IRC log for #asterisk on 20070406

00:04.12knarfly8-)
00:05.51*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-75-1-17-69.dsl.hstntx.sbcglobal.net)
00:08.44*** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)
00:09.04*** join/#asterisk neoalex (n=neoalex@user-0ccengj.cable.mindspring.com)
00:09.21codefreezeknarfly: use aelparse!!
00:09.24neoalexhi guys, can anyone recommend a good ATA to work with asterisk?
00:09.30neoalexgood as in sound quality
00:09.40mmartinnAnyone know why ChanSpy would say "NOTICE[28525] app_chanspy.c: Attaching SIP/152-50beaf80 to "Caller ID Name" <3525556666>" and then crash Asterisk, even though I don't *ever* set anything but outgoing caller id to that value?
00:10.07neoalexATA and Asterisk will be in the same lan so NAT traversal features are not a concern
00:10.41*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
00:11.51knarflycodefreeze: can you tell me more...how do I run it?
00:13.01CrashHDis ${UNIQUEID} built from the unix timestamp?
00:13.06CrashHDand if so why would it be wrong?
00:14.03mmartinnHow about this... How would ChanSpy(SIP/20${EXTEN:3}) *ever* output "app_chanspy.c: Attaching SIP/150-50b50458 to Zap/5-1"
00:14.29*** join/#asterisk rubber_chicken[] (n=blitzrag@CPE000fea3dbc27-CM0012c9db3d2e.cpe.net.cable.rogers.com)
00:14.43jqlbecause the SIP/xxx in the latter part usually contains the bit after the @
00:15.15jqlSIP/foo@example.com => SIP/example.com-deadbeef
00:15.35jqlsince you had nothing after the @, it did something even cooler
00:15.36jql:)
00:15.38*** join/#asterisk doug (i=doug@zaxxon.telerama.com)
00:15.43dougmmm
00:15.46mmartinnjql: Are you answering my question?
00:15.53jqlyes, mmartinn
00:16.29dougwhat's a "dream" asterisk setup
00:16.30doug?
00:16.40rubber_chicken[]doug: that one that works?
00:16.49mmartinnjql: ah, but how would someone every type anything but digits on these ATA's that would get ${EXTEN:3} to have that?
00:16.50rubber_chicken[]doug: you need to be more specific
00:16.59mmartinnjql: *someone ever, even
00:17.14jqlyou don't
00:17.45jqlit has to be in the dial plan, if anywhere
00:17.48mmartinnjql: I'm confused... I want people to only spy on channels that start with SIP/20, and ChanSpy(X) says that it will only chanspy on channels that start with X
00:18.01mmartinnjql: right, right... well, could this be a bug then?
00:18.25mmartinnjql: ChanSpy(SIP/20) ends up chanspying on things that are my outgoing callerid, which is definitely wrong
00:19.44jqlchannel names are supposed to correspond with the sip.conf [section]s
00:24.21d00gsterguys, I have a sip client  (eyebeam) overseas connecting to my asterisk. since they have high latency, I dud a qualify=5000 in one instance and =no in another. I also forced the client to register every 300 seconds (eyebeam option). the client can pickup the line and call me anytime  of the day. when I call the client, they don't see the call come in and I go to vm. I asked the  client to dial 7777 and dial his extension and tha
00:24.36knarfly;exten => s/,2,Goto(blocking,s,1)exten => s/_866,2,Goto(blocking,s,1)           ; Block Calls from 866 area code
00:24.36knarfly;exten => s/_877,2,Goto(blocking,s,1)           ; Block Calls from 877 area code
00:24.36knarfly;exten => s/_702818XXXX,2,Goto(blocking,s,1)    ; Block Calls from 702818XXXX
00:24.49knarflywhy doesn't this work?
00:25.23rubber_chicken[]what is the s/ for?
00:25.32rubber_chicken[]matching on CID?
00:25.52*** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com)
00:25.58knarflyYes, cept it ain't matching...it hangs up
00:26.00mmartinn_877 matches only "877", no?
00:26.06mmartinnYou'd have to dial only 877
00:26.25rubber_chicken[]mmartinn: correct
00:26.26doughm
00:26.31douga "dream" setup
00:26.32rubber_chicken[]same with _866
00:26.46knarflyokay I can change that but with these lines uncommented it hangs up and doesn't answer
00:26.53rubber_chicken[]need at least _877! and _866!
00:26.56dougguess that'll be highly application-dependant
00:27.03doughow 'bout for a personal setup?
00:27.13rubber_chicken[]doug: exactly -- vPBX system is gonna be different than a home setup
00:27.25rubber_chicken[]doug: any P3 will do what you need for home
00:27.42dougor, even better, for a small CS operation...
00:28.33dougwhat's The Absolute Coolest VoIP hardware handset?
00:28.33dougstandalone..
