdi and press enter, nothing.  However I can do all the dahdi {tab}{tab} and get options
22:37.59jayteecarl689, I said type HELP DAHDI
22:38.32carl689wow, sorry dont' know how i missed that
22:38.32carl689<PROTECTED>
22:38.32carl689<PROTECTED>
22:38.32carl689<PROTECTED>
22:38.32carl689<PROTECTED>
22:38.32carl689<PROTECTED>
22:38.32carl689<PROTECTED>
22:38.44jayteeok. type dahdi show channels
22:39.03jayteebut don't paste the output here, use a pastebin
22:39.06jaytee~pb
22:39.07infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
22:39.54carl689http://pastebin.com/d43e6a269
22:40.34jayteecarl689, what distro are you running?
22:40.43carl689ubuntu server 9.10
22:40.56jayteedid you install from packages or compile?
22:41.01carl689compile
22:41.58jayteeyour module for the TDM844 isn't loading from the looks of it. at a terminal prompt type lsmod | grep dahdi
22:42.34carl689http://pastebin.com/daadbbfa
22:42.44carl689I did load it via modprobe
22:43.03carl689if thats what your referring to
22:43.14jayteeand it looks like it's loaded now
22:44.46carl689weird huh?
22:45.15jayteeyeah, did you configure your /etc/dahdi/system.conf and your /etc/asterisk/chan_dahdi.conf files?
22:45.36carl689Yes, would you like me to pastebin those files?
22:46.55jayteeno
22:47.37jayteeif asterisk is running as a service stop the service and then stop dahdi, then start the dahdi service and see if there are any errors at the terminal
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22:49.44carl689I get 2 warnings about .conf being ignored in a future release and   wctdm24xxp: done
22:51.09jayteecarl689, when you compiled and installed did you run sudo su first?
22:52.00carl689I did it via "sudo" in front of each command as opposed to sudu su, but as far as my linux knowledge goes that should be the same thing?
22:52.20jayteeyeah, it should be
22:52.44jayteeI don't use Ubuntu for *. I pretty much stick with RHEL and CentOS
22:53.13carl689I just don't get why it was working fine the majority of the day, then after a restart, nothing
22:53.52jayteethat is odd
22:54.06carl689I MUST have changed something, though I can't even remotely imagine what it was
22:54.21jayteeif you type ls /dev/dahdi do you see all the channel devices?
22:54.42carl689yes
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22:55.39jayteecarl689, well I'm fresh out of ideas at this point.
22:56.03jayteeyou haven't done any kernel updates this afternoon did you?
22:56.11carl689Well thanks for the attempt, I am almost to the point of doing a fresh install
22:56.15carl689No no updates today
22:56.50jayteeif you update your kernel (something Ubuntu likes to do about twice a week at a minimum) you have to recompile dahdi and asterisk
22:58.11carl689If I recompile dahdi and asterisk right now, is there something I should do first or can I just redo the ./configure,make,etc...
22:59.00jayteeif you're not changing settings you can just redo ./configure, make, etc. if you change anything you need to do a make clean first
23:00.23carl689Even though I tried that earlier, I will give each a recompile first before wipe the system
23:00.30jayteecarl689, go ahead and pastebin your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf files just so I can make sure it's not a config issue there.
23:02.04carl689http://pastebin.com/d145a67e1   -< chan_dahdi.conf
23:04.01carl689http://pastebin.com/d616ce325   <- system.conf  (that one is long, still has all comments in it)
23:04.25jayteedid you type the chan_dahdi.conf file or was it generated?
23:04.37carl689type
23:04.48carl689plus a little copy and paste from a guide ;-)
23:06.20carl689Something wrong with it?
23:07.08jayteethe channel defines have to last for each group so the FXO portion looks ok to me but the FXS portion looks messed up.
23:08.10carl689which part?
23:09.07jayteechannel = 1-4 should go last after the mailbox statement
23:09.26carl689ok, I will update and see what happens
23:11.07carl689::sigh:: Got nothing.  Got really excited there as its usually something small like that , that gets me
23:12.31Qwellwaves to dwayne
23:12.47dwayneQwell, hey how's it going ?
23:13.44Qwellnot bad
23:13.46dwaynethis thunderstorm keeps making me have to restart dahdi
23:13.51*** join/#asterisk javb (n=javb@190.94.43.75)
23:13.55Qwellwent and saw Dr. Steve today
23:14.15drmessanoSteve Tortaro?
23:14.17Qwellactually Dr. Charles, but you'll get the reference I hope :p
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23:14.41jayteecarl689, I'd rename your /etc/dahdi/system.conf and create a new one with just loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8 and echocanceller=mg2,1=8
23:14.59javbI have a laptop, testing asterisk there. I would like to connect a analog line, obuosly have no pci fxo card for the laptop.. any option? USB adaptor? For testing purpose? External 56K Modem that would be considered as a Generic Clone Board?
23:17.22carl689jaytee, nope, still nothing
23:17.43carl689jaytee, I have some mates waiting on me for dinner now (patiently) so I am going to have to try this again later tonight
23:17.49carl689thanks for trying to help me out
23:18.16jayteecarl689, here's my chan_dahdi.conf http://pastebin.com/d5c654309
23:18.34jayteeI have 1 FXS module and 1 FXO module on a TDM410 card
23:19.07carl689I will give it a shot when I get back tonight, thanks again. Will update you if you are on
23:19.18jayteecarl689, after doing any changes to the conf files you have to stop * and restart dahdi and then asterisk in that order
23:19.31jayteeok, I'm usually here until around midnite
23:19.40jayteeand it's 7 pm here
23:20.03carl689Thanks again, btw that is what I am doing after changes
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23:24.17Qwellnobody got the Dr. Steve reference.  That makes me sad.
23:24.52jayteemakes me sad too, cuz I didn't get the reference
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23:29.26dwayneQwell, the snake bite guy ?
23:29.42dwaynewere you letting snakes bite you ?
23:29.45Qwellthe chiropractor on The Simpsons
23:30.28dwaynehrm.  Missing a simpsons reference really upsets me.  :-(
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