IRC log for #asterisk on 20080906

00:00.28_zoomy_What i know of there is only /etc/odbc.ini
00:00.30kdasanyone ?
00:00.45[TK]D-Fender_zoomy_:  res_odbc.conf <- pastebin it.
00:00.56[TK]D-Fenderkdas: ManxPower has already answered you.
00:01.17jayteekdas, how does the analog phone connect? a digium card with FXS or an ATA adapter?
00:01.25kdas[TK]D-Fender, then how do you set rxgain?
00:01.46_zoomy_[TK]D-Fender: sorry im sort of new to the slang. pastebin?
00:01.53Qwell~bs
00:01.54jbotmethinks bs is Banal Superficiality. So be smart: /ignore bs.
00:01.56Qwellerm, wow
00:01.58Qwell~pb
00:01.59jbot[~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
00:02.03kdasjaytee, its a normal comcast phone which i call into a DID number that acess my sip which in turn is registered with asterisk :D
00:02.12Qwellwhy was I thinking bs?  /peer
00:03.05jayteekdas, Comcast VOIP to another SIP provider for DID to your * box?
00:03.07*** join/#asterisk loca|host (n=tux@196.203.53.221)
00:03.26*** join/#asterisk sircco (n=sircco@dh207-102-87.xnet.hr)
00:03.34LemensTSTK: the xlite didnt have that option, my polycom had a spot under "Sip" tab, under "local sip port" i changed it to something different and it still does the same thing after 'asterisk restart'
00:04.15kdasjaytee, Comcast normal pstn that calls a DID that is to a SIP account and my * box regesters to that SIP
00:04.55sirccohello people, i have misdn trunk on openvox card with 4 bri.When I dial out i want outgoing cid to be some number and when call goes out it's number assigned to bri port. Trunk is mISDN/g:out/$NUMBER$
00:05.19sirccohow can i get right outgoing number?
00:06.08kdasjaytee, does that make sence ?
00:06.20jayteesense enough
00:06.50_zoomy_[TK]D-Fender: http://paste.debian.net/16489/. So I only use lines 12-19 really
00:07.27kdasjaytee, so is it my * box config ? or thc maybe the SIP don't allow user input or is it the rxrate ?
00:07.29jayteekdas, what do you have set for dtmfmode in the general section of your sip.conf and in the config for the softphone
00:08.02*** join/#asterisk genji1981 (n=gej@203.152.122.50)
00:08.06jayteekdas, dtmfmode=rfc2833 should work for 99.9% of sip connections.
00:08.49[TK]D-Fender_zoomy_: there is no section named 'asterisk-connector' in there.  You have entries that refer to a DSN by that name, but that is not the name of an entry
00:08.54kdasjaytee, but does that apply even if i am calling in through a pstn line ?
00:09.19[TK]D-Fender_zoomy_: [asterisk] , [mysql2] , [sqlserver]
00:09.25jayteekdas, yes because the last leg of the journey into your * box is SIP not analog
00:09.35LemensTSd
00:10.13*** join/#asterisk kj4acm (n=kj4acm@24-183-225-98.dhcp.kgpt.tn.charter.com)
00:10.25_zoomy_ok tnx ill try and change it. i thought it got the connection details from /etc/odbc.ini, no?
00:10.51kdasjaytee, well just tried that but no luck
00:11.03jayteereally? did you do a reload?
00:11.05[TK]D-Fender_zoomy_>Hello. Im trying to connect to a postgresql database and do a write via Set() in the dialplan. But I get this error message: [Sep 6 01:42:28] ERROR[8809]: func_odbc.c:128 acf_odbc_write: No database handle available with the name of 'asterisk-connector' (check res_odbc.conf). <-- READ YOUR OWN ERROR.  iT TELLS YOU TO YOUR FACE WHERE YOU SHOULD BE LOOKING
00:11.07kdascalling in with my soft phone and analog still same
00:11.09[TK]D-FenderDarn caps
00:11.19kdasdtmfmode=rfc2833 in the general setting of my sip.conf
00:12.30jayteedo you have relaxdtmf=yes or is that commented out? usually it's a waste of time playing with it but you could try setting it yes or comment it out if it's already set to that.
00:12.36*** join/#asterisk moy (n=moy@189.169.91.147)
00:12.54kdasjaytee, in my sip.conf ?
00:12.59jayteeyup
00:13.07kdaslet me try to put it in
00:13.27jayteeI'm not sure I like that way that was phrased
00:14.00kdasjaytee, why whats wrong ?
00:14.22*** join/#asterisk mog (n=mog@216-83-246-98.static.networktel.net)
00:14.22*** mode/#asterisk [+o mog] by ChanServ
00:14.25jayteeit was sarcastic humor for your last statement
00:15.22[TK]D-Fenderkdas: Don't worry, jaytee will help you out.  He'd practically bend over backwards to help people here ;)
00:15.35kdasjaytee, http://pastebin.com/d3bbe38de
00:15.40kdasthat is my sip.conf
00:16.08kdas[TK]D-Fender, that is re-assuring
00:16.24jayteekdas, line 3, comment it out, reload and test again
00:16.59kdasthe dtmf mode line ??
00:17.02[TK]D-Fenderkdas: Voipbuster uses "dtmfmode=inband" and requires ulaw <-
00:17.13jayteethe relaxdtmf=yes line
00:17.32jayteeoh, he's using Voipbuster?
00:17.52kdasvoipbuster for incoming
00:17.54[TK]D-Fenderjaytee: Read his PB will ya....
00:17.57kdasterrasip for outgoing
00:18.24jayteeah, ok
00:18.31kdas[TK]D-Fender, i am using ulaw its under the [kdas] context
00:19.28[TK]D-Fenderkdas: wrong MODE for them.
00:19.33kdasalso i am using x-lite for my softphone if that is any help
00:19.40_zoomy_[TK]D-Fender: sorry man and thanks... of course it works now. what i dont really get though is why i from func_odbc.conf have to redirect to a dns in res_odbc.conf which sort of has an "in-dns" and an "out-dns" which redirects to odbc.ini where the connection info is. is this just how its set up or have i really misunderstood something?
00:19.42[TK]D-Fenderkdas:  15. - Does TerraSip support the DTMF Mode (Touch Tone)?  NO. DTMF (Dual Tone Multiple Frequency) or Touch Tone is unusual within Sip-Telephony, because caller and destination have to support the same SIP DTMF Mode in order to communicate with each other.
00:19.47kdas[TK]D-Fender, i switched to inband but no help
00:20.12[TK]D-Fenderkdas: http://www.terrasip.com/xoshop/faqdesk_index.php?nav=2&faqPath=2&language=ru&t_country=ru
00:20.34kdas[TK]D-Fender, i am not using terrasip to access my voicemail though!!
00:20.35[TK]D-Fenderkdas: Terasip doesn't seem to want to support DTMF.  I have seen a few other fuck-off carriers like this
00:21.01[TK]D-Fenderkdas: PB your failed call and your new sip.conf
00:21.47jaytee[TK]D-Fender, he also said his softphone won't work with DTMF either.
00:22.06kdas[TK]D-Fender, there is no failed call. i dial into voicemail just fine but when it asks me to press 1 or enter my password it don't regonize my input
00:22.26[TK]D-Fenderkdas: Show us the call.
00:22.44[TK]D-Fenderkdas: And what softphone?
00:22.52kdas[TK]D-Fender, how ?, x-lite
00:23.08[TK]D-Fenderkdas: PB the CLI output at verbose 10, SIP DEBUG enabel;d
00:24.20kdas[TK]D-Fender, how do i do that ? in the logger.conf file ?
00:24.32[TK]D-Fenderkdas: Asterisk CLI <-------------------
00:24.51kdas[TK]D-Fender, yea how do i set the verbose to 10 and sip debug enable ?
00:25.01[TK]D-Fenderjaytee: He's all yours
00:25.08kdaslol
00:25.09[TK]D-Fendersteps out for a while
00:25.14kdas[TK]D-Fender, i got it
00:26.18kdas[TK]D-Fender, the sip debug output is a crap ammount you want one bit of it or should i copy it all ?
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00:27.58kdas...?
00:28.06jayteekdas, he's stepped out.
00:28.12kdasoh
00:28.17kdasjaytee, so no ideas ?
00:30.24jayteewell, you can do what he asked and set verbose 10 and sip debug and pastebin a call and I'll look at it but I'm not even close to being in the same league as [TK]D-Fender for debugging.
00:30.59kdasjaytee, but i  mean the sip debug output is crazy it every single packet
00:31.46kdaswell i will play arround and try some more and then come back till then thanks
00:32.01jayteekdas, just copy and paste from when the call starts to when you try a couple keystrokes and hangup
00:34.14*** part/#asterisk dawalama (n=dawalama@wsip-68-105-250-226.sd.sd.cox.net)
00:42.08*** join/#asterisk CybDev (i=cybdev@unaffiliated/cybdev)
00:43.16[TK]D-Fenderjaytee: phew... saved myself on that one....
00:43.47jaytee[TK]D-Fender, haha, yep got out just in time.
00:44.33jaytee[TK]D-Fender, did you checkout that link I posted earlier with the guy's ad?
00:45.06[TK]D-Fenderjaytee: If I had a nickel for every twit who wouldn't "show me the money" in this channel, I'd put them in a bag an use it to beat the next one FUCKING SENSELESS as a warning to others...
00:45.20[TK]D-Fenderjaytee: Yes, I watch DIGG like a hawk too....
00:45.39[TK]D-Fenderbreaths out.....
00:45.40jayteethat guy was really pissed, it was so funny
00:45.42[TK]D-Fenderouuuuuuuummmm
00:46.03jayteeom mani padme hum, breathe! om
00:47.42[TK]D-Fenderjaytee: padme... yup, Nathalie Portman... I'd hit that :)
00:47.47*** join/#asterisk althiom (n=pirch@189.237.8.67.cfl.res.rr.com)
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00:47.55jayteeI loved V for Vendetta
00:48.03[TK]D-Fenderjaytee: Ditto....
00:48.08tristanbobanyone work with Vertical sip phones?  will they work with asterisk?
00:48.12[TK]D-Fenderand on that note I'm off to play pool for the night
00:48.13tristanbobhttp://www.vertical.com/index.html
00:48.24jayteeand I'd put up with Jar Jar just to see Natalie
00:48.43jaytee[TK]D-Fender, have fun
00:54.09althiomdoes *now support fax over ip??
00:54.36ManxPoweralthiom: I guess it depends on how you define "support".
00:54.43jayteealthiom, not reliably but then no other SIP based VOIP system does either
00:54.52ManxPowerThere's T.38, with all the limitations Asterisk has with T.38, but that's about it.
00:55.36ManxPoweroh, look at that.  newbie wrote "*now" instead of AsteriskNow.  Well, bless his heart.
00:56.07althiomI have a voip connectin and would like to use FOIP on it
00:56.14UDyou did too
00:56.16ManxPoweralthiom: give up now.
00:56.34ManxPowerUD: got your problem solved yet?
00:56.41UDya
00:56.45althiomand was wondering if it was supported
00:56.47UDi had to walk away from it for a while
00:56.51ManxPowergood.
00:56.56jayteealthiom, get a POTS line and a real fax machine or teach people how to attach documents to email.
00:57.08ManxPoweralthiom: perhaps you could ask on the #AsteriskNow channel?
00:57.24jayteeI'm running AsteriskLater, it's much better :-)
00:57.25althiomalright and thanks
00:57.34UDsorry to have been a pita
00:59.42UDi didnt even know there was a debug command
00:59.43UDheh
01:00.06ManxPowerIf you had read the damn book you would have.
01:00.11althiomalright, then I am having installing asterisk.  I can get to the change the password screen and it wil not allow me to do anything
01:00.19UDdug this hunk of junk computer out of the attic and had so many other issues with it that it was frustrating
01:00.58*** join/#asterisk Corydon76-dig (i=gray@pdpc/supporter/bronze/Corydon76-home)
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01:01.12UDi read through the old version a few times
01:01.49UDit isnt as educational when you dont have a working config to test stuff out on
01:03.14UDthe book isnt the most clear IMO
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01:04.50UD1st edition i mean
01:05.02UDim going to to burn it
01:06.03*** part/#asterisk althiom (n=pirch@189.237.8.67.cfl.res.rr.com)
01:07.15UDManxPower do you run asterisk on your myvzw connection?
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01:13.45jayteeI found a link to a pdf of The Asterisk Handbook by Mark Spencer and boy that was really old.
01:14.09Qwelljaytee: you aren't kidding
01:16.26jayteeI was googling for hotline and asterisk as keywords and that was one of the hits. I want to setup an analog phone with an FXS ATA adapter that when you go off hook it autodials our security department.
01:16.28*** join/#asterisk Yourname (i=Yourname@unaffiliated/yourname/x-837320)
01:16.56Qwelljaytee: zapata.conf, immediate=yes
01:17.55jayteeQwell, yeah but I need this to be remote on an ATA like a GS Handytone or a Linksys SPA2102, preferably the latter.
01:18.53*** join/#asterisk beighto (n=chatzill@57.155-62-69.ftth.swbr.surewest.net)
01:19.39beightoAnybody have experience trunking Cisco Call Manager and Asterisk?
01:21.28*** join/#asterisk Entr4nced_ (i=IMG001@102.sub-75-218-197.myvzw.com)
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01:22.06jayteeI've been digging through the Linksys documentation which is pretty crappy looking for a way to do the same thing as immediate=yes but haven't found anything like it so far.
01:22.14jayteemay end up having to call Linksys.
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02:00.34KylerI need to get a unique (or even close to unique) ID for a call...without a dot in it.  Is there a simple way to do that?  I see the patch for a negative substring length but I don't want to patch.
02:01.34KylerOh!  Is there a float to integer conversion?  Looking...
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02:30.35rpmanyone know where i can find the mitel sip firmware 7.0.0.8? i can only find 6.0.0.19...
02:43.24*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)
02:46.47jameswf-homeneat http://www.nathanpralle.com/software/ast_masterlist.html
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02:50.44jayteejameswf-home, that's a great link, thanks!
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03:15.03jameswf-home~itsplist-us
03:15.03jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
03:16.45*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
03:16.45*** mode/#asterisk [+o mog] by ChanServ
03:17.06jeevmy friends office's people are complaining about hang ups.. a lot lately
03:17.10jeevi dunno if it's my ITSP or not :/
03:23.34techman97jeev:  I'm in the same ballpark, just in the last 4 weeks
03:25.35jeev:/
03:25.41jeevwhat is that ITSp that mitnick faggot used
03:26.20ManxPowerjeev: what protocol are you using?
03:26.40jeevManxPower, from the local server to the main server, iax
03:26.48jeevthen iax2 goes to ITSP via SIP
03:27.35jeevso -internal network with 30 phones- - -internal asterisk- iax - nat - internet - iax - asterisk server - sip - internet - sip - itsp
03:28.15ManxPowerjeev: I have personally experienced situations where IAX2 randomly disconnected calls.  Also, if you keep everything as SIP you can do reinvites internally
03:28.55ManxPowerNow,this "personal experience" was several years ago and I would be shocked if they had not fixed the problem by the recent releases.
03:28.58jeevManxPower, i dunno what to do.. i couldn't for all my life, figure out how to use the internal asterisk box to connect to my external one via SIP
03:29.08jayteeplus you've got two internet hops and routing through 2 asterisk servers. lotta opportunity for latency
03:29.12jeevif the problem is my dual wan, i will remove one of the ports.
03:29.23jeevnat just isn't working.
03:29.34jeevdual wan isn't helping, no matter how my nat is set up.
03:29.37ManxPowerNAT is hard to do in complex situations.
03:29.51jeevyea, it's so weird, it's like i'm trying a hosted pbx but it's not working via sip..
03:29.54jeevand i know it's fully possible
03:29.54vipcarrierdoes any one has a php script to provision cisco 79XX
03:29.58ManxPowerjeev: people frequently end up NATing internal traffic by mistake.
03:30.01jeevjaytee, what's that ITSP that needs a $35 deposit ?
03:30.05vipcarrierdoes any one has a php script to provision cisco 79XX
03:30.32jayteejeev, you mean FWD? the one that used to be free.
03:30.33jeevManxPower, what do you mean by that? internal traffic is just fine
03:30.36jeevhmm
03:30.40ManxPowerjeev: can you set up a GRE tunnel between the sites or are you using consumer routers?
03:31.11jeevisn't that a type of tunnel? like vpn /
03:31.22ManxPowerjeev: VPN without the encryption
03:31.28jeevjaytee, flowroute.
03:31.33jeevany luck with flowroute?
03:31.36jeevManxPower, definitely.
03:31.40jeevthe dual wan router at the office is a freebsd box.
03:31.43jayteejeev, never used it.
03:31.45jeevso is my asterisk box at the datacenter.
03:31.51ManxPowerjeev: Ah.
03:32.08jeevyou suggest?
03:32.16jeevvpn would be great too..
