IRC log for #asterisk on 20100121

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01:24.45staffmemberanyone awake ?
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01:26.07p3nguinIs that all you want to know, or did you have a question about Asterisk?
01:26.58carrarI'm your huckleberry
01:27.38paulcI'm awake, but fading fast
01:27.45staffmemberjust awkward silence, but yeah there is a question.. anyone know where i can find some docs on nortel and asterisk
01:27.48staffmemberwhitepapesr
01:27.54staffmember*whitepapers
01:27.58carrargoogle
01:28.03staffmemberdid that and got what i watned
01:28.04voipmonkstaffmember: t1?
01:28.08paulcWhat kind of docs? like Nortel using Asterisk? Integration between BCM or DMS and Asterisk?
01:28.11staffmemberbut wanted to know if anyone else had some sources i dont know about
01:28.15carrarlike the unistm driver?
01:28.17staffmemberpaulc: yeah
01:28.34staffmembervoipmonk: AEX800
01:28.52carrarI'm installed asterisk between a nortel and the PSTN
01:28.59carrarI've
01:29.13paulcI've got a friend who's done that too.. PSTN <--> Asterisk <--> Option 11C
01:29.29paulcI recall the Opt11 was a bitch to configure, but once done, all was fine and happy
01:29.42staffmemberpaulc: how about PSTN -> Asterisk -> Nortel Compact ICS + callpilot
01:29.46carrarThis was PRI on both sides
01:34.16paulcstaffmember: Compact ICS is like "Norstar" right?
01:34.20paulcthat should be easy
01:34.23paulcI love those systems :)
01:36.49paulcstaffmember: What do you want to do with it exactly? Can you give me the 1 or 2 sentence "It should work like this..." ?
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01:53.39simplydrew_I've been wanting to get a DID for my asterisk box that I can have forward to my cell phone, just as a "work only" number essentially to reach me on the go. Although I know I can configure this without a problem, is it also possible to forward text messages to my cell if sent to the DID?
01:54.13*** join/#asterisk cadmium (i=mike@217.194.139.22)
01:54.41cadmiumhi what channel variable can I use to get the peername of the phone which is placing the call
01:56.00paulcsimplydrew_ It's been talked about but SMS forwarding is still a bit up in the air it seems. Hard to find a DID provider that supports it.
01:56.08paulcThat said, if you find one, let me know
01:56.28carrarSIPCHANINFO(peerip)
01:56.28simplydrew_paulc: ah gotcha, so the dependency is all reliant on the DID provider as well then?
01:56.35carraroh
01:56.37carrarpeername
01:56.38paulcmy understanding is that most will ping a URL when you get sent a text message (but maybe some provide it as SIP messaging?)
01:56.59simplydrew_hmm. I see. I'm with vitelity. I'll have to ask them.
01:57.00paulcsimplydrew_ yes - The DID provider has to be a recognised "end point" for SMSs to be sent to
01:57.01carrarSIPCHANINFO(peername)
01:57.17paulcI'm with them too - they're pretty good (but don't do SMS as far as I know?)
01:57.42p3nguinMaybe the SIPPEER() function would be better?
01:58.27cadmiumaghh got it myself ${CALLERID(num)}
01:58.54p3nguinThat's not the PEER INFO
01:59.37p3nguinThe CALLERID() function is only as reliable as the person who set his caller ID info.  It is easily spoofed.
01:59.57p3nguincarrar is right -- SIPCHANINFO(peername) is probably the best.
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02:00.23random_mikegreetings p3nguin :)
02:00.25carrarwell that this point the user doesn't even know what he's looking for
02:00.41p3nguinrandom_mike: hi
02:02.15random_mikethanks for your ongoing assistance
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02:02.16*** join/#asterisk Kevin10258 (n=airmadne@173-112-45-135.pools.spcsdns.net)
02:02.22Kevin10258Hello all..
02:02.25random_mikemy asterisk box is nearly complete :)
02:02.32Kevin10258Is SipFoundry Asterisk based?
02:02.41Kevin10258Does anyone know
02:03.04p3nguinrandom_mike: Are you confident enough in your doings to reveal your works to the world?  Specifically, to #Asterisk?
02:03.38random_mikeperhaps once its 100%
02:03.51random_mikejust ironing out the last few bugs, adding the last features
02:04.08random_mikeive no doubt i'll be hassling for more assistance when I get stuck again :P
02:04.15Kevin10258random_mike : What did you use to create your PBX?
02:04.33carrarrpm -i
02:04.41carrarheh
02:05.03random_mikeKevin10258, I'm running Asterisk 1.4.28 under CentOS 5.4
02:05.03Kevin10258I am trying to decide on using a distro like Elastic or creating my own on Defora
02:05.07Kevin10258Fedora
02:05.22Kevin10258Did you install all the RPMs or was it an ISO install?
02:05.35random_mikeI enabled the atrpm's repo, and did a yum install
02:05.35p3nguinneither
02:05.45Kevin10258I c
02:06.02Kevin10258I used to used TrixBox, and I loved it.
02:06.20random_mikealthough to get everything correct, I had to enable testing and bleeding edge to the base repo.
02:06.24random_mikeI've not used TrixBox
02:06.27Kevin10258But the server was to loud, drowning out the TV, so it's down until I find a new one that doesn't sound like a 747 taking off in my tv room.
02:06.37p3nguinI run two asterisk servers -- one is ArchLinux, the other is FreeBSD.
02:06.55Kevin10258Just another layer over the top of Asterisk to make it easy.
02:07.29p3nguinMaybe you want an Atom system so you can run fanless.
02:07.35*** part/#asterisk cadmium (i=mike@217.194.139.22)
02:07.48Kevin10258I have thought about that.
02:07.49random_mikeMy server is virtualised on a Dell m1000 blade chasis :P
02:08.19Kevin10258I have also thought about a small p3 based system without a fan.  Cheaper, and the same performance as the Atomless.
02:08.37Kevin10258But I really want a RACK mounted solution.
02:08.49Kevin10258To go in my rack with my HP Procurve 2948. :D
02:09.14random_mikeany rack server is going to be noisy, blame the industry for that.
02:09.29random_mikeperhaps you can Atom it, and mount some super big ears to it, and rack it up :P
02:09.40carrarKevin10258, you have our permission. Please make it so
02:10.06Kevin10258huh?
02:10.07Kevin10258lol
02:10.18Kevin10258Yeah, I am working on paying off the credit cards first.
02:10.29carrarThats not very American
02:10.40Kevin10258Yeah, well my wife and kids are cherokee
02:10.46Kevin10258so we don't americans very much.
02:10.47carrarFree money!
02:10.47Kevin10258:D
02:10.52Kevin10258yeah right
02:10.55Kevin10258that's a MYTH dood.
02:10.59carrarhaha
02:11.05carraryou 1/4 or more?
02:11.08Kevin10258My wife is certified, as are our kids.
02:11.11carrarerr 1/8
02:11.19Kevin10258My wife is, and my kids are, I am not certified.
02:11.31carrarDo they call you 'Running with Asterisk'
02:11.31random_mikeyou americans are weird :P
02:11.32Kevin10258They have their RES cards and all, there is no free money.
02:11.44Kevin10258I am not American, I am a leftist.
02:11.45carrarheh
02:11.49Kevin10258Sorta
02:12.01Kevin10258Well mate..
02:12.19Kevin10258We could say the same about Ausie, and what was done to the natives there.
02:12.30random_mikeLets not walk down that alley
02:12.34Kevin10258LOL
02:12.48random_mikeit's despicable. And continues to be.
02:12.59Kevin10258We don't "celebrate" thanks giving.
02:13.30Kevin10258My oldest daughter and I talk about how Columbus trickd the indians, enslaved them, and killed those who would not comply.
02:13.44Kevin10258Never forget.
02:13.59carrarIndians had weapons of mass desctruction
02:14.05Kevin10258ROFL
02:14.07carrarthey had to be shut down
02:14.18Kevin10258<PROTECTED>
02:14.22Kevin10258Wait, I am white.. ;)
02:15.02carrarbe proud of you who you are
02:15.06random_mikeim sorry I dont know the history of the Indian's in the US
02:15.13random_mikeOther than there's not many left
02:15.25*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
02:15.34Kevin10258Yeah, it's TERRIBLE.  Watch "A peoples history" and you will find the truth.
02:15.44Kevin10258And that's only some of it.
02:15.52random_mikeTerrible as in Aboriginal Australians terrible?
02:15.55Kevin10258Kill de white people... He he he o.
02:15.57Kevin10258lol
02:16.09Kevin10258I know nothing of what has happened to them.
02:16.36Kevin10258But the indians were mass murdered, forced into slavery in some cases, and then forced onto reservations because they were hard to look at.
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02:16.58Kevin10258They were forced to abandoned their beleifs, language etc.
02:17.19Kevin10258I get pretty pissed when I think about it, not much different that what is happening to us as a people today.
02:17.29Kevin10258Minus most of the murdering.
02:18.03random_mikeright - well the Aboriginal Australians were mass murdered, chased off into the bush, and then in the 50's and 60's the Australian Gov had a great idea to go and take every Aboriginal Child and place them in Foster Care, to teach them white australian ethics.
02:18.15random_mikethis ofcourse didnt work
02:18.44Kevin10258LOL
02:18.51Kevin10258Asterisk is hard man.
02:18.55Kevin10258but awsome when it works.
02:19.17random_mikeagreed - all this damn history - wait... asterisk? I thought I was in #history for a second then.,...
02:19.24random_mikeAsterisk IS awesome when it works
02:20.59random_mikeAnyone know a good wakeup call script/scripts/service to stick on my asterisk box? :D
02:21.19random_mikeI've experimented with a few but didnt really find them that appealing.
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02:26.39simplydrew_paulc: I got a response from vitelity about the DID SMS. They said they are in the final stages of SMS platform completion. So sounds like it'll be available son
02:26.40simplydrew_*soon
02:29.54Kevin10258Does anyone know of a GOOD cost effective SIP provider?
02:30.50carrar~itsp
02:30.50infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
02:30.55random_mikeheh - too bad you're not in Australia or I'd suggest my company
02:31.18carrarWhat state are in?
02:33.15Kevin10258Missouri
02:33.19Kevin10258AKA MYsery
02:34.39p3nguinkevin10258: VoIP.ms
02:35.05p3nguinkevin10258: I would say Flowroute, but I am told they are declining new sign-ups right now.
02:45.01random_mikequery regarding extensions.conf
02:45.27random_mikeI have my extension set up like: exten => *99,1,Answer
02:45.27p3nguin!ask
02:45.46random_mikeif the user then press's 1, their directed to that area of the conf
02:46.09random_mikeexten => 1,1,Set(SUCCESSCOUNT=$[${SUCCESSCOUNT} + 1])
02:46.13p3nguin"area of the conf"
02:46.48random_mikei think this call's for a paste bin
02:47.06p3nguinWhat's with the random apostrophes?
02:48.40random_mikeim random
02:48.45random_mikei sec I broke something
02:48.50random_mike*tears out hair again*
02:51.23jayteeever notice that most older I.T. professionals are balding or have very thin hair? ever wonder why?
02:51.28*** join/#asterisk RobH_ (n=robh@cpe-173-169-30-118.tampabay.res.rr.com)
02:54.34Kevin10258They don't have "unlimited" plans.
02:55.13Kevin10258Yeah, "F" that..
02:55.20Kevin10258Callcentric is at least unlimite
02:55.21Kevin10258Callcentric is at least unlimited
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02:56.27Kevin10258jaytee : Because we have been watching it ALL FALL APART for years.
02:56.28Kevin10258:D
02:58.38p3nguinkevin10258: CallCentric also isn't that good.
02:59.27p3nguinkevin10258: Well, flowroute is letting me sign up, so I don't know what's up with them.
03:00.33p3nguinkevin10258: Flowroute's Flat Rate Unlimited plan - Bundles a DID with two channels for unlimited free incoming calls to that number, but no more than two simultaneous calls.
03:00.44p3nguinkevin10258: $6.95/mo
03:01.53*** join/#asterisk netpro25_ (n=mmanning@64-238-176-248.ksg.apt.gru.net)
03:04.02netpro25_Hello, I am having issues with hold music. Asterisk is saying the hold music is started but I hear nothing. I am using mp3s.
03:04.12netpro25_Asterisk says: Started music on hold, class 'default', on channel 'SIP/desktop-ec3b3628'
03:04.41netpro25_I have asterisk 1.6
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03:07.15hobbes615join #freepbx
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03:15.18_ShrikEIn 1.4, will asterisk match the username or ip address first if type=friend for sip inbound call?
03:31.00Kevin102586.95 per months?
03:31.01Kevin10258What?
03:32.05p3nguinkevin10258: what?
03:33.15Kevin10258<p3nguin> kevin10258: Flowroute's Flat Rate Unlimited plan - Bundles a DID with two channels for unlimited free incoming calls to that number, but no more than two simultaneous calls.
03:33.16Kevin10258<p3nguin> kevin10258: $6.95/mo
03:33.32p3nguinOkay, so what's the problem?
03:34.08Kevin10258I don't beleive it
03:34.13Kevin10258They don't state it on their website
03:34.22Kevin10258I am paying nearly 4 times that for 1 line
03:34.27p3nguinActually they do.  That's how I was able to give you the info.
03:34.27Kevin10258as it stands right nowe
03:34.36Kevin10258Where are you seeing this?
03:34.40p3nguinVoIP.ms is about the same rate as Flowroute, too.
03:34.44Kevin10258I see a point click run around nightmare
03:35.01p3nguinhttps://www.flowroute.com/services/inbound/
03:35.43Kevin10258That's inbound, what about outbound?
03:35.55p3nguinhttp://voip.ms/dids.php?action=statesearch&state=MO
03:36.20p3nguinOutbound with flowroute is $0.0098/minute
03:36.40p3nguinor $19.95/mo. with CallCentric.
03:37.12Kevin10258Yeah, I will stick with CC, I use the phone WAY to much to have that over my head.
03:37.20Kevin10258Their support is pretty good as well.
03:37.43p3nguinYou use more than 2000 outbound minutes each month?
03:37.49Kevin10258WAY
03:37.57Kevin10258I have 5 kids, a wife and myself.
03:38.15Kevin10258I do A-LOT of side work and testing for other people that reuires me to be on the phone.
03:39.02Kevin10258Even if I didn't I am not big on "limited" plans anyway, call it the "American" in me.
03:39.16Kevin10258I ALWAYS get screwed at some point.
03:40.39*** join/#asterisk d00gster (n=doughant@77.31.197.112)
03:42.56dlynesKevin10258, vitelity has an all you can eat plan
03:43.12p3nguinfor termination?
03:43.27p3nguinIt should be 1.4c/minute for termination.
03:43.29ChannelZHow many minutes do you use?  .0098/min is nothing
03:44.16p3nguinHe uses far more than 2000 minutes, so the $19.95/mo. plan with CallCentric seems to be more cost effective.
03:44.49p3nguinThat's $19.95/mo. for unlimited outbound calling.
03:45.13p3nguinMost ITSPs don't do unlimited outbound rates.
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04:25.08titterAnyone know of a good IAX provider?
04:36.14random_mikeanyone know what RFC asterisk prefers?
04:42.39p3nguintitter: VoIP.ms
04:43.50*** join/#asterisk FirstSgt (n=cheney@cpe-72-181-174-190.tx.res.rr.com)
04:44.33FirstSgtDoes anyone have a network topology map for how SIP Clients, Softphones, PBX, SIP, DID, and DOD work?
04:44.55p3nguinI don't get it.
04:46.38FirstSgtNeither do I... it's a complex subject.
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04:46.52FirstSgti dont even know what to google for
04:51.35p3nguinCan someone verify the stock price on Charter Communications for me, please?
04:51.44p3nguinIf it is what I think it is, I think I am rich.
04:52.09*** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com)
04:53.59ChannelZdepends what market and counyty, seems they are listed in a bunch of places
04:54.30p3nguinUSA, NASDAQ
04:57.19daidoji31.75
04:57.21daidojiyou rich?
04:58.16daidojialthough its been in decline for the past couple of days
04:58.25daidojiand didn't you get wiped out in the bankruptcy?
04:58.34daidojior did you buy like 5 days ago?
04:59.03p3nguinI lost $500 on it just today alone.
04:59.22p3nguinI bought months ago when it was under $0.25 per share.
04:59.35daidojio well then you are rich
04:59.47daidojicause 31.75 is a lot higher than .25
05:00.04daidojialthough I thought you would have gotten wiped out in the bankruptcy
05:00.38ChannelZonly rich if you sell it
05:00.49daidojiehhh I'd keep it
05:01.09daidojiCharter's a good company, they re'orged, and they've turned nice profits in the past
05:01.22daidojinow that cheap credit is over and they don't have to try and expand everywhere they'll do all right
05:01.29ChannelZwe're shooting an internal 'rah-rah' video for them
05:01.33daidojiespecially if you bought at .25
05:03.04daidojiactually if you wanna go really crazy on it and make some serious money
05:03.13dlynesrandom_mike, your question is totally random, and makes absolutely no sense
05:03.15daidojishort it for about 3 days on margin
05:05.10random_mikedlynes, I know that - but where I sit... We have a SIP server running (slipper) and an upstream SIP provider (isphone) and they are very odd about which SIP RFC they support
05:05.48random_mikefor a call to be marked as a private number, they ignore the P-Asserted-ID tag, and the Privacy: id tag, and only rely on the FROM field being re-written to Anonymous
05:06.18random_mikeit's my understanding that re-writing the from field is an older standard, although good practice now, it's not the only method for making a call as private.