00:28.33rubber_chicken[]you're asking the wrong question
00:28.33*** join/#asterisk wax408 (n=matchbox@chello084113018116.7.12.vie.surfer.at)
00:28.34rubber_chicken[]it's not about being cool -- it's about functionality
00:28.34rubber_chicken[]~phones
00:28.40jbotextra, extra, read all about it, phones is http://bani.anime.net/phones/.  SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom
00:28.40doughah.  screw functionality
00:28.40dougi want Hip and Happenin'
00:28.42rubber_chicken[]that's the whole point of this VoIP thing
00:28.42dougGroovy and Outta Site
00:28.42rubber_chicken[]then get a 7970
00:28.45wax408hello...is there a way to change the payload type in rtp packets on asterisk?
00:28.54dougsuggestability?
00:28.58douglike, you can talk them into doing stuff?
00:29.03rubber_chicken[]yes
00:29.08hadsHah, who's opinion of order of quality is that.
00:29.13rubber_chicken[]one let me do things to it you don't want to know about
00:29.21dougactually, what i *really* want is a nice wireless headset that does sound like ass
00:29.23rubber_chicken[]hads: the general population
00:29.29hadsRiiight
00:29.34dougi don't care if it's bluetooth, as long as it works.
00:29.47rubber_chicken[]Polycom is rated highest generally
00:29.51dougi'd love to have something that would be barely noticable, wearable all day, and won't fall off my ear.
00:30.02hadsYeah, I agree Polycom make nice phone
00:30.04rubber_chicken[]doug: get something BT then
00:30.04hadss
00:30.09rubber_chicken[]Aastra makes good phones too
00:30.16rubber_chicken[]in fact, I own almost every one of those phones
00:30.18hadsYeah, they aren't too bad.
00:30.24dougbt for phones or headsets?
00:30.33hadsI wouldn't put the snoms last though
00:30.46rubber_chicken[]I would -- they seem to only be popular in Europe, and the handset is crappy
00:30.57rubber_chicken[]I'd place the Mitel 5220 just behind the Snom because of the same issue
00:31.05hadsI've sold loads of them and people are really happy with them.
00:31.29rubber_chicken[]the fact the Snom is listed means it is suggested -- not unsuggested
00:31.37rubber_chicken[]notice the lack of Grandstream for example
00:31.41rubber_chicken[]it's what works
00:31.48rubber_chicken[]because the handset is useless on the Snoms
00:31.51hadsThe Linksys are OK, but you can't do much with them unless you use the SPA9000
00:31.54rubber_chicken[]impossible to hold to your ear
00:32.03hadsWorks for my ear :)
00:32.13Dirk-it is a little large
00:32.15rubber_chicken[]I have the Linksys SPA-942 -- only bad thing about it is the speakerphone
00:32.35rubber_chicken[]I could never hold the Snom headset on my ear for 3 hours while programming
00:32.46rubber_chicken[]Cisco has the best handset
00:32.53Dirk-well, for 3 hours, you want a headset, not handset
00:32.55hadsYeah, The SPAs aren't too bad. But like I said you can't do anything much with them.
00:33.13rubber_chicken[]what do you need to do with them other than transfer and make calls?
00:33.31rubber_chicken[]the less things it does, the less things there are to fuck up
00:33.46hadsYou've got those line keys that are just itching to have something done with them but you can't unless you use the SPA9000
00:33.59Dirk-doug, does it have to be an ip unit, or would an analogue device suffice?
00:34.06rubber_chicken[]separate line keys seem to work fine for me
00:34.24hadsWhereas with the Polycom or snom etc. you can watch other extensions and whatnot
00:34.46rubber_chicken[]anyways, I'm going to watch the hockey game
00:34.46rubber_chicken[]lates
00:36.46Dirk-doug, I have personal experience of the following link, and for quality, range and comfort, it hits all with a great score: http://plantronicsheadsets.evocal.co.uk/index.cfm?event=catalogue.product&productID=15363&categoryID=0
00:40.00dougi'm checking out the GN 9350 ...
00:40.09dougnot sure about DECT in the US tho
00:41.02dougi remember once reading about an (experimental) headphone system where they would glue a tiny bit of ferrous material to your eardrum
00:41.03Dirk-why's that?
00:41.21dougyou'd then wear a magnetic coil around your neck, low like over your shoulders under your shirt
00:41.40dougsupposed to have a very good frequency range
00:41.54doug4-40KHz or so
00:41.55*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
00:42.16*** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net)
00:43.43dougthat looks nice tho
00:43.49Dirk-cool, outside of the human range.... useful
00:44.19*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
00:45.48hadsheh
00:47.00dougthere's some debate about that
00:47.13dougdunno that <20hz inside your ear would be detectable tho
00:47.44*** join/#asterisk cr4z3d (n=cr4z3d@168.158.222.2)
00:47.57dougthis looks kinda like what i had in mind: http://www.geocities.com/examear/elite.html
00:48.45*** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org)
00:49.52douggonna take a chance on nit...
00:50.29dougthen buy that cool fan-based 8Hz subwoofer driver...
00:51.32Dirk-well, that's a geocities page, which bodes less than well, bus aside from that the device looks technically feasible.  Regardless, what looks worse, a borg style boom mike or a hippy style pendant?
00:51.52Dirk-each to their own, but I'd buy plantronics sooner than an unknown make
00:53.05hrmphhanyone have an example using AEL with gotoiftime?
00:53.36Dirk-plus, you need to factor in the discomfort and audio isolation problems of a canalphone device, some people find them nice, others find hot pokers in the eyes better than having a plug deep in their ear
00:54.28hadsI'd buy a known brand from a known store myself too.
00:54.33codefreezehrmphh: Isn't there one on the AEL2 voip-info wiki?
00:54.39codefreeze~wiki
00:55.57Dirk-Reading further, the device is an induction loop driven unit, the bluetooth or other radio function is driven from the pendant, this looks like it would be very compatible with hearing aids, which is a plus, but other than that...
00:56.37Dirk-bluetooth is 10m (or the alleged 100m) whereas dect is (also supposed to be) 1000m
00:57.01hadsGAP isn't very popular outside Europe though unfortunately
00:57.10Dirk-frankly, I'd pay no more than $40 for that thing
00:57.35Dirk-hads, yeah, but for a single device, does it matter?
00:58.20*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
00:58.32hadsNot at all, just that they access point (or whatever they are called) may be darer. Or is it provided as part of the unit?
00:59.40hadsYeah, looks like you would need a GAP base station, which may be dare - I'm not sure.
00:59.54*** join/#asterisk l0rdr0ck (n=l0rdr0ck@adsl-70-231-138-121.dsl.snfc21.sbcglobal.net)
01:00.07hrmphhyeah found iftime()
01:01.58polerinok stupid newb question. what's the * development cycle nominclature.  I'm more used to debian/linux kernal terminology... would 1.2 be stable or would 1.4?  Looked but can't actually find any notation on that on asterisk.org :P
01:02.07Dirk-hads, unit I posted is a complete analogue device, base stn and headset
01:02.11*** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net)
01:02.28Dirk-polerin, good question
01:02.28hadsDirk-: Doesn't appear to be from the description.
01:03.05hads"designed to be registered to GAP* compatible DECT bases or wireless PBXs, which is supplied with a fast desk charger and headset holder"
01:03.06Dirk-had's, you may be right, I actually thought it was cheap, last one I bought was closer to £180 (~$360)
01:03.48Dirk-polerin, I'll go on a limband say 1.2 branch is considered stable and 1.4 branch is considered 'good'
01:06.50Dirk-hads, they are expensive, I'll concede to that, but I;ve used them (in business, cant afford one for home) and they are really very good indeed. You get what you pay for with these things
01:07.03polerindebian is still distributing 1.0 with sarge
01:07.04hadsOh I agree
01:07.20polerin:P
01:07.21hadsThis looks like it might be nice http://www.gnnetcom.com/US/EN/MainMenu/Products/Wireless+Solutions/GN+6210.htm
01:07.31polerinthanks dirk.
01:07.33Dirk-checking...
01:07.51Dirk-ooh, pretty!
01:08.46Dirk-ssure looks nicer than the plantronics, likely to be the same internally, should perform as well. not sure about the in-ear design though
01:09.15hadsYeah. It appears to be a bluetooth base station that has a rj11 plug to connect to your deskphone.
01:09.20hrmphhwould be neat if asterisk had an ifTodayIsNYSEHoliday() func :)
01:09.22Dirk-but again, bluetooth, gonna be great if you dont need to move far from your desk, in which case, would a long curly cord not be cheaper?
01:09.51hadsDirk-: True, but that appears to be able to handle two seperate bluetooth devices.
01:10.00hadsWhich is kind of cool
01:10.23Dirk-ooh cool
01:10.53Dirk-ah, I see how it's doing that
01:11.58Dirk-the bt is in the base station, so it's simply two chips or a nice bt subset so it can link to more than one device, the kicker is, can it signal the second to the headset, in other words can it do call waiting for the second device?
01:12.11Dirk-I doubt it, but it's not really an important feature
01:12.43hrmphhcan you do ifTime(blah) {} else {}?
01:12.44Dirk-That's the nice thing with this market, there are a great many ways to accomplish the same thing
01:13.16Dirk-at the end of the day though, it's still a 10m range, with aint that far
01:13.28hadsIndeed.
01:13.44Dirk-fine for home use though
01:13.56polerinwhats the usuall range for the ear hook BT headsets?
01:13.57hadsMy bluetooth here can do ~40M
01:14.14Dirk-but for business use, with modern mezzanine offices that can be over100m in any direction, you;d be luck to reach the coffee machine!
01:15.02hadsThat's from a class 1 (100M) dongle to a class 2 (10M) device.
01:15.07Dirk-hey, sure, bt can do a theoretical 100m
01:15.21Dirk-great for data
01:15.42Dirk-but when it comes to walking round an office, I'll take 1000m theory over 100m theory any day:)
01:15.54hadsNo need to convince me :)
01:16.08Dirk-dect will give around 220m in real world, and bt, as you prove, 40m
01:16.14Dirk-oops, sorry :)
01:16.16Dirk-ranting
01:16.19Dirk-:D
01:16.29polerinI'm actually helping my wife set up her home buisness, and so I'm looking at doing a softphone+a BT headset for the inital phone, so I'm looking at options :P
01:16.33InnatechI just read a bunch of scrollback. A couple remarks:  (1) Snom has been good to me, so far.  (2) I have yet to find a BT headset that I think is decent in terms of features/quality that isn't totally obtrusive and/or ergonomically terrible.
01:17.01polerinwe already have a headset running around from an old cell, so I was hoping to go with that
01:17.03Dirk-Innatech, snom.....  I have to ask....
01:17.19hadsInnatech: Yeah, I like snom phones too. The general channel population doesn't seem to much though.
01:17.19Dirk-when you hang up the handset, does it 'fit' nice?
01:17.33InnatechDirk: my needs are not elaborate. I find them totally acceptable thus far.
01:17.34Dirk-try the aastra 57i and see how that 'feels'
01:17.51hadsIs the 57i IP?
01:17.55Dirk-I tell ya, the snom is stubborn and difficult compared to the smoothness of the 57i
01:18.09Innatechhads: I agree, they don't have the most stellar rep. I understand that their earlier builds of some models were really buggy, which might have something to do with the negative opinions from the vets here.
01:18.12Dirk-yeah
01:18.21hadsInnatech: True
01:18.30Dirk-dont get me wrong, I LOVE the way the snom's look
01:18.34hadsDirk-: Interesting, I haven't seen it in this part of the world.
01:18.45Dirk-but in practice, they just dont, well.... work
01:18.57mmartinnDoes the Dial command return when the call ends, or when it is bridged?
01:18.59hadsBut then we are about the last to get most things :)
01:19.04Dirk-hads, the aastra 57i range is seriously new
01:19.18*** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net)
01:19.22Dirk-I have one one trial and that arrived whiteboxed and is a demo model
01:19.29hadsLooks nice. Kinda polycomish in design
01:19.33InnatechWell, I haven't tried to make them jump though many hoops yet.
01:19.38Dirk-very, yes
01:20.01Dirk-but the handset piece in itself is enough to make me class that as the 'one'
01:20.15Dirk-I'm trialling many for work, and so fact, well 57i's all round
01:20.36hadsThe thing I like about the snoms is they are quite configurable quite easily. I must get onto my distributor about those new Aastras :)
01:20.39Dirk-fact = far*
01:21.01Dirk-gods, aastra is so much easier to configure than snom
01:21.12Dirk-but  that mayjust be me being dense, of course
01:21.21fluffyfluffydirk: Just popped in in the middle of your conversation. I'm in need of opinions for sip wireless handsets. which handset are you talking about?
01:21.33Innatechno, the snom setup could be a little more intuitive. I agree with that, for sure.
01:23.33Dirk-fluffyfluffy, wired, so far!
01:23.49fluffyfluffyahh. ok :)
01:24.09Dirk-although there is a 57i CT which is the 57i base with a dect phone 'attached'
01:24.21InnatechHere's a kinda offtopic equipment question: anyone know of one of those POTS basestation/speakerphone + wireless extension combo systems that (a) doesn't have an answering machine (b) uses handsets of a comfortable size (most are too small) and (c) doesn't suck? I could use one to slap into a jack on my ATA at home, rather than using the house wiring to distribute dialtone (which makes me nervous since I never disconnected it from t
01:24.22Innatechhe TNI panel--can't find the pair.)
01:24.26hadsLike the old 480iCT
01:24.31*** join/#asterisk _lanlv_ (n=chatzill@58.186.172.21)
01:25.00Dirk-exactly
01:25.34hadsI can't wait to hassle the distributor after easter now :)
01:25.52polerinInnatech: ugh.  tell me about it... my house wiring is attrocious.
01:25.56Dirk-hads, I troubled mine for a weeks!
01:26.04techiepolycom is going to release some wireless handsets this year
01:26.06fluffyfluffy$380!!!
01:26.32InnatechYeah...I could tolerate the poor quality, but I *really* don't like having a network device attached to circuitry that the telco could theoretically light up like a xmas tree if the urge so moved them.
01:26.33Dirk-eventually he sent me one, and like I say, it was 'used' already, I'm convinced  that I'm on of very few in the UK with a 57i
01:26.34polerinI need to do the cable too, but I've got way more experience with that :P
01:26.48hadsDirk-: What's the pricing like compared to the old models? If you know.
01:26.57Dirk-pretty much the same
01:27.01polerinInnatech: ... can you not get into the demark?
01:27.01hadsnice
01:27.05Dirk-arounf £150 per unit
01:27.09Dirk-ish
01:27.30hadsPricing all gets out of whack by the time it gets over here.
01:27.31Innatechpolerin: I can, but I'm in a sizable condo building and the pairs aren't marked. I don't have any equipment to identify it.
01:27.38Dirk-seriously though, it arrived and I'm like oh yeah, that's the one
01:27.44Dirk-white back light too
01:27.49hadsnice
01:27.52Dirk-you know that's a deal clincher!
01:28.12polerinoh
01:28.13polerinheh
01:28.43Dirk-hads, if you can lay your hands on one for a trial, do so. It's completely different to the 480i
01:28.53polerinyou could always do what I did when I was out in the field for comcast....  have someone inside and talk, then disconnect one at a time :P
01:29.01*** join/#asterisk pkey (n=pkey@216.248.143.76)
01:29.11Dirk-and yeah, I know, I sounds like a bloody aastra rep, but....
01:29.12Dirk-bah
01:29.23hadsDirk-: I sell phones retail as part of my business so I'll be getting some in to play with as soon as I can.
01:29.50Dirk-then you needto nag and nag, seeing as the US should have them before the UK
01:29.56InnatechHeh. My neighbors might not like that. And, anyway, since I don't have POTS, I'd need to attach a handset to each pair and listen....which is *koffkoff* not entirely legit.
01:30.04hadsDirk-: I'm in NZ :)
01:30.16polerinInnatech: no no no, call them on a cell
01:30.26Dirk-ah, then you are......... screwed
01:30.28Dirk-:)
01:30.35hads:) Yeah
01:30.36Dirk-(joking, of course)
01:31.03polerinInnatech: of course this also depends on something else... who's owns the wiring to your condo?
01:31.17Innatechpolerin: the HOA, which I'm a member of.
01:31.34*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
01:31.36polerinInnatech: with condo's it's always weird, because you have to traverse other people's property half the time
01:31.38hrmphhis there an example dial by name extensions example?
01:31.45Dirk-hads, it's my first day in the channel, I carry precisely zero weight, but trust the random guy on the internet, get hold of a 57i for testing, you will justlove it right out of the box
01:31.45hrmphherr one too many "examples"
01:32.13hadsDirk-: heh, it's always good to trust random people on the 'net.
01:32.18Innatechpolerin: the TNI is in a common storage closet. The problem is that since I don't have POTS, I'd have to attach to each pair to listen for a convo on my internal wiring. Disconnecting my lines from the TNI won't cause any noticable change from my end.
01:32.21Dirk-:)
01:32.39Dirk-hrmphh, dial by name? well.....
01:32.47Dirk-there are scripts for that yeah
01:33.00hrmphhDirectory() is what i need heh
01:33.17polerinInnatech: oh, hmm.. well you could still do another version of the same thing.  They still give you dialtone right?
01:33.26Dirk-go have  look at the scripts offered by freepbx, iirc that has the function built in
01:33.29Innatechpolerin: nope. I provide dial tone from my ATA.
01:33.43Innatechpolerin: If I had dialtone, I never would have plugged in my ATA to the same copper.
01:33.57polerinnot what I meen
01:34.10polerinif you plug a phone into it, you still can dial nine eleven right? ;P
01:34.35Dirk-depends where you are
01:34.37Innatechmm...over POTS? No. The pair is *dead*.
01:35.06polerinhmm.  I thought they had to provide 911 regardless.  **shrugs**
01:35.23polerinok I'll hush now.  I can do cable, but phone i'm not used too ;P
01:35.26Dirk-could be a fun test
01:35.42fluffyfluffyAnyone have opinions on the UTstarcom F3000 + asterisk. Or UTstarcom products in general?
01:35.50Dirk-but it's 999 here. Regardles though, it;s not the best way to test a line is live
01:36.05*** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net)
01:36.10hrmphhany repositories of extensions.conf and .ael?
01:36.14hrmphhjust want to see some examples
01:36.23Dirk-actually, I lost sight of the original question
01:36.25Innatechhehe. Trust me, the pair is dead. I'm just concerned that some voltage might come across it unexpectedly (testing at the local exchange, a bad tech working on the panel, etc.) So I'd prefer to ID the pair and physically tape over the contacts, attach a sign, etc.
01:36.36hadshrmphh: there's the config examples in the source
01:36.49fluffyfluffyhrmphh: Did you check out the "make samples" ?
01:36.52hrmphhhads; beyond that
01:37.00hadsSearch the net
01:37.02Dirk-Innatech, a decent precaution
01:37.17polerinInnatech: yeah trust me I understand.  I've been a field tech ;)
01:37.42Dirk-Innatech what is it you are trying to do, find the pair at the DP that feeds (or fed) you rline?
01:37.45hrmphhwhats the 'a' extension?
01:37.52InnatechDirk: yeah. I just don't know how to isolate it w/o real equipment, or attaching a field handset to each of the lines in the building, which is not-so-nice.
01:38.00InnatechDirk: yes, exactly.
01:38.13Dirk-your gonna need a tone & anp unit really
01:38.30Innatechyeah. Which I'm not going to shell out for just for peace of mind on my home LAN.
01:38.40hadshrmphh: *
01:38.50polerinwell.. you could actually do it by putting a know ammount of impedence in line and look for that
01:38.56Dirk-do you have a decent suspicion of which pair may be yours?
01:38.56hrmphhthanks
01:39.02Dirk-or is it completely unmarked?
01:39.12InnatechDirk-- not really. The TNI is a gawd-awful mess.
01:39.26Dirk-ok
01:39.36Dirk-well, you could ask the telco to id them for you
01:39.43InnatechYeah, I might do.
01:39.45Dirk-or build your own amp (easy)
01:39.56InnatechIs it? That's not a bad idea.
01:40.04Dirk-or put 9v down the pair and meter it at the dp
01:40.14Innatechhrrm. That's an even better idea.
01:40.32InnatechThanks!
01:40.40Dirk-or short your own pair at the inbound point and isolate yourself from the building wiring completely
01:40.51Dirk-or... or.... or... :)
01:40.59Dirk-we have our ways :)
01:41.10polerinDirk-: Or put resistance on the line and look for... yeah
01:41.10InnatechVE HAFF VAYS OF ISOLATING YOUR PAIRS!
01:41.11polerinhehe
01:41.37*** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br)
01:41.38Dirk-I've done things in my time :) I used to install analogue pbx's
01:41.40polerincounterstrike time :)
01:41.42Dirk-back in the day
01:41.50Dirk-C&C 3 ?
01:42.08polerinDirk-: I just do coax :P
01:42.09InnatechHeh. I was just explaining to some higher ups why its not a good idea to pay someone to move our ancient Nortel *again.*
01:42.29InnatechEven if I do like the desk sets....*sniff*
01:43.55[TK]D-FenderNorstar = Dionsaur
01:44.05hrmphhso if you Dial() from somewhere in your dial plan, and you dont catch the ${DIALSTATUS}, what happens if someone picks up?
01:44.07*** join/#asterisk codazoda (n=chatzill@ip69-223.konnections.com)
01:44.08[TK]D-FenderDitch cat 3 and through cat5e+ all around.
01:44.13hrmphhit just stops executing next priorities/
01:44.16Dirk-mmmmm, nortel/meridian, those things were my bread & butter
01:44.35[TK]D-Fenderhrmphh: Typically if they pick up there is nowhere to continue TO.
01:44.43hrmphhok tk
01:44.45hrmphhso execution stops there
01:44.49mmartinnHow does ChanSpy define a bridged call? Would a non-voice call count?
01:44.50[TK]D-Fenderhrmphh: And hence dialstatus is kinda irrelevent
01:44.57InnatechThey still work nicely....I just can countenance the expense to move them, or for all the physical lines that feed it.
01:45.01hrmphhwell
01:45.01[TK]D-Fenderhrmphh: Unless you tell Dial to do otherwise
01:45.06hrmphhthere is a DIALSTATUS ANSWER
01:45.23[TK]D-Fenderhrmphh: And GUESS what you have to do to have hopes of SEEING that? ;)
01:45.25Innatechthe Meridian desk phones really are nice, too. Nightmarish to configure, but they feel good.
01:45.35[TK]D-Fenderhrmphh:  Starting hint : "show application dial"
01:45.35Dirk-yeah, heavy
01:45.54Dirk-like the snom actually, good to hold
01:45.59Dirk-but crap to hang up
01:45.59[TK]D-FenderInnatech: Meridian = block plastic crap.  Durable which feeling crappy.
01:46.09Innatechyeah, and big neck/shoulder friendly handsets. I just don't understand the incredible shrinking handset phenom.
01:46.10[TK]D-Fender~phones
01:46.25jbotphones is, like, http://bani.anime.net/phones/.  SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom
01:46.26InnatechDirk: yeah, the feel is one of the better qualities of the Snoms.
01:46.26[TK]D-Fender^- Read & Obey
01:46.35Dirk-telling ya, aastra 57/55 all the way :)
01:46.40Dirk-it's gonna be huge
01:46.42hadsOr just make your own opinions.
01:46.48[TK]D-FenderDirk-: Actually... NO.
01:47.00[TK]D-FenderDirk-: The 57i CT has some serious piss-off flaws, like the rest of their line
01:47.05Dirk-I love the snom hand piece, it's just a shame I hate the phone
01:47.08InnatechD-Fender: >shrug< I've always liked em, myself. But, yes, they are generic black plastic.
01:47.13[TK]D-FenderDirk-: Still higher than most though.
01:47.19Dirk-fender, name one
01:47.27Dirk-please, I'm about to buy 50
01:47.44[TK]D-FenderInnatech: 2 line shit display, and too light/blocky but otherwise functional.
01:48.16InnatechFender: kee-rist, we bought 'em around....crap...'92? '94? c'mon now.
01:48.36Innatech2 line display was HAWT.
01:48.49[TK]D-FenderDirk-: I hould name a BUNCH.  only 7 chars for soft-key labels, RUBBER FUCKING KEYS!!!, If you try to configure a handset thinking it'll be "independant" of the base, FORGET IT, itll RING ON THE BASE.
01:49.47[TK]D-FenderDirk-: And a few other points if you want me to continue.
01:49.47Dirk-please do
01:49.47[TK]D-FenderDirk-: For general use Polycom still wins hands down.
01:49.47Dirk-:)
01:49.48Dirk-I'm in two minds about the polycom, I guess I need to buy one and test it
01:49.48[TK]D-FenderDirk-: I haven't found how to do attended VS blind transfer, the process of flowing through the keys to take such actions sucks a bit.
01:49.59Dirk-whats what closest poly to the 57i ?
01:50.17mmartinnCan someone define a bridged call for me?
01:50.28[TK]D-FenderDirk-: Poor use of LCD (pixel based, but they still think in char matrix mode.  Usability expert should BEAT THEIR ASSES UP)
01:50.43Innatechdinner time. cyas, all. :)
01:50.44Dirk-that's true
01:50.53Dirk-it's a gfx display in char mode
01:50.55[TK]D-FenderDirk-: What kind of call volume is this post going to need to handle?
01:51.17[TK]D-FenderDirk-: And are you planning all posts to be about the same needs?
01:51.27Dirk-he phones? 50 -60 handsets in the bulding, with 13 of them taking 2000 call per day
01:51.38Dirk-the rest is low outbound
01:52.07[TK]D-Fendermmartinn: Bridged = 2 channels bridged together like a SIP-SIP call, SIP->ZAP, etc.  if you are just running the Voicemail app for instance, that is not "bridged, its just YOU -> Asterisk
01:52.39[TK]D-FenderDirk-: How many simultaneous calls ON a phone?  Need speakerphone?  PoE?
01:53.13[TK]D-FenderDirk-: Multiple reg's per phone, or just multiple calls on a single reg?
01:53.23Dirk-poe, not required. speakerphone, not on the high volume xtn's, only pne call per phone
01:53.42Dirk-single office use, call ctr environment for the high volume phones
01:53.47[TK]D-FenderDirk-: Nee pass-through for PC?
01:54.00[TK]D-Fenderneed*
01:54.01Dirk-seperate app used for CTI
01:54.20Dirk-so, no, but handy if available
01:54.28[TK]D-FenderDirk-: to clarify, do you want an extra ethernet port so you can plug it IN-LINE with your PC or will if have an independant jack?
01:54.34Dirk-current phones connected over cat5 anyway
01:54.36mmartinn[TK]D-Fender: Would TAPI device speaking SIP that calls out with a ZAP channel be considered bridged? Or only voice calls?
01:55.24[TK]D-Fendermmartinn: Yes... * is bridging your TAPI's SIP to Zap.
01:55.25Dirk-[TK]D-Fender, go with, no. A switched port on the phone is not a requiremtn
01:55.41[TK]D-FenderDirk-: Then your star budget choise = Polycom IP 320
01:55.52Dirk-compar to a 57i
01:55.53mmartinn[TK]D-Fender: I think my ChanSpy crashes when it hits one of those calls; is there a creative way to avoid that?
01:55.56Dirk-+e
01:56.06[TK]D-FenderDirk-: 57i is MASSIVE overkill
01:56.17[TK]D-FenderDirk-: And massively more expensive
01:56.22*** part/#asterisk codazoda (n=chatzill@ip69-223.konnections.com)
01:56.24Dirk-let me check a local price......
01:56.32[TK]D-FenderDirk-: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-54293451520.htm
01:56.34[TK]D-Fenderthere
01:56.50*** join/#asterisk codazoda (n=codazoda@ip69-223.konnections.com)
01:56.52[TK]D-FenderDirk-: Ah.. a brit :)
01:57.09[TK]D-FenderDirk-: Right proper and all... well... polycom IS more expensive wher you are for sure....
01:57.20hrmphhdo i need to convert wav to gsm or can i simply place recorded wavs in /var/lib/asterisk/sounds ?
01:57.36[TK]D-FenderDirk-: if you want to be frugal about it, Linksys may do if you value money over quality.
01:57.49Dirk-actually, polycom 320 is close to half the price of the 57i
01:58.00[TK]D-Fenderhrmphh: * can transcode most normal wav formats
01:58.03Dirk-but I work in a blame culture
01:58.13Dirk-so if it's wrong it'm my fault
01:58.14fluffyfluffyhrmphh: I use gam. sox <sound>.wav <sound.gsm
01:58.20fluffyfluffygam=gsm
01:58.22[TK]D-FenderDirk-: Its 40% of the price in North America
01:58.24Dirk-so I'd rather over spec than under spec
01:58.49hadsWhat about the 430 then?
01:59.01hrmphhthanks
01:59.03[TK]D-FenderDirk-: can you link an on-line retailer for me to evaluate for you?
01:59.19[TK]D-Fenderhads: IP 430 is a rare phone to suggest now.
01:59.19Dirk-that's what 3 line lcd display?
01:59.34Dirk-www.myphonecall.co.uk
01:59.39hads[TK]D-Fender: I see. Has something replaced it already?
02:00.04hadsOr just that the 320 is better value
02:00.06[TK]D-Fenderhads: Depends on what you would consider a "replacement".  It is still a "current" product.  It hasn't even been out for a year now...
02:00.07Dirk-I understand the tech req's but I have to factor in the comany's oddness
02:00.30[TK]D-Fenderhads:  430 supports more call per line-key, bigger display I believe, and 4 soft-keys
02:00.47hadsI'm not really up on the newer polycom kit as we are slow to get it over here.
02:01.03hadsAnd I hardly sell any unfortunately
02:01.06Dirk-57i has a huge display (you try to convice a sales advisor that t aint so)
02:01.20Dirk-I work in an odd place !
02:02.08Dirk-considering what we would save going asterinsk rather than cisco or mitel et all, the phone cost is close to irrelivent
02:02.28Dirk-we were quotes £65,000 for 45 handsets
02:02.45Dirk-and that was without all-line call recording function
02:02.49[TK]D-FenderDirk-: Ok.... tell you what... the IP 430 has power bricks and 4 soft-keys, and still supports PoE.... might be a good idea.
02:03.12[TK]D-FenderDirk-: Or... actually you know....
02:03.22[TK]D-FenderDirk-: I'd sooner pump the $ into a PoE Switch
02:03.24[TK]D-FenderBRB, 5 mins...
02:03.28Dirk-k
02:05.50Dirk-the ploy 430 compared to the astra 5i, well, there's no comparison, the astra wins for the small amount of extra cash. and both can be driven from PoE
02:06.34Dirk-given the enviromnet I work in, if you know of a phone more over specced that that 57i, then we'll likely buy it !!
02:07.42[TK]D-FenderDirk-: What are your users going to do with the phone?
02:07.58[TK]D-FenderDirk-: Do they need to call tons of people or monitor who's on the phone ?
02:08.21Dirk-around 10-15 will take 200 calls per day and make 1000 outbound
02:08.41[TK]D-FenderDirk-: But not heaving INTER_OFFICE use tracking people, etc?
02:08.46Dirk-the other 30 will take and make perhahps 500 beteen them
02:08.47[TK]D-Fenderheavy*
02:08.55*** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net)
02:09.13Dirk-perhaps 100 internal calls over all 50 phones
02:09.19Dirk-1000*
02:09.40Dirk-ugh, take 2000* calls per day
02:09.54[TK]D-FenderDirk-: OH... yeah forgot to tell you the 5I series handset has NO weight to it and pisses me off.  I run an almost all-polycom office, am 100% polycom @ home, and have recently swapped my work desk for a 57i CT.
02:09.55Dirk-typo's are getting worse, it must be time for sleep!
02:10.20Dirk-I have one on my desk
02:10.43Dirk-and yeah, it is a little light
02:10.53[TK]D-FenderDirk-: As Polycom is cheaper, Polycom's call-handling capabilities really beat Aastra if you aren't doing stuff like Presence, lots of in-call dynamic recording etc.
02:10.56Dirk-but the high volume users will be on headsets anyway
02:11.05Dirk-right
02:11.23Dirk-I'll order up a polycom on tuesday (holidays etc) and see what it' like
02:11.34[TK]D-FenderDirk-: There's reasons for both....
02:11.35Dirk-you recommend a 430?
02:11.45hrmphhhmm any idea why wthe system is trying to send me to w-80-myextension for voicemail?
02:11.48Dirk-what' sthe best?
02:12.08Dirk-polycom 601?
02:12.59Dirk-the 501 looks like it's enought
02:13.01hrmphhhttp://www.pastebin.ca/426322
02:13.20Dirk-I'll get one of thse ordered in and see what it's like
02:13.22hrmphhit should be sending to 20104
02:13.26hrmphhnot w-80-20104
02:13.27hrmphh?
02:13.57Dirk-omg, it's gone 3am
02:14.01[TK]D-FenderDirk-: Everything depends on usage & budget
02:14.24Dirk|sleepin our work, it all depends on function
02:14.28Dirk|sleepbudget be damned
02:14.35[TK]D-FenderDirk|sleep: 501 is nice, but 430 offers PoE.  501 is a better PHONE though.
02:14.52[TK]D-FenderDirk|sleep: no sane user needs anything more than that...
02:15.02Dirk|sleepthat's not to say we have money to burn, more of a case of we must not allow a tight budget to comprimise function
02:15.13[TK]D-FenderDirk|sleep: Its a great phone (for the record I have a 301, 430, 501, 600, and 601)
02:15.15hrmphhor bad spelling to