03:32.22jeevbut i just feel like it'd add massive latency
03:32.26jeevbut the crap part is the instability.
03:32.28jayteelotta people running * on freebsd without trouble, doubt if it's your * boxes.
03:32.38ManxPowerjeev: About 5 suggestions, none of them easy to explain in my somewhat intoxicated state, but I'll try.  Give a min.
03:33.08ManxPowerjeev: not VPN.  A gre tunnel would not have encryption latency but would allow you to talk between your networks WITHOUT NAT
03:33.09jeevjaytee, i know..
03:33.17jeevManxPower, i will do that.
03:33.20jeevi've got to read on it
03:33.25jeevjaytee, i'm only saying it could be an IAX issue
03:33.27ManxPowerIt would have a little added latency from the tunnel.
03:33.47jeevtotally ok, the connection to my servers is ~10ms.
03:33.58ManxPoweralso if you did that and switched to sip (since there's no nat) you could have phones doing reinvites anywhere except to the ITSP.
03:34.11jeevhm
03:34.22ManxPowerYou won't notice less than like 150ms
03:34.30jayteejeev, yeah possibly. any weird errors for IAX in the logs?
03:34.46jeevi can go back to the log, i have debug enabled in both places
03:34.50jeevbut i've got to step out REALLY soon
03:34.55jeevi can't believe i'm being helped but have to leave
03:34.56jeevthis sucks
03:35.29ManxPowerjeev: I'll be around this weekend
03:35.34ManxPowerusually late mornings for a while
03:35.36jayteeManxPower's suggestion sounds like a more reliable solution with your setup.
03:35.38jeevhmm
03:35.41jeevyea yea
03:35.52jeevi'm aboutto deposit $35 into flowroute to give them a full shot for about a week
03:35.57ManxPowerMy suggestion should be much much simpler, assuming you can easily to a tunnel
03:36.01jeevit's already annoying, they text me every day saying this got hung up, that got hung up
03:36.04jeevthey can't hear me.. this and that
03:36.14ManxPowerjeev: welcome to MY world.
03:36.16jeevi'm sure tunneling via the two bsd boxes would be great.
03:36.22jeevi'm almost to the point of getting ATT linse here
03:36.22jeevlines
03:36.30ManxPoweryou can easily encrypt the tunnel later if you want.
03:36.32jeevif you get an unmetered $50
03:36.35jeevper month line
03:36.39jeevhow many concurrent's could there be? 2?
03:37.22ManxPowerAs much as I think FreeBSD is way over rated, it's far, far better then a linksys or netgear.
03:37.41jblackfor $50/unlimited, 2 concurrent is common
03:37.50jeevhmm
03:37.55jeevi dunno what to do man, i'm getting TIRED of this crap.
03:37.59jblackManxPower: yeah, I can believe that.
03:39.11jayteeI had a real weird one today. Guy at DID number 630-2031 keeps getting calls that aren't for him and the callers have said they get 630-203-1111 on their callerid which is Hinsdale, Illinois. I'm in Indianapolis 317 area code. All I can think of is they see the number and try to call it back without a 1 and matches 7 digits before they can finish dialing the whole thing. but if I dial from a cell either with a 1 or without I get a message from MC
03:39.11jayteeI that the number isn't in service. If I go through my telco I get my telco's not in service message.
03:40.07ManxPowerjeev: try sip, it can't hurt and will be easy with a tunnel
03:40.42jayteeI think it's MCI's routing screwing up or some call center is calling people in my area code and they're trying to dial the number back without a 1
03:41.16ManxPowerhe could ask him what number they dialed.
03:41.24jeevweird
03:41.26jeevok i've got to go
03:41.32jeevManxPower,i will try SIp asap and let you know.
03:41.32jayteeManxPower, he did
03:41.33jeevbbiab
03:42.56jayteeI told him to get me more specifics. Did they dial a 1? did they dial the full 11 digits? what area code are they calling from, etc. says if he gets another call like that he'll get more info for me. Started happening 3 days ago and has happened several times now.
03:42.59vipcarrierguys
03:43.20vipcarrierany one has a php scrtip that can generate configs for cisco 79XX
03:43.29vipcarrierI'm willing to pay for it
03:43.34vipcarrierit is an urgent
03:43.40ManxPowervipcarrier: I doubt anyone different from the 10 other times you asked.
03:43.54jayteelol, I was just thinking that
03:44.19vipcarrieri can't belive that u guys provisiong every singel phone manualy
03:44.26ManxPowervipcarrier: not that many people use Cisco phones here.
03:44.45jayteevipcarrier, I don't need php scripts for Polycoms
03:45.01ManxPowerPolycoms are just as good quality and they don't have weird firmware restrictions.
03:45.07vipcarrierjaytee how do u provision them every phone manyaly
03:45.29ManxPowervipcarrier: you change 3 lines in a template config file and plug in the phone
03:45.33vipcarrieri don't like polycom's becouse they have a web gui and it is not secure
03:45.36jayteevipcarrier, I provision them using FTP and the built in firmware logic in the phone.
03:45.45jayteeyou don't have to use the web gui
03:46.00vipcarrierbut it is not secure in corporate envirement
03:46.12jayteeand the web gui doesn't give you the ability to modify all the phone's capabilities.
03:46.14ManxPowerin fact, the web gui on the polycoms is only good for doing your first phone config and generate the template file.
03:46.27ManxPowervipcarrier: you can disable the web gui.
03:46.37vipcarrierhow can u disabel it?
03:47.04vipcarrieri remember all Asterisk@Home had a provisioning script for Cisco's but I can't find any where that source
03:47.10ManxPowerread the docs
03:47.11jayteeyeah, like if you want to enable ring-answer so the phone will autoanswer a call for paging.
03:47.20ManxPowerthe polycoms have extensive docs for their phones
03:47.26jayteecan't do that in the damn web gui
03:47.41ManxPowerjaytee: I've not touched the Polycom gui in YEARS.
03:47.53vipcarrierI have few polycoms that I'v got for testing but not really happy with them
03:47.54ManxPowerI suppose we really should disable it.
03:47.58vipcarrierI hav 330 and 650
03:48.19jayteeManxPower, I've only done it for a couple phones for initial testing but so much of the feature set capabilities are lacking in it.
03:48.19ManxPowervipcarrier: the more you use the polycoms the more you will like them.  The more you use Ciscos the less you'll like them.
03:48.43vipcarrierlook I had an aastra for 2 weeks and i cracked it
03:48.47vipcarrierpolycom 650 dead
03:48.51vipcarriercisco 7960 works
03:48.58ManxPowerbest of luck
03:49.18vipcarrieru can traw the phone in to the brick stone wall and it will keep working
03:49.27vipcarrieraastra 57i CT dead
03:49.34vipcarrierafter i droped a had set on it
03:49.42jayteeif you add a block in the phone's config to handle the ring-answer you can setup paging using SIPAddHeader and the Page application and it'll ring all the phones once, they'll autoanswer and voila. real sweet
03:51.40vipcarrier$200 for web gui script to provision Cisco 79XX
03:52.23jayteeif I knew Cisco phones and PHP I'd take you up on that offer but I doubt if you'll find anyone in here with both those skillsets at the level you need.
03:52.44Qwellwhere's bkruse..
03:53.27jayteegood question, he's not here or in asterisk-gui but then you already know that
03:53.56jayteewe should "chip" that dude so we can track him with a GPS unit.
03:54.30Qwellvipcarrier: trixbox has such a script.  Feel free to send the money via paypal to my email address
03:55.53*** join/#asterisk sah-work (n=Bawbatos@adsl-76-227-17-18.dsl.pltn13.sbcglobal.net)
03:56.45Qwellcrickets...
03:57.35jayteewell, it's night. rather have crickets than cicadas
04:01.11*** join/#asterisk moy (n=moy@189.169.91.147)
04:03.59Qwellguess not
04:06.39*** join/#asterisk Thorn (n=thorn@unaffiliated/thorn)
04:08.58*** join/#asterisk c4t3l (n=root@74.95.210.124)
04:09.14*** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec)
04:13.16*** join/#asterisk kamanashisroy (n=kamanash@119.30.34.7)
04:14.45jameswf-homeQwell: any personal itp  recomendations
04:14.51jameswf-home*itsp
04:14.52Qwellitp?
04:15.03Qwelldunno, I've always liked nufone
04:15.35Qwell(Jeremy isn't paying me to say that)
04:16.48jameswf-homeshould do this " * Asterisk configuration assistance is $99 per hour, one hour minimum."
04:16.53mchoujameswf-home: all recs are based on your call volume
04:17.45jayteejameswf-home, where did you see that?
04:18.12mchou$99 is probably a deal :)
04:18.13jameswf-homeI like les net but wife wants a local DID
04:18.27mchoujameswf-home: where you live?
04:18.31jameswf-homeaz
04:18.42mchoujameswf-home: call volume?
04:18.44jayteeMcCain country
04:19.18mchoujameswf-home: minutes/month or week?
04:19.23jameswf-homewell she talks to me which would be over an iax trunk to work... heh probably ledss than 500 min/mo
04:19.29jameswf-home*less
04:19.46mchoujameswf-home: diamondcard.us
04:20.04mchoujameswf-home: strictly paygo
04:20.26mchoujameswf-home: they have DIDs nationally for all of UD
04:20.30mchouUS*
04:20.32jameswf-homeI should just trunk in to work and use company lines...
04:20.39jblackI'm using callwithus, diamondcard and voipstreet.
04:20.56mchoumore than what's listed on their (crappy) web site
04:21.17jblackCWU is .0125, diamon is .015 and voipstreet is .017. CWU has less DIDs, diamond and voipstreet had 570 area code numbers, which are hard to get.
04:21.29mchoujblack: of the three, which is most feature complete?
04:21.37techman97voipstreet...*shudder*
04:21.47jameswf-homeI found callwithus to be latency ridden
04:21.49jblackThey're all quirky.
04:22.04techman97I've been good with callcentric for the last year, but their fax service is broken permanently
04:22.15mchoudiamondcard is a bit sucky becase they dont set the CNAME field in callerID
04:22.17jblackThey all seem reliable, but none of them have that professional sheen to 'em.
04:22.36jblackThat's true. Diamoand card doesn't do CNAME. I forgot to mention that.
04:22.48mchouCNAM*
04:22.57jblackyah ayah yah. That's still the case.
04:23.02jameswf-homewhat is fax?
04:23.05mchoulol
04:23.08jblackas in, "this morning".
04:23.23mchouI dont understand why ppl still use faxes
04:23.24techman97the only kickers about CallCentric are that they don't do CNAM and their DIDs are done via SIP RFC, not traditional DNIS/ANI
04:23.26jblackjameswf-home: A legally binding way to electronically transfer images of paper.
04:23.35mchouI bought a house w/o using a fax :)
04:23.40techman97jblack:  exactly
04:23.41jblackmchou: because the law doesn't know what teh internet is yet.
04:24.00mchoujblack: wha???
04:24.05jblackRemember the lawmakers themselves, a "series of tubes"?
04:24.05techman97either it's legal, or your customer base are 3,000 60+ year old geezers who are scared of the internet
04:24.07techman97rofl
04:24.19mchoujblack: you mean all that stuff I ordered on line I can have for free?
04:24.21jameswf-homeI hear email hasn't caught on and no one really has an email address
04:24.31techman97yeah, the interwhat?
04:24.35techman97that's just a fad.
04:24.36jblackLaws move slow.
04:24.58techman97I drove my sherman tank across Europe at 30MPH without no "interweb"
04:25.02mchoujblack: so internet mechanidizing is legalized robbery? :)
04:25.16mchoumerchandizing*
04:25.19jayteeit is technically impossible to attach a digital document to an email, just ask any higher up exec that is not in a technical field, they'll tell ya!
04:25.30jblackWhy are you guys picking on me? I'm not in congress!
04:25.36jblackHell, I know how to use all 110 keys!
04:25.47mchoujblack: cause you bought up the law
04:25.57techman97jblack:  where's the "any key"?
04:26.00jblackbecause someone asked why fax hasn't died. What's why.
04:26.50jblackIt hasn't died because legal documents remain legally intact over a fax.
04:27.17mchoujblack: I bought a house w/o using a fax :)
04:27.36mchoujblack: that better be legal!
04:27.51jayteemy first PC was a piece of crap Packard Bell 286 12mhz machine. when most systems had 104 key keyboards this one came with a 105 key keyboard, the extra key labeled Macro. Nothing in the owners manual. Called their tech support. "What's it do? How do I use it?" "Um, it doesn't work and we don't ship those anymore"
04:27.55mchoujblack: or you're gonna hear from my lawyers :)
04:28.22*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
04:29.05jameswf-homeok so voipstreet what are the cons
04:29.09mchoutechman97: callcentric doesnt do CNAM either???
04:29.23techman97you're hard pressed to find a provider that does
04:29.34mchoutechman97: ipkall does
04:29.37techman97I had one that did, but they dumped their commercial support - voiceeclipse
04:29.44mchoutechman97: and they are "free"
04:29.58mchouok, I take that back
04:30.07mchouCNAM is geographic info
04:30.18mchounot "full name"
04:30.42mchoubut even geographic info is better than just the phone #
04:31.00jameswf-homeeh just use agi who needs cnam
04:31.10mchouwhat???
04:31.27jblackcallwithus, voipstreet and teliax do CNAM.
04:31.34mchoujameswf-home: tell me how that works
04:31.50mchoujameswf-home: how does agi provide CNAM?
04:31.51jameswf-homevoip street doesnt says so in their faq
04:32.00jblackYou want me to call you right now and verify?
04:32.15mchoujblack: stop smoking dope
04:32.32mchoujblack: we are talking inbound CNAM
04:32.35jblackPardon? Why do you think I'm smoking dope?
04:32.42mchoujblack: we are talking inbound CNAM
04:32.53mchoujblack: that's why you smoking dope
04:33.22jameswf-homemchou: Google, whitepages.com
04:33.22jblackOk. I know that voipstreet does incoming CNAM, as does Teliax. IPKall cid is completely broken. I don't know about the other two.
04:33.36mchoujameswf-home: screw that
04:33.52mchoujameswf-home: is outdated and unreliable
04:33.59jameswf-homewhocalled.us is valuble
04:34.14mchoujameswf-home: on whocalled.us we can agree
04:34.17jblackWe need to make a voipinfo table.
04:34.24jameswf-homeif your in whocalled.us you go to telemarketer torture
04:34.48mchoujblack: wtf are you talking about?? Ipkall has geographic CNAM
04:35.34jblackIt's badly broken. One of their internal hops eats incoming CID and replaces it with SEATTLE   WA, and the number with a seattle number.
04:35.47mchoujblack: in fact it works better than other *paid* ITSP as far as CNAM is concerned
04:36.04jameswf-homegeographic wouldnt be to hard
04:36.19mchoujblack: maybe, but I've NEVER had that happen
04:36.26jblackYou can find more information on voxilla.
04:37.13jblackThey tend to work when calls are originated with pstn, and tend to break when they come in on voip.
04:37.35mchoujameswf-home: yeah, except apparently no ITSPs seem to do it
04:38.14mchoujblack: maybe, but I have those calls and it has NEVER happened to me
04:38.34jblackCall IPKall with callwithus, voipstreet or teliax if you'd like to verify.
04:38.52mchoujblack: it could be DID exchange related but I'm dubious
04:39.06mchoujblack: lol
04:39.12jblackI had trouble believing it too. It took me several months to become certain.
04:39.45jblackI found out the hard way, when I was using IPKall to test which providers did CID, and they were all breaking.
04:39.49mchoujblack: how the hell do you KNOW that's not an issue with the othe ITSPs you mentioned?
04:40.05mchoujblack: put down the vrack pipe man
04:40.23jblackI'm getting tired of your insults. Be nice.
04:40.41mchoujblack: especially since we pretty much concluded that those ITSP dont do CNAM
04:40.51mchouwho is insulting you?
04:41.03jblackKNOW is a big word. Those three ITSPs work fine when calling other ptsn's, and when calling each other.  When IPKall is the destination, all the CID breaks down to SEATTLE    WA, with a seattle number.
04:41.07mchouI'm challenging you on points of fact
04:41.24jblackForget it. Plonk
04:41.38mchoujblack: good riddance
04:42.07techman97ok, so here's a weird problem guys....so I set my SIP phone to register to my asterisk box via IP, works JUST fine.  I set it to register by the internal DNS name, "asterisk", and it doesn't register.  Syslog shows me that it like 1/2-way registers, but when I dial, an address of 0.0.0.0 shows with an error.  I know the DNS is setup right, because I can ping it from other computers by name...but the phones just don't deal with
04:42.53techman97I have 3 other PBXs up and running in the same manner (on different networks)...no problems there
04:43.17techman97I have sip debug on, and I see messages coming from the phone in question with the DNS name in the SIP registration info...
04:43.17mchoutechman97: I'm dubious your DNS is working right based on your description
04:43.26jblacktechman97: Perhaps you forgot to tell the phones where the dns server is, or what domain to default to.