05:08.07random_mikeyou'll have to forgive me, after being thrown in the VoIP deep-end I'm struggling to float ;)
05:08.34dlynesrandom_mike, it's not a matter of which rfc, but rather which version of sip, and how fully they support that version
05:08.48dlynesrandom_mike, asterisk supports all of sip v2 (afaik), and part of sip v3
05:08.56dlynesafaik, again
05:09.01random_mikethankyou dlynes
05:09.03dlynesthat being said
05:09.40dlynesoej is working on a complete sip v3 stack for asterisk...I heard rumors it was going into asterisk 1.6.x, but I'm not so sure that ever happened
05:10.43dlynesrandom_mike, however, a quick way to see which method it's using for the particular behaviour you're talking about is to enable sip debug during a call where you expect to see that, so you can see how asterisk does it
05:10.55dlynesrandom_mike, or use pcap or pcapsipdump
05:11.09dlynes~pcapsipdump
05:11.21dlyneshrm...i guess the bot doesn't know about pcapsipdump
05:11.38dlynesrandom_mike, pcapsipdump.sf.net, iirc
05:11.58dlynesit'll dump every single sip conversation into a separate pcap file
05:12.10dlynesgreat for debugging sip traffic
05:12.12[TK]D-Fenderthrows a pineapple at dlynes
05:12.17dlynes[TK]D-Fender, ?
05:12.39[TK]D-Fenderdlynes: Check your history...
05:12.47dlynes[TK]D-Fender, which history is that?
05:12.54[TK]D-Fender*
05:13.04dlynes[TK]D-Fender, regarding sip version numbers?
05:13.40dlynes[TK]D-Fender, is the sip v3 stack going into asterisk 1.8, then?
05:15.21dlynesrandom_mike, anyways...here's a sip compliance document for your perusal:  http://svn.asterisk.org/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt
05:16.23dlynes[TK]D-Fender, hrm...you throw a pineapple at me, and then run off like a little girl.... :(
05:16.46random_mikesorry was afk
05:16.50ChannelZmaybe he went to get more pineapples
05:17.00dlynesChannelZ, or maybe some mangoes
05:17.19dlynesor maybe he's just not fruity enough
05:17.51dlynesrandom_mike, is that file what you were looking for, then?
05:18.18random_mikeyes dlynes, this looks deliciously juicy.
05:18.38random_mikewhilst you're here, and you seem to know your stuff
05:18.48dlynesrandom_mike, i don't know my stuff that well :)
05:19.04random_mikeis it possible, using extensions.conf to find if a "Privacy: id" tag is present?
05:19.07dlynesrandom_mike, at least not the super technical protocol level stuff
05:19.18[TK]D-Fenderdlynes: http://www.google.ca/#hl=en&source=hp&q=asterisk+SIP+pineapple&btnG=Google+Search&meta=&aq=f&oq=asterisk+SIP+pineapple&fp=ada65163c2197ad6
05:19.44[TK]D-Fender"Psst. Top Secret... Codename Pineapple | Asterisk 24 Nov 2006 ... So what is the chan_sip3 project (codename pineapple) about? "
05:20.07[TK]D-Fenderdlynes: There have been 3-4 simultaneous complete rewrites in the roks for almost half a decade
05:20.12[TK]D-Fenderworks*
05:20.33dlynes[TK]D-Fender, so what you're saying in a roundabout way, is that you have absolutely no confidence it'll get completed in our lifetime?
05:20.49random_mikeI've found this regarding privacy headers:   exten => _9.,1,SIPAddHeader(P-Asserted-Identity: <sip:+1${CALLERID(num)}\;user=phone>)
05:20.56random_mikebut what I seek is Privacy: id header
05:21.16dlynesrandom_mike, why not use sipaddheader for it, too?
05:21.30random_mikeI dont know what sipaddheader is :)
05:21.40random_mikealthough i've now consulted the google bible
05:21.47random_mikeso hopefully this paints a pretty picture for me
05:22.19random_mikeok - clarification - sipaddheader only add's a header?
05:22.35random_mikeit cant be used in a GotoIf, if the header is present?
05:23.22dlynesrandom_mike, how about the function, 'CALLERPRES'?
05:23.55*** part/#asterisk nny (n=scott@64.203.239.83)
05:24.01dlynesrandom_mike, type 'core show function CALLERPRES' at the asterisk cli
05:24.21random_mikebf1*CLI> core show function CALLERPRES
05:24.21random_mikeNo function by that name registered.
05:24.26random_mikethat's not fun
05:24.28dlynesrandom_mike, yes...sipaddheader only adds a header
05:24.40random_mikedlynes, perhaps I should further clarify
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05:25.13dlynesrandom_mike, which version of asterisk are you using?
05:25.30dlynesrandom_mike, CALLERPRES as a function only exists in asterisk 1.6
05:25.53dlynesrandom_mike, in earlier version it was SetCallerPres() (a dialplan application)
05:26.03random_mikemy server accepts incoming calls only. Based upon the existance of the Privacy: id, I can tell whether or not the call has callerid suppression enabled or not. And depending on whether or not callerid suppression is enabled, I wish to read "CALLER ID SUPPRESSION ENABLED" or CALLER ID SUPPRESSION DISABLED" as such., to the incoming call.
05:26.12random_mikeI'm using 1.4.28
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05:28.16dlynesrandom_mike, i can't remember...refresh my memory...does 1.4.28 have SIP_HEADER function?
05:28.22dlynesrandom_mike, core show function SIP_HEADER
05:28.51random_mikebf1*CLI>
05:28.51random_mike<PROTECTED>
05:28.51random_mike[Syntax]
05:28.52random_mikeSIP_HEADER(<name>[,<number>])
05:28.52random_mike[Synopsis]
05:28.52random_mikeGets the specified SIP header
05:28.54random_mike[Description]
05:28.56random_mikeSince there are several headers (such as Via) which can occur multiple
05:28.57*** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-zdrnotdnckwzqruo)
05:28.58random_miketimes, SIP_HEADER takes an optional second argument to specify which header with
05:29.01dlynesrandom_mike, PID=${SIP_HEADER(Privacy)}
05:29.02random_mikethat name to retrieve. Headers start at offset 1.
05:29.04random_mikebf1*CLI>
05:29.14dlynesrandom_mike, wouldn't that do it?
05:29.31random_mikeYes, that appears to do what I'm after
05:29.36dlynesrandom_mike, also, if you're pasting 3 or more lines, please use pastebin in the future
05:29.37dlynes~pb
05:29.38infobot[~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://bin.cakephp.org/ , or apt-get install pastebinit
05:29.38random_mikeis that not valid in 1.6.x?
05:29.45dlynesrandom_mike, it is
05:29.58dlynesrandom_mike, it is valid, that is
05:30.39dlynesrandom_mike, so ${PID} will now contain the value of your privacy id
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05:30.59dlynesrandom_mike, that being said, i would use a different name than 'PID', because PID usually means process ID
05:30.59random_mikesorry - have an outage at work
05:31.00random_mikeafk
05:33.10drmessanoYou may have more friends on Facebook than I, but what about enemies?
05:33.32drmessano</thought of the day>
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05:36.20random_mikethanks dlynes
05:36.21random_mikesorry
05:36.30random_mikeour upstrean VoIP provider just had an outage
05:36.34random_miketalk about good timing
05:37.03random_mikeso dlynes, in your example: PID=${SIP_HEADER(Privacy)}
05:37.23random_mikewhat would happen if the sip header: "Privacy" didnt exist?
05:38.56random_mikewould it be like dividing by 0?
05:39.41random_mikeI cant test this current, as our SIP server which trunks calls to my Asterisk box is not correctly sending the invites. This is under investigation by our vendor atm.
05:44.14random_mikecan anyone assist?
05:45.16Corydon76-digrandom_mike: nope, just blank
05:45.23random_mikethanks Corydon76-dig
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05:45.48Corydon76-digThere should not be anything that you can do in the dialplan to make Asterisk crash, short of consuming all memory
05:46.20random_mikeYes, but then again, I have all sorts of magical powers!
05:46.30random_mikeEspecially crashing things that shouldnt crash :P
05:46.38Corydon76-digand there's a few other things, like turning off safety checks in Macro, that enables you to overflow the stack
05:47.27Corydon76-digbut really, you shouldn't be able to crash Asterisk from the dialplan
05:47.37random_mikethat's reassuring :)
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05:48.11Corydon76-digDividing by zero might do it, though.  Dunno if there's a safety check for that in the expression parser or not
05:48.47*** part/#asterisk sjobeck_ (n=sjobeck@valdisere.sjobeck.com)
05:49.26Corydon76-digYep, there is.  Division by zero returns INT_MAX.
05:50.01random_mikeBrilliant!
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06:47.33sun28moin \o/
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06:55.27ChannelZyawns
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07:04.28CheBuzz_HomeIs there a way to specify what verbosity asterisk should use instead of issuing "core set verbose 3" every time on the CLI?
07:05.14reptileschange the default in the source and recompile?
07:05.19reptiles8^)
07:06.00ChannelZwell you can run asterisk initially with many v's if you want it always to be like that
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07:06.12ChannelZIf you are reconnecting to the console, you can just do it there to:   asterisk -rvvvvv
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07:32.14ChannelZdoes the safety dance
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08:54.46Polysicshello
08:55.04Polysicshow do i know if * is registering correctly with a provider? incoming calls only
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08:59.23Polysicsplease? :-)
09:01.14angryuser_Polysics, 'sip show registry"
09:01.30Polysicsnice, thanks
09:05.37Polysicsone more thing... what would be a quick and dirty way to hook up the number to a SIP account that is already registered?
09:05.47Polysicsfor an incoming call
09:05.54Polysicsdo i need to use the "s" extension?
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09:11.22Polysicsok, * is registering correctly with te provider, and i have a simple s extension configured, yet calling the number gets me an "user is unreachable" message from the provider
09:11.31Polysicsis there anything i can do in * to diagnose the problem?
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09:18.09Polysicsi can't tell if it is something about * or the provider
09:19.17Polysicsi suppose that since i hear the message from the provider, it's the provider
09:19.20Polysicsany pointers?
09:25.55*** join/#asterisk tigerdyr2 (n=bach@217.195.185.143)
09:26.13tigerdyr2Hi all. Nice to be here :-)
09:32.33*** join/#asterisk DelphiWorld (n=Miranda@41.110.0.199)
09:32.37DelphiWorldhi
09:32.52DelphiWorldany asterisk business channel? i know that this is a support channel... no business talk here
09:33.55tigerdyr2Im looking for a way, to easy disable the MOH function in an asterisk system. All things i could find relates to the speciffic dial command or user. Any hints ?
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09:52.59klochanhi, is there some way to use followme app using sip/realtime-mysql? =)
09:53.15fiddurHi.  I have a problem using my new sip trunk; it's the first time for me where login-name differs from the number I'm calling from.  I have set it up with fromuser = 0317195500 and authname = MyLogin, but in the INVITES asterisk sends, the authname isn't used.  I've tried with auth = MyLogin@trunkserver  but still nothing mentioned of "MyLogin" in the sent INVITE, and the trunk server responds with SIP/2.0 401 Unauthorized...   Anyone versed in sip that
09:55.20fiddurBtw, I'm using 1.6.1.4...
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09:57.05TSMwhat is the best way to dial people into a conference room
09:57.27TSMi want someone to call into the system and then go into the room, but they need to be able to pull people into the room
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10:17.26oktayhi
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10:37.34benngardwhat's the default value for "x" in allow=alaw:x statement, i use :40 in my ooh323.conf but cant remember why :( think it was left when may and i did som ooh323 tests...
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10:43.47Polysicshello
10:43.56Polysicshas anyone ever used red5phone with their asterisk?
10:44.08Polysicsi have a weird problem and can't seem to solve it
10:44.20Polysicscalls go through but only stay up for 2 seconds
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11:39.18benngarddamn, i did receive the fax but asterisk crashed :(
11:40.33benngardbest attempt so far
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12:02.59benngardguys, can u give me some help plz... i try to receive fax, (i actually receives the fax) but i stop a step 4 in dialplan, look here:
12:03.08benngardexten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff)
12:03.08benngardexten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf)
12:03.08benngardexten => 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf magnus.b@inputinterior.se < /dev/null)
12:03.23benngard<PROTECTED>
12:03.23benngard<PROTECTED>
12:03.23benngardsip*CLI>
12:03.39benngardwhy am i not executing step 5?
12:04.07benngardif i run the system command from linux prompt, it send the fax to me as an attachment
12:12.23klochanmaybe someone can advise the best way to use followme with realtime sip-users? =)
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13:03.19DelphiWorldhi, do asterisk support H.248?
13:12.43coppicethe megabucks protocol
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13:13.06DelphiWorldcoppice: ;)
13:13.56DelphiWorld[TK]D-Fender: hi, do asterisk support h248?
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13:15.29Omorikahi guys
13:15.45Omorikais asterisk capable of detecting pressed pulse digits?
13:15.51Omorikain IVR? :D
13:16.23DelphiWorldOmorika: you mean dtmf?
13:16.37Omorikanope
13:16.43Omorikai mean pulse dialing
13:16.44Omorika:D
13:17.03coppicewow, someone still uses that? :-\
13:17.11Chainsawcoppice: The red phone in the office does.
13:17.22Chainsawcoppice: My FXS gateway port is set up for pulse dialling there. Works fine.
13:17.36Chainsawcoppice: http://www.vroon.org/ringring.jpg
13:17.41[TK]D-FenderDelphiWorld: What does "core show codecs" or "show translation" tell you?
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13:17.53DelphiWorldOmorika: lol, got it/................
13:18.14DelphiWorld[TK]D-Fender: i can't read it
13:18.29Omorikaso... is there a way to detect if some1 turns the dial?
13:18.34coppiceChainsaw: that's an old BT phone, isn't it?
13:18.35Omorikaon that phone
13:18.44Chainsawcoppice: Correct :)
13:18.47Omorikalike "press 0 for menu"
13:18.51Omorikaand u turn 0 :)
13:19.03[TK]D-FenderDelphiWorld: Because?
13:19.33DelphiWorld[TK]D-Fender: console is plin of text... and my screen reader can read only 1 line at time and can't navigate troug it
13:20.40coppice[TK]D-Fender: how would show codecs help him?
13:21.17[TK]D-FenderAh, protocol.
13:21.23[TK]D-FenderDelphiWorld: Sure looks like mgcp to me.
13:21.45[TK]D-FenderDelphiWorld: For which yes, * does, but only as a central, not as an endpoint
13:21.59coppicedoes asterisk support MGCP these days? it used to only support a tiny subset
13:21.59DelphiWorld[TK]D-Fender: hehehe;)
13:22.11[TK]D-Fendercoppice: Hasn't grown
13:22.17dmastg'morning
13:22.42DelphiWorld[TK]D-Fender: i want to use it a a media gateway controler, is it pocible?
13:22.45DelphiWorldfrom sip to MGCP
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13:24.34[TK]D-FenderDelphiWorld: Again depends whats ont he OTHER side of that MGCP connection
13:26.39DelphiWorld[TK]D-Fender: user dial from sip to asterisk, asterisk pass / contril it troug the media gateway, media gateway to pstn
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13:40.14hlueseaanyone know the hotdesk pins are the sip configured secret ?
13:40.14[TK]D-Fendergrrr
13:40.15[TK]D-Fenderhluesea: huh?
13:40.17hlueseai am trying to configure hotdesk feautera and i am wondering to hotdesk users pin ? where can i configure the pins for agents ?
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13:40.17[TK]D-Fenderhluesea: What agents?
13:40.17[TK]D-Fenderhluesea: Configured where?
13:40.28DelphiWorld[TK]D-Fender: http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/VoIP/Q_22882892.html
13:40.28hlueseamy used agents mean phone users using hot desking.
13:40.28DelphiWorld[TK]D-Fender: is mgcp... no h248 is difere;)
13:40.28hlueseamy real question is where i can configure the hot desking users pin ?
13:40.28[TK]D-Fenderhluesea: What PIN?  Where in * do you configure PINs?
13:40.40[TK]D-Fenderhluesea: There is no such thing as "hot-desking".  It is a concept you have to implement
13:41.05[TK]D-Fenderhluesea: What you use to auth state-changes for this is upt o you
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13:43.11Polysicshello
13:43.19Polysicsanyone is using red5phone or some web phone?
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13:55.25benngardSendFAX(filename[,a]) - Send a given TIFF file to the channel as a FAX. how do u define/set the channel?
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13:58.42[TK]D-Fenderbenngard: What channel?
13:58.59[TK]D-Fenderbenngard: You're IN ONE when you call it.
14:00.08benngardoki, think i got u
14:00.21dmastAre there any good CDR reporting apps for * out there that support MSSQL?
14:00.49Gido-Emssql /me is going to bark
14:02.57benngardif i have a fax attached to an ata (with extension 033211101), then exten => 033211001,1,SendFAX(test.tiff) will convert test.tiff to a fax an send it to the ata/fax?
14:05.07Amorsenbenngard: Nope. You need to Dial() first.
14:05.27AmorsenOr am I confusing myself?
14:05.32Gido-Eyep, or use callfiles.
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14:05.49AmorsenSorry, I'm just causing confusion. Forget what I said.
14:06.53[TK]D-Fenderbenngard: You don't seem to be thinking about where the call is starting from.
14:07.19Holosdmast: Are you writing directly into mssql or parsing the logs through an insert script?
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14:11.31dmastHolos: Writing directly to SQL
14:13.09dmastI'm not going to die on the mssql hill, either... I would just rather not put up another db server if it can be avoided.