04:43.48jblackthey may be looking up "asterisk" versus "asterisk.yourdomain.com", to be more clear.
04:43.53techman97phones get address via DHCP, and I have all of the options set.
04:43.59techman97hmmm, let me try the FQDN
04:44.03techman97maybe you're right jblack
04:44.05jblackdhcp doesn't set the domain, iirc.
04:44.05mchoulol
04:44.13jblackor is that ppp.....
04:44.17rob0ping(1) does not necessarily use DNS. In fact it doesn't use DNS directly, it uses system gethostbyname() function.
04:44.27techman97jblack, yeah it does, DHCP option 15
04:44.43jblackYou may not be setting it, or it may be overridden in the phone?
04:44.50jblackAnyways, using fqdn will verify.
04:45.02techman97I pinged the hostname from another linux box and a few other misc. windows boxes - works fine as far as I can see
04:45.03techman97:S
04:45.06techman97yeah, FQDN test we go.
04:45.30mchoutechman97: dont use ping.  use dig :)
04:45.44techman97ok, phone rebooting...
04:45.56techman97syslog and CLI debug rockin'
04:46.09techman97c'mon you pig
04:46.45techman97crapsticks...that was the deal.  FWDN
04:46.48techman97FQDN*
04:46.52jblacksuspected as much.
04:46.55techman97man I feel stupid...ROFL
04:47.00jblackNah, don't.
04:47.12mchoustupid is as stupid does
04:47.25techman97now to figure out what that whole deal is about
04:47.26techman97rofl
04:47.29jayteewhat does baffled mean?
04:47.35jblackconfused.
04:47.40jayteehehe
04:47.48jblackOr drapped with layers of cloth in a rippling fashion.
04:48.11techman97jblack: Nice Trading Spaces reference
04:48.25jayteethat was one of my favorite lines from Highlander
04:48.31*** join/#asterisk obnauticus_ (n=obnautic@about/windows/regular/obnauticus)
04:48.33jblackis confused.
04:48.46jblackTrading spaces? Is that a stock market movie or something?
04:48.53mchoulol
04:49.01mchouthat's Trading Places
04:49.18techman97hehehe
04:49.44jblackturns pink
04:49.45techman97Trading Spaces = home redesign show on TLC.  My wife is nuts over those shows.  :S
04:49.52jblackYeah. Just googled it.
04:50.11jblackMy idea of fashion is "Hey kid. See those spiderwebs on the ceiling? I'll let you eat dinner if you get rid of them"
04:50.25techman97amen to that.
04:50.42techman97hell, my daughter redecorated our house recently with crayons on the walls....=/
04:50.49techman97I'm perfectly fine with that
04:50.52jblackDid you whip her with a chain?
04:51.11jblackOh geeze. You're such a wimp. :)
04:51.11techman97not initially no, started with the cat...then moved to the chain
04:51.27techman97cat makes great noises while using it as a beating stick
04:51.31jblackYou whipped your kid with a cat? That's cruel! Those things have claws
04:51.56techman97not mine...it would fight back, but we got rid of his balls too...
04:52.08jblackYou want to cut into their phsyche, not their skin.
04:52.13jblackpsyche, that is
04:52.30techman97=)
04:52.45jblackI trained mine to jump up and put her shoes on every time my keys rattle.
04:53.12techman97irl, I just smiled and started looking at buying some crayon-resistant paint...yeah, wuss am I.  My daughter has me wrapped around her little 3yr old pinky
04:53.27techman97pavlovian kid - love it
04:53.30jayteeawwww
04:54.00jblackYeah, it's great.
04:54.21jblackShe's learned that I have no problem leaving her (she's 14, after all). She also knows that if she can beat me to the car, I'll probably get a milkshake on the way.
04:54.37techman97hahaha
04:55.05jblackSo yeah. A tiny jingle, and she's a lightning bolt.
04:55.09techman97oh man...boy phase.  I thankfully have a decade before that really starts.  I swear I'm going to chain my kid to the basement floor until she's married
04:55.41jblackAll you have to do is tell her that anyone that gets her pregnant will die. And then convince her it's true.
04:56.11techman97obedience through intimidation...uncle buck style.  "Hey bug, I have a hatchet in the trunk, wanna see it?"
04:58.15mchoutechman97: lol.  dude, that wont work.  You can be Governor of some State and it still wont work :)
04:58.27techman97=)
04:58.51jblackAbsolutely. I'm calm and mellow when she gets straight As, is respectiful, polite and does what she needs to do.
04:58.58mchoutechman97: apparainte you can fire state troopers though :)
04:59.05techman97ha!
04:59.07mchouapparantly*
04:59.27jblackI work hard to be a great dad when she's a good kid. Deviation, though, results in a 240 force of nature with a bad attitude.
04:59.43techman97as it should.
04:59.53jblackShe once took a dollar out of my wallet for ice cream without telling me. That resulted in her bedroom being turned upside down.
05:00.05jblackI have no idea how she righted the bed.
05:01.04jblackShe learned quick that she can have a dollar from dad if she asks. Not asking is not a good idea.
05:02.20jblackThat only works because she's a good kid. If she weren't naturally good... <shudder>
05:03.36techman97=)
05:04.46*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
05:09.32mchouI wonder how telemarketer equipment works.  Apparently they have some tech that knows a fake SIT from a real one.
05:10.21mchoutaht effectively renders Zapateller function relatively useless
05:10.27mchouthat*
05:14.46jblackWhy won't you move, Mr Bubbles?
05:14.46techman97damn, I was kinda hoping for a Wootoff tonight...o well.
05:14.55jblacknope. no wootoff. Not even a bag of crap
05:15.04techman97telemarketing stuff - they dump amazing amounts of money into ensuring that you hear their blather.
05:15.41mchoutechman97: yeah, but _how_ does that equipment work?
05:15.51techman97that's the magic, and how they can get companies to buy it
05:15.52techman97lol
05:16.02techman97some of it is the PRI result code that is sent back...
05:16.15techman97Asterisk as far as I know just plays audio
05:16.24techman97it doesn't actually return a PRI code
05:16.47mchoutechman97: yup
05:17.29*** join/#asterisk CrazyTux[m] (n=CrazyTux@ip68-111-67-4.oc.oc.cox.net)
05:17.31techman97I've installed my fair share of those kind of dialers
05:17.49CrazyTux[m]techman97, what kind is that :)
05:17.49mchoutechman97: pls stop :)
05:17.53scooby2with ChanSpy() do you just hangup when done?
05:18.25techman97the majority of what i did was for a company (been bought and sold 5+ times since 2001) was CellIT in Miami - their own dialer.
05:18.32techman97then some Buffalo dialers and whatnot
05:18.46mchoutechman97: tapeworms have more value than telemarketers
05:18.46techman97they became Concerto I think...then was bought by Davox maybe?
05:18.59techman97mchou: correct!
05:19.27techman97you know what has even less value than telemarketers?
05:19.44mchounot really
05:19.53techman97hard account debt collectors...man, being a fly on the wall in some of those places makes you really never want to be late on a bill
05:20.43techman97they have a value to the people they're collecting for, yes....but I swear those people ENJOY being dicks on the phone LOL
05:20.45mchouwell, I'm sure those folks have their uses
05:21.03mchouthey at least purport to go after deadbeats
05:21.32mchouwho do telemarketers go after?  Innocent bystanders
05:21.40techman97the industry does have value...especially in the current US economy...but the PEOPLE that perform the task are just...interesting people
05:22.02mchoutechman97: that's bullshit
05:22.14techman97what, the economy comment?  lol
05:22.15mchoua fake economy is NO economy
05:22.54mchouthat's like saying prositutes do good since they let men let off steam.
05:23.00techman97this could be an all-nighter discussion...hehehehehehehe....
05:23.33*** join/#asterisk MrNaz (n=naz@ppp118-208-176-116.lns10.mel4.internode.on.net)
05:24.29mchoualso especially considering even in this economy all the telemarketer calls I get is "reduce your mortgage" or some such BS
05:24.47techman97going after the perceived pain points....
05:25.33mchouthat and the dish network :)
05:25.37mchouhaha
05:25.41techman97LOL!
05:26.17mchouPITA is more like it
05:26.24techman97I just can't wait for the Cialis / Viagra telemarketer calls @ dinner time
05:26.44mchouI'm BUSY right NOW! :)
05:26.52techman97"Oh NOW you call..."
05:27.44mchouspeaking of which I just found out how those drugs work
05:27.50mchounitrous
05:28.13mchouI'd rather go to the dentist, thanks :)
05:28.42techman97I have this idiot "eternal frat boy" friend of mine that brags about a Viagra and Red Bull drink he does
05:28.48techman97why oh why
05:29.09mchousigh
05:29.31techman9732 years old, drank himself though college
05:29.42techman97"Frank the Tank" from Old School?  yeah.
05:29.42rue_mohrsorry, can you repeat that in context of a voip phone system?
05:30.06jblackBioshock is a great fps.
05:30.16techman97Goto(toilet_fast,s,1)
05:31.01mchouexten => s,1,DrivePorcelainBus
05:31.12[TK]D-Fendertechman97>I have this idiot "eternal frat boy" friend of mine that brags about a Viagra and Red Bull drink he does <- so he's proud that he needs help getting it up or even staying awake?  Must be the stud of the retirement community
05:31.28techman97aye aye
05:31.49techman97he could absolutely fit on this website:  http://www.hotchickswithdouchebags.com/
05:34.36*** join/#asterisk vipcarrier (n=vipcarri@ool-44c65236.dyn.optonline.net)
05:34.57*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
05:36.46oilinki<PROTECTED>
05:44.59*** join/#asterisk vipcarrier (n=vipcarri@ool-44c65236.dyn.optonline.net)
05:55.52[TK]D-Fenderok, bedtime.
05:55.55[TK]D-FenderLater all
05:56.08techman97I think I should head out too - l8r all
06:12.04*** part/#asterisk c4t3l (n=root@74.95.210.124)
06:47.16scooby2hrm so I'm supposed to use exten   =>      *96,n,Playback(agent-loggedoff)
06:47.16scooby2exten   =>      *96,n,Playback(goodbye)
06:47.16scooby2exten   =>      *96,n,Hangup()
06:47.21scooby2ack
06:47.33scooby2sorry about the paste
06:49.26*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
06:49.28*** join/#asterisk juanjoc (n=juanjoc@host120.190-226-84.telecom.net.ar)
06:55.28vipcarrierany one wanna job at clifton NJ/
06:55.39vipcarrierlooking for expert Asterisk/SER
06:55.48vipcarrierfull time
06:55.54vipcarrieror part time
06:57.00*** join/#asterisk Levonk (n=lk@adsl-75-62-136-40.dsl.lsan03.sbcglobal.net)
06:57.30jblackvipcarrier: Hmm.
06:57.46jblackchecks the distance from wilkes.barre
06:58.27jblackCan you take a part timer that would primarily telecommute?
07:02.47*** join/#asterisk af_ (n=getsmart@88-149-240-50.dynamic.ngi.it)
07:04.59scooby2man why does AgentCallbackLogin() say agent logged in when I am trying to logoff?
07:08.17jblackwow. ser looks pretty low level.
07:08.37*** join/#asterisk sah-work (n=Bawbatos@adsl-76-227-17-18.dsl.pltn13.sbcglobal.net)
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07:36.41FabiOnehi all
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09:45.00sirccoanyone here using trixbox with multiple bri ?i need idea how to work out something
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11:40.33BarthezZhmmfg weird, I've updated my asterisk from 1.2 to 1.4 (bristuff version 0.3 to 0.4) and now i get some additional warnings i can't really place
11:40.51BarthezZ[Sep 6 13:39:46] WARNING[3647]: app_queue.c:3930 queue_exec: Unable to join queue 'mainqueue'
11:40.53BarthezZfor example
11:41.22gr0mita lot has changed from 1.2 to 1.4
11:41.24mvanbaakwell, that message is very clear isn't it ?
11:41.48BarthezZmvanbaak not really, the queue is there when using show queues etc....
11:42.07mvanbaakBarthezZ: are there any agents logged in ?
11:42.23BarthezZyes 6
11:42.28mvanbaakBarthezZ: if not, check the queueconfig. the thing you want to check is joinwhenempty
11:42.52mvanbaakany free agent ?
11:43.53BarthezZall are free
11:44.19BarthezZhmmfg, i commeted out joinempty and it now seems to work
11:44.24BarthezZthanks for the pointer :)
11:45.16mvanbaak:)
11:45.31mvanbaakif only all problems would be that easy to fix
11:45.41BarthezZWARNING[3548]: chan_sip.c:13003 handle_response: Remote host can't match request NOTIFY to call <-- isn't rining any bell inside of me ;p
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11:51.12BarthezZhmmfg, seems to be coming from one of my dect handsets
11:52.02BarthezZcorrection :x all my dect handsets
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12:40.04knarfly8-)
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14:26.49fordfroghi, anybody knows why I get this error when starting asterisk? /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: SQLFetch
14:27.06fordfrogI tried with asterisk 1.2.17 and 1.2.27 and it behaves the same
14:28.01synchrisconfigure options?
14:29.28fordfrogwhat exactly? I am not asterisk expert so not sure what you exactly mean :-) anyway I can connect to the database using isql from unixodbc
14:30.13synchrisu used any package? distro specific?
14:30.19fordfroggentoo
14:30.25fordfrogunixODBC 2.2.12
14:31.24synchrisuse flagas?
14:31.27fordfrogI recall I had the problem last time I installed asterisk too but do not recall what was the problem
14:31.28synchrisflags*
14:31.31fordfrogmmt
14:31.55fordfrognet-misc/asterisk-1.2.27  USE="hardened mmx odbc postgres ssl -alsa -bri -curl -debug -doc -gtk -h323 -lowmem -nosamples -osp -pri -speex -sqlite -zaptel"
14:32.18synchristhis is probably
14:32.25synchrissomething missing from odbc
14:33.29synchrisfordfrog, try revdep-rebuild and some more extensive option
14:34.27fordfrogsynchris: the backend database is postgresql, could that be related to the pg driver?
14:34.31synchrisfordfrog, plus check with ldd
14:34.48synchrisi think is odbc problem
14:35.19synchrisas the odbc ask for this undefined symbol
14:35.47synchrisfordfrog, be sure check with ldd too
14:35.56synchrisldd the library which problem exist
14:36.54fordfrogsynchris: just `ldd lib`?
14:37.38fordfrogsynchris: revdep-rebuild rebuilt nothing
14:39.22fordfrogsynchris: this is what I got from ldd: http://pastebin.osuosl.org/22013
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14:49.16jayteemornin [TK]D-Fender
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15:07.06BarthezZHey, I'm trying to recieve a fax over an zap channel (junghanns quad bri card) using rxfax... After some time my log looks like http://paste.barthezz.name/?show=318 and obviously my fax isn't received :x
15:07.52BarthezZwhen i debugged an open zap channel trying to receive a fax it only displayed Null Frame(5) Subclass: n/a alot
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15:25.15jeev:<
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15:48.13justinnnnnnhey ppls
15:48.35justinnnnnnif i have 2 asterisk servers.. whats the best way to share the filesystem for redundancy ?
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15:51.37Alton2Not sure what you're getting at there.....
15:51.50Assidanyone suggest a good termination provider? around 1.2-1.5c/min?
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15:57.55Assidi mean i do see broadvoice and others which have unlimited.. but i prefer going a-la-carte
16:02.55justinnnnnnalton2 me?
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16:18.54Alton2justinnnnn yes
16:19.40fordfrogsynchris: seems noload res_config_odbc.so solved the problem
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16:47.37jeevwarez russellb
16:48.36outtolunctrying to make russell_jr .. who knows
16:49.40jeevouttolunc, how can i take a variable that the headers pass to me and then set another variable with the results?
16:49.48*** join/#asterisk tobias (n=tobias@cpe-076-182-118-165.nc.res.rr.com)
16:50.38lmadsenSet(HEADER_VAR=${SIP_HEADER(<name_of_header>)})
16:50.44kdasI am linking ipkall to a voipbuster account and then ii register on that account with my * box and i have my *box act as a voicemail server, but when i call the server ethier with a analog phone or softphone my *box dosent reconize my input. If some one can help it would be much apreshated.
16:51.09jeevjhmm
16:51.58jeevleif, can i msg you the example? it's one line
16:52.24lmadsenif its one line, I'd rather you just do it here
16:52.30lmadsenI'm doing a system upgrade, so I might not be able to help
16:52.36lmadsenI'm barely look at this window :)
16:52.43lmadsens/look/looking
16:52.57jeev2,Set(HEADER_VAR=${SIP_HEADER(${CALLERID(num)})
16:53.00jeevi dont know which to replace
16:53.16lmadsenuhhhh... CALLERID(num) is not a header
16:53.22lmadsenthat is another function
16:53.23outtoluncsip header is a function
16:53.48lmadsendon't use CALLERID(num) -- that is already giving you a value
16:54.04jeevi have SipAddHeader(P-Asserted-Identity:<sip:${CALLERID(num)}@hostname>)
16:54.05jeevwhich works
16:54.23jeevi'm then wanting to insert the caller id unmber and send it to the guys cell phone
16:54.27lmadsenSet(HEADER_VALUE=${SIP_HEADER(P-Asserted-Identity)})
16:55.18jeevok
16:56.37jeevtghanks, now i have to fix the incoming call acceptance, jesus
16:56.47lmadsenallah
16:56.54lmadsenbuddah
16:57.01lmadsenleif
16:57.06lmadsenall the same shit
16:57.07lmadsen:)
16:57.15jeevhahah
16:57.16jeevbastard
16:57.21jeevleif ERICCSON for life
16:57.21jeev!
16:57.22jeevericson
16:57.36jeevi can't seem to remember why it does this thing
16:57.50QwellLeif must be what the L. stood for in LRH
16:57.53jeevit's ilke the call from the 'user' to the extension '1DID' rejected because extension not found
16:58.01jeevi always fix it
16:58.04jeevthen forget next time
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16:58.20lmadsenQwell: LRH?
16:58.24Qwellnothing
16:58.26lmadsenQwell: I am the way, the truth, and the Leif!
16:58.31Cresl1nPsssh
16:58.37outtolunclooks like someone has a DID var that isn't wrapped in ${}
16:58.41lmadsenCresl1n: oh no you di'ant
16:58.42Cresl1nlmadsen, you sure think a lot of yourself
16:58.43Cresl1n:-)
16:58.51lmadsenCresl1n: lol
16:58.58Cresl1nLong time no see
16:59.01Cresl1nhow's CA doing?
16:59.02lmadsenCresl1n: I think more of you than me
16:59.05jeev[Sep 6 09:57:10] NOTICE[22791]: chan_sip.c:14035 handle_request_invite: Call from '9xxx' to extension '1xxxxxxxxxx' rejected because extension not found.
16:59.06Cresl1nheh
16:59.07jeevi hate that crap
16:59.08lmadsenCresl1n: CA is beautiful today
16:59.14jeevlmadsen, wehre in cali
16:59.14Cresl1nGood to hear
16:59.17jeevi'll come and steal your next damn book
16:59.18Cresl1nwe're overcast today
16:59.18lmadsenCA == Canada
16:59.22jeevBOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO
16:59.27lmadsenCresl1n: slightly overcast, but still sunny
16:59.36jeevthat just made leif ericson not proud
16:59.46lmadsenCanada > California
17:00.09outtoluncuse CAN for can, us california guys like CA
17:00.10Qwelllmadsen: You are mistaken.
17:00.10Cresl1nif you're talking about land masss
17:00.10lmadsenQwell: I've been to Cali :)
17:00.13Cresl1nGDP might not be so
17:00.26lmadsenheh
17:00.32Cresl1nalthough, that might be incorrect
17:00.38lmadsenwikipedia!
17:00.41Cresl1n<-------- likes people from canada
17:00.49lmadsen<------- likes people from HSV
17:00.58Cresl1nhugs lmadsen
17:01.10lmadsenw00t
17:01.17lmadsenCresl1n: gonna be at Astricon?
17:01.22Cresl1nYessir
17:01.25lmadsenw00t
17:01.25Cresl1ngot 3 talks to do
17:01.26jeevi'm gonna go to astricon and beat every canadian
17:01.29lmadsendamn G :)
17:01.40jeevlmadsen, you, twice.
17:01.40lmadsenjeev: uh huh... I doubt that
17:01.50lmadsenwonders how big jeev is... :)
17:02.07jeev6'1 194 now
17:02.18outtoluncjust put a dialplan in front of him, that'll slow him down
17:02.24jeevlol
17:02.27lmadsenkick to the nuts will bring you down to 5'3"
17:02.35lmadsenouttolunc: lol
17:02.36jeevlol
17:02.45lmadsengoes back to his system upgrade
17:02.49*** join/#asterisk Nicolas\ (n=nicolas@91.178.231.211)
17:02.50jeevok
17:02.52jeevNo
17:02.53jeevleif!
17:02.54jeevSet(HEADER_VALUE=${SIP_HEADER(P-Asserted-Identity)})
17:02.55jeevlast thing
17:03.01jeevhow can i set caller id with that value now
17:03.07jeevscreams adrian.... er leif
17:03.35*** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org)
17:03.42jeevguessing: Set(CALLERID(number)=$SIP_HEADER) ?
17:04.28jeevalmost.
17:04.54outtolunc{almost}
17:05.13jeevhahah
17:05.26jeevok, i got it working kind of... except it adds crap to it
17:05.29jeevwhich i've got to remove
17:05.50lmadsenjeev: before you ask... think for just a few seconds, and you'll probably figure most of your questions out
17:06.55jeevyea, i know bro.
17:07.15lmadsenbo knows
17:07.16jeevCALLERID(number)=<sip:1xxxxxxxxx@xxxxxxxxxxxxx>") in new stack
17:07.21jeevi've just got to fix the thingy
17:07.23jeevand it'll work
17:07.33lmadsenCUT() is a wonderful thing
17:07.47jeevyea, i know
17:07.56jeevi mean, i helped youw rite the book in the time of your need
17:07.57lmadsenactually... I'd just use the variable string control
17:07.57jeevyou help me with this
17:07.59jeevit's mutual
17:08.20jeevSet(HEADER_VALUE=${SIP_HEADER(P-Asserted-Identity)})
17:08.22jeevwghen i do that..
17:08.36jeevi dunt understand variable string control
17:08.40jeevoh well, i've got to step out
17:08.41lmadsenSet(CALLERID(num)=${HEADER_VALUE:5:10})
17:08.44jeevi dunno what to do but i'm sure i'll get it
17:08.45jeevahh
17:08.47jeevis that not cut ?
17:08.51lmadsennope
17:09.00lmadsenCUT() is unecessary there probably)
17:09.05jeevahh
17:09.19lmadsen${VAR:offset:length}
17:09.22jeevnice
17:09.23jeevworks
17:09.31jeevthanks so much leif
17:09.34lmadsennp
17:09.38jeevi wont beat you up
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17:09.39jeevat astricon
17:09.47lmadsenok... I appreciate that :)
17:09.50jeeveveryone else is going to hell!
17:09.53jeevhow big are you ?
17:10.03lmadsen5'9" 170 lbs
17:10.04jeevahh
17:10.16jeevok dood, i've gotta go pick up my friend and bring him home
17:10.16lmadsenI'm farm boy strong :)
17:10.18jeevhaha
17:10.21jeevwith a farmers tan
17:10.46outtoluncus farm kids got to stick together
17:11.23outtolunchas a perm faint farmers tan
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17:12.32lmadsenmy black belt would help too :)
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17:15.02beastyhi there
17:15.20beasty[Sep  6 19:17:13] WARNING[21864]: pbx.c:2968 ast_register_application: Already have an application 'Directory'
17:15.27beastycan anyone tell me what that mean ?
17:15.28beastys
17:16.36tzafrir_laptopbeasty, you're trying to register Directory from both the module app_directory.so and app_directory_odbc.so ?
17:16.37outtoluncwaits for the next book 'Asterisk-Foo: the Black Arts'
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17:17.16tzafrir_laptopHappen to use the Debian package? If so, see the included modules.conf that by default disables the odbc variant
17:17.33F00JINhi!
17:18.16Qwelltzafrir_laptop: freepbx
17:18.34tzafrir_laptopwhat about it?
17:18.41Qwellit replaces modules.conf
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17:28.34jayteelmadsen, did jeev really help you write the book?
17:30.04lmadsenjaytee: ummm... not that I know of :)
17:30.22jayteeanother one of his grandios claims then.
17:30.37lmadsenI was drunk most of the time, so who knows, heh
17:30.43lmadsenis being sarcastic there
17:31.22lmadsenouttolunc: when I meant black belt... I meant the black leather belt I typically wear
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17:32.09jaytee"Must stay focus,  madsen-san!"
17:32.50lmadsentotally... this system upgrade is... not the most fun :)
17:33.02lmadsenbut now I get to install my clustering stuff into another production server, so that kinda is exciting
17:33.16jayteelmadsen, system upgrade? hardware or OS version?
17:33.39lmadsendialplan
17:33.51jayteeuh-huh
17:33.52lmadsenmulti-site asterisk cluster
17:35.59jayteein a couple months I'm going be setting up a second * server as a failover for redundancy. I'm thinking of just using rsync from the "hot" system to the "cold" system and having eth1 active on cold with eth0 inactive with the same IP. then I can just flip them after moving cables. Eventually I'll get a T1 failover box and fancy it up with HA and figure out how to run both hot with failover.
17:36.00kdasI am linking ipkall to a voipbuster account and then ii register on that account with my * box and i have my *box act as a voicemail server, but when i call the server ethier with a analog phone or softphone my *box dosent reconize my input. If some one can help it would be much apreshated.
17:36.19lmadsenjaytee: yep, that sounds like a good plan
17:36.34kdasjaytee, :)
17:36.36jayteekdas, appreciated not apreshated
17:37.00jayteeslaps kdas on the hand with his "teacher's ruler"
17:37.03kdasjaytee, thanks i knew that looked wrong, sorry i just woke up
17:37.11jayteemore coffee dude
17:37.16kdasouch i am sirry
17:37.26kdassorry*
17:37.32kdaslol see i told you!
17:37.41lmadsenok, off to grab some lunch!
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17:37.53jayteekdas, when we were going over your problem last night [TK]D-Fender asked for some stuff and you didnt' want to pastebin it.
17:38.51kdasjaytee, i don't mind pastebining it just i asked him if he is sure we wants like all the packet data etc. and then he didn't reply and you told me he was afk or something
17:39.38jayteequestion I have regarding your problem. when you call from the softphone is it on the same local net as the * server? and if so, what is it's configuration set for in regards to dtmfmode?
17:41.10kdasjaytee, well yesterday we established that voipbuster is dtmfmode=inband and then yes it is on same local network and i use the local ip 192.168.1.117 in this particuliar case
17:42.37F00JINI'd like to know if there's an easy way to upgrade asterisk from 1.4.2 to 1.6
17:42.48F00JINi'm using ubuntu 8.04
17:42.59jayteeyes, voipbuster is inband and I wasn't aware of that till [TK]D-Fender pointed that out but what about the softphone itself? calls from it don't go through voipbuster so I want to know what that softphone is set to for dtmfmode.
17:43.15jayteeF00JIN, not with packages
17:43.34jayteeyou're better off uninstalling and then compiling the source
17:44.00F00JIN:-\
17:44.12jaytee1.6 is still beta
17:44.35lmadsenactually release candidate
17:44.40F00JINin fact I want to use an mp3 on hold music
17:44.42lmadsenrc9 I believe
17:44.46jayteeand 1.4.2 is pretty old which means whoever packaged it for Hardy isn't keeping up to date.
17:44.54lmadsen1.4.2 is REALLY old
17:44.57lmadsenlike... almost 2 years
17:44.58jayteelmadsen, yup
17:45.08jayteeon both old and rc9
17:45.17*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
17:45.33F00JINand with asterisk-gui 2.0 a message notify me I need 1.6 to upload and use mp3
17:45.39lmadsenF00JIN: btw -- 1.4 --> 1.6 is a major upgrade
17:45.46jayteevery major
17:45.58lmadsenits like going from a 1.0 to a 2.0
17:46.13lmadsenor going from kernel 2.4 to 2.6
17:46.38kdasjaytee, i have x-lite softphone and dosent seem to give me the information or the option for switching the dtmf mode
17:46.44lmadsenthere is no *easy* path -- read UPGRADE.txt and CHANGES in the 1.6.0-rc9 source as well
17:47.06jeevfarmers
17:47.07jayteekdas, it would be under the phone's SIP settings
17:47.07jeevhee haw
17:47.27lmadsengoes for lunch for real this time
17:47.30jeevNO
17:47.33jeevyou can't go when i just came
17:47.34jeevbastage
17:48.04jayteekdas, you wouldn't happen to have a hard IP phone to test with would you? it might be nice to know if dtmf detection is working period on your * server.
17:48.54kdasjaytee, no and the still can't find no dtmf setting in x-lite
17:49.09kdasjaytee, let me do a google search to see if that soft phone support that
17:50.01kdasbut if i am using a analog phone to dial in wouldn't not matter about the dtmf modes?
17:50.09jayteeit's been awhile since I've used x-lite. When I do use a softphone I use Ekiga but that only allows RFC2833, it won't let you set it to anything else. probably the same in x-lite.
17:51.07jayteeif you're configuration is set to inband for your incoming calls from voipbuster then it should work calling from an analog phone.
17:51.26jayteewhat version of * are you running?
17:52.10kdasummmi think 1.4 that what you get when you do a apt-get install with ubuntu
17:52.26kdasand on my other test system i am running the windows version of it
17:52.38jayteeof ffs!
17:52.58kdaswhat ?
17:53.23jayteewindows? with cygwin? why not just go out and spend the money on bubble gum?
17:53.23kdaswould it solve the problem if i used something other then voipbuster ?
17:53.48kdashttp://www.asteriskwin32.com/
17:53.54kdasthe other system is ubuntu
17:54.06jayteekdas, I don't really know at this point. you can't even get dtmf detection working right with a softphone. I'd say you might have a bad * package.
17:54.08kdasi have 2 but both fail
17:54.35jayteekdas, I'm familiar with asteriskwin. I'd rather use two cans and some string
17:54.59kdasjaytee, what about this ? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPdtmfmode
17:55.24kdasjaytee, ok got the point but what about the ubuntu box ?
17:55.49*** join/#asterisk ManxPower (n=manxpowe@195.sub-75-201-155.myvzw.com)
17:56.14ManxPowerwaves to everyone
17:56.26*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:57.04kdaswaves to ManxPower
17:57.10jayteewith the 'buntu box I don't know. you've tried relaxdtmf=yes already and that didn't help (usually doesn't) but you haven't proven that dtmf detection works AT ALL so trying to point the finger at voipbuster at this stage isn't going to accomplish anything.
17:57.30jayteewaves back at ManxPower
17:57.44*** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk)
17:57.52kdasjaytee, right i understand, but if i switch up servers then maybe it will be easier ?
17:58.07jayteeswitch up servers?
17:58.15ManxPowerkdas: it is not a hardware problem.
17:58.15jayteeyou mean providers?
17:58.23jayteeit's a software problem
17:58.34ManxPowerjaytee: what is kdas's problem (I ask you because you would be better at explaining it)
17:58.39kdasjaytee, wait a second i see what you are saying sorry, because the softphone dosent even use voipbuster
17:59.04kdasjaytee, server, provider its all the same lol i am so sorry maybe i should go make my self some coffe or something
17:59.16jayteeManxPower, it's the same as what you pegged it for last night. dtmf problems with voicemail, a mismatch in settings or his * package for Ubuntu isn't working for dtmf detection at all.
17:59.19kdasManxPower, well :P i can explain just fine
17:59.22ManxPowerkdas: yes.  do that.  step away from it for a few mins.
17:59.29jayteehe can't even get a softphone on his local net to work with it.
17:59.59ManxPowerAh.  Well getting the local phone and the local server seeing the correct DTMF needs to be solved before ANYTHING else.
18:00.10*** join/#asterisk drcode (n=Gadi@85.65.12.156.dynamic.barak-online.net)
18:00.14drcodehi all
18:00.21jayteehe uses voipbuster as an incoming ITSP and that uses dtmfmode=inband but his softphone is x-lite.
18:00.37jayteeManxPower, exactly. I told him to test locally first
18:00.58drcodeAsterisk with x100p can answer fax and voice (with iaxmodem)?
18:01.07drcodeor send fax also
18:01.14ManxPowerkdas: if the softphone is on the local LAN (and will stay on the local lan), I would use ulaw as the codec and RFC2833 as the dtmf mode.  If you just can't get rfc2833 working between your phone and the server, then look at inband for phone/server link
18:01.20jayteeEkiga only does RFC2833. don't remember if x-lite offers any additional options. he doesn't have a hard phone to test with.
18:01.47ManxPowerEven with ulaw you can get audio blips occasionally so even with ulaw rfc2833 is best, but inband would work well enough for most people.
18:02.07kdasManxPower, i was thinking i should remove all my sip server settings and just test a loca softphone to server connection till we can get this thing right sounds good ?
18:02.27ManxPowerkdas: make sure you have a backup copy of sip.conf.
18:02.45ManxPoweronce you do that put the sip.conf on pastebin.ca masking ONLY the passwords.
18:03.05ManxPowerkdas: you have multiple problems.  It is good you are willing to solve them one at a time.
18:03.11jayteeand you'll still need settings for the softphone in sip.conf. I'm assuming you meant the section that refers to your voipbuster account settings in sip.conf. just comment them out with semicolons.
18:03.21kdasyea
18:03.32kdasone second please gentlemen
18:03.35ManxPowerjaytee: x-lite allows at least rfc2833 and inband I think.
18:04.19jayteeand if you really want to make progress, quit wasting time with asteriskwin and focus on * on linux. Most of us here swear by compiling over package installs.
18:04.40ManxPowerHuh?  He's trying to run Asterisk on Windows?
18:04.58kdasManxPower, no i have 2 servers a windows and ubuntu
18:05.00jayteeManxPower, I thought it did but I haven't messed with it in almost a year. I had a copy of the free version but can't find it anymore. just have eyebeam at work.
18:05.29lesouvagekdas: if you use xlite don't to disable the option "silence surpression". I'm nt sure under what name it is part of the configuration option but as I remember it is something like "don't send silence".
18:05.33ManxPowerAsterWin was an april fools joke that got out of hand.  It is and never was an actual port.
18:05.37*** part/#asterisk drcode (n=Gadi@85.65.12.156.dynamic.barak-online.net)
18:05.43lesouvagedon't = don't forget
18:06.04jayteelesouvage, good point
18:06.09ManxPower"transmit silence" I think was the term
18:06.16ManxPowerIt would not have an effect on DTMF, however.
18:06.52lesouvageMaxPower: that is through I guess but it results in a general sound quality problem.
18:06.54jayteeno, but it sure messes with voice recognition.
18:07.43kdasok here is how sip.conf looks now to edit my extensions
18:07.44kdashttp://pastebin.com/d3ca12548
18:08.25ManxPowerkdas: why are you using non-standard port?
18:09.35kdasManxPower, should i change from 5070 to 5060 ?
18:09.41ManxPowerkdas: http://pastebin.com/m2ba3a5d6
18:09.44ManxPowerleave it out.
18:10.10lesouvageI would start with allow=all and if that works narrow the codes available.
18:10.18ManxPowerlesouvage: then you would be wrong.
18:10.31ManxPowerallow=all can easily make things not work.
18:10.47lesouvageManxpower: why is that?
18:10.48ManxPowerIn fact it is the worst thing you can do during troubleshooting, next to formatting your system.
18:11.09kdasok here is extensions.conf http://pastebin.com/d7da4647b
18:11.13ManxPowerlesouvage: because Asterisk supports several codecs in passthru only modes and if your call uses one of those it will fail.
18:11.45kdasManxPower, LOL @ formatting system
18:11.50lesouvageManxPower: thanks for the expenation. I think I have been lucky in the past.
18:11.51ManxPowerkdas: that should work.  Now you need to go into your softphone and set it up for the port and the DTMF mode.
18:12.00ManxPowerlesouvage: you have.
18:12.04lesouvageand the codec
18:12.22kdasManxPower, ok well my softphone don't have a dtmf setting unless i havent disconvered the easter egg yet
18:12.36ManxPowerlesouvage: no, just leave all the codecs enabled one the softphone and let Asterisk's allow/disallow settings determine the codec.
18:12.38jayteedamn! I didn't even spot the non-standard port number last nite.
18:12.53kdasjaytee, you need coffee too!
18:12.53ManxPowerkdas: all softphones have dtmf modes that I know of.
18:13.02jayteesucks when you're old and tired
18:13.07ManxPowerwhich softphone are you using?
18:13.14kdasManxPower, i know i just don't know were to find setting or which dtmf mode it uses
18:13.23kdasManxPower, x-lite
18:13.41jayteex-lite for linux or windows?
18:13.49ManxPowerlesouvage: if you manually force your phone to a specific codec then allow=all does not matter.  I prefer to force the stuff on the asterisk side and leave as much of the phone config alone as I can.
18:14.43kdasManxPower, how am i to call my systems vmail if i don't have any outgoing rule in the extensions.conf ?
18:15.00jaytee?
18:15.12jayteefrom a softphone the call isn't outgoing
18:15.13kdashttp://pastebin.com/d7da4647b is my extensions.conf
18:15.32ManxPowerkdas: because you are not making an outgoing call
18:15.45ManxPowerkdas: however "s" is not what you want.
18:15.56ManxPowerI would make it exten => 1234 or something like that.
18:16.06kdasok i am a little confused i have x-lite up and it is registered to my * box now what ?
18:16.09ManxPower"s" is almost never used unless you have FXO signalled ports.
18:16.21ManxPowerkdas: looks like x-lite defaults to rfc2833.
18:16.21kdasok let me change
18:16.48ManxPowerkdas: http://pastebin.com/m1cd22534
18:16.56ManxPowerthen dial 66 from your x-lit
18:17.13ManxPowermake sure to restart Asterisk as we made some sip.conf changes that should require it.
18:17.40jayteethat looks like an 88 to me with these bifocals. maybe I need a new prescription?
18:17.54kdasit is 88 last time i checked
18:17.56ManxPowerjaytee: you do.
18:18.15ManxPowerhttp://pastebin.com/m1cd22534  <-- has 66
18:18.45jayteeI'm goin to Walmart next time. Lenscrafter raped me for 300 bucks for the last pair and their scratch resistant coating is worthless.
18:18.46kdasok when i dial 88 on my x-lite phone it gives me the "the person you calling is unavaliable" message
18:19.01ManxPowerkdas: correct.  it's 66 not 88
18:19.18ManxPowerkdas: and is that a text message on the screen or an audio message?
18:19.45ManxPowerdo you get anything on the CLI?
18:19.47kdasManxPower, sorry it is 66
18:19.59ManxPowerext message on the screen or an audio message?
18:20.04ManxPowertext message that is.
18:20.06*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
18:20.06jayteeswear to god it's 88 in the extensions.conf he pasted
18:20.08kdasok one sec
18:20.17ManxPowerjaytee: it is 66 in the extensions.conf *I* pasted.
18:20.21kdasYAY it works!!!!
18:20.41ManxPowerkdas: now try logging into voicemail.
18:20.50ManxPowersee if it recognizes your DTMF
18:20.54jayteeah, ok that one. yes. I was looking a the one he pasted with the s extension
18:20.56kdasit does
18:21.07jayteewell, that's progress.
18:21.28kdasyea i was confused because i didn't notice my self the differance
18:21.33ManxPowercreate a mailbox and password that is more than 2 digits.  Sometimes DTMF can work 4 out of 5 digits and stuff like that.
18:21.47ManxPowerJust to be sure and it takes only a few moments
18:22.19ManxPowerWe want to make 100% sure that we have good DTMF on the local side before we try the service provider side.
18:22.22kdaswell it is working now yesterday and just a lil before it was dead like a corpse
18:22.37kdasok so i am to make a password longer then 4 didgits ?
18:22.56kdasjust for a side note you kept the |s option in your pastebin ;)
18:23.15ManxPowerkdas: you can make the password as long as you want (subject to Asterisk's internal limits which I suspect is something abour a 64 digit password
18:23.34kdasok but what did you want me to test now ?
18:23.48ManxPowerkdas: |s doesn't really matter either way.  you should use , instead of | as pipe(|) is going away in 1.6
18:24.18*** join/#asterisk rconnect__ (n=rconnect@85.138.128.75)
18:24.23ManxPowerI guess now we should work on your provider.  Do you have a cell phone or analog telco phone to test incoming calls with?
18:24.46ManxPowerThe provider part is the hard part.
18:24.53ManxPowerwhat provider do you use?
18:25.21kdasok thanks for head ups with the |
18:25.43jayteehe uses voipbuster
18:25.51kdasi have normal tele phone
18:26.02kdasi am using voipbuster temp
18:26.03jayteewhich needs dtmfmode=inband
18:26.34kdasbut i am looking for a good sip provider ... if you know any ?
18:26.53kdasterrasip and voipbuster i am using now because they are free for a while...
18:27.10*** join/#asterisk trumee (n=trumee@cpc1-seve11-0-0-cust723.popl.cable.ntl.com)
18:27.23trumeeguys, i am trying to setup * using pbxinaflash. I have setup an IAX trunk. I can call out successfully, but  do not receive any incoming calls. any ideas?
18:27.30ManxPowerI don't use voip over internet (I work in the business industry and they want rock solid and reliable).
18:27.47jayteePRI is the only way to go :-)
18:27.48ManxPowerWhen I have to use an ITSP I usually recommend Vitelity
18:27.53trumeei am using freepbx btw.
18:28.03TJNII~freepbx
18:28.03jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
18:28.04jayteetrumee, wrong channel
18:28.05ManxPowerjaytee: I don't do installs unless they are PRI
18:28.12kdasManxPower, so what do you use asterisk for hooking up normal lines ?
18:28.16trumeejaytee:why is that?
18:28.18ManxPowerkdas: PRI
18:28.32kdasPRI = what ? forgive my ignorance
18:28.33jayteeManxPower, if I was doing installs I'd be take the same approach.
18:28.51ManxPowerPRI is a type of voice T-1 with advanced signalling on it.
18:28.55trumeejaytee:is there a way to debug iax2 incoming calls.
18:29.07jayteeITSPs are too unreliable for the most part for real world businesses
18:29.20ManxPowerjaytee: It's less the ITSPs and more the internet.
18:29.28jayteeManxPower, true
18:29.29trumeejaytee:i did do a iax2 debug on. But if call my DID it doest give any trace at all.
18:29.31TJNII~freepbx
18:29.32jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
18:29.36TJNIItrumee: ^^^^^^
18:29.42trumeei did do a iax2 debug on. But if call my DID it doest give any trace at all.
18:29.42ManxPowerheck even without voip my customers frequently have problems even connecting to some sites because some carrier is having a spat with another carrier.
18:30.18ManxPowertrumee: If #freepbx support is so terrible maybe you should consider a different product?
18:30.21kdasManxPower, well i was thinking if i shutoff my normal phone i can upgrade my internet speed to 16mb download and X upload should i try using that and something differnt then voip ?
18:30.42jayteeeven with SIP over TCP instead of RDP the internet isn't setup to prioritize voip traffic so using the PSTN network is the only way to guarantee a higher reliablity.
18:30.49kdasManxPower, I love your subtle little jokes like the formating and the another product LOL
18:30.57trumeeManxPower:freepbx irc channel is so dead. is there any other GUI which has better community support?
18:31.04jayteequick and funny, ain't he cool!
18:31.05ManxPowerkdas: voip is OK for home use.  You are not gong to be screaming at yourself because you screwed up and you are going to lose a 4 million dollar contract becaus the phones are down.
18:31.15ManxPowertrumee: none of us here use GUIs.
18:31.21kdaspoint
18:31.50jayteegui's are sticky and you end having to wash your hands and recode your dialplan constantly. yuch!
18:31.53ManxPowerMy significant other uses a SIPura ATA, a cordless phone, and VoIP to Vitelity.
18:32.05trumeeManxPower:ok.  what basic files should i replace to get the basic asterisk back. i have another system (gentoo) on which i have asterisk setup
18:32.15ManxPowerworks 9 out of 10 times he picks up the phone.
18:32.32ManxPowertrumee: you must replace everything in /etc/asterisk
18:32.36kdas* uses 5060udp correct, i am trying to setup my DID to contact my *box directly
18:33.11trumeeManxPower:ok, i guess i will gzip /etc/asterisk and copy over vanilla asterisk files
18:33.29ManxPowerkdas: correct.  You need to port forward 5060/UDP and 10000 - 10020 and make sure that range is reflected in /etc/rtp.conf
18:34.07ManxPowertrumee: but almost nobody here will want to help you.  The reason most people use a GUI is they don't know enough to not use a GUI.
18:34.29ManxPowerso if you want to take the time to learn some stuff then go read the Asterisk book
18:34.31ManxPower~book
18:34.32jbotAsterisk: 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
18:34.52ManxPowerkdas: all voip with Asterisk is UDP, BTW.
18:35.29kdasManxPower, ok so 10000 -10020 udp is for what ?
18:35.35lmadsenRTP
18:35.38lmadsen(media)
18:36.21lmadsenjbot: no, 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
18:36.22jbotlmadsen: okay
18:36.38lmadsenthat comma after the tfot link seemed to break for me
18:36.49kdasManxPower, to use ipkall i just need to put HOSTNAME:5060 no need 10000-10020 right ?
18:36.54ManxPowerkdas: Audio
18:37.19ManxPowerkdas: huh?  Don't get ahead of yourself.  we havea lot of work to do before we start making calls.
18:37.36jeevhip hip, HORRAYYYYYYYYYYYYY
18:37.39jeevManxPower, i'm gonna do GRE
18:37.41jeevin a smidge
18:37.44ManxPowerdo you have the ports forwarded in your firewall and have you set the /etc/asterisk/rtp.conf to match?
18:37.58jeevhttp://www.pointless.net/~jasper/consume/docs/my-docs/tunneling.html looks easier than jaytee's sister
18:37.59ManxPowerjeev: Awesome!  Let me know if it works well for you.
18:38.36kdasManxPower, well on loacal net all ports open so no problem i was just trying to deal with ipkall
18:38.57ManxPowerkdas: now we need to set the localnet= and externip= in sip.conf
18:39.02kdasManxPower, i was just going to test the analog phone
18:39.21ManxPowerkdas: we are not even registered to a provder yet, no incoming calls will work.
18:39.30ManxPowerkdas: what is your local lan network and netmask?
18:39.54ManxPowerlets get outgoing calls working first and get registration working to your provider.
18:40.01kdasManxPower, isn't if you have ipkall go directly to your asterisk box  you don't need a provider right ?
18:40.28ManxPoweripkall IS a provider
18:40.40kdasipkall = DID provider
18:40.43ManxPowerbut it's NOT GOING TO WORK until we have the NAT issues worked out.
18:41.00ManxPowerkdas: you did so well, don't screw up by getting ahead of your self.
18:41.00kdasok well lets do that then you know best :)
18:41.25ManxPower(1:39:29 PM) ManxPower: kdas: what is your local lan network and netmask?
18:41.33kdasok so i am on 192.168.1.X 255.255.255.X network
18:41.43ManxPowerand with is your external IP address
18:41.58ManxPowerso you would put localnet=192.168.1.0/255.255.255.0
18:42.09ManxPoweras the last line in [global] in sip.conf/
18:42.14kdas67.161.44.222 = externip
18:42.27kdasexternal*
18:42.33ManxPowerwe also have to set externip=67.161.44.222 in sip.conf right before or right after the localnet line
18:43.42kdasok
18:43.49ManxPowerkdas: looks like IPkall does not give you any setup instructions.
18:44.08kdasdoes asterisk respect the placement of the lines?
18:44.34ManxPowerkdas: sort of.
18:44.39kdaslol ok
18:44.52ManxPowerisipkall your only provider?
18:44.59ManxPowertheir setup seems like a nightmare
18:45.14drmessanoLOL
18:45.18kdasipkall is DID provider and as far as the sip goes i need to find a reliable good one
18:45.19drmessanoIPKALL is easy to set up
18:45.49ManxPowerstandby, I'm looking at some stuff.
18:45.51drmessanoYou tell it which extension to send the calls to, and which proxy..
18:46.05kdasbut as of now i am just using voipbuster and terrasip becasue it is free as of now
18:46.07drmessanoAll calls come from voiper.ipkall.com
18:46.21drmessanoJust need to accept the calls
18:46.43ManxPowerdrmessano: how do they know your IP address?
18:46.48drmessanoFrom the Web UI
18:47.00drmessano<drmessano> You tell it which extension to send the calls to, and which proxy..
18:47.15kdasManxPower, you set it on there site on voip-info it tells how to set it up
18:47.15ManxPowerkdas: ok, first you say you are only using ipkall, now you are saying you use voipbuster and terrasip.  Which is it?
18:47.24ManxPowerkdas: how 1980s of them
18:47.30kdasManxPower, lol
18:47.43ManxPowerkdas: put your current sip.conf on pastebin as well as your current extensions.conf.
18:47.52kdasManxPower, will you be here like in 10 mintues ?
18:47.59ManxPowerkdas: maybe.  why?
18:48.08drmessanoIPKALL works great for free
18:48.10kdasManxPower, i need to do something real fast
18:48.19drmessanoHe has to poop
18:48.21ManxPowerkdas: sure.  can you put up the pastebins first?
18:48.25kdasok
18:48.44kdasso far only sip.conf changed i will put it up
18:48.58ManxPowerOK.
18:49.08kdashttp://pastebin.com/m4acfb9a
18:49.18kdasok brb
18:55.44*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
18:55.52jeevManxPower, you suggest doing ipsec over gre for this?
18:56.09ManxPowerjeev: not at this time.  Maybe later.
18:56.42ManxPowerjeev: we are not doing the tunnel for security.  We are doing it to make your stuff 10x less complicated. 8-)
18:57.06ManxPowerjeev: ipsec would just add latency at this point.
18:57.24jeevhm
18:57.34jeevonly problem is that
18:57.40jeevthe internal asterisk is dynamic.
18:57.48jeevip
18:58.15drmessanoNO-IP
18:58.24ManxPowerjeev: that sucks.
18:58.30*** join/#asterisk shinao1 (n=shinao1@62.173.48.176)
18:58.50ManxPowergetting a static IP would make things easier as well.
18:59.14jeevhmm, or maybe this will help ease the dual wan situation
18:59.16jeevlet me see
19:00.10ManxPowerjeev: there are other ways to accomplish what you want without tunnels, but they are much more complicated on the Asterisk setup.
19:00.51kdasManxPower, ok back so what is your verdict on ipkall?
19:00.57ManxPowerhttp://pastebin.com/m707d7690  use this sip.conf
19:01.05kdasme?
19:01.13ManxPowergive me the url for the past extensions.conf so I can modify it.
19:01.15ManxPowerkdas: yes
19:01.33jeevyea, i know
19:01.59kdasok cool
19:02.40kdasyour sip phone number here = my DID number?
19:03.12ManxPowerkdas: I don't think so, the docs I found are vague.  It says "Where 508 is the SIP phone number you specified when setting up IPKall."
19:03.20ManxPower[508] is on the example.
19:03.49ManxPowerkdas: if you don't know what it is, we can find out when we debug a failed call
19:04.02kdasoh you must have read wrong docs one sec let me find the ones i read
19:04.14jeevk, pf is blocking the packets, let me add the rule and :D
19:04.54ManxPowerkdas: doesn't really matter, just use the sip.conf I gave you and get me a copy of your extensions.conf so I can add the required lines for that.
19:04.54hardwireAnybody ever used Alepo RBS?
19:05.00kdashttp://www.voip-info.org/wiki-IPKall
19:05.14kdasok if you read that it will tell you how to set it up
19:05.30ManxPowerkdas: the exact same document I was looking at.
19:05.36ManxPowerand you see the [508] in the example.
19:05.42ManxPowerand I quoted off that page as well.
19:05.42kdaslol your right
19:05.52kdasi was using the top half though
19:05.55ManxPowerso we are back exactly where we started, me waiting for your extensions.conf.
19:05.58kdaslet me re-read it real fast
19:06.17kdasmy extensions.conf is same as before rememeber ?
19:06.32ManxPowerkdas: yes, but I don't remember the 100 char URL for me to find it on pastebin again.
19:06.41*** join/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com)
19:06.50ManxPowerso either tell me the URL again or paste it again.
19:06.58ManxPowerI really don't care which.
19:07.41jeev1. 001027 rule 102/0(match): block in on gre0: 192.168.20.38 > 192.168.20.37: ICMP echo request, id 58689, seq 210, length 64
19:07.42jeevyayayay
19:08.08jeevyayyyyyyyyy
19:08.16jeev64 bytes from 192.168.20.37: icmp_seq=250 ttl=64 time=7.016 ms
19:08.17jeev:D
19:08.27jeevManxPower, i try the rest later.
19:08.32kdasManxPower, http://pastebin.com/m7152c559
19:08.32jeevthe reconfiguration
19:08.33jeev:(
19:08.41ManxPowerjeev: you are making major progress,.
19:08.44jeevhmm wtf
19:08.53jeevi am confused on my next step
19:08.57*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
19:08.58jeevi'm converting from iax to sip, that's it ?
19:09.00kdasManxPower, let me get the extension.conf for you
19:09.04rhombusI have a callcentric trunk that is working for inbound calls, but for outbound calls, I hear no sound.
19:09.21ManxPowerkdas: I'm not doing or saying anything more until I get extensions.conf
19:09.29rhombus... in either direction.
19:09.35jeevrun a sip debug
19:09.39jeevand look to see for errors
19:09.46kdashttp://pastebin.com/d2bfe5d9 = extensions.conf
19:11.15rhombusjeev: ok
19:12.22*** join/#asterisk ElSonico (n=tav@xdsl-179-75.nblnetworks.fi)
19:12.33ManxPowerhttp://pastebin.com/m394c3063  notice all the extra stuff, make sure all the . and , are correct when you copy it to your system
19:13.00ManxPowerkdas: then do a reload and try calling your number.  chances are it won't work, but we are doing things step by step.
19:13.21kdasManxPower, i love your way of thinking ;) step by step
19:13.21ManxPowerkdas: ignore the stuff about _. not being good when you do the reload.
19:13.31kdaslol ok
19:13.54ManxPower_. is almost never a good idea, except when it is a catch all and no other extensions exist in the context.
19:14.08TJNIIManxPower: [INVALID} <- Typo?
19:14.22lmadsenyes
19:14.23ManxPowerTJNII: good catch!  yes it should be [INVALID]
19:14.31kdasok
19:15.05ManxPowerkdas: if you get any CLI output when you try calling the DID, pastebin it.
19:15.35kdasok sec
19:15.45*** join/#asterisk kj4acm (n=kj4acm@24-183-225-98.dhcp.kgpt.tn.charter.com)
19:15.55kdasmanx it works!!!
19:16.06kj4acmmy googlefoo isn't very strong here... is there a way to subscribe 1 extension to multiple mailboxes ?
19:16.09kdasManxPower*
19:16.11ManxPowerkdas: nifty!
19:16.33ManxPowerpastebin the cli output so I can see if I can find any potential future problems.
19:16.52ManxPowerkdas: I've been using Asterisk heavily since late 2001.  I've done this a few times before. 8-)
19:17.11kdasbows to ManxPower the * guru
19:17.12kdaslol
19:17.20kdasu sure know your shit
19:18.15ManxPowerkdas: Asterisk pretty much accepts any incoming connections, that's why we use INVALID in [general] to make sure those calls NEVER get into your real dialplan.
19:18.48kdasManxPower, yea i understood that i just never thought of it, its a nice security touch ;)
19:18.50ManxPowerkdas: [TK]D-Fender is also a guru, but he sometimes likes sip debug information a little too much.
19:19.03ManxPowerIf it had not worked, we would have used sip debug to diagnose it further.
19:19.39kdasManxPower, umm the CLI output is prettly clear it just has the verbose msg that we have about the incoming call and then it gives us a waring on using "_."
19:19.40jplankwhich side of the call does pickup() pick up?
19:19.53ManxPowerjplank: the asterisk side.
19:20.14ManxPowerof whatever the first call that is ringing in the ring/call group
19:20.31jplankI can't use it to pick up a specific extension?
19:20.49*** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net)
19:21.04ManxPowerjplank: not with Pickup().  If your versions as DirectedPickup() use that.  I'll double check for you
19:21.33ManxPowerkdas: the verbose message is something like Noop(Call to XXX)
19:22.10kdasexten => 66,1,Verbose(1|--Random Incoming Call--${CALLERIDNUM})
19:22.35jplankI thought pickup() was a directed pickup?
19:22.35ManxPowerjplank: I was wrong.  Pickup is directed pickup.  I was thinking of park/pickup.
19:22.57ManxPowerkdas: nifty.
19:23.02jplankI can grab a call from a pickup group
19:23.14ManxPowerkdas: now MAKE A BACKUP COPY of the files.
19:23.14jplankI just can't pick up a specific extension
19:23.17kdasManxPower, ;) ok so what sip/voip provider do you suggest for me ?
19:23.30ManxPowerjplank: remember extension@context
19:24.01kdasManxPower, backup complete
19:24.24jplankI get a unknown, no matter what context I use
19:24.25*** join/#asterisk UD (i=OHNOUDID@unaffiliated/underdawg)
19:24.30ManxPowerkdas: I really don't have a suggestion other than Vitelity, but there are many providers out there.  Vitelity's web account management and DID management is a little wonky at first, but you get used to it.
19:24.41ManxPowerjplank: weird.
19:24.55kdasManxPower, does it support CID spoofing ?
19:25.09ManxPowerAnd they were VERY helpful when I totally screwed up my account settings.
19:25.09jplankif I don't put a context, doesn't it use the context that I'm calling from?
19:25.14ManxPowerkdas: I have no idea.
19:25.30ManxPowerjplank: always specify the context 8-)
19:25.38kdasManxPower, i read alot about CID/ANI spoofing and i have a quick qestion, a lil off topic but you seem to have the knolege..
19:25.51kdasknowledge*
19:26.14jplankthe context the ringing extension is in, or the one where the call came from? (I assume the ringing extension)
19:26.21ManxPowerjplank: once you get it working, try it without the context.  The "calling context" can be pretty complicated when you have includes, etc.
19:26.35ManxPowerno, pickup does not know what the ringing extension is
19:26.40ManxPoweruntil you tell it.
19:26.44kdasManxPower, when a cell provider offers free incoming and out going to mobiles on the same network is that based on ANI or CID ?
19:26.51ManxPowerkdas: no.
19:26.57jplankthis is whats confusing me
19:27.04jplankwhen I do extension@context
19:27.16jplankdo I do the ringing extension @ the calling from context?
19:27.19kdasManxPower, huh ?
19:27.23ManxPowergive us the ACTUALL stuff
19:27.40jplankor the ringing extension @ the ringing extensions context
19:28.01ManxPowerlets say a call comes in and is sent to exten 666 in context [hell]  If you want to pick up that extension using pickup you would use Pickup(666@hell)
19:28.30Strom_M666 hell? sounds familiar
19:28.34ManxPowerkdas: no, the carriers do not use CLID/ANI to determine if the call is on the same network
19:28.34jplankI'll try it again, but I tried that and kept getting a unknown
19:28.49ManxPowerjplank: I've not used pickup much.
19:29.06ManxPowerStrom_M: yes, I just copied my customer's info. 8-)
19:29.12jplankso something like exten => _**.,1,Pickup(${EXTEN:2}@from-test) *should* work
19:29.13Strom_Mheheh
19:29.24kdasManxPower, what do they use ?
19:29.34jplankassuming ${EXTEN} is in from-test
19:29.51ManxPowerkdas: the tower, the ESSN, the PRL, any number of things cell networks have that users never see.
19:30.04ManxPowerjplank: that's what the docs say.
19:30.34jplankyea thats what I read
19:30.38jplankalso doesn't work :(
19:30.41ManxPoweris the ACTUAL exten => line in [from-test] or is there an include => realcontext in the [from-test] context?
19:30.47kdasManxPower, i see so no way in tricking the bill system... smart
19:31.41jplankthe actual exten => line is in from-test
19:31.50ManxPowerkdas: a few carriers use CLID/ANI to bypass your voicemailbox auth requirement (the idea is that if you are calling voicemail from your cell, you must want to log in and a password would be just so much of a bother for the customer).  Not many these days do it that way.
19:31.58ManxPowerjplank: I'm stumped.
19:32.07jplankas am I
19:32.21jplankthe last thing I need to do to get BLF working perfectly :(
19:32.29kdasManxPower, ok i understand thanks
19:33.35kdasManxPower, i can't seem to find the rates for USA and stuff on the Vitelity website
19:33.36ManxPower~manxpower
19:33.36jbotsomebody said manxpower was NOT an employee of Digium.  He is looking for a training/teaching job in networking and/or Asterisk.  Currently doing Asterisk and WAN consulting.  Contact: eric@fnords.org
19:34.22ManxPowerkdas: all of use is one rate.
19:34.28ManxPower..er.. all of the USA are one rate.
19:34.41ManxPower1.9 cents, I think
19:35.00ManxPowerhttp://vitelity.com/index.php?p=retailserv
19:35.19ManxPower1.44 cents/min actually.
19:35.37kdasManxPower, but what about services like skype or voipbuster witch is free as long as you have 5$ on account ?
19:35.49ManxPowerkdas: what about them?
19:36.17ManxPowerSkype is a closed protocol closed source, proprietary protocol.  You get what you pay for.
19:36.19jeevflowroute.com is cheap
19:36.26kdasManxPower, lol just asking why pay 1.44 rather then free ?
19:36.38ManxPoweryou get what you pay for.
19:36.51kdasManxPower, point
19:37.23ManxPowermy significant other's phone bill is like $5-$10/month and that includes a DID, incoming/outgoing, e911, CNAM, and several other features.
19:37.46kdasManxPower, so my next step is in getting a Voip provider and using it for my outbound calls and then fixing up my dialplan so people don't just get my vmail each time correct?
19:38.05ManxPowerkdas: yup.  Go read The Book.
19:38.07ManxPower~book
19:38.08jbotwell, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
19:38.21*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)
19:38.54kdasManxPower, just the DID is 7.XX$ and everything else might be extra 5$ so i am guessing a 14-20$ bill right ?
19:38.56j0on samgona analog cards with an additional remora board, should the red led's turn on for the additional boards right away? i've hooked power up to the main board too
19:40.06kdasManxPower, i would to thank you sooooo much for your help you kick ass i been wondering from channel to channel and person to person and they all could not help me and you just straight up kicked ass with out making me reporgram the source to asterisk so 2 thumbs up for you
19:40.20Strom_Mkdas: send him some cash
19:40.22jplankI can't think of a logical place to set pickupmark to use with pickup(), seems like it would be a pain to use with DID routing, would have to add it to ever DID translation
19:40.32ManxPowerkdas: huh?  dids are lile $1.99/month
19:40.38kdasStrom_M, how much and how ?
19:40.49Strom_Mask him
19:40.51ManxPowerpaypal donations are welcome to eric@fnords.org
19:41.06*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
19:41.07kdaserrrg need to setup paypal account
19:41.40ManxPowerkdas: the $7.xx/month is for unlimited inbound.
19:41.49kdasManxPower, yea
19:42.08kdasManxPower, so what she gets the 1.2cents a minute incoming ?
19:42.37jplankresidential DID's charge inbound?
19:42.51ManxPowerso to break even you would have to have 64 incoming mins per month to break even on the $7.95/month
19:43.05ManxPowerjplank: yes, in the voip world.
19:43.12jplankhmmmm
19:43.17jplankthats kind of messed up
19:43.54jplankmost telco's make money off the inbound call from the sending carrier
19:43.58kdasManxPower, that is 1hr 4min for 7$
19:43.59jplankand they are charging the end user
19:44.03jplankscrewed up
19:44.13kdasManxPower, i talk more then that every month
19:44.18ManxPoweroh wait!  you would need 662 mins per month to break even on the $7.95 plan
19:44.30kdasManxPower, that is better
19:44.39ManxPoweryou can calc it if you want.  7.95 / 0.012
19:44.57ManxPowerMy first number did 12 cents/min, not 1.2 cents/min
19:45.23jplank0.12 and 0.012 biiiig difference :)
19:45.47ManxPowerjplank: yeah, I should know better.
19:45.54jplankeveryone does it
19:45.59jplankI find myself doing it all the time
19:46.16ManxPowerjplank: in the telcom world a simple mistake can cost you thousands of dollars
19:46.29jplankforget about it
19:46.44ManxPower8-)
19:46.50jplankVZ missed rated 200 of our calls by one decimal place
19:47.07jplankturned 400000 minutes into 1.1 million
19:47.11ManxPowerjplank: how much extra did they try to bill you for?
19:47.16jplanka lot
19:47.36jplankturned calls that lasted minutes into hours
19:48.11jplankwe only noticed it because when we did our bill run our minutes jumped almost 300% in one month
19:48.16ManxPowerjplank: you have to wonder how many people had the same thing happen to them and did not catch it.  Did you at least report the issue to the PUC/PSC?
19:48.54jplankit was a simple mistake I guess, my company is a CLEC and VZ is our interconnect for local calls
19:49.01jplankso we just had them fix it
19:49.14jplankI could only imagine how many people get screwed by it
19:49.22jplankwe only noticed because we have a high volume of calls
19:49.30jplanksomeone with a small volume would never notice
19:49.33vipcarrierwhat do u use for CDR mediation
19:49.53jplanktimes that by 1000's of end users and VZ makes out with $$$
19:50.04jplankwell we use a billing company calls profitec
19:50.14ManxPowerjplank: and VZ should be nailed to the wall by the PSC/PUC
19:50.27Strom_M"times that by"?  what the fuck ever happened to the word "multiply"
19:50.52vipcarrierwhat do u use for CDR mediation
19:51.03rhombusStrom_M: "
19:51.10jplankI can't say if it happened to everyone, I just know it happened to us, actually twice, first time it was on one call, that happened to be international, which was easy to spot, and the next time it happened on a ton of calls
19:51.27jplankvipcarrier: if your talking to me, like I said, we use a company called profitec
19:51.28rhombusStrom_M: "times" is shorter than "multiply" -- easier to type, too.
19:51.32kdasManxPower, do you use any Instant messengers ? yahoo google etc?
19:51.43Strom_Mrhombus: and dumber-sounding, to boot!
19:51.45drmessanoStrom_M: "times that" is from 4th grade
19:51.46ManxPowerkdas: a few, occasionally
19:51.54jplanklol
19:51.54rhombusStrom_M: but gets the point across :)
19:52.02drmessanoStrom_M: Where most people seem to stop learning nowadays
19:52.09kdasManxPower, so it would be better to catch you here ?
19:52.10vipcarrierhow does it work?
19:52.12Strom_MI didn't know stupidity was in such vogue these days
19:52.27jplankrhombus: what did I just say about drmessano :)
19:52.29drmessanoStrom_M: You think that's bad, times that by 6 billion people
19:52.37jplankROFL
19:52.37Strom_Moh em gee
19:52.43rhombusjplank: yes
19:52.55vipcarrierslaps jplank around a bit with a large trout
19:53.01drmessanozee oh em gee
19:53.14drmessanoSorry Canada
19:53.18drmessanoZED oh em gee
19:53.22kdasManxPower, i want to send some money i just got to wait for my friend who is in the bathroom to use his paypal don't expect alot but it is something if i had more to spare i would give more ;)
19:53.39kdasso i will come back in a few to get your details
19:55.14jplanklol
19:57.20*** join/#asterisk kdas (n=ME@c-67-161-44-222.hsd1.ca.comcast.net)
19:57.39kdasok ManxPower what info do i need to send you money to your paypal ?
19:58.15jplankyou can send money to my paypal
19:58.27*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
19:58.31drmessanoSend $10 to all of our paypal's
19:58.37drmessanotimes that by 241
19:58.42jplanklol
19:58.49drmessanoThat's almost $10000 or something
19:59.04Strom_Mno, just divide it by .005 and send the result directly to me
19:59.10jplankyou paypal me $10 a month and i'll set you up with a DID with free incoming :)
19:59.29drmessanoStrom_M: What is that times by 100?
19:59.58drmessanoA nice greasy pork sandwich, served in a dirty ashtray.. that's what..
20:00.38*** join/#asterisk wiscados (n=mint@81.25.184.155)
20:00.42jplankerrr I can't get pickup to work, this is really getting to me, what use is BLF if you can't pick up the call
20:01.40jeevsets up manxpower@hisdomain.com and on paypal as an alias and msgs to kdas.
20:02.14ManxPowerjeev: I've received less than $300 in paypal donations since 2002
20:02.27ManxPowerI'm not in this channel for the money 8-)
20:02.38drmessanoI'm here for the Ho's
20:02.39jeevheh
20:03.02drmessanoStill waiting for the women to arrive :(
20:03.09drmessanoSausage fest ------------------------------------------------------------------>
20:03.18drmessanoAll 241 of them
20:03.43SpaethCohopefully your MoH is working so you won't sit in silence while you're waiting for them to show up
20:04.12drmessanoI got that working first.. I knew it was gonna be a while
20:04.24drmessanoMark Spencer promised me free telephony and Ho's
20:04.34drmessanoI'm beginning to dislike him :(
20:04.43jayteehe promised me a Winnebago
20:04.51rhombusdrmessano: only him? I thought you disliked everybody.
20:05.03jplanklol
20:06.05ManxPowerdrmessano: I don't think you'd like the hos mark would send your way/
20:06.26drmessanorhombus: No.. I dislike a lot of people, bit everyone.. So if you think I dislike you, and are wondering if it's just me disliking everyone or if it's personal, it is indeed personal.  Sorry.
20:06.50drmessanoerrr
20:07.08drmessanoI also dislike thinking 3 or 4 words ahead too
20:07.37jayteeI hate everyone equally without regard to race, creed, nationality or sexual orientation
20:07.59ManxPowerjaytee: that's what I do.
20:08.09jayteeit's all about the equality :-)
20:08.33jplankjaytee: does that include hot female blonds who are bi?
20:08.53jayteejplank, well......there are exceptions to every rule :-)
20:09.57ManxPowersends Major Iceborg to jaytee
20:10.07jayteeOMG!!!
20:10.07jplanklol, except the rule of exceptions I suppose
20:10.59jaytee"Leeloo Dallas Multipass"
20:11.04ManxPower8-)
20:11.17[TK]D-FenderManxPower: SIP debug does wonderful stuff like show that calls are even mathing te peers we think they should....
20:11.20jayteeI love that movie
20:11.33rhombusdrmessano: well, forgive me. It does indeed seem like you hate everybody. Personally.
20:11.34ManxPowerjaytee: 5th Element, Matrix, and Dogma are what I call "jesus movies"
20:11.39ManxPower[TK]D-Fender: I know.
20:11.55ManxPower[TK]D-Fender: but you love sip debug enough to be...creapy. *tease*
20:11.55jayteeManxPower, jesus movies?
20:12.11ManxPowerjaytee: main character has to save the world, but has to die first.
20:12.25jayteeI loved all three but the Matrix sequels sucked.
20:12.27[TK]D-FenderManxPower: and it usually eve saves you looking at useless dialplan contexts people think should work but are never being used
20:13.06jayteeonly part of Matrix 2 that I liked was Merovingian when he said, "I love ze french language, it's excellent to curse with. Like wiping your ass with silk."
20:13.40jplankfender: do you have much experience with pickup()?
20:14.02[TK]D-Fenderjplank: nope, but what's your actual question
20:14.03*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)
20:14.12jayteeManxPower, when I think of jesus movies in that context The Omega Man with Charlton Heston is the first thing that comes to mind.
20:14.28ManxPowerLeeloo dies right at the beginning, Neo has to die before he can control the matrix, in Dogma Bethany has to die when she saves the world.
20:14.30jplankno matter how I use it, I get a error inside the console
20:14.34jayteespeaking of which I guess now we can finally pry his gun from his cold, dead hands.
20:14.49jplankI'm pretty sure I'm using it correctly, but it doesn't work
20:14.58jplankpickupexten works though
20:15.27[TK]D-Fenderjplank: "pretty sure" doesn't mean much.  pastebin <-
20:15.32jplanki know
20:15.55jayteeDogma was excellent. I actually own a Buddy Christ dashboard figure.
20:16.09jayteeand now they have Buddy Christ bobbleheads.
20:16.50ManxPowerI'm a devout atheist, but I'd consider a Buddy Jesus on my dashboard
20:17.03jayteethen go to www.viewaskew.com
20:17.08jplankhttp://pastebin.com/m77904f3e
20:17.09[TK]D-Fender%$##ing sliced my thumb at martial arts practice today.  Just messed my whole week up...
20:17.13[TK]D-Fenderdammit
20:17.21jplankbot extensions are inside the from-test context
20:17.23jayteeouch!
20:17.24jplankboth*
20:17.57UDgcc is god
20:18.00[TK]D-Fenderjay took most of my thumb-print though 3 layers but left a flap so it should reseal....
20:18.37*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
20:18.38jaytee[TK]D-Fender, yeah but it's like a paper cut but worse and they take longer to heal than a scrape or gouge
20:19.15jayteeis it on your writing hand?
20:19.19[TK]D-Fenderjay it was real clean fortunately.... Mr. Pointy was very good at its job...
20:19.19jplankfender: did you catch the pastebin?
20:19.28[TK]D-Fenderjaytee: left fortunately
20:19.40UDhey ManxPower I can't seem to get a 2 person voip conversation going
20:19.43jayteethat's not as bad but still
20:19.44UDon this connection
20:19.57UDdoes your wireless connection suck as bad as mine?
20:19.59[TK]D-Fenderjplank: it doesn'rt match
20:20.19jplankwhat do you mean?
20:20.48[TK]D-Fenderjplank: exten => _**.,1,Pickup(${EXTEN:2}@from-test)   and   -- Executing [**2003@from-test:1] Pickup("SIP/2002-08e02888", "2003") in new stack  <-- no context in execution
20:20.50UDnot that i was actually planning on using this connection for real world asterisk use
20:20.58jayteeI've finally rewritten my code for the speech recognition IVR so I'm not using the n+101 deprecated crap anymore. Yay!!!!
20:21.19[TK]D-Fenderjaytee: whats the vr component?
20:21.19jplankerr
20:21.20jplanksorry
20:21.28jaytee[TK]D-Fender, Lumenvox
20:21.29jplankthat wasn't the right error
20:21.42jplankthat was the error from when I tried doing the pickup without a context
20:21.45jblackHeh. Looks like New Orleans is gonna have to evacuate again.
20:21.49jplankhold on, I'll update the pastebin
20:21.59ManxPowerjblack: maybe, still too far in the future to know.
20:22.09[TK]D-Fenderjaytee: how accurate is it performing?
20:22.11jayteeaw, man! those poor people in LA just can't catch a break
20:22.19j0the additional remora board on my sangoma a200 doesn't power up (yes power is connected). any ideas?
20:22.39*** join/#asterisk Levonk (n=lk@adsl-75-62-133-138.dsl.lsan03.sbcglobal.net)
20:22.49*** part/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com)
20:23.20jaytee[TK]D-Fender, actually very well so far. I need to test with people who have odd accents but for the simple grammars I'm using the recognition is usually 80% or better so it matches properly.
20:23.46jplankfender, I updated the pastebin http://pastebin.com/m7fc41949
20:24.08jayteeI wished I could have imported the grammars from my Locus Liaison system but those are compiled grammar files and Lumenvox only seems to support raw text grammar files.
20:24.44jblackLike I said, "It looks like", not will. N.O. is dead center for the predicted path for a H3 on Friday.
20:25.23[TK]D-Fenderjplank: please include more cli incuding proof the exten you are trying to pickup is ringin
20:25.28jayteeso I've got the menu tree tested for both voice rec and dtmf and both options work for each keyword. Just need to record the custom prompts and I'm good to go.
20:25.33jplankok
20:26.38jplankfender: http://pastebin.com/m20b93f0b
20:27.17jaytee[TK]D-Fender, if you ever decide to mess with Lumenvox their starter kit for one port license is only 50 bucks but comes "without support" and even then their reps will answer questions anyways if you're evaluating it.
20:28.10jayteeand they've got alot of example configs and online training vids for it but I haven't really had to dig into the documentation to deep to make it work.
20:28.18jplankfender: the actual ringing is line 81
20:30.41[TK]D-Fenderjplank: "This application can pickup any ringing channel that is calling the specified extension"
20:31.02[TK]D-Fenderjplank: the extension being diales i "s"
20:31.10[TK]D-Fenderjplank: not gonna work
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20:31.18jplankahhh
20:31.47jplankso if I did a plain old dial it should work?
20:32.39[TK]D-Fenderjplank: if you used a valid target as per the apps instructopns, I'm sure it does its job
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20:38.46ManxPowerjplank: I think you are confused.  An extension is the first field after exten =>
20:39.13ManxPowerthe extension is not the sip account name.  If you set the sip userid and the extenson to be the same....well you see the confusion that results.
20:40.39jplankI get it, thank you
20:46.28jplankaastra 57i ct is a pretty sick phone
20:48.12[TK]D-Fenderjplank:  I hated mine.  made me yearn for my bedside Polycom IP 301
20:51.35jplankreally why?
20:52.03jplankI've been testing it all day, and everything seems to work perfectly (aside from direct call pickup from the phone)
20:52.47jplankso theres not way to grab a call from the device thats ringing directly?
20:52.51jplankno way*
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20:56.38[TK]D-Fenderjplank: sure there is.  "pickup", "pickupexten", etc
20:57.02[TK]D-Fenderjplank: unfortunately FreePBX owns your sorry ass.
20:57.12jplankdoesn't pickupexten pickup the ring group?
20:57.20jplankerrr your telling me
20:57.47*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
21:00.54jplankI guess I could use pickupmark, but that seems like a pain in the ass
21:01.45jplankhmm maybe not
21:04.10[TK]D-Fenderjplank: I'm saying that FreePBX'S SPAGHETTI
21:04.18jplankyea it is
21:04.22jplankI totally agree
21:04.48[TK]D-Fender& meatball dialplan and you lack of understanding of it, and those dial plan apps = failure
21:06.01jplankwell, not as much as not understanding it, I use very few of freepbx's dialplans, but DID routing is one of them, and I forgot that, so I forgot that the inbound call was going through 100 macros before ringing the extension
21:06.23*** part/#asterisk Cresl1n (n=matt@nat/digium/x-1751bd4058c47317)
21:06.36jplankbut if I set pickupmark at the last macro, it *should* work I think
21:08.08ManxPowerjplank: the PROBLEM with FreePBX is that all calls go thru like 100 macros.
21:09.10jplankI really use freepbx for creating extensions, all the context's I've created in extensions_custom, I know its stupid, but I originally started with freepbx, so now I'm so deep into it, its easy working around it then rebuilding
21:09.29jplankeasier*
21:10.07[TK]D-FenderI love it when people send good money after bad...
21:12.13*** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.207.184)
21:13.26ManxPowerjplank: At least you are not a total idiot, unlike the vast majority of GUI users.
21:15.09_trinewell that's a backhanded compliment if ever I heard one
21:15.15_trine:))
21:15.30jplankmy problem is I jumped into asterisk before learning exactly how it worked, so I was stuck with a gui for a while, and then I realised all of freepbx's limitations so I had to start compensating for them, luckily with the custom conf's its not THAT hard to do
21:15.54jplankthe plan is to sooner then later rebuild the whole box on a test box without freepbx, get it running and them swap them
21:16.02jplankunfortunately easier said then done
21:18.19j0i started with raw asterisk for a few years.. but got tired of editing configs and need to have some way for un-educated techs to make changes... so now i use ...... trixbox! barf
21:19.15*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
21:19.40jplanktrixbox is good if you don't want to do anything really cool, or pretty much anything but make a couple phone calls
21:19.50[TK]D-Fendersalmonella : the state of constantly swimming upstream only to get fucked & die.
21:20.44j0jplank: yeah.. and it *just works* as long as u dont try to add all the cool features.. :) i just started using aastra phones with trixbox and it's made my life so much easier
21:21.30jplankI just started testing aastra today, fendor doesn't seem to like them, but I'm loving it
21:22.13jplankdon't know if we are going to start swapping out polycom users any time soon, but if the testing keeps going so well, might give us another option
21:22.31jplanktried grandstream, but I was ready to throw them away after a day
21:22.58j0haha. i started with grandstream for my personal system just to test with... they've been terrible
21:23.51ManxPowerwe tested GS BT101, Cisco 7940, Polycom 500, and some Uniden SIP phone  (aastra was not an option at the time -- their IP phones did notexist)
21:23.57ManxPowersettled on Polycom
21:24.02[TK]D-Fenderaastra 5i's have many disappointments, and nothing I'd pick one over a polycom for except in extremely specific scenarios.
21:24.16[TK]D-FenderI wouldn't want one for any of my users
21:24.49[TK]D-FenderManxPower: uniden uip-200 = z0mg kill me now
21:25.18ManxPower[TK]D-Fender: It was a UIP somethething, it was about as good as a grandstream
21:25.53[TK]D-FenderManxPower: I have 2.  glad they run and I don't have to think about them ever
21:26.11jplankfender: what disappointed you about 5i's?
21:27.25jeevFender is the canadian chuck norris, everything disappoints him and then he will head butt it
21:28.44*** join/#asterisk Yourname (i=Yourname@unaffiliated/yourname/x-837320)
21:28.56[TK]D-Fenderjplank: no handset weight, tinny audio, no base weight, slippery rubber base that gets yanked around by the handset, shit viewing angle on the LCD, pixel screen used in dumb char matrix mode designed by retards, cryptic button icons to name a few
21:29.23[TK]D-Fenderjeev: no, I have almost no complaints about polycom....
21:29.37RypPnsounds like my first phone, the atcom 530
21:29.56[TK]D-Fenderjeev: atcom = crap, just like GS
21:30.00jeevFender, i've been getting a lot of complaints about hang ups.. so ManxPower figured i should try a gre tunnel and use SIP..
21:30.13jeevi heard grandstream's = satan
21:30.23YournameHi guys, sometimes an agent is logged into a queue via SIP, and his internet connection drops. The agent is logged in for several minutes afterwards. And so when the agent comes back online in 2-3 mins, he cant log back on cuz Asterisk thinks he's still logged on
21:30.24[TK]D-Fenderjeev: nobody wants to talk to you... it isdn't the phone ;)
21:30.25jplankfender: I can't speak for the 5i, but the CT version doesn't seem to have (all of) those problems
21:30.30jplankthe weight is fine
21:30.30jeevhahah
21:30.32jeevyou suck
21:30.39jplankthey replaced the icons with labeled
21:30.43ManxPower[TK]D-Fender: I'm helping him as the networking piece is interesting.
21:30.46[TK]D-Fenderjplank: 5i = the whole SERIES.
21:30.47jplankthe feet have grips
21:30.53jeevhopefully that is it.
21:30.57[TK]D-Fenderjplank:  and I specifically had the 57i CT
21:30.58ManxPowerand most of what I *do* is networking.
21:31.06jplankthats what I'm testing here
21:31.24jplankthe manual shows different buttons and feet then came with it
21:31.35[TK]D-Fenderjplank: yet it has grips.... THEY SUCK
21:31.39jplanklol
21:31.42jeevManxPower, so now i just change all iax crap to sip and viola, eh?
21:32.09jplankI def can kind of agree with the LCD comments, but the backlight helps a lot I think
21:32.11ManxPowerjeev: sounds simple, huh?  It's not, but I can help, as I have a similar Asterisk SIP to Asterisk SIP in production.
21:32.28jeevyea, i think i'll get that fine, i haven't tried.
21:32.48ManxPowerjeev: /join #asterisk-cli so we don't start flooding the channel with talk.
21:32.49[TK]D-Fenderjpbacklight gives you something to look at when the writing disappears as of 10deg view
21:35.16jplankI need to get some more time under my belt with this phone to experience your dislikes it seems, but features wise, it seems solid so far
21:35.23jplankthats a lot more then I can say about most phones
21:36.36YournameSo....nobody
21:41.50jplankI can't remember how to trim a string from the right instead of left, anyone know off the top of their head?
21:43.12[TK]D-Fenderjplank: channelvariables.txt
21:44.11jplankthanks
21:44.51jplankheh, that should of been obvious, thanks
21:46.29*** join/#asterisk knarfly (n=knarfly@c-75-74-155-198.hsd1.fl.comcast.net)
21:47.44*** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net)
21:55.14jplankaha!
21:55.21jplankscrew freepbx and their crazy marco's
21:55.26jplankgot pickup to work!
21:55.39jplankpain in the ass though
21:56.17jplanknow if I could just get the aastra to send **+exten when pressing the BLF button, I'd be in business
21:57.55*** join/#asterisk Levonk (n=lk@adsl-76-230-111-142.dsl.lsan03.sbcglobal.net)
21:57.55[TK]D-Fenderjplank: it sends what you tell it to
21:58.05jplankdoes it?
21:58.14[TK]D-Fenderjplank: yup
21:58.23jplankthe value?
21:58.49jplankput if I put the value as **+exten, wouldn't it request to notify for that?
21:59.22jplankyup
21:59.26jplankerrr
21:59.29*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
21:59.29*** mode/#asterisk [+o mog] by ChanServ
21:59.58[TK]D-Fenderjpchange the hint
22:00.49jplankerr backs to hints, thats where I lost it with the polycoms
22:01.35[TK]D-Fenderjplank: hint = notify....
22:01.51jplankoh thats it?
22:04.35jplankwait I still don't get it...
22:04.35*** join/#asterisk rconnect__ (n=rconnect@85.138.128.75)
22:05.00jplankhow would I send the hint?
22:05.31jplankif I want to do **2003
22:07.00[TK]D-Fenderjplank: then thats the exten that should have the hint
22:07.55jplankI get that, but how would I send the hint, exten => hint,SIP/2003 ?
22:07.59jplankthat doesn't make sense
22:08.32jplankohhh
22:08.43jplankexten => hint,SIP/**2003 ?
22:10.09rconnect__what is one of the best linksys routers to use with Asterisk?
22:10.11*** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com)
22:10.28jameswf-homelinksys ewww
22:11.00ManxPowerexten => _**XXXX,hint,SIP/2003
22:11.00[TK]D-Fenderjplank: see the problem... you don't even know how one is DEFINED
22:11.23[TK]D-FenderManxPower: through 1.4 you cant use patterns
22:11.31rconnect__<jameswf-home, which one you recomend then? :)
22:11.43jameswf-homemake your own
22:11.45tvirusIs it possible to change the outgoing caller ID on an analog line (Asterisk with a TDM410). The line is from Verizon.
22:11.47[TK]D-Fenderrcwhicever
22:11.55[TK]D-Fenderrconnect__: whichever
22:11.56jplankfender: yea I knew that was the problem
22:12.02ManxPowertvirus: no
22:12.13rconnect__ok
22:12.15[TK]D-Fendertvirus: did verizon tell you a way?
22:12.23ManxPower[TK]D-Fender: Ah!
22:12.35tvirusWe have yet to contact them, figured it would have to go through them. Thanks.
22:12.48[TK]D-Fendertvirus: 99.99999999999999999999999999999999999999% = impossible
22:12.56ManxPowerso it would be exten => **2003,hint,SIP/2003 would be correct
22:13.10tvirusSo there is a chance.
22:13.13tvirus:D
22:15.43jplankManxPower: let me see if I follow
22:16.05jplankthat, thrown into my context, should translate the aastra's notify **2003 message to 2003?
22:16.42[TK]D-Fenderjpno
22:16.45[TK]D-Fenderjplank: no
22:16.55jplankit did work btw
22:17.13jplanknow the question is why?
22:17.34jameswf-homehow to fix any asterisk issue:http://jantman.dyndns.org:10011/wiki/index.php/Generic_Problem_Solving_Method
22:18.57jplankdo you mind explaining?
22:22.14*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
22:22.14*** mode/#asterisk [+o mog] by ChanServ
22:25.06jplanktvirus: you'd have to be some pretty good telephone freak to spoof your POTS line's caller-ID (without redirecting the call of course)
22:25.22jplankphreak*
22:26.35[TK]D-Fenderjplank: like the kind that can spontaneously create services that don't exist, and then take advantage of them...
22:27.42jplankit was sarcasm
22:28.06jplankand its not something that doesn't exist
22:28.28jplankbut most telco's wont do it for you
22:28.42jplankyoud have to break into their switches and set it up
22:36.20rconnect__anyone know why accessing the gui, this message is displayed? The GUI does not have necessary privileges.
22:36.30rconnect__i have configured both manager and http
22:36.44rconnect__Please check the manager permissions for the user !
22:41.35jplankmaybe its a NAT problem?
22:45.28[TK]D-Fendernope, and GUI's are not supported in this channel
22:46.09rconnect__ok, i have asked in the proper channel as well, but no answer so far, so i thought maybe someone here would know... sorry
22:48.38jplankgrrr no one gets my sarcasm, I actually thought he was asking a sarcastic question himself
22:49.20jplankrconnect__: what gui are you using?
22:51.04rconnect__digium asterisk gui 2.0
22:54.42jplanknever used it, so can't help you there, but if its anything like trixbox or freepbx, most privileges issues come down to changing passwords in the gui, and not changing the proper password on your system (example mysql) or vice versa.
22:55.05jplankcan I change the value of ${EXTEN}?
22:55.09rconnect__right. thanks jplank
22:55.17rconnect__have to go. good night
22:55.31*** join/#asterisk Levonk (n=lk@adsl-76-243-67-57.dsl.lsan03.sbcglobal.net)
22:55.31jplankI can't find anywhere it says I can't, but it doesn't seem to work
22:56.17jayteejplank Set(${EXTEN} = 1234567)
22:57.12jplanktried that, it doesn't seem to work
22:58.08jplankhttp://pastebin.com/m237285cf
22:58.44[TK]D-Fenderlol
22:58.56fileyou can not change EXTEN, it is a special variable used to return the current executing extension
22:59.14[TK]D-FenderAND you're refernecing the variabl
22:59.28[TK]D-FenderAND adding BS whitespace
22:59.36jayteeah, yeah
23:00.11jplankI was hoping there was a way
23:00.38jayteeso you'd do something like Set(${newexten} = ${EXTEN:2}) ?
23:01.04jplankI'm going to have to rewrite a few scripts now to get the ** out of there, and I thought it would be easiest to do it before it hit those scripts
23:01.46jplankwhy would it matter if I was referencing the variable?
23:02.01jplankand where am I adding a whitespace?
23:02.18jayteehe probably meant my bad example
23:02.27jplankahhh
23:02.40jplankohhhh
23:02.59jplankhe meant me ${EXTEN}=${EXTEN:2}
23:03.17jplankI only added that because jaytee said it, it was originally EXTEN=
23:04.18jayteetry using a unique variable name for the new extension number you want to dial
23:06.54jplankI'm trying to remove the ** so I can match for 2003
23:07.19jayteeso use a new variable name that isn
23:07.25jplankI'd hate to have to rewrite all my ext lists to account for **
23:07.27jayteeisn't a reserved system variable
23:08.19jplankhow would I match for that?
23:09.22jayteeyour paste didn't have any kind of code for matching anything
23:09.43jayteeyou're trying to strip the ** out and dial?
23:10.03jplankyea
23:10.12jplankno
23:10.26jayteewell that's frikken clear as glass
23:10.28jplanki thought you said ** out of the dial
23:10.30jplanklol
23:10.36jplankI'll show you the full code
23:11.05jayteeno please don't
23:11.21jplankhttp://pastebin.com/m39e0539b
23:11.23jayteethe last time someone showed me the full code I couldn't sleep for weeks and needed therapy
23:11.25jplankno two mroe liens
23:11.42jplanknot the full code
23:11.48jayteelook if your bank has liens on stuff that's not my problem
23:11.49jplankI wouldn't want to scare you away
23:12.34jplankis that clearer?
23:12.48*** join/#asterisk sanjayb (n=sanj@59.183.52.45)
23:13.11jayteejplank, those two pastes look identical to me
23:13.40jplankadded two lines
23:13.44jplankmatching for 2003 now
23:13.55jplankjust to show you why I'm trying to set EXTEN
23:14.35jayteeyes, I understand and I see those two line IN BOTH pastes and they're identical
23:15.42outtolunclast time i looked EXTEN was one of those you couldn't change
23:16.44jayteeso just try Set(${YOURMOMMA} = ${EXTEN:2} and on the next line NoOp(${YOURMOMMA}) and then do a Goto(samecontext,${YOURMOMMA},1)
23:17.00jplankoriginal pastebin http://pastebin.com/m237285cf
23:17.01jayteeon the next line
23:17.10jplanknew http://pastebin.com/m39e0539b
23:17.22jplankhow can you say they are the same?
23:17.27outtolunchttp://svn.digium.com/view/asterisk/branches/1.4/doc/channelvariables.txt?view=markup
23:17.36jplankouttolunc: they told me already
23:17.37outtolunc<PROTECTED>
23:17.37outtolunc<PROTECTED>
23:17.37outtolunc<PROTECTED>
23:17.49outtoluncok
23:17.49jplankI was just trying to show jaytee WHY I was doing it
23:18.05jplankthank you thought
23:18.47jplankis there any way to change the digits dialed then?
23:19.17outtoluncgoto(yadda,${your mod'd exten},1)
23:19.25YournameHi guys, sometimes an agent is logged into a queue via SIP, and his internet connection drops. The agent is logged in for several minutes afterwards. And so when the agent comes back online in 2-3 mins, he cant log back on cuz Asterisk thinks he's still logged on
23:19.32jayteejplank, ok my mistake on the same stuff. I had too many pastebin tabs open in Firefox.
23:19.38jplankn/p
23:19.46jplankI thought I was going crazy at first
23:20.02jplankI got it though
23:20.04jplankthank you both
23:20.30jplankI didn't read jaytee's message at first because I thought he was being sarcastic
23:20.43outtolunci missed his also
23:20.50outtolunchaha
23:22.36outtoluncneeds to build a modified roomba for steam cleaning the carpets
23:24.00jplankheh I left samecontext thinking that was some cool thing I didn't know about
23:24.07jayteecompletely forgot you couldn't actually change the value of ${EXTEN}, probably because I'd already read that and never atttempted it. I use unique variables to do stuff.
23:24.39jayteeno, the name of the context, which is optional I believe
23:25.15jayteeunless you WANT to jump to another context and then it's required
23:25.32jayteeanyone feel free to jump in and correct me if I'm wrong on that point.
23:25.46outtoluncyes, context in goto is optional
23:26.13outtoluncobviously if goto'ing a diff context then its required <G>
23:26.22jayteewhew!
23:27.05jayteebe back later
23:28.19outtoluncnote: never let the woman talk you into renting a rug doctor thing
23:36.27*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
23:37.54jplankis this the correct syntax exten => **2003,Transfer(SIP/2003) ?
23:38.10jplankI'm trying to redirect a SIP REF for **2003 to 2003
23:39.11*** join/#asterisk kdas (n=ME@c-67-161-44-222.hsd1.ca.comcast.net)
23:39.22kdasManxPower, you arround still?
23:39.36kdasjaytee, alive?
23:40.23kdasi have my extensions.conf file setup as _XX.,1, blah but i don't think that is too good is there a better way of doing this ?
23:42.04[TK]D-Fenderkdas: pick real patterns based on what your telco requires
23:42.45kdas[TK]D-Fender, reall patterns meaning 9 or 10 digets etc?
23:42.59[TK]D-Fenderkdas: is that proper?
23:43.16kdaswhats the use of 's' like s,1,blah it dosent catch anything ?
23:43.44kdas[TK]D-Fender, you tell me i mean i am not going to be expecting 1-235-2345-2342342 numbers just usa numbers
23:44.12[TK]D-Fenderkdas: "s" is not a catchall
23:44.57[TK]D-Fenderkdas: no i mean to tell you that YOU should know what kind of numbers are valid and not use wide reaching patterns like you have been to do the job
23:45.21kdas[TK]D-Fender, "s" is the start
23:45.49ManxPowerkdas: "s" is almost never used except for in macros and FXO signalled ports.
23:46.05[TK]D-Fenderkdas: you clearly do not understand the "s" exten
23:46.10kdasManxPower, oh thanks
23:46.42kdas[TK]D-Fender, well thanks for pointing out the obvious
23:46.54[TK]D-Fender~stdextens
23:46.54jbot[~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf.  "s" is also used to make IVRs & macros.
23:47.10kdasdamn i hate geeks who think they all that because they know just that tad more
23:47.24[TK]D-Fenderkdas: When you dial a number on a phone that lands on a NUMBERED extensions. "s" does not factor into this
23:47.37[TK]D-Fenderkdas: You jsut finished trying to
23:47.38kdas[TK]D-Fender, thank you
23:47.39ManxPowerkdas: [TK]D-Fender knows at least as much as I do.
23:47.51[TK]D-Fender"tell me how it is".  And then you you FAILED
23:48.07[TK]D-Fenderthen*
23:48.14[TK]D-FenderAnyways, forget "s" right now
23:48.24jblackHe doesn't have to help you, and you don't have to ask for his help.
23:48.28[TK]D-Fenderkdas: And go pick some sensible patterns
23:48.41jplankfender: how can I compensate for the aastra sending **+extension inside the SIP REFER on a BLF xfer
23:48.55kdasManxPower, i am not saying he isn't just i hate when "people" try to show off there knowledge like "you clearly don't understand what hacking into nasa means.. HHAHA.. N00b i am so 31337
23:49.30jblackjplank: Look at offsets. Such as ${EXTEN:3}
23:49.35kdasjblack, fair enough i never asked for his help but he is kind enough to help ;)
23:49.56kdas[TK]D-Fender, once again thank you i will do some more research and forgive my ignorance
23:51.00[TK]D-Fenderjplank: aastra sends exctly what you tell it to subsribe to.  its your job to make sure * matches
23:51.06jplankI know
23:51.12jplankthats what I'm trying to do
23:51.13ManxPowerkdas: I get as pissed off at people as [TK]D-Fender when they don't just let the person with experience drive.
23:51.56jplankhow can I get the variable from SIP REFER?
23:52.04jplankto look at the offsets
23:52.05[TK]D-FenderjpAnd the problem becomes your FreePBX dialplan.  As I said, it owns your ass... If you can dodge this bullet, good luck.
23:52.21jplankI've gotten pretty far so far
23:52.24[TK]D-Fenderjplank: You don't get aceess to traffic like that
23:52.58jplankso then how can I tell my * to refer calls destined for **2003 to 2003?
23:53.07jplankoh
23:53.23jplankI could just tell my * that **2003 and 2003 is the same thing, can't I
23:53.28jblackexten => _**XXXX,1,Goto(${EXTEN:2},1)
23:53.30ManxPowerjplank: If you were trying to get the info out of a TRANSFER, that might be able to be done, but not during a REFER that I'm aware of.  That requires much lower level access to the SIP protocol that Asterisk allows
23:53.58jplankjblack: I used the same thing with transfer
23:54.18jplankManxPower: so theres no way?
23:54.54jplankgrrr
23:55.29ManxPowerjplank: I suspect you have the wrong approach.  Have you checked the wiki?
23:55.38*** join/#asterisk mog (n=mog@74.95.48.254)
23:55.38*** mode/#asterisk [+o mog] by ChanServ
23:55.49jplankvoipinfo? I doesn't have much about REFER
23:55.57ManxPowerjplank: until the call hits the dialplan, you can't know anything about it.
23:56.04jplankhmmmm
23:56.06ManxPowerjplank: No.  What you are trying to accomplish.
23:56.24jplankbasically, i'm working on BLF with a aastra phone
23:56.28ManxPowerWhich I suspect is direct accesss to call parking lots using a a BLF for each slot.
23:56.42ManxPowerso that is what you should look on the wiki for.
23:56.48jplanklol
23:57.05ManxPowerIt doesn't matter HOW you do it, as long as you can do it.  don't get stuck in the mechanics of it.
23:57.16idodoes anyone here have a good all-software solution for fax detection from incoming SIP or IAX channels?
23:57.27*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
23:57.33idopreferrably in tutorial format? :)
23:57.49ManxPowerido: NVFaxDetect if you can find the code.
23:57.57idoManxPower: where might one find the code?
23:58.11ManxPowerjust remember chances are it won't work on anything except ulaw and alaw codecs.
23:58.19idoulaw is fine
23:58.22ManxPowerido: I have no idea.  check the wiki, the link is dead

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