14:13.38Polysicsagh, this web-based phone thing is killing me
14:13.51Polysicsno one is using red5phone or another web-based phone?
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14:14.11[TK]D-FenderPolysics: Why would we?
14:14.30Polysics[TK]D-Fender, maybe your boss wants one like mine does :-)
14:14.37[TK]D-FenderPolysics: Only the smallest subset of * users could really care for such a thing.
14:14.54Polysics[TK]D-Fender, i was hoping to meet someone from such subset here :-)
14:14.58[TK]D-FenderPolysics: So whats his reason to?
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14:16.02Polysicsproject is to have translators provide services by voice, and the potential callers often have network/system setups where they can't use/install a softphone
14:16.41PolysicsLeif Madsen himself? sign my book :-)
14:16.49leifmadsen:)
14:17.03Polysics[TK]D-Fender, and the reasoning was, let's look for a browser-based alternative
14:17.24leifmadsenleifmadsen: if you check my website, you can select the signature you want. Signatures range from $5-$100 plus shipping and handling.
14:17.25[TK]D-FenderPolysics: Again, what makes this the preferred solution?
14:17.39[TK]D-FenderPolysics: Sorry, just got the "why".
14:17.47[TK]D-FenderPolysics: Good luck with that :)
14:17.54leifmadsenerrr.... s/leifmadsen/Polysics/
14:18.29coppiceleifmadsen: does that include you providing a cheque from your current account for the signature to go on?
14:18.39Polysics[TK]D-Fender, i don't know if it is a good "why", but it does not look that bad of an idea
14:18.49Polysicstoo bad for it being nigh-unimplementable
14:18.56leifmadsencoppice: that's why I don't sign cheques, because they tend to end up on eBay
14:19.08Polysicsdue to ANY clients i found having some sort of problem
14:19.48[TK]D-FenderPolysics: have you considered having people CALL in with...oh I dunno.... PHONES?
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14:20.28Polysics[TK]D-Fender, what you are saying SOUNDS silly, but in fact, it is not... what can i say :-)
14:20.45leifmadsenwhat was the original question?
14:21.06[TK]D-Fenderleifmadsen: he's got woes with 2 web-soft-phones
14:21.12Polysicsif anyone knows about a browser-based SIP client that actually works
14:21.28Polysicspossibly open source
14:21.43Polysicsthat's the skeleton of the question, then there are some others that are related
14:22.05Polysicsred5phone works... for 2 seconds each call, it's a reported and not solved issue
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14:22.45leifmadsenPolysics: have you looked at the web phone from Zoiper?
14:22.56leifmadsenI haven't tried it, but Zoiper in general is a very good softphone
14:23.04leifmadsenand they have a web phone now too
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14:23.34Polysicswow
14:23.41Polysicssee? i knew that someone would know
14:23.50Polysicswhat i find interesting is the "no interface at all" option
14:24.01Polysicsso i can brand and integrate it whenever
14:28.26russellbcheck with phonefromhere.com, too
14:28.57russellbif SIP isn't a hard requirement, anyway, as I believe their main product is IAX2 based (which is a better choice for a browser client IMO)
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14:29.21leifmadsenand the Zoiper softphone will likely do IAX2 as well, so now you have 2 options :)
14:29.45russellbyup, and good people behind both of those options
14:30.44leifmadsenhttp://moziax.mozdev.org/  <-- never used, but a third option
14:30.44leifmadsendamn you net split!
14:30.44russellbleifmadsen: fail
14:30.44leifmadsenrussellb: your fail
14:30.44leifmadsenrussellb: and your face
14:30.44russellbyour face's fail
14:30.44russellbdamn
14:30.44leifmadsenHA!
14:30.44leifmadsenI need a beer
14:30.44russellbsame here
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14:43.58Polysicsi suppose it's because the other option is cumbersome stuff like the Flash thing i've beent testing
14:43.58leifmadsenok, so freenode is useless again -- time to go work on something else
14:43.58Polysicsleifmadsen, the fact that i have no idea how to configure IAX2 access and SIP works is a problem :-)
14:43.59leifmadsenPolysics: you configure IAX2 just like SIP in Asterisk -- and IAX2 only uses a single port
14:43.59leifmadsenPolysics: but if SIP works, then enjoy :)
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14:44.49[TK]D-FenderPolysics: Skype web client < -
14:44.49leifmadsenthe options are too numerous!
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14:44.50phasipI don't think that's what I want, I want something like a java application that translates calls to 222 to 5551123 and then sends the tones 222 so that the asterisk server on 5551123 calls it's internal number 222
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14:44.51[TK]D-Fenderphasip: huh?
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14:44.51phasipSo that i could use a cellphone as a local extension
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14:46.58phasipI have a cellphone number 777333, then I set something like GotoIf($[${CALLERID(num)} = 777333]?local) and in local i have something like WaitExten(10), but instead of having to type 777333 on my phone, wait for asterisk to answer and then type 222 i want a java application to do this for me so I only have to type 222
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14:47.42Polysicsok, while i mull over the web client options, how does one troubleshoot a SIP provider connection?
14:47.53Polysicsregistration is ok (sip show registry shows it)
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14:48.20Polysicsbut i get the "customer is unreachable" message from the provider (not from my *)
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14:48.52Polysicsmight be a problem with the firewall?
14:48.58[TK]D-Fenderphasip: You are missing part of your description.  Just because you say you have a cell phone doesn't tell us where the call STARTS, what actions you want * to take, and what you what it to do from there
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14:49.35[TK]D-FenderPolysics: You've given no description as to what that is a response TO.
14:49.58Polysicsto dialing the number from a PSTN phone
14:50.12[TK]D-FenderGo look in SIP debug
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14:52.05phasip[TK]D-Fender: What i want is a cellphone application that instead of dialing 222 on the phone network it will dial my home number and then make asterisk understand I want to dial a local extension, just as if it was one of the sip phones connected to the internal network.
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14:52.42Polysics[TK]D-Fender, SIP debug doesn't budge, which i equate to something else not working
14:53.07[TK]D-Fenderphasip: Ok, so far this is nothing VoIP related, or * related.
14:53.26[TK]D-FenderPolysics: Bad firewall, registeration, etc
14:53.34[TK]D-Fender~sipnat
14:53.35infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:53.36[TK]D-Fender^^^^^^^^^
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14:53.57Polysicschan_sip.c:19043 handle_request_invite: Call from '' to extension '0159526021' rejected because extension not found
14:54.09Polysicsmight it simply be a bad extension?
14:54.31[TK]D-Fenderphasip: Just "I want a generic java app for a phone platform I didn't even bother to specify to interpret my dialed numbers and script locally on the phone"
14:54.46Polysicsi have an "s" extension defined, but apparently i get the number
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14:54.49[TK]D-FenderPolysics: How much more does it have to tell you right to your face? :p
14:54.58[TK]D-FenderPolysics: "s" is not magic.
14:55.05[TK]D-FenderPolysics: Nor a catch-all
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14:55.10[TK]D-Fender~stdextens
14:55.11infobot[~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.
14:55.11Polysicsneither are my * skills, it seems :-)
14:55.32[TK]D-FenderPolysics: Sure they are... they've disappeared ;)
14:55.42[TK]D-Fenderpwned
14:55.49Polysicsyou win :-)
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14:56.14Polysicsbut someday you will need something about PHP, Ruby, CSS or JS, then i'll get my revenge .-)
14:57.00phasip[TK]D-Fender: I am sorry, i don't understand what you ment by that, but yes what I am looking for is a generic java app that works for most java enabled phones. I was hoping that someone here had used something like that since it would be a very powerful thing to combine with asteirsk. I am sorry if I upset you.
14:57.26phasipasterisk*
14:57.31[TK]D-Fenderphasip: As an attempt to be lazy this one falls short or being worthwhile.
14:57.53Polysicsphasip, i would say that is not much very related to *, but as i said my skills with * aren't that great :-)
14:58.12[TK]D-Fenderphasip: Most phones have 1-though dialing capability to which you could point to a number to reach your *.  So basically its an app to remove dialing a single digit (or quick sequence)
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15:00.11phasip[TK]D-Fender: That's a solution I haven't tought about, I look if it'll work, I hope you don't think I came here before trying to search for all terms and combinations I found relevant to this on google.
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15:01.56[TK]D-Fenderphasip: There is no "term" for this.  You gave no indication on the phone platform, made an assertion that java can actually cause the phone to dialout like that.  That it would also be a RUNNING app all the time (otherwise what's the convenience?), and provided all this as a BROKEN description.
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15:04.20[TK]D-Fenderphasip: On my HTC Touch phone I grab my phone, swipe my thumb up, and press on the middle button of a 3x3 grid and it dials my home answering with a menu and awaiting input. Swipe + push.  Is that worth replacing with a Java app that I'd have to leave running all the time?  What about exiting it so I could do something else?
15:04.31fogrideranybody here worked with superfecta CID?
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15:04.55[TK]D-Fenderfogrider: Wrong channel ---> #freepbx
15:04.57[TK]D-Fender~freepbx
15:04.58infobot[~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, or fpbx
15:05.12fogriderok thanks
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15:07.18MAbbasHi everyone, my WinSip generated sip calls are disconnected by Asterisk every 20 sec.
15:07.19MAbbashttp://www.voip-info.org/boards/index.php?t=19148
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15:08.26[TK]D-FenderMAbbas: You've provided NO details about where these 2 systems are relative to each other, provided CLI & SIP debug or much else.
15:09.07MAbbasThese two system lie in the same network ..
15:09.18Polysicstricky but interesting question: is the ringing sound ITSELF part of the Dial() app, or do i need to call something else?
15:09.24[TK]D-FenderMAbbas: Where is the failed call to look at?
15:09.39[TK]D-FenderPolysics: Depends
15:09.48MAbbaslet me dump the sip debug .. if its not against forum rules?
15:09.50Polysicsthe call from cellphone to * goes through fine, only the caller does not hear any ringing sound till it is answered
15:09.59PolysicsMAbbas, pastebin and we'll love you mroe
15:10.02Polysics*more
15:10.10[TK]D-FenderMAbbas: PASTEBIN <_  wtf are are you bringing forums up in here for?
15:10.22[TK]D-FenderMAbbas: If you want our help just PB stuff like normal
15:10.57[TK]D-FenderPolysics: I also don't see a call I can help debug.
15:11.06Polysicsthere is nothing to debug
15:11.09Polysicsno errors at all
15:11.13[TK]D-FenderPolysics: And I'm quite certain there is no call going from cellphone to cellphone
15:11.28[TK]D-FenderPolysics: it has issues.  I'd call that something to "debug"
15:11.30MAbbasI have updated the message with SIP debug, can you chk there?
15:11.34Polysics[TK]D-Fender, not from cellphone to cellphone
15:11.48Polysicsi am simply dialing a dial-in number i got from my provider
15:12.00Polysicswhich is hooked up to a simple Dial() extension
15:12.12Polysicsand EVERYTHING works, but the caller does not hear ringing sound
15:12.28[TK]D-FenderPolysics: and I still don't see the call.
15:12.45[TK]D-FenderMAbbas: Retransmitting #3 (no NAT) to 10.110.32.44:5161:  <--- packets aren't getting back.  check your FIREWALLS
15:13.19MAbbas<[TK]D-Fender>: I have updated the sip debug in http://www.voip-info.org/boards/index.php?t=19148
15:13.57[TK]D-FenderMAbbas: Your other side is likely FIREWALLED
15:14.28MAbbasWinSip is generating calls on Windows, which is not firewalled ..
15:18.26Polysicsone thing i have always wondered: is there a way to log all and only one call's SIP debug?
15:18.33[TK]D-FenderMAbbas: Whatever the reason, packets aren't getting answered by their side
15:18.49[TK]D-FenderPolysics: One call's?  No.  1 IP's, sure
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15:19.56donnibhi
15:20.17donnibhave a stupid question :) What's the difference between CLI termination and non-CLI termination ?
15:20.39donnibis it callerid where the call comes from ?
15:20.58donnibi mean when it's CLI termination the callerid get's sent to the end caller ?
15:21.31[TK]D-Fenderdonnib: that makes no sense.
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15:21.47donnib[TK]D-Fender: you tell me :)
15:22.00[TK]D-Fenderdonnib: No, you come up with a better phrased question.
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15:22.38donnibi just looked up some prices for VoIP termination in Romania and the pricelist has two difference prices. One CLI Terminaion and one non-CLI so i am asking what's the difference ?
15:22.52MAbbas<[TK]D-Fender>: I have tried without WinSip, using SipPhone and its works fine .. "BTW same environment"
15:23.22donnib[TK]D-Fender: http://bit.ly/7B3O7L
15:23.24cuscohi
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15:24.01[TK]D-FenderMAbbas: Maybe their software is broken.  Either way, its not *
15:24.36MAbbas<[TK]D-Fender>: Any other tools you might have in mind that will do bulk call generation like WinSip?
15:25.12Polysicsok, i was just missing the "r" option
15:25.21[TK]D-Fenderdonnib: Maybe you should ask THEM what their wording means.
15:25.57donnib[TK]D-Fender: ok i just thought that this is something i should have known before i talked with them. i thought this was a normal term.
15:26.32cuscoWe hae a prefix for outbounds like: _7878XXXXXXXXX and that Dial(DAHDI/g1/${EXTEN:4});
15:26.53[TK]D-Fenderdonnib: Acronyms thrown around here tend to be ENGLISH-origin ones.  Coming frmo a clearly non-english language page it could mean ANYTHING, and right now doesn't seem to say anything
15:27.06cuscoall good, now I would like to add a 78782XXXXXXXX because most landline numbers start with 2
15:27.17cuscotho there are some that strt with 3
15:27.33cuscoshould I add 78783XXXXXXXX and 78782XXXXXXXX separatelly?
15:27.50cuscoor can I do it any other way?
15:28.05[TK]D-Fendercusco: Whats wrong with the X instead?
15:28.23donnib[TK]D-Fender: ok thanx
15:28.25cusco[TK]D-Fender: they can dial mobile numbers
15:28.31cuscoand we want to prevent that
15:28.41cuscomobilenumbers start with a 9
15:28.44[TK]D-Fendercusco: 78782 + 8 X ---vs---- 7878 + 9X
15:28.52[TK]D-Fendercusco: What is the processing difference?
15:29.25keith4so... polycom phones come with wall brackets. right?
15:29.31cusco78782 the number start with a 2 + 8 digits
15:29.40Naikrovekkeith4: yes
15:29.46cuscoall phone numbers have 9 digits in here
15:29.51cuscoincluding mobile
15:29.54Naikrovekmy 321s did, anyway
15:30.02cuscofixed numbers start either with a 2 or with a 3
15:30.02keith4and they're designed to hang on those faceplates with the studs for standard phones, right?
15:30.42cuscoso if operator dials on his softphone 78789XXXXXXXX it won't dial... or somehting
15:30.43keith4but... i can't find anyone that sells keystone faceplates with phone studs
15:30.55[TK]D-Fenderkeith4: many are inhreently wall-mountable
15:31.01cuscoelse I need an IF evaluating if the number start with a 9
15:31.17cuscoI thought there would be a way to do 78782-3XXXXXXXX
15:31.19cuscoor soemthing
15:31.32[TK]D-Fendercusco: there is.... go read your chaper on dialplan patterns again
15:31.36keith4[TK]D-Fender: yah, looks like the 320 is. as is its replacement
15:32.13[TK]D-Fendercusco: exten => _78789[23]XXXXXXXX,1,Dial(BLAH)
15:32.23cuscothanks
15:32.53keith4[TK]D-Fender: but it seems like nobody makes a faceplate with phone studs that takes a regular keystone
15:33.02keith4so what good is the wall bracket?
15:33.28[TK]D-FenderkeyPardon?  I'd wall-mounted IP 301's, 430's, etc
15:33.35[TK]D-Fenderkeith4: Pardon?  I'd wall-mounted IP 301's, 430's, etc
15:33.57keith4right. i want to mount them on the wall boxes
15:34.01MAbbasAny tools to generate SIP calls other than WinSip?
15:34.04[TK]D-Fenderkeith4: keith4 and I have faceplates with them.
15:34.12keith4[TK]D-Fender: where did you get them?
15:34.27[TK]D-Fenderkeith4: Local interconnector who did this place's wiring
15:34.29Naikrovekhttp://qdb.us/301260
15:34.34[TK]D-Fenderkeith4: this is common stuff...
15:34.42Naikrovekyeah
15:34.49keith4[TK]D-Fender: that's what i thought! but none of the usual places that i get stuff from have them
15:34.51[TK]D-FenderNaikrovek: 403
15:34.56keith4[TK]D-Fender: something like this? http://www.twacomm.com/catalog/dept_id_1138/model_IC107FWP.htm
15:35.03[TK]D-Fenderkeith4: Maybe your usualy places suck :p
15:35.23[TK]D-Fenderkeith4: Looks like.
15:35.34keith4[TK]D-Fender: but $2.77 each is armed robbery
15:36.56[TK]D-Fenderkeith4: http://tinyurl.com/sulfm
15:37.05keith4nice
15:37.13[TK]D-Fender:D
15:37.20keith4for "common stuff", even google can't find 'em
15:38.12keith4... and the Polycom documentation suggests wall anchors for wall mounting
15:41.32phasipexit
15:41.52*** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel)
15:42.24[TK]D-Fenderkeith4: http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=348893&CatId=745
15:42.36[TK]D-Fenderkeith4: No, $100 is robbery.  $2.77 looks like a BARGAIN
15:43.03keith4[TK]D-Fender: how would I wall mount on that plate?
15:43.42keith4http://www.monoprice.com/products/product.asp?c_id=105&cp_id=10517&cs_id=1051703&p_id=6263&seq=1&format=2
15:43.51keith4I would say.... $0.40 is a bargain
15:44.12*** join/#asterisk Cain (n=Geek@unaffiliated/cain)
15:44.14keith4but no studs
15:44.23keith4or maybe I just need special "stud screws"? hmmm
15:44.40Naikrovekyou mount a polycom the same way you mount a traditional analog phone
15:44.55Naikrovekto go lowe's, find the mounting plate, install it, plop the phone on there
15:45.00Naikrovekdone
15:45.12keith4Naikrovek: right. but Lowe's doesn't have keystone mounting plates with phone studs
15:45.32Naikrovekwith phone studs? yes they do
15:45.39Naikroveki don't know wtf you mean by keystone
15:45.48keith4they have little kits that have RJ11 jacks, and face plates that screw to them
15:46.32keith4keystone: http://images.google.com/images?q=keystone jack
15:46.51Naikrovekmy lowe's has wall-mount phone plates, with studs, with holes for rj45 jacks and the metal studs that phones hang on
15:47.28keith4yah? where is it?
15:47.34Naikrovekand they have the jacks, too
15:47.41Naikrovekeast peoria, il
15:47.48keith4ooh, that's a bit far
15:47.53Naikrovekyeah i figured
15:48.07keith4man. their website's search function is absolutely worthless
15:48.17Naikrovekyou should be able to call any union electrician and find out where they get their stuff.  they do wiring like that for new offices, electricians do
15:48.20Naikrovekours do anyway
15:50.39*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:51.24keith4what the hell? CDW seems to have them... for $24 each?
15:51.30keith4http://www.cdw.com/shop/products/default.aspx?edc=1525596
15:51.36keith4that'd better be a 10 pack
15:51.38*** join/#asterisk jasonwoot (n=some@69.73.89.233)
15:52.03high-freqhas anyone successfully used fonosip to ipkall to your pbx and able to receive calls?
15:52.37*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
15:55.52*** join/#asterisk mbranca (n=matteo@host139-217-static.224-95-b.business.telecomitalia.it)
15:58.06*** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil)
15:58.38*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:58.38[TK]D-Fenderhigh-freq: Whats the point of having them together?
15:59.05high-freqsip connection to ipkall setup then ipkall gives number to use
15:59.24high-frequnless if i'm doing somethin wrong here
16:00.06[TK]D-Fenderhigh-freq: No, these are 2 separate services.  Why are you mentionging them TOGETHER?
16:00.24[TK]D-Fender[10:59]<high-freq>sip connection to ipkall setup then ipkall gives number to use <- this makes no mention of fonosip
16:00.53high-freqoooook..um ipkall website said to create a sip account then use that sip account registering it with ipkall
16:01.08*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
16:01.08*** mode/#asterisk [+o putnopvut] by ChanServ
16:01.10[TK]D-Fenderhigh-freq: No need.
16:01.18high-freqkk so just use ipkall settings then
16:01.34Polysicsis there such a feature as "emergency call" that would disconnect a busy SIP account and connect it to the entrant call?
16:01.54high-freqi'll try that then..hell i'm trying everything over here ;)
16:02.18Polysicsie. my 1000 user is talking, a call comes that has the precedence, he gets disconnected from what he's doing and connected to that call - with or without warnings
16:04.02*** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)
16:06.00high-freqnope i don't understand. ipkall setup is straight forward, has to have sip phone number, sip proxy, setup via once logged in.  This service uses SIP config which then ipkall gives out a free number for others to call so using this one service is not gonna work
16:09.53high-freqomg if i had hair i'd pull it out already lol this just aint right
16:10.02*** join/#asterisk dahl (i=dahl@4.84-234-222.customer.lyse.net)
16:10.05*** join/#asterisk fofware (n=chatzill@190.7.25.160)
16:14.47[TK]D-Fenderhigh-freq: number = exten on your system.  proxy = IP or hostname for your server
16:14.56[TK]D-Fenderhigh-freq: this ain't Raw-Cat Sigh Hence
16:15.17high-freqk i'll try that
16:16.10dahlis it possible to check if outgoing calls can be made every 5 minutes or so with a simple cronjob, just to see that things work properly?
16:16.27[TK]D-Fenderdahl: Sure
16:16.39*** join/#asterisk jo8330 (i=d04149c9@gateway/web/freenode/x-xmhqymmgiokbigzp)
16:16.45jo8330hello!
16:17.23*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:20.33*** join/#asterisk vitaminx (n=vitaminx@89.130.31.1)
16:25.00high-freq[TK]D-Fender, for the proxy do i put my external ip because i'm behind a vlan i setup and putting that ip in doesn't seem right
16:26.49chilicuilHi there, I'm totally new to asterisk and I'd like to know what're the softphone you're currently using, right now I've tried ekiga, SJPhone and Xlite but I'd know the opinion of the experts
16:27.29PolysicsXlite is good
16:27.33[TK]D-Fenderhigh-freq: Of course your external IP.
16:27.57[TK]D-FenderChithey all suck (as all softphones do).  Some suck a bit less than others
16:29.03high-freqk
16:29.34UQlevchilicuil: depends on protocols you want use SIP or IAX
16:29.57chilicuilI'm using currently only SIP, UQlev
16:30.37*** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net)
16:30.45UQlevchilicuil: for SIP you may have a way more options
16:32.35UQlevchilicuil: twinkle (for Linux) and Phonelite (for Win)
16:32.46*** join/#asterisk youngproguru (n=youngpro@74.10.229.45)
16:32.50UQlevPhonerlite
16:32.53*** join/#asterisk markwaters (n=mark@weloveit.info)
16:33.45chilicuilOk, I'm gonna give them a try, thx UQlev
16:36.01*** join/#asterisk manxpower (n=ewieling@226.sub-70-214-193.myvzw.com)
16:38.49*** join/#asterisk mrchrisadams (n=mrchrisa@87-194-125-43.bethere.co.uk)
16:40.01mrchrisadamshi guys - where would I find stats on how popular the various commercial and noncommercial asterisk based distros like SwitchVox, FreePBX, Trixbox et al are in relation to each other?
16:40.31bmoraca_workyou probably wont
16:40.48bmoraca_workis this for a study or is it because you're interested in getting in to asterisk?
16:40.53manxpowerThey will all claim they have the largest "market" share,
16:41.17manxpowermrchrisadams: just remember we don't support Asterisk distros or Asterisk GUIs here.
16:41.19mrchrisadamsbmoraca_work: I've been tasked with sorting out a decent phone setup for the company I work  for
16:41.39p3nguinmrchrisadams: Don't forget non-asterisk ones, such as FreeSWITCH.
16:41.48mrchrisadamswe're using a badly configured setup of elastix right now
16:42.06bmoraca_workmrchrisadams, that's all well and good.  asterisk is a very capable platform.  i would recommend vanilla asterisk.
16:42.12mrchrisadamsp3nguin - freewitch does interesting
16:42.28bmoraca_workmrchrisadams, you're going to be sadly disappointed if you think that switching platforms is going to solve your planning/configuration woes
16:42.29mrchrisadamsbut I'm nowhere familiar enough with voip to be able to dive into it
16:43.39*** join/#asterisk vitaminx (n=vitaminx@89.130.31.1)
16:43.41manxpowermrchrisadams: All Asterisk GUIs are "badly configured"  No way to avoid that when you let something else write your dialplan and configs
16:43.54NuggetThe best asterisk gui is gvim.
16:44.10mrchrisadamsNugget: try telling the receptionist that!
16:44.37p3nguinShe shouldn't be writing dialplans in the first place.
16:45.31Nuggetor even adding phones or extensions or anything like that.
16:45.34dmastIf she is, she needs more to do.
16:46.47*** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it)
16:47.04bmoraca_workthere's nothing wrong with letting users modify their own find-me-follow-me settings or call forward or things like that...but they only need a very minimal GUI (or even no GUI at all) to do that
16:48.26mrchrisadamsbmoraca_work: that's more in line with the use case I was thinking about
16:49.19*** part/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net)
16:51.32Polysicshmm
16:51.48Polysicswhat if a user is behind a firewall that does not let him connect on port 5060?
16:51.56*** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net)
16:51.57Polysicsi mean, he has outbound connnections blocked
16:52.05Polysicsnat=yes isn't enough, i'd say
16:52.20p3nguinThen you had better look into a VPN or something where he CAN connect out.
16:52.42p3nguinIs he allowed any ports at all?
16:52.48p3nguinYou can change which port to use.
16:53.37Polysicsbasically only port 80 and 113
16:53.43PolysicsSkype somehow works
16:53.55Polysicsi think i will have to look into a STUN proxy, do I?
16:55.12ChainsawPolysics: STUN would be your best chance. But if the outbound firewalling is too aggressive it might be intermittent it or might not work at all.
16:55.50Polysicsin case I decide to investigate STUN, what is the nmost widely used server?
16:55.54manxpowerAsterisk is a PBX, not a firewall busting voice system
16:55.56*** join/#asterisk fenrus (i=fenrus@oklart.com)
16:55.59fenruso/
16:57.14ChainsawPolysics: I believe ekiga has a STUN server or two. I remember stun.xten.com working for us.
16:57.21voipmonkadd a Damnit Jim in front, manxpower
16:57.30Polysicsso you mean i can use someone else's stun server?
16:57.40Polysicsi thought i had to install one :-)
16:57.42ChainsawPolysics: I've tried hosting my own, I could never get it to work.
16:57.59ChainsawPolysics: There's only one public codebase and it seemed... rather abandoned.
16:58.40high-freq[TK]D-Fender, thx for help, managed to make it ring lol but i picked up and it quit on me haha...just another thing to work on
16:59.09*** join/#asterisk daemon (n=paul@cpc1-linc11-2-0-cust594.12-1.cable.virginmedia.com)
16:59.14*** join/#asterisk mrchrisadams (n=mrchrisa@78-105-1-158.zone3.bethere.co.uk)
16:59.22daemonhey all im having a bit of a problem with asterisk, it seems that it is getting registration time outs from my provider
16:59.32daemonbut I cannot see anything wrong with my setup
16:59.39daemonI verified the username and password etc
16:59.41daemonand im not behind nat
16:59.49daemonapparently
17:00.00daemon[Jan 21 16:59:56] NOTICE[54443]: chan_sip.c:12997 handle_response_register: Failed to authenticate on REGISTER to '__SOMEUSER__@voiptalk.org' (Tries 3)
17:00.19daemonfollowed by: [Jan 21 17:00:16] NOTICE[54443]: chan_sip.c:7849 sip_reg_timeout:    -- Registration for 'USER@voiptalk.org' timed out, trying again (Attempt #24)
17:03.40manxpowerdaemon: set srvlookup=no in sip.conf [general] and "module reload chan_sip.so"
17:03.49*** join/#asterisk k4tana777 (n=roni@190.196.71.206)
17:04.04k4tana777hi everybody !
17:04.09daemonmanxpower, thanks ill give it a whirl
17:04.24p3nguin"sip reload" isn't enough?
17:04.46manxpowerPolysics: a STUN server must be OUTSIDE the network
17:04.49manxpowerp3nguin: it should be
17:04.51Kobazaww
17:04.54Kobazkatty's not here
17:04.56Kobazhttp://www.whowantsabalti.com/funpics/mild/squirrel-surprise.jpg
17:05.04p3nguinJust wondered if there was a difference.
17:05.06Naikroveklol
17:05.34daemonmanxpower, ah ha now im getting failure to authenticate no timeout though
17:05.43daemonstrange though I know im using the right username and password
17:05.43daemonhmm
17:06.09k4tana777somebody can give me a little help ? i need that a call, rings in all extensions that i have here ..
17:06.28manxpowerdaemon: srvlookup=yes is the default and your provider must have screwed up their SRV records.
17:06.40k4tana777i now that many of you gonna send me to google :) , but i just wanna know how can i look for that
17:06.52manxpowerk4tana777: Dial(SIP/123&SIP/124&SIP/125)
17:07.03k4tana777really ?
17:07.04manxpoweryou don't dial extensions, you dial DEVICES
17:07.06daemonmanxpower, actually just got reg timeout
17:07.10k4tana777so simple ?
17:07.21manxpowerk4tana777: read the Asterisk book
17:07.22manxpower~book
17:07.23infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
17:07.25k4tana777damn .. i gonna try that ..
17:07.29k4tana777thanks
17:08.05manxpowerdaemon: timeout means "no response from server, not even a "sod off" message"
17:08.23*** join/#asterisk paulc (n=paulc@unaffiliated/paulc)
17:08.36*** join/#asterisk TimeRider (n=steve@78.32.26.1)
17:08.36manxpowerWhich means 1) provider is dead 2) your firewall is blocking 3) you put in the wrong server host/IP
17:08.50daemonmanxpower, logged into it via telnet not a single thing in the way
17:08.50Nuggettelnet is eeeeeeevil!
17:09.01*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
17:10.18manxpowerdaemon: you can't connect to a sip server over telnet
17:10.34Kobaznc -u host 5060
17:10.36daemonmanxpower, my fault I meant I could connect to their http port over telnet, so the host is up
17:10.46manxpowerSIP (normally) uses UDP.  Telnet uses TCP.  You did not actually test ANYTHING when you try telnetting to a SIP port
17:10.46daemonI dont have anything to conenct via udp, unless I quickly hack something together in perl
17:10.54Kobazsip is generally udp, but sip tcp is coming along in asterisk slowly i heat
17:11.00Kobazdaemon: netcat
17:11.06Kobazheat/hear
17:11.10Naikroveksip over tcp is in 1.6 and up
17:11.20Naikrovek1.6.0 i should say
17:11.23manxpowerdaemon: remember ITSP provided configs are almost ALWAYS wrong.
17:11.25daemoncool ill try that
17:11.46Kobazheh
17:12.07daemonmanxpower, yeah I tried there config, it did not work so I installed the ajax api
17:12.18daemonadderd it in my self, getting same problem though
17:13.23daemonfixed it
17:13.30*** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50)
17:13.34manxpowerwhat the hell does an ajax API have to do with registration?
17:13.49Naikrovekwow ajax
17:13.54daemonit suddenly occured to me, that if I connect to my mail server I use mail/pop/smtp . host, so I wondered why they was using 'voiptalk.org' as the host
17:14.00daemonI changed it to sip.voiptalk.org and wallah
17:14.01daemonit works
17:14.06daemonI should really email them about that
17:14.09Naikroveks/wallah/viola/
17:14.21comradeb14cksup
17:14.32QwellNaikrovek: viola is a musical instrument
17:14.38Naikrovekit is
17:15.02Naikrovekit's also how you spell the word that sounds like "wallah"
17:15.13QwellNo it's not
17:15.18Naikrovekthen tell us
17:15.20Naikrovekhow do you spell it
17:15.23Qwellvoilà
17:15.35Naikrovekwell aren't you just the biggest nitpicker of all
17:15.47Naikrovekmy keyboard doesn't have that, and &agrave; doesn't work
17:15.49Qwellremove the accented a.  It's still different.
17:15.53beekapplauds Qwell's fast fingers.
17:16.07Naikrovekor &aacute;
17:16.15Qwelloi != io
17:16.26Naikrovekfine
17:16.30k4tana777<manxpower> thanks man ... works perfect... and was so easy ..
17:16.35Naikrovekyou could just correct me instead of being a dick
17:16.44QwellI did correct you.
17:16.46Naikrovekbut the @ infront of your name apparently gives you any rights you want
17:16.55*** mode/#asterisk [-o Qwell] by Qwell
17:16.56Qwellclearly
17:17.05*** join/#asterisk mrchrisadams_ (n=mrchrisa@78-105-1-158.zone3.bethere.co.uk)
17:17.08Qwellno @, and I'm still merely correcting you.
17:17.13Naikrovekokay
17:17.28Naikrovekeveryone is so fucking fast to correct me
17:17.33manxpowerNaikrovek: You are correct.  @ means he can do anything he wants.  Deal with it.
17:17.37Naikrovekbut never anyone else
17:18.38manxpowerNaikrovek: many people here have the ability to get ops (@) any time they want,  Most of them don't unless there is a need.
17:23.32QwellNaikrovek: I seem to recall you correcting somebody (incorrectly..).  Saying I'm fast to correct you isn't really going to fly.
17:23.59Qwelland seriously, you're getting upset over this?
17:30.23comradeb14ckdrama llamas
17:30.25comradeb14ck:s
17:34.00*** join/#asterisk Yedidya (n=chatzill@host86-142-22-34.range86-142.btcentralplus.com)
17:34.38drmessanoI dare say he's the one showing bias against someone for being an OP
17:34.51drmessanoLike Qwell can't correct someone or have an opinion because of the @?
17:34.54drmessanoHello?
17:35.00hardwirethis thing isn't on.
17:35.04hardwiref.y.i.
17:37.49*** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net)
17:41.00EmleyMoorhas finally added a feature his partner has been asking for to the dialplan, without compromising existing facilities
17:41.38*** join/#asterisk carrar (i=tim@osburn.com)
17:42.47comradeb14ckQuestion: How many good asterisk developers do you think are out there in the wild?
17:43.05comradeb14ckAnd by good I mean someone who knows and can write dialplan, asterisk modules, agi, and ami code.
17:43.13QwellThat isn't an "Asterisk developer"
17:43.19Naikrovekright
17:43.21Naikrovekbut there are a lot
17:43.28Naikroveka lot more than you'll find in here i'd think
17:43.35comradeb14ckI don't really see many.
17:43.39*** join/#asterisk hluesea (n=hulusika@88.247.127.66)
17:43.42Qwellyou haven't looked
17:43.43Naikrovekdoesn't mean they're not there
17:43.45comradeb14ckI see a lot of people who work with Asterisk, but who aren't very good at it.
17:43.46carrarbecause they are too busy to be in here
17:43.47*** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com)
17:44.06Naikrovekthe really good people come in here once in a while with a specific, well worded question
17:44.27Naikrovekone guy came in a couple weeks back, had an asterisk system with 20,000 end points and 5,000 simultaneous calls
17:45.07Naikroveki asked him about all that then he left, and hasn't been back
17:45.21Qwellcomradeb14ck: "I've been searching around this rehab clinic for days, and I haven't found any upstanding citizens!"  It's a place people (usually..) only go when they have problems.
17:45.45Naikrovekif you hang out in here long enough you'll find many very very smart people running very very large, complex systems
17:45.45drmessanoAsterisk and or mental problems
17:45.46QwellI suspect you aren't on any of the Asterisk-related mailing lists
17:45.46comradeb14ckQwell, I didn't ask how many were in here. My question was: "How many good asterisk developers do you think are out there in the wild?"
17:45.58Naikrovekcomradeb14ck: many
17:45.58comradeb14ckAs in, everywhere.
17:45.59drmessano3,500
17:46.09comradeb14ckJust an opinion question =p
17:46.11Naikroveka number?  ... dunno.  but a lot
17:46.13QwellMy point is - you're looking in the wrong place.
17:46.27comradeb14ckI'm not looking anywhere, I asked for an opinion.
17:46.31Qwellwhether it be here on IRC, trixbox forums, etc
17:46.35comradeb14ckIt's just for my own curiosity.
17:46.39QwellYou said "I don't see a lot"
17:46.54drmessanoGive him a frickin number, @master
17:46.56comradeb14ckRight, that's why I was asking you!
17:47.01drmessano6,786
17:47.07comradeb14ckI figured some people in here may have seen more than myself =p
17:47.15Qwellcomradeb14ck: it's unknowable
17:47.35comradeb14ckIs it less than a million?
17:47.40Naikrovekprobably
17:47.45comradeb14ckIs it less than 999,999?
17:47.45drmessanoIts less than 6 billion
17:47.51seanbrightjust shy of
17:48.05comradeb14ckfair enough :)
17:48.06drmessanoId say 50,000, +/- 50,000
17:48.10Naikrovekheh
17:48.17Naikrovekthey're everywhere
17:48.24seanbrightjust pedophiles
17:48.30comradeb14ckit is always possible to approximate something. Regardless of whether or not it is correct =/
17:48.30seanbrightlike pedophiles
17:48.41seanbright'just' = 'like' in na'vi
17:48.44comradeb14ckIt's like getting the limits in calculus =p
17:48.50drmessanocomradeb14ck: Thats not approximating, thats inventing
17:49.11seanbright70% of statistics are made up
17:49.16seanbright35% of people know that
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17:51.55*** mode/#asterisk [+o Cresl1n] by ChanServ
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18:09.22*** mode/#asterisk [+o Qwell] by ChanServ
18:09.56FinboySlickAnyone has a good example for calling dialplan apps from a shell using asterisk -rx ?
18:10.30*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
18:10.44[TK]D-FenderFinboySlick: There is no such thing
18:11.30FinboySlick[TK]D-Fender: That settles it.  What should I look into as an alternative?  AGI?
18:12.05seanbrightasterisk -rx "originate SIP/1234 application MeetMe 1234,d"
18:13.12FinboySlickseanbright: Interesting, that should be able to do just what I want.
18:13.42seanbrightthat will call SIP/1234 and when it is answered send it into MeetMe
18:14.40[TK]D-FenderFinboySlick: make an extensions that answers, and plays silence for a while like 5 min or something
18:14.47[TK]D-FenderFinboySlick: And originate against it
18:14.58[TK]D-FenderFinboySlick: Thats as close as you'll get
18:15.09*** join/#asterisk benngard (n=benngard@213.88.138.230)
18:17.43*** join/#asterisk lanning (n=lanning@208.87.235.224)
18:17.52carrarTK got SERVED!
18:17.58carrar:)
18:17.59*** join/#asterisk ruben23 (n=AGENT@122.55.48.243)
18:19.02carrarscrambles away from the cluebat
18:19.54benngard#"&¤(/"&¤/&"(/
18:19.58benngardsorry
18:20.08bmoraca_workom nom nom...i ate your random characters
18:20.58benngardi hve been doing some tests with latests dahdi, spandsp and asterisk trunk
18:21.41benngardi can get fax to work, but not with the same settings i need for outgoing calls
18:22.01*** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com)
18:22.02benngardso i have to choice, either fax or voice :(
18:24.40stones_Anyone here needing a dedicated server?
18:25.14Qwellstones_: Please do not advertise here
18:25.37FinboySlick[TK]D-Fender: Given the lack of 'obvious' way to do what I'm trying to do, I'm tempted to consider that I'm not taking the right approach to my problem.  I'm trying to setup something that re-sets all the right agents into all the right queues every morning at 8am.
18:25.48*** join/#asterisk bsaxon (n=bsaxon@12.68.234.174)
18:26.08FinboySlick[TK]D-Fender: What would you suggest I look into for something like that?
18:27.09FinboySlickI was figuring cronjob calling AddQueueMember() apps
18:27.26seanbrightasterisk -rx  "queue remove member SIP/1157 from queue"
18:27.30seanbrightheh
18:27.40*** join/#asterisk korcan (n=kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net)
18:27.58*** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk)
18:28.42titter`Say you have _70XX in your dial plan doing one thing, but you need say _700[0-5] do something else ... what would be the best way to write that?
18:28.52Qwelltitter`: just like that
18:28.56FinboySlickseanbright: I'm starting to like you.
18:29.04*** join/#asterisk ruied (n=ruied@92.250.95.7)
18:29.13*** join/#asterisk toddejohnson (n=toddejoh@ppp-70-226-205-249.dsl.spfdil.ameritech.net)
18:29.15titter`When I dialout with that, it dials both since they both are in the same context
18:29.18[TK]D-FenderFinboySlick: Generate the dialplan to do all actions.  Issues a dialplan reload, then a single originate, or call-file to trigger the batch
18:29.27ruben23hi how do i setup h323 on asterisk, im currently using SIP.
18:29.37[TK]D-Fenderruben23: Go read the sample configs
18:29.53carrartitter: _70X[6-9]
18:30.20carrarerr
18:30.21carrarno
18:30.22carrarhaha
18:30.24titter`lol
18:30.27titter`bout to say
18:30.39carrarmust find coffee
18:31.34carrarjust create 5 extension 7000 - 7005
18:31.39carrarthey will match first
18:31.44carrarerr 6 extensions
18:31.54Qwellso would _700[0-5]
18:32.06*** join/#asterisk theron (n=theron@ip244.scolloc.lh.net)
18:32.13titter`yup
18:32.27titter`it shows first, but because they are in the same context, it dials both
18:32.35Qwellit doesn't dial both...
18:32.38[TK]D-Fendertitter`: DIALS BOTH?  PARDON?
18:32.51[TK]D-Fendertitter`: PAStebiN the failed call and your dialplan
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18:36.34titter`dp - http://pastebin.com/d67447949 cli - http://pastebin.com/d2d99efb7 - dp show in cli - http://pastebin.com/m18de54fb
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18:37.28titter`nvm, im an idiot ... i just realized what it did
18:37.57[TK]D-Fendertitter`: Yeah... COULD match... not WOULD.
18:38.03*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
18:38.07Kattyherroes.
18:38.10[TK]D-Fendertitter`: And clearly isn't executing anything extra
18:38.13[TK]D-FenderkettMew.
18:38.26[TK]D-FenderKatty: Even...
18:38.27Kattytab tail.
18:38.29Katty...
18:38.34Kattyi fail, too.
18:38.42*** join/#asterisk Alagar (n=Administ@122.164.18.195)
18:38.43titter`the firt two times i saw it in the cli someone else made a call in the 700x range, so i was like wtf is it doing ... it's been a long night.
18:38.56titter`see it's and its ...
18:39.49p3nguinlooks like you have correctly used "it's" in your sentence.
18:41.19titter`ROFL
18:41.33p3nguindoesn't get it
18:41.34titter`I give up.
18:42.30titter`needs a nap
18:42.55Kattyapplies pillow and blanket to titter`
18:43.15benngardguys/girls i ran into a strange problem today: was playing around with trukn and and faxing: i hade i dialplan like below
18:43.27benngardexten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf)
18:43.27benngardexten => 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf magnus.b@inputinterior.se < /dev/null)
18:43.48*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
18:44.09benngard101,4 was executed without error bout 101,5 didnt got executed
18:44.18benngardchanged to:
18:44.31benngardexten => 101,4,System(fax.sh)
18:44.55benngardtiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf
18:44.55benngardmutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf boras@inputinterior.se < /dev/null
18:45.23benngardin fax.sh and all starts to work...
18:45.24*** join/#asterisk Alagar (n=Administ@122.164.18.195)
18:45.56p3nguin< /dev/null is going to take input FROM /dev/null.  Is that really what you meant?
18:46.07benngardyes
18:46.13p3nguinokay, just making sure.
18:46.35Kattyyawns
18:46.43p3nguinthrows popcorn in
18:47.00Kattyso so so tired
18:47.09benngardon me?
18:47.22Kattyit's not always about you.
18:47.22p3nguinIt's one of those days, apparently.
18:47.28benngard:)
18:47.52ErestarDoes anyone else experience an issue when dialing out to cell phones where Answer() occurs way to early? If so, can someone point me to a place to resolve the issue?
18:47.52benngardhad ones of those days today :(
18:48.40Kattyi would think you'd be happy the cellphones answered promptly.
18:48.51p3nguinerestar: If you are dialing out, why would you have Answer() in your dialplan?
18:49.06Kattyp3nguin: call file maybe.
18:49.09ErestarThey're originated calls
18:49.21titter`benngard: I have had the most success with http://caspian.dotconf.net/menu/Software/SendEmail/ for sending e-mails via System() on my fax2email script
18:49.35Kattyoh, i guess i could link my fax page.
18:49.39ErestarWell, the issue is that I'm playing a sound file, and it starts playing before the call is actually answered
18:49.40Kattybut it's not on sending faxes.
18:49.57titter`Katty: link anyways, ill probably do that next
18:49.58dustybinis AGI the only way to execute shell commands from a dialplan in asterisk 1.6 ?
18:50.00p3nguinerestar: Originate it the other way around.
18:50.14benngardtitter`: yes, but why isnt 101,5 not even got executed
18:50.22p3nguinerestar: Make the cell phone answer before going to the dialplan.
18:50.52benngardwhen i moved both "system" commands to a shell script it worked
18:50.55Erestarp3nguin, Hmm. I'm afraid I'm not sure how to do that
18:50.57titter`benngard: pastebin the CLI of the failed attempt and somone could help
18:51.33Kattytitter`: http://42ndgeekstreet.blogspot.com/2009/11/asterisk-faxing.html
18:51.44p3nguinerestar: originate SIP/1234567@itsp.com extension 123@yourcontext
18:51.47ruben23hi
18:51.49p3nguinerestar: Something like this.
18:52.02benngardi cant send a fax now, i am at home, can just receive
18:52.31titter`Katty: thanks, I have had that bookmarked form my research in the past
18:52.38benngardbut i can do it tomorrow when i am back at office
18:52.47*** join/#asterisk lupine_85 (n=lupine_8@unaffiliated/lupine-85/x-7392152)
18:52.48Kattytitter`: orly, i wonder how you got your hands on it.
18:53.01titter`Katty: the magical power of google
18:53.13p3nguinerestar: In my experiences, the tech/data leg of the call must answer and be 'up' before extension will execute.
18:53.23Kattyohiseethenk
18:53.35Erestarp3nguin,  I believe that's what I'm doing... Originating on Channel DAHDI/1/5555555 with the extension 500 (which is where the sound file gets played)
18:53.45benngardi can do alot of things remotly, but i cant put a paper in a fax and press the buttons :(
18:53.54ErestarAnd where 5555555 is the cell phone number I'm calling
18:53.58titter`:>
18:54.01p3nguinerestar: Oh, dahdi... that's why it is behaving that way.
18:54.11Erestarp3nguin, ??
18:54.28p3nguinerestar: DAHDI channels go up as soon as the number is finished -- no waiting for an answer.
18:54.39Kattywatches the squirrels
18:54.47ErestarOh
18:54.50ErestarYuck
18:55.01p3nguinerestar: So consider some magic with WaitForSilence() or AMD.
18:55.03ErestarSo should I be waiting for silence or something?
18:55.19Kattywaiting for the awkward silence.
18:55.27Kattyit happens, you know
18:55.32[TK]D-Fender[13:54]<p3nguin>erestar: DAHDI channels go up as soon as the number is finished -- no waiting for an answer. <- many assumptions here
18:55.35Kattywhen the little old lady asks you if you've seen the tapicoa.
18:55.49p3nguinerestar: [tk]d-fender is willing to help you solve your issues.
18:55.54[TK]D-Fender:p
18:55.55Erestar:-p
18:56.13Kattyyou did it now fenrus
18:56.22Kattys/fenrus/ [TK]D-Fender
18:56.26titter`Katty: interesting blog ... asterisk advice and diet tips
18:56.27ErestarThe reason I thought it was only cell phones is because the last time I messed with asterisk back in the zapata days this only occured when calling cell phones
18:56.28ErestarI think
18:56.33[TK]D-FenderKatty: "Fenris"
18:56.34Kattytitter`: and product reviews.
18:56.41Katty[TK]D-Fender: i tried to tab complete fender
18:56.45Katty[TK]D-Fender: fenrus
18:56.46titter`Katty: yup, can't forget those
18:56.55Kattytitter`: what can i say.
18:56.59Kattytitter`: i have a wide range of interest.
18:57.17Kattytitter`: there's also 42ndrecipestreet
18:57.19[TK]D-FenderErestar: What card are you using?
18:57.46Erestar[TK]D-Fender, e159:0001 Wildcard TDM400P REV I
18:57.52titter`Katty: same, but my diet has consisted of lemon pepper chicken i make, and brown rice portioned into a 1:1 carb:protien diet
18:58.06titter`Katty: sonow your blog has made me hungry
18:59.02Kattytitter`: i was doing a 350-400 per meal and 100-150 on the snacks.
18:59.14Kattytitter`: but i've since just been doing 1500 a day
18:59.43[TK]D-FenderErestar: If you use "callprogress = yes" for your channels 8 will not consider them "answered" the moment theya re placed.  this runs the risk  however of your calls being randomly disconnected
19:00.28titter`Katty: I try not to count calories as much as concetrating on the 1:1 ratio per pound of weight ... tends to keep the calorie intake more random, but also makes you focus on clean food. can't accomplish that goal without eating clean.
19:00.31*** part/#asterisk toddejohnson (n=toddejoh@ppp-70-226-205-249.dsl.spfdil.ameritech.net)
19:00.37*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net)
19:00.50Kattytitter`: well i don't just calorie count, i also nutrient count.
19:00.59Erestar[TK]D-Fender, Thanks... I'm taking a look at that now
19:01.04Kattytitter`: g and mg of vitamins, minerals, etc.
19:02.31titter`Katty: that's good, good job on the goals
19:03.10Kattytitter`: gotta keep that girlish figure.
19:03.11Kattyi mean.
19:03.16Kattythat 50 year old man in the basement figure.
19:03.57titter`heh, i just try to avoid the typical "i.t. guy" figure
19:04.33titter`living in florida also motivates me ... can't be the fatass of the bunch when going to the beach
19:05.00Kattywhat's wrong with Roy and Moss?
19:05.35Kattyespecially moss.
19:06.26Erestar[TK]D-Fender, Thanks, that worked perfectly for the first test. Do you know what circumstances the random disconnects start happening? Is it when there are lots of concurrent calls or something?
19:06.53[TK]D-FenderErestar: No... * detects silence, etc.... lack of ringing, et
19:07.16titter`Katty: havent watched the show enough to say, watched one episode
19:08.30Kattytitter`: it's lovely.
19:09.09titter`Katty: ill have to watch more ...
19:09.19dustybinhow can one pass "${CALLERID}" into a shell script ?
19:09.40Kattyecho?
19:09.50Kattyand system()
19:10.02dustybinasterisk 1.6 doesnt have system
19:10.05dustybini only have AGI
19:10.14Katty1.6 doesn't have System()?
19:10.20dustybinnope :(
19:10.21Kattyi find that very hard to believe.
19:10.34dustybinill have another look :D
19:10.45Katty[TK]D-Fender: was System() taken out of 1.6?
19:11.25dustybinoh it worked!
19:11.28dustybinyou are correct!
19:11.29*** join/#asterisk daidoji (n=daidoji7@adsl-99-27-128-66.dsl.pltn13.sbcglobal.net)
19:11.31titter`I am using it now :< so if it was, news to me
19:11.34Katty[TK]D-Fender: nevermind.
19:11.41dustybinhugs Katty
19:12.22[TK]D-Fender:)
19:12.24dustybinnow its time to test this
19:12.25dustybinexten => s,1,System(/usr/local/bin/mythtvosd --template=cid --caller_number="${CALLERID(num)}" --caller_date="${STRFTIME(${EPOCH},%F)}" --caller_time="${STRFTIME(${EPOCH},%r}" --caller_name="${CALLERID(name)}")
19:12.35dustybinif this works, this will be the best thing ever
19:12.37KattySubject: Fire... Dear Sir/Madam, I am writing to inform you of a fire that has broken out at the premises.... wait, that's too formal.
19:13.49dustybinIT WORKS!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
19:13.59Erestardustybin, =)
19:14.00dustybinwhen somebody rings me, a OSD appears on mythtv
19:14.05fenrussexy
19:14.07dustybinfalls off chair
19:14.08ErestarOh nice
19:14.13fenrusdoes it pause the movie aswell ?
19:14.16fenrusor mute the tv ?
19:14.18dustybinnope
19:14.18*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
19:14.25fenrusturn on the lights ?
19:14.27fenrus=))
19:14.30dustybinheh
19:14.31*** join/#asterisk TimeRider (n=steve@78.32.26.1)
19:14.34benngardmake coffee
19:14.38dustybinim in shock
19:14.49fenrusthen you have work to do
19:14.50Kattybenngard: throw coffee.
19:14.51dustybinrings himself
19:14.52fenrus=)
19:15.00*** join/#asterisk Ad-Hoc (n=nimbus@62.1.172.179.dsl.dyn.forthnet.gr)
19:15.39benngardKatty: catch!
19:15.43Kattyi need to velcro a pager to my remote
19:15.51Kattyso i can find it when the dog runs off with it
19:15.55fenrushaha
19:15.56*** join/#asterisk minotaur01 (n=minotaur@24.215.3.50)
19:16.47*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:17.29beekKatty: don't your ferrets ever take it from you?
19:18.07Kattyno
19:18.41Kattyguess it's not interesting enough
19:18.42*** join/#asterisk minotaur01 (n=minotaur@24.215.3.50)
19:19.10titter`If my dog ran off with my remote, I would not be happy ...
19:19.28ErestarSo if you guys had to call a bunch of people and play a message, but you only had 4 analog lines to do it, how would you make it work?
19:19.36[TK]D-Fender[14:14]<benngard>make coffee <- Mine does
19:19.52benngardlucky u!
19:19.55fenrus=))
19:20.04KattyErestar: i guess the first question is why are we calling a bunch of people, and what is the nature of the message.
19:20.12titter`lol
19:20.23fenrusKatty, you don't like voice-spammers? :)
19:20.25ErestarKatty, Its for an opt-in alert system
19:20.39KattyErestar: now put that in english.
19:20.42dustybini spend my friends 5 days solid with .call_me
19:20.46dustybin*spammed
19:20.48*** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net)
19:20.49*** join/#asterisk stan (n=stan@87-194-216-82.bethere.co.uk)
19:21.03ErestarKatty, People can sign up to receive phone messages when their kid's school has a delay or is cancelled
19:21.17KattyErestar: okay.
19:21.25KattyErestar: how many people are we talking about here, roughly
19:21.31Erestarhttp://communitycrieronline.com/
19:21.40ErestarIn some cases, 15k ;)
19:21.54KattyErestar: and how often do you really call them
19:22.04ErestarWe don't know yet ;)
19:22.19KattyErestar: my personal suggestion would be txt messages, not phone calls.
19:22.20*** join/#asterisk minotaur01 (n=minotaur@24.215.3.50)
19:22.28ErestarWe do texts too
19:22.41ErestarI've already written a system using the manager API that monitors channels and what not to find available lines
19:22.47ErestarAnd places calls based on what's available
19:22.47KattyErestar: if you absolutely need phone calls, with audio, you need more than 4 analog lines
19:23.07ErestarI just had to ask to see if there was some built in way of doing it
19:23.17benngardthought about erlang?
19:23.26ErestarOh, in some cases we'll have 24 analog lines, and sometimes we'll be using a VoIP provider
19:23.32KattyYou mean a magically way to place a phone call without using a phone line?
19:23.48ErestarNo, no
19:23.59fenruslike a built in system queueing up outgoing calls ? :)
19:24.07Erestarfenrus, Yeah
19:24.07fenruswaiting for available lines
19:24.15Erestarfenrus, Exactly that
19:24.17Kattyi'm sure you can write one
19:24.21ErestarOk
19:24.22ErestarI did
19:24.26ErestarAnd it works
19:24.31fenrusi've not seen anything - but google a bit :)
19:24.32Kattyhoray!
19:24.38titter`Anyone know off the top of their head how to redirect a call different when a Polycom is forwarded? I see it gives a diversion ;reason=deflection, then places a new call to the number ... but I would like to set the callerid of that outbound call
19:24.56bmoraca_workErestar, unless you're prepared to pony up for the capacity to call 15K people, you might be best served outsourcing this (15,000 people at 24 people at a time, even if it's only 1 minute per call will take 10 hours to call)
19:25.06ErestarWe're testing it this weekend, but I'm completely new to Asterisk and phone systems, so I _had_ to check before we move to production just to make sure I didn't just waste a week ;)
19:25.08Kattytitter`: status
19:25.21titter`tahnky
19:25.22daemonhey guys, im using the asterisk 'ajax' gui, all is working excellently but non of my logs seem to be appearing
19:25.26titter`thanky*
19:25.36daemonthey are in the database, but nothing ever gets displayed in my browser
19:25.47Erestarbmoraca_work, I know it. I've got a spreadsheet that figures that stuff out. I was most asking about the out going queue system that fenrus was talking about
19:25.51Kattydaemon: we do not support gui's here. please visit whatever irc chanenl supports your box.
19:25.54ErestarEr, mostly*
19:26.29*** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
19:26.47*** join/#asterisk puzzled (n=patrick@535335AA.cable.casema.nl)
19:27.04*** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk)
19:27.21*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
19:27.26daemonKatty, asterisknow, is on asterisk.org... so the channel would be here I think
19:27.41Kattydaemon: we DO NOT SUPPORT GUIS HERE.
19:27.50bmoraca_workdaemon, #asterisknow
19:27.53Kattydaemon: We are NOT asterisknow.
19:27.56bmoraca_workdaemon, it's right in the topic
19:28.06*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
19:28.07Kattydaemon: Please go to the appropriate IRC channel.
19:28.35*** join/#asterisk minotaur01 (n=minotaur@24.215.3.50)
19:28.46seanbrightjeez
19:28.59devmodHello, If i wanted to setup a very simple asterisk setup at home, what voip provider would u guys recommend ?
19:29.03[TK]D-Fendergathers the pitchforks, torches, and villagers
19:29.15[TK]D-Fender~itsplist-us
19:29.16infobotmethinks itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
19:29.16titter`I have this GUI.... *channel bursts out in rage*
19:29.19[TK]D-Fenderdevmod: ^^^^^^
19:29.20seanbrighti can't believe you would ask about a GUI in here you SON OF A BITCH
19:29.30seanbrightheh
19:29.40ErestarMan, I miss IRC ;)
19:29.45bmoraca_workseanbright, it's not that.  the problem is (especially with asterisknow) that there's almost no way for anyone in here to support it
19:29.51Kattyseanbright: exactly.
19:29.51seanbrighti know i know
19:29.55Kattyseanbright: YOU KNOW
19:30.04seanbrightthe goal of this channel is clear
19:30.06devmod[TK]D-Fender, thanks :)
19:30.34seanbrightto pawn people off on other channels so there is more "hug time"
19:30.36seanbrightheh
19:30.58seanbrightproblems with extensions.conf?  try in #asterisk-extensions-dot-conf
19:31.24seanbrighti may be trolling
19:31.30Kattydo you need a hug, mister cranky pants.
19:31.39theharneeds a hug
19:31.42theharvery cranky
19:33.19seanbrighti require no hugs
19:33.51Kattyyou're gonna be a bitter old cranky man.
19:33.57seanbrightgod willing
19:34.10Kattykay
19:34.18titter`needs food
19:34.29Kattytitter`: spaghetti!
19:34.34benngardneeds sleep
19:34.59Kattynap would be good
19:35.12Naikroveknaps are always welcome
19:35.14Naikrovekeven while sleeping
19:35.16Kattyjosh groban looks like Roy from the IT crowd
19:35.19Naikroveki dream about sleeping
19:35.25Naikrovekwhile sleeping
19:35.28Kattyi dream about having to pee
19:35.31Naikrovekhehe
19:35.35titter`Katty: hmm, good idea ... but nothing at my office to make it, so im thnking those new spicey nuggest from wendys will hold me over.
19:35.45Kattyfast food :<
19:35.47KattyDON"T DO EET
19:36.05titter`lol
19:36.30Kattywhat is nearby, besides wendy's
19:36.35*** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
19:36.55titter`everything basically ...
19:36.55NaikrovekHeartClogger's, Fatty McGee's, McDonalds, uh, and Best Buy
19:37.31titter`actually havent had fast food in amost 2 months ... but the spicey nuggets look good
19:37.36Erestartitter`, Today feels like a Chinese food day to me
19:38.18titter`81 and sunny, with a bit of Chinese ... but ive been eating so much brown rice, don't know if I want more rice
19:38.20*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
19:38.45Kattyblack beans, kidney beans, seasonins, and cornbread.
19:38.46p3nguinYou must be in Miami.
19:39.14titter`Other side of the state
19:39.49Kattytitter`: http://www.tonychachere.com/recipesapp/ViewRecipe.asp?RecipeId=68
19:39.52p3nguinNaples?
19:40.00Kattyhas relatives in Naples
19:40.06titter`very close to Naples
19:40.09titter`Fort Myers
19:40.12Kattytitter`: i'm making that this weekend, no rice tho. substituting cornbread.
19:40.13p3nguinSt. Pete?
19:40.16p3nguinoh
19:40.27titter`St Pete and Miami are two places I wouldn't live
19:40.36titter`St Pete Beach is nice
19:40.40Kattywhat's at Fort Myers
19:40.47Kattyi'm guessing military personell?
19:40.50Kattypersonnel
19:40.52titter`Nope
19:40.56titter`it's a city
19:41.00Kattyah.
19:41.07*** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com)
19:41.11titter`north of naples by about 20 minutes
19:41.28Kattyi don't remember much being in Naples
19:41.29p3nguinso like 3 minutes from Cape Coral.
19:41.31titter`couple colleges, and the airport
19:41.43titter`yup 3 minutes from Cape Coral
19:41.49titter`depending how close you are to the bridges
19:42.02titter`and how many northerns are getting lost on the roads
19:42.10titter`northeners*
19:42.43titter`It was 28 last Sunday night ... now its 81
19:42.47titter`Love the climate
19:42.52ErestarYou guys made me hungry. I'm off to lunch. Thanks for all of your help and advice :)
19:44.04titter`http://www.wunderground.com/history/airport/KRSW/2010/1/11/DailyHistory.html?req_city=NA&req_state=NA&req_statename=NA
19:44.09Kattyi could so nap at my keyboard
19:44.21p3nguinI might vacation in New Smyrna Beach this summer.
19:44.28titter`I have done that before
19:44.32titter`jacket on my desk
19:44.34titter`and nap away
19:44.43titter`to be woken by the sound of the polycom
19:45.35*** join/#asterisk tzafrir (n=tzafrir@212.179.75.202)
19:46.16Kattyp3nguin: what's at New Smyrna Beach?
19:46.43Kattytitter`: i usually just lean back in my chair, and play some relaxing ambient music free rain/wind video.
19:47.02Kattytitter`: after the 30 minutes, the slight change from ambience to quiet is enough to wake me up.
19:47.09titter`hmm
19:47.13p3nguinkatty: Condos... and a beach.  With less people than Daytona Beach.
19:47.29Kattynot much of a beach person, personally
19:47.31titter`Daytona can get crazy
19:47.53p3nguinI used to spend a lot of time there.
19:48.14titter`Summer isnt as bad as Spring Break
19:48.50titter`off to find some food, ill be back ... thinking a turkey wrap
19:49.16bmoraca_workoh food...how great thou art
19:49.35dustybini really got to stop ringing myself, this is getting silly
19:49.57Kattylol
19:50.26titter`how nice of dell to send me a whole new lid, bezel, and switchpanel for free for my xpls ... <3 warranties
19:50.27benngardspring break at pompano beach could be .... iíntresting ;)
19:50.40*** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com)
19:50.47Kattytitter`: shoo
19:50.49Kattytitter`: go eat.
19:50.54titter`:>
19:51.34*** join/#asterisk ChrisWi (n=admin@mx2.wwserver.net)
19:52.01dustybinif somebody rings me, the caller name is the number, is there a way to replace numbers with peoples name?
19:52.53*** join/#asterisk cadmium (i=mike@217.194.139.22)
19:52.57p3nguindustybin: Of course.
19:53.17cadmiumhi. what channel variable can I use to obtain the peername
19:53.18dustybinp3nguin: i mean from the outside
19:53.39dustybinp3nguin: this would require me setting up some kind of database?
19:54.02p3nguindustybin: You could, but you can also run a script to do free lookups via internet.
19:54.16p3nguinThat's what I do for my home number.
19:54.20dustybinoh nice
19:54.33cadmiump3nguin how can i reference the peername as a channel variable in the dialplan ?
19:54.37p3nguinFor work, I let the ITSP do queries and they send me the name.
19:54.43dustybincan the internet find the name for any number
19:54.52p3nguindustybin: no
19:55.57dustybinp3nguin: i only have 2 friends, so all i need to do is create some kind of table with name and number, if there is a match then use the name
19:57.22bmoraca_workdustybin, astdb should be more than easy enough.  or even just global variables.  just set CALLERID(name) to whatever you want.
19:57.48bmoraca_workor hell, just use callerid matching on your extension priority
19:57.50dustybinbmoraca_work: if my asterisk server rings my cellphone, my cellphone says 'private'
19:57.56dustybindisplays no name or number
19:57.59bmoraca_workdustybin, no way to get around that.
19:58.07dustybinaye ok
19:58.20dustybinbmoraca_work: is that down to my VOIP provider?
19:58.21bmoraca_workdustybin, best you could do is have it display a number.  you cannot send callerid name.  CNAM is ALWAYS performed on a lookup basis by the CALLED party
19:58.25p3nguincadmium: ${SIPCHANINFO(peername)}
19:58.53dustybinright lets check the outgoing first
19:58.56bmoraca_workdustybin, if your VOIP provider allows you to specify any callerid number, yes you could send number.  but you will never be able to send name
19:59.13cadmiump3nguin thanks
19:59.32bmoraca_workmany (probably most) providers do not allow you to specify CID to anything other than numbers you own.
19:59.42bmoraca_workthere's been a big crackdown on that lately...callerid spoofing is bad
19:59.43*** join/#asterisk guax (n=guax@unaffiliated/guaxinim)
20:00.08[TK]D-Fender[14:58]<bmoraca_work>dustybin, if your VOIP provider allows you to specify any callerid number, yes you could send number. but you will never be able to send name <- I've seen PRI's I can do this on... not a stretch to imagine its possible for them to apss the same on via SIP
20:00.34guaxGot a attended transfer that when hung up turns into a blind transfer, when that happens the transfered channel stops playing music on hold and gets muted even if the other channel should be ringing.
20:00.56dustybinaye ok thanks, i will try and set the number
20:01.05guaxeven muted the other leg keeps ringing and if its answered the call place normaly
20:01.44*** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca)
20:02.19bmoraca_workaaaaaand...lunchtime
20:02.57*** join/#asterisk joako (n=ston3d@opensuse/member/joak0)
20:04.02*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
20:04.04dustybinbefore the dialplan rings my VOIP provider, all i need before is:
20:04.14*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
20:04.17dustybinSetCallerID(number)
20:05.10dustybinwhoops
20:05.11dustybinSet(CALLERID(num)=value)
20:05.23dustybintests
20:12.41dustybinoh it was working anyway, i think i had a problem with .call_me showing a private number
20:13.00dustybintests
20:16.00titter`adventures of dealing with tourists on the road
20:19.38minotaur01im setting up a new building in a month or so with a fresh install on asterisk... they will require external paging... dose anyone have any recommandations
20:19.47*** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110)
20:20.19*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
20:20.29Kattyi recommend...
20:20.39Kattyhaving a healthy breakfast.
20:20.45minotaur01lol
20:22.53Kattywe don't use external paging.
20:22.54Kattyj
20:23.01Kattyjust the add sip header stuffs.
20:24.04minotaur01i use a CyberData Voip Amplifier in one building now but im wondering if there is anything better out there
20:24.12*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
20:24.50Kattyhi raden
20:25.43titter`really needs a nap now
20:25.57*** part/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
20:26.04Katty2.5hrs before i go home still.
20:26.09Kattybut a nap does sound pretty awesome.
20:26.17titter`I could go now
20:26.39titter`I just like the idea of working, and getting things finished lol
20:27.37Kattywell currently nothing here is broken.
20:27.48Kattyjust waiting for someone to call, email, or otherwise run screaming to my office
20:28.28titter`lol, I know the feeling ... I restarted 11 switches and almost 400 shoretel phones last night after making a system wide change ... and today has been quiet ... maybe because they can't call me
20:28.44titter`ponders...
20:28.51titter`oh well.
20:29.33Kattysounds good to me
20:30.02minotaur01titter: thats the best phone call
20:30.05*** join/#asterisk ruied (n=ruied@bl7-217-61.dsl.telepac.pt)
20:30.39titter`yes ive gotten calls from people saying their phone isn't working
20:31.11minotaur01titter: how are those shortel phones to work with?
20:31.56titter`since we run ipsec tunnels from some office to access internal websites, ive gotten emails stating my internet is down, as they e-mail me.
20:32.24minotaur01lol
20:32.25titter`minotaur01: it's a pig. it's a nice system, but the costs of it isn't worth it
20:33.09Kattyyou know.
20:33.14Kattywe're actually a shoretel partner
20:33.21Kattybut we've never sold a single shortel system
20:33.24titter`it was from acquisition, and making asterisk work with shoretel ... bleh.
20:33.41Kattyseems like everyone around here wants or already has, toshiba.
20:34.06minotaur01i use Aastra's 57i my self but always looking for the next thing
20:35.23titter`shortel phones are proprietary to their hardware ... sort of
20:35.27*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
20:35.42minotaur01ah no good for me then
20:35.49titter`shoretel only uses one rtp stream on port 5004
20:35.57titter`and only supports MGCP
20:36.29minotaur01i had a proprietary 3Com Nbx at one time... what a pice of crap...
20:36.34simplydrew___does anyone know if there's a script out there or addon for asterisk that will let you dial into asterisk and define a time to call a number?
20:37.08minotaur01like a hotel wake up service?
20:37.11minotaur01then yes
20:37.18simplydrew___pretty much, but to call an external number
20:37.27simplydrew___I have a wake up call script integrated, but it won't dial out
20:38.02bmoraca_workit should be fairly simple to modify the script you have
20:38.07minotaur01you have to change the channel its calling on
20:38.09bmoraca_workto do what you want
20:38.24*** join/#asterisk hfb (n=hfb@96.247.53.139)
20:38.26p3nguinWake up calls from outside the system?
20:38.36minotaur01find the spot that says SIP/[ext]
20:38.51simplydrew___essentially yes, I just want to be able to tell it to call my cell at a certain time or something
20:38.54minotaur01and change it to TRUNK/[number]
20:39.13p3nguinminotaur01: You sound like a GUI.
20:39.23minotaur01lol
20:39.29minotaur01i use one
20:39.33Kattyhow does one sound like a GUI
20:39.33p3nguinIt shows.
20:39.41Kattydoes it make specific, gurgly noises?
20:39.46titter`lol
20:39.48Kattyor maybe whirly sounds?
20:39.52Kattymwrrrrrrrhhhhfff
20:40.06minotaur01lol
20:40.08titter`pew pew
20:40.57bmoraca_workthe sound of my belly trying to digest the McDonalds double cheeseburger i just ate
20:41.21titter`bleh
20:41.22*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:42.14bmoraca_workmeh...it was cheap and quick and it's been pouring rain here for 4 days straight...didn't feel like going anywhere else
20:42.15titter`why is it old people have a hard time seeing a bright red car? that they much pull out of whereever they are, and cut me off, with no notice of their action
20:43.21Kattythat's not strictly old people
20:43.30bmoraca_workmost people drive like morons
20:43.49ChainsawI think you'll have better luck with a bright yellow car.
20:43.50titter`true, but where I am it is mostly old people, or out of state plates
20:44.07p3nguinThe problem with this country is that there is no death penalty for incompetence.
20:44.07titter`has road rage
20:44.30bmoraca_workright now, we're getting the morons who slow down to 1 MPH to go through a puddle that's not even 6" deep
20:44.34*** join/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net)
20:45.01Kattytitter`: eww.
20:45.08Kattytitter`: you need to work on that
20:45.15Kattytitter`: very very bad for your health
20:45.48titter`lol
20:45.52titter`I have a lead foot too
20:46.05p3nguinI'm not too good to run someone off the road for driving like a twit.
20:46.10Kattyi assure your, your foot is not lead. you just like to drive fast.
20:46.12*** part/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net)
20:46.15dlynesI'm just wondering what I'm missing?  The following two lines tell me that I'm missing the trailing parenthesis?
20:46.16dlynesexten => s,2,Read(CHOICE,/var/spool/asterisk/autoattendants/default/${ARG2},1,,,20)
20:46.17dlynesexten => s,n,GotoIf($["${READSTATUS}"="OK"]?${CHOICE},1:1)      ; if the user picked something, branch to the choice
20:46.25*** join/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net)
20:46.33*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
20:48.22dlynesIt's the second of those two lines where I'm getting the error
20:48.30voipmonkhrmm
20:48.34dlynes${CHOICE} has a value of '1', if it matters
20:48.37minotaur01was anyone come across a problem with the queues conf not loading propery in version 1.6.0.20 on start/restart but loading fine on reload??
20:49.28dlynesminotaur01, could be the order in which you're loading your modules
20:49.46bmoraca_workdlynes, your destination syntax for gotoif is wrong.  it's looking for a label, as such ${CHOICE},1:1 is wrong.
20:49.59Kattydlynes: just stick another parenthesis at the end
20:50.00titter`Katty: I guess
20:50.08*** join/#asterisk mlr78 (n=mratliff@74.5.197.138)
20:50.20titterKatty: I can't go 2 days without see 1xx in my car
20:50.22dlynes${CHOICE},1 is one label (exten, priority); 1 is another (priority)
20:50.33minotaur01hey dlynes, how do i change the order?
20:50.51Kattytitter: i very rarely see my car up in the 3 digits.
20:51.15bmoraca_workdlynes, it's not looking for an exten,priority, it's looking for a LABEL
20:51.18dlynesminotaur01, autoload => no in your modules.conf file, and then do a load => module_name.so specifically for each module you want to load, in the order you want
20:51.23p3nguinI usually only go that fast when everyone else is also going that fast.
20:51.38dlynesbmoraca_work, so i can't use exten, priority or context, exten, priority there?
20:51.46dlynesbmoraca_work, only priority and n(label)?
20:51.57bmoraca_workdlynes, i don't believe so, no.  your second question would be an affirmative
20:52.05minotaur01dlynes: thank you
20:52.09dlynesthat sucks, immensely
20:52.13bmoraca_workdlynes, if you want to do it the way you have been, Background() would be more appropriate
20:52.14p3nguindlynes: I use exten,priority in 1.4 and it works fine.
20:52.21titterp3nguin: http://www.radarbusters.com/radar-detectors/escort-radar-detectors/escortpassport9500ix.cfm
20:52.21dlynesp3nguin, i'm using 1.6.1
20:52.40p3nguinI can't imagine they would take out that functionality.
20:52.52dlynesbmoraca_work, how is background any different from read(), goto()?
20:53.08Kattyoooh. it appears the sun is out.
20:53.17Kattythe crittercam is BRIGHT for the first day in....days!
20:53.18bmoraca_workdlynes, background() goes directly to the extension entered in...it does not store the captured digits in a variable
20:53.21p3nguintitter: Is that what you have?
20:53.27titterYUP
20:53.32titteryup*
20:53.40p3nguintitter: Have you ever used a Valentine One?
20:53.46titterI have
20:53.55p3nguinHow would you compare the two?
20:54.13p3nguinI find the Valentine to be bulky and ugly.
20:54.16dlynesbmoraca_work, yeah, but read gives me a bit more flexibility
20:55.16p3nguindlynes: n,GotoIf($[${BLACKLIST()}]?misc,blocked,1)  for example... misc context, blocked exten, priority 1.  No problems.
20:55.20titterHonestly they are pretty close
20:56.42p3nguintitter: If you had an extra $500 sitting around and wanted to buy another detector for another car, would you buy either of the two or something different?
20:57.03titter1 sec smoeone made a 911 call -.-
20:57.11Kattyi can think of better things to buy with 500 bucks
20:58.13dlynesp3nguin, I just said forget it, and called it ch1, ch2, and ch3 instead
20:58.20p3nguinSome people want to have a good radar detector, though.
20:58.31dlynesp3nguin, so it should go to 1,n(ch1) and 1,n(ch2) and 1,n(ch3) instead
20:58.34titterI would get the Passport again
20:58.38tittermore features
20:58.40tittersame range
20:58.45titterV1 is better on X band
20:58.48titterbit overkill
20:58.48*** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek)
20:58.50titterand has arrows
20:59.04p3nguindlynes: So your labels are working, but con,ext,pri would not?
20:59.08titterI dont use arrows much, since I know where cops are probably sitting
21:00.22p3nguinThe arrows seem like a novelty to me.  If you know you are speeding, it shouldn't matter where the signal came from... SLOW DOWN anyway.
21:00.47titterA good detector goes a long way, knowing when tos peed, and not to speed plays a big roles
21:01.43Kattyor you could just....not...speed.
21:01.48Kattyor go 75 in a 70 zone.
21:01.58*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
21:03.00[TK]D-FenderHere the cops don't car if you go like 60-65 in a 50 zone... its when you pass by 20+
21:05.07titterI avg 90 in a 70
21:05.10titteron 75
21:05.22p3nguinI normally don't go over 10 over on the freeway.
21:05.32p3nguinunless other traffic is also going over.
21:05.40Skeeter-p3nguin, its a called a freeway from some reason
21:05.43bmoraca_workI go 80 on freeways
21:05.48Skeeter-100 neeeds to be around 140
21:05.54Skeeter-kmh here
21:06.00p3nguinI prefer the freeways where there is no daytime speed limit.
21:06.04bmoraca_workspeed limit on most of them is 65, but we're starting to get some 70 and 75s around
21:06.31titterit's easy with my car ... but most of the time I cruise around 90, and if get a blip ill slow down to 85ish until I know if its real or not
21:06.39titterthe 9500ix hasnt failed me yet
21:06.48p3nguinOn my way home from Phoenix, I didn't drive under 110 unless impeded by other traffic.  That's 1600 miles.
21:08.44p3nguinLots of speed corridors, too, but never say any highway patrol in any of them.
21:08.50titterive picked up FHP almost 3 miles in front of me
21:09.28p3nguinHow well does your detector do when the radar power is turned way down?
21:10.00p3nguinI've approached cops and their signal is so weak that even right on top of them, my detector isn't showing full bars.
21:10.36titterthey are proabably using pop radar or instant on
21:10.41p3nguinno
21:10.57p3nguinThose are totally different than what I described.
21:11.57p3nguinPOP isn't detected by a lot of units, and instant-on goes from off to on in a fraction of a second and stays on.  They just often turn down the power on the normal mode.
21:12.06titterhttp://guysoflidar.com/august-2007/radar-detector-test.html
21:12.09bmoraca_workthey've started using LIDAR around here
21:12.26titterlaser is used a lot of places
21:12.31bmoraca_workradar detectors don't work on them
21:12.44titterthey do... just have to have more than the detector
21:12.55p3nguinMost radar detectors don't come without laser built-in.
21:13.08titterhttp://www.radarbusters.com/radar-detector-accessories/laser-veil/laserveil.cfm
21:13.20titterI have his on my headlights
21:13.39titterand ive been through a few laser traps without a ticket
21:13.53p3nguinI can't imagine how that could even work.
21:14.16p3nguinIt's not like lidar only reflects off headlights or something.
21:14.56titterthats the first place they aim
21:15.02titterand the only place on my car they can aim
21:15.08titterif they aim higher
21:15.08p3nguinNo, that's not how it works.
21:15.11titterit is
21:15.15p3nguinIt's not.
21:16.21*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
21:17.00*** join/#asterisk glasshut (n=chatzill@67.200.59.158)
21:17.32titterhttp://www.youtube.com/watch?v=jdeIj1KKL0w&feature=fvw
21:18.19titterleo is trained to aim at front plates and headlights, they can get a reading back from the car itself, but the better response is from highly reflective surfaces
21:19.01p3nguinbetter response... does not mean LIDAR relies on headlights nor license plates.
21:19.17p3nguinIt just means the light is more likely to reflect off those surfaces.
21:19.21*** join/#asterisk ZX81 (n=ZX81@121-74-230-221.telstraclear.net)
21:19.26titterit doesnt rely on it, but with "MY" car it helps a lot
21:19.35titterbecause where my detector is
21:19.39p3nguinIf you take off your plate and remove your headlights, you are not invisable to LIDAR.
21:19.41titterit will get the lidar hit
21:19.49*** join/#asterisk settntrenz (n=joe@162.226.189.72.cfl.res.rr.com)
21:20.03titternot invisible, but the range is a lot less for them
21:20.04Kattyyawn.
21:20.05*** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
21:20.09Katty1.5hrs
21:20.23p3nguinWhy are you still awake?
21:20.24titterthe idea with lidar is to slow down before they get you
21:20.42titterveil and my detector make that happen
21:21.09titterhttp://i109.photobucket.com/albums/n58/dotdotkeek/01ss/IMG_0360-1.jpg?t=1264108849
21:21.12Kattywell it looks like i'm gonna be driving to The Country tomorrow
21:21.18titterfront of my car
21:21.22tittermy headlights are huge
21:21.23p3nguinIf I were to build a LIDAR unit, I would change the wavelength of light that is used to circumvent the detectors.
21:21.38*** join/#asterisk jelly-bean (n=mikesmul@75-148-103-190-Utah.hfc.comcastbusiness.net)
21:22.01jelly-beanhow could i get asterisk to notify me when a call has been waiting in queue for more than 1 minute?
21:22.13p3nguinI thing my headlights have more area, but are probably less reflective than yours.
21:22.47[TK]D-Fenderjelly-bean: Write a script that monitors the queue and checks member wait time
21:23.02[TK]D-Fenderjelly-bean: AMI or CLI via "-rx"
21:23.13jelly-beanwhat kind of script--any kind? AMI?
21:23.43*** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net)
21:24.16settntrenzSay I have a staff that logs into a queue to handle inbound support calls. The staff (supporting athletic events that don't have a set ending time) leaves for the night when the last event is over. We also have a nighttime answering service. Can I configure asterisk so that the last employee out can use their handset, dial an ext, input a password which would automatically forward calls destined for the support queue to the night extension
21:25.14*** join/#asterisk luminblade (n=luminfor@cpe-24-28-78-39.austin.res.rr.com)
21:25.16bmoraca_worksettntrenz, of course.  use astdb to store which mode (day/night) the inbound calls should be treated as
21:25.29*** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net)
21:25.30mlr78anyone have a good example of a config file for a linksys spa942?
21:25.44dlynesah...found out what the real issue was
21:25.54Carlos_PHXmlr78: What do you want in your config?
21:25.59dlynesIt has an issue with a trailing parenthesis on the line:  exten => 1,1,Dial(${ARG1}/${DB(${ARG2}/option1},${ARG3},ktx) ; enable call parking, call transfer, and call recording
21:26.07p3nguindlynes: missing a "close paren"?
21:26.07Carlos_PHXmlr78: You know the phone can spit out its current one?
21:26.13settntrenzbmoraca_work: Thanks, I'll do some googling. I didn't know what this type of feature was actually called making it difficult to google.
21:26.34dlynesI'm guessing it has something to do with the DB function
21:27.10bmoraca_worksettntrenz, it's just an alternate type of routing.  you probably won't find anything prebuilt, but it owuld only take 6-7 lines to actually put something together.
21:27.16luminbladequestion: running 1.6.2.1.  i have multiple peers (carriers) connected running g729 pass-through.  if i enable T.38 (t38_udptl=yes) asterisk segfaults from time to time...  is there anything else/special i need to do to set up Fax/T.38 pass-thru correctly?  i am not trying to connect directly to any fax endpoints/devices.
21:28.00bmoraca_workdlynes, you are missing the closing parenthesis for your DB() call...but it's impossible for me to tell you where it should be because i don't have any other context for what you're trying to do
21:28.38[TK]D-Fendercheckout time, later all
21:29.07bmoraca_workdlynes, but i suspect it should be after "option1"
21:29.53settntrenzbmoraca_work: thanks for pointing me in the direction, now I just gotta see how to do this using trixbox. Before this gig I've only worked on standard asterisk. I'm trying to figure out if I'll break things or have them overwritten by editing extensions.conf directly
21:29.55p3nguindlynes: vim is your friend in a case like that.  It highlights the matching open or close paren, brace, and bracket.
21:30.43bmoraca_worksettntrenz, uhg...trixbox...let me just tell you that FreePBX (what generally handles trixbox's dialplan) has a module that will do day/night routing for you.
21:30.59settntrenzyup, just found it. http://www.freepbx.org/support/documentation/module-documentation/day-night-mode-control
21:31.26settntrenzit looks like it handles doing it from the handsets too
21:31.38bmoraca_workyes it does.
21:31.42settntrenzin the past I'd just get a call saying we're closing early and edit the gotoiftime's
21:32.26dlynesbmoraca_work, oh shit....just realized I'm missing the ')' on DB
21:32.26settntrenzok, i'm going to read some docs. Thanks again
21:32.30*** part/#asterisk settntrenz (n=joe@162.226.189.72.cfl.res.rr.com)
21:32.43glasshutOur company has an editor who listens to long winded stories from people on the phone then needs to transcribe them into text.  To do this we set up recordings.  The only problem is that on an outbound call if she hits mute, the recording pauses even though she can hear the other person.  When you listen to it the times when she pushes mute are chopped right out of the recording. Asterisk...
21:32.44glasshut...1.4.22. Any ideas?
21:33.28bmoraca_workglasshut, make sure that there's no VAD or comfort noise of any kind enabled on the endpoint
21:34.05*** join/#asterisk [Outcast] (n=anonymou@64.202.62.5)
21:34.15*** part/#asterisk stan (n=stan@87-194-216-82.bethere.co.uk)
21:35.03glasshutI believe I did. I tried to make sure it was turned off in /etc/asterisk/phone.conf silence suppression as well as /tftpboot/SIP* files.  I may have missed something.
21:35.08[Outcast]so what is the deal with g.729 and 1.6.2?"
21:35.48bmoraca_work[Outcast], it works fine?
21:36.17[Outcast]no I have been told g.729 does not work with 1.6.2
21:36.33[Outcast]There is no module fo rit
21:36.42p3nguinThat would be a bother.
21:36.57bmoraca_worki'm using 1.6.2.0 right now with g729...
21:37.02bmoraca_workmust be voodoo
21:37.24glasshutbmoraca_work: I have cisco 7960 phones, enable_vad: "0" in SIPDefault.cnf.  Shouldn't that do the trick?
21:37.28[Outcast]yea it is pass through, but it will no terminate
21:37.51bmoraca_workonce again, i'm using it just fine to terminate as well
21:38.10luminbladebmoraca_work: do you have fax / T.38 working as well?
21:38.30bmoraca_workluminblade, fax yes, but I don't use T.38
21:39.34[Outcast]so you can have a g.729 call be answered by application in the dialplan
21:42.17glasshutI am trying to make sure VAD is not on to fix my recordings issue.  If I make a call and select mute for both the Cisco 7960 using SIP as well as a cell phone.  With the volume the whole way up on the desk phone, should vad make noise if it is turned on?
21:42.36glasshutIt seems dead silent.
21:42.56p3nguinIf you have set vad to be off, and reloaded the phone, it is off.
21:44.06glasshutOk, so maybe that isn't the issue as to why pushing mute on the phone cuts that part out of the recording :-( any ideas?
21:44.21p3nguinWhat are you using to record?
21:44.32glasshutMixMonitor I think
21:44.35mlr78for the linksys spa phones...is the correct; spaMAC.cfg or spa$MAC.cfg to do auto provisioning via option 66?
21:45.16glasshutexten => s,1+998(record),MixMonitor(${MIXMON_DIR}${CALLFILENAME}.${MIXMON_FORMAT},,${MIXMON_POST})
21:46.06KattyAHHHHHHHHHIEEEEEEEEE.
21:46.10Katty1hr.
21:47.41glasshutIf I look at the call length and the recording length in audacity, the recording is significantly shorter if I use the pause button.
21:48.02glasshut*Mute button
21:51.45Kattyso many pretty birds in the front yard.
21:53.23*** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net)
21:53.49*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
21:53.53LemensTSIs there a way to get the username of a sip client on an outgoing call? I dont see it in predefined channel variables
21:54.35p3nguinlemensts: Yes.  SIPCHANINFO(peername)
21:54.37[TK]D-FenderLemensTS: where would you grab this anyway?
21:55.32*** join/#asterisk ctooley (n=ctooley@c-98-193-232-154.hsd1.tn.comcast.net)
21:55.40LemensTSTKD: Id pass it into an agi script from the dialplan.
21:55.54ctooleyHow do I use tshark to output just the SIP messaging from a pcap file?
21:56.10[TK]D-FenderLemensTS: parse out sip.conf
21:56.16LemensTSp3nguin: cool never knew that cmd before
21:56.35p3nguinlemensts: core show functions
21:57.29[TK]D-FenderLemensTS: Also isn't what you asked for.....
22:02.27*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
22:06.23titter[TK]D-Fender: If Asterisk gets a 302 Moved Temp request from a Polycom, is there anyway to change the outgoing Contact it recieved from the Polycom to append say a * to then dialout via a specific pattern matach. I want to forward on incoming callerid, to my forwarded phones.
22:06.48*** join/#asterisk slima (i=slima@unaffiliated/slima)
22:07.05*** join/#asterisk sflemming (n=stefan@85.183.40.40)
22:11.43voipmonkhey titter
22:11.52titterhey
22:14.26*** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de)
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22:18.08Kattypokes voipmonk
22:18.50*** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131)
22:20.21titterbbl, time to diassemble my laptop
22:23.45*** join/#asterisk rossand (n=aross@dhcp-233-179.tb-classrooms.carleton.ca)
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22:27.12dustybinits time for a cup of tea
22:29.22dustybinmy lord
22:29.23dustybindatabase put cidname 8005555555 "Asterisk Caller #1"
22:30.00p3nguinhmm?
22:30.12dustybinp3nguin: this is exactly what im looking for
22:30.15p3nguinoh
22:30.25dustybintest
22:30.26p3nguinI thought someone suggested astdb earlier.
22:33.26Katty20 minutes!!
22:34.34dustybinadding a entry in the asterisk database didnt change this
22:34.36dustybin--caller_name="${CALLERID(name)}"
22:34.50dustybinthat still shows the incoming number, not name
22:37.17dustybinoh i need to change my dialplan
22:37.18dustybin<PROTECTED>
22:37.18dustybin<PROTECTED>
22:37.18dustybin<PROTECTED>
22:37.21dustybin<PROTECTED>
22:37.50*** join/#asterisk KaneHau (n=KaneHau@133.40.166.155)
22:38.37dustybinexten => s,n,Set(CALLERID(name)=${IF($["${DB(cidname/${CALLERID(num)})}" = ""]?${CALLERID(name)}:${DB(cidname/${CALLERID(num)})})})
22:38.45KaneHauquestion... I am using BackGround to play a message while waiting for a user to hit any DTMF tone.  I don't care what key they hit... is there a way to label an extension so that any key (0-9) goes to the same extension (without using extension 'i')?
22:38.47dustybinman that is ugly
22:39.03dustybintests
22:39.26KattyKaneHau: yes.
22:39.34KaneHauwonderful
22:39.37seanbright_X
22:39.41Katty^-
22:40.00seanbrightexten => _X,1,NoOp(they hit ${EXTEN})
22:40.07dustybinIT WORKS!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
22:40.32dustybinMY NAME SHOWED ON MYTHTV OSD WHEN I CALLED FROM MY CELLPHONE :D
22:40.39fenrus:D :D :D
22:40.42fenruscongratz
22:40.57KaneHausean: thank you very much
22:41.11seanbrightyup
22:41.21Kattygosh.
22:41.25Kattysimmer down now.
22:41.45KaneHausean:  didn't work... I got this:  Channel 'DAHDI/1-1' sent into invalid extension '8' in context 'outgoing', but no invalid handler
22:42.00KaneHauI used the exact syntax you showed
22:42.05KattyKaneHau: it did work.
22:42.17KattyKaneHau: else how would you get 8?
22:42.25KaneHauI didn't get the NoOp line
22:42.35KaneHauexten => _X,1,NoOp(they hit ${EXTEN})
22:42.41KaneHaunever said "they hit 8"
22:43.03KaneHauoh, I bet my '1' priority should be 'n'
22:43.09Katty^_-
22:43.26Kattyi am not following your thought process.
22:43.34Kattyfeel free to share some pastebin.
22:44.19KaneHauno, it wasn't the priority either... guess I'll just use 'i' as the extension
22:44.27*** join/#asterisk JraNil (n=JraNil@85.133.218.30)
22:44.37seanbrightor you could pastebin your extensions.conf
22:44.41seanbrightand we could tell you what is wrong
22:44.53KaneHausure, all 4 lines of it... hold on
22:44.54Kattypersonally, i'm not a mind reader.
22:44.57Kattyseanbright here might be tho
22:45.07Kattyseanbright: WHAT AM I THINKING
22:45.21seanbrighti have no idea
22:45.28Katty:<
22:45.31Kattytacos.
22:45.35Kattythe answer is always tacos.
22:45.40seanbrightnoted
22:45.50QwellWho was the 32nd President of the United States?
22:45.52Qwelltacos.
22:46.10KaneHausean:  http://pastebin.com/m46625b30
22:46.23KattyI'm pretty sure mister Roosevelt liked tacos.
22:46.32seanbrightthe priority should be 1
22:46.35seanbrightnot n
22:46.45Qwelland Background doesn't wait
22:46.45KaneHauI tried '1' - also failed
22:46.47p3nguinkanehau: Well there's yer problem.
22:46.47seanbrightand make sure you do a 'dialplan reload' after making changes
22:46.52QwellIt will fall through immediately
22:47.05KattyWaitExten() <-
22:47.06KaneHauahhh, forgot the dialplan reload... thanks
22:47.09seanbrightand there is that as well
22:47.13p3nguinkanehau: _X doesn't have a 1 priority.
22:47.22KattyWaitExten(Tacos)
22:47.23seanbright17:46 < seanbright> the priority should be 1
22:47.25seanbright17:46 < seanbright> not n
22:47.59sflemmingHi all, is there someone who uses dahdi and has a problem with the dahdi_dummy timing source? It simply does not tick with my setup. Someone there who can help me?
22:48.00p3nguinkanehau: And the dialplan CANNOT flow from 's' to _X without a goto or something else to get it there.
22:48.08Kattyexten => _X,1,Goto(Tacos,s,1)
22:48.27KaneHauok... it was because I forgot the dialplan reload, thanks!
22:48.30Kattysflemming: it does not tick?
22:48.37Kattysflemming: have you replaced the batteries?
22:48.48Qwellsflemming: what version of dahdi?
22:48.53seanbrightKatty: excellent
22:48.54sflemmingno, when I use dahdi_test it hangs
22:49.08sflemmingdahdi 2.2.1, the same with dahdi_tools
22:49.21KaneHaup3: I know that... I didn't want it to flow - I just wanted to get it to detect the DTMF tone and go to the _X extension... I need to drop a "hangup"between those
22:49.23Kattyi hear that replacing batteries fixes a lack of ticking.
22:49.35Qwellsflemming: unload all the modules, load only the dahdi module
22:49.38p3nguinkanehau: Perfect.
22:49.40Qwell(not dahdi_dummy)
22:49.50Katty10 minutes!
22:49.55Kattyi will survive another day of work!
22:50.08KattyI WILL SURVIVE
22:50.12Kobazkatty: http://www.whowantsabalti.com/funpics/mild/squirrel-surprise.jpg
22:50.14sflemmingWhich batteries? I had a look into the sources of dahdi_dummy and it seems to use the HPET timer of the kernel, I tried to rise the frequency from 250hz to 1000 but same problem
22:51.03KattyKobaz: mmmm, schmirnoff
22:51.18Kattysflemming: what do you mean which batteries?!
22:51.34sflemmingI tried nearly all loading and unloading scenarios. Have a discussion about the problem since 3 month, there are three people that have the same problem. (All on 64 bit machines) On 32 bit there is no problem with it
22:51.36KattyKobaz: what is that other little baggy thing?
22:51.47*** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com)
22:51.48KattyKobaz: sugar/salt?
22:52.02citywoki'm having issues with callerid names, and asked bandwidth.com to help me out, and they asked me this "Can you try and enable the use of Remote-Party-ID or P-Asserted-Identity and give me the call examples of either of those fails?"
22:52.12QwellKatty: looks like a chocolate
22:52.25Qwellchocolate covered cherry perhaps
22:52.33Kattyooh
22:52.35Kattyaww.
22:52.36sflemmingkatty asked if i replaced the batteries, but there are no batteries, okay there is a battery for the rtc but this seems to work correctly
22:52.49seanbrightsflemming: she was making a joke
22:52.49Kattysflemming: just...nevermind...you didn't get it.
22:52.57dustybinyou guys are clever sods :D
22:53.01dustybin*gals
22:53.49Kattycitywok: I believe they're talking about something Native to toshiba systems.
22:53.59sflemmingkatty: ;-) sometimes this is not easy to recognize when english is not your motherlanguage. nevermind, any ideas?
22:54.05Kattycitywok: what they're asking you for is information off the asterisk CLI that might give them a clue as to what's going on.
22:54.30citywokso i googled and there is a voip-info document about p-asserted http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header
22:54.34seanbrightno they aren't
22:55.44Qwellsflemming: scroll up to what I said
22:56.01seanbright<@Qwell> sflemming: unload all the modules, load only the dahdi module
22:56.06seanbright<@Qwell> (not dahdi_dummy)
22:56.12seanbrighti've reproduced it here
22:56.16seanbrightheh
22:56.34Qwellwith dahdi_dummy 2.2.1 loaded?
22:57.21*** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com)
22:57.54sflemmingwithout dahdi dummy i won't have a timing source since my card does not provide it
22:58.04Qwellsflemming: please do as I asked. :)
22:58.39sflemmingi already did
22:58.44KaneHauQwell:  You said that "BackGround" doesn't wait... but the Asterisk manual states:  Plays the specified audio file(s) while waiting for the user to begin entering DTMF digits. Once the user begins to enter DTMF digits, the playback is terminated.
22:58.45Qwellnow run dahdi_test
22:59.14seanbrightKaneHau: it waits for the sound to finish, but it doesn't wait beyond that
22:59.25Kattyhe said, she said.
22:59.43seanbrightKaneHau: if you want to continue waiting after the sound plays, you need to put a WaitExten() after the Background()
22:59.50Kattyi'm going to go back to lurking and trolling about batteries.
22:59.57KaneHauyes, that's fine... if no DTMF is detected by the time the audio plays, I want to hangup
23:00.04seanbrightthen you're done
23:00.07sflemmingUnable to open dahdi interface: No such device or address
23:00.12KaneHauer, by the end of the audio I meant
23:00.33KaneHauok, got it... so it should work as I want
23:00.39seanbrightcorrect
23:00.59Kattyohhhhhhheyyy
23:01.00Kattyit's 5 :>
23:01.03KattyBAI!
23:01.04Kattypoofs
23:01.05KaneHau(this is to get around the fact that I can't easily detect a PSTN line that has picked up)
23:01.52sflemmingQwell: what do you suggest now?
23:02.07*** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be)
23:02.57seanbrightsflemming: can you pastevin the output from lsmod?
23:03.00seanbrightpastebin*
23:03.32QwellI thought all the timing stuff got pushed into dahdi-base..  maybe that's scheduled for2.3
23:04.32seanbrightno, it's in 2.2.1
23:04.51sflemminghere it is http://pastebin.com/d2acb64ba
23:04.52seanbrightoh, but it's off by default
23:04.57p3nguinI had to install dahdi to get timing so I could use MeetMe().
23:05.06seanbrightCONFIG_DAHDI_CORE_TIMER
23:05.11Qwellseanbright: where are you seeing the disabling?
23:05.24seanbrightin the commit message
23:05.49KaneHauThe extension __X doesn't seem to do what I intend... it thinks the pickup of the line is the same as DTMF '1'
23:05.53Qwellright
23:05.54KaneHauer _X
23:06.07KaneHauput using extension 'i' works
23:06.10KaneHaubut
23:06.12KaneHaudamn
23:06.28p3nguinkanehau: You know that 'i' is the invalid extension, right?
23:06.30seanbrightQwell: http://svnview.digium.com/svn/dahdi?view=revision&revision=6863
23:06.38KaneHauyes
23:06.51KaneHauI just want to detect ANY tone, I don't care what one
23:07.02KaneHauit tells me to "go ahead" with the rest of the announcement
23:07.09Qwellsflemming: How did you install dahdi?
23:07.12seanbrightsflemming: that is lspci.  lsmod.
23:07.22sflemmingoh, I'm sorry
23:08.04seanbrightNetwork controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI]
23:08.13seanbrightwhy does that sound familiar to me?
23:08.27QwellmISDN.  it wouldn't be providing timing
23:08.30*** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net)
23:08.34sflemmingit is only echo, dahdi, r8169 and forcedeth loaded
23:08.44seanbrightah.
23:08.46KaneHauThanks once again for all your help!  Mahalo nui loa!
23:08.54Qwellsflemming: how was dahdi installed?
23:09.36sflemmingI know that hfc-s does not provide a timer, therefore i would like to use dahdi_dummy. I built an ebuild for gentoo, so it is compiled from source
23:09.49Qwellwalks away
23:09.57*** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil)
23:10.00seanbrightheh
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23:10.48sflemmingdahdi_scan and dahdi_speed works when all the drivers are loaded, but dahdi_test shows there is no timing
23:12.36p3nguinI guess KaneHau doesn't care that it works here and I was going to pastebin the dialplan.
23:14.27sflemmingokay, i reloaded everything and now i have a response the very first time. but it looks like it is quite unreliable. sometimes i get a response, sometimes it simply hangs, sometimes i get a segmentation fault
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23:20.19chilicuilHi there, I'm trying to setup a local sip account and it doesnt matter what softphone I use the sound is very low, I've configured the sound in the local machine (with alsamixer) as high as possible but it doesn't change anything I've tried with diferent audio devices and looked at the asterisk log but I see nothing, so right now I'm a bit lost, any ideas?
23:22.47*** part/#asterisk bsaxon (n=bsaxon@12.68.234.174)
23:23.14krytzzhi, i am interested in using the fxs ports on the t-com speedport w701v on openwrt, i found no documentation if that is even possible, looking on the board a "cpc5621" chip is there and connected to the fxs ports, and i found the source for a driver, would it possible to use them?
23:24.46Gido-Ethe source is a kernel driver?
23:25.12krytzzyes, but only 300 lines of code
23:25.25Gido-Eok, normally there is a kernel part, and a asterisk part.
23:25.31krytzzbut i dont know if that is enough
23:25.38bmoraca_workyou'd be way better off in an openwrt channel
23:25.44krytzzah ok, sounds reasonable
23:26.12krytzzok, thank you, but can you tell me for the asterisk part, i would probably have to write it by myself?
23:26.29Gido-Echeck openwrt
23:26.31krytzzalso i am not sure if there are other asics involed with the ports
23:26.35krytzzok, thanks :)
23:27.10bmoraca_workkrytzz, that depends how you want to use them...are you trying to run asterisk on the device itself or do you want to use the device as a media gateway?  two very different scenarious that will require you to modify two different codebases
23:27.31krytzzbmoraca_work: i would like to run asterisk on the device
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23:27.50bmoraca_workyou'll need to build a dahdi driver for them
23:28.13krytzzmy setup would be: analog phone - speedport - iax to another asterisk
23:28.23krytzzok, thanks ill look into the source
23:33.02*** part/#asterisk ruied (n=ruied@bl7-217-61.dsl.telepac.pt)
23:35.52sflemmingdoes not work but will continue tomorrow, thank you for your help, bye
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23:49.26titterit's alive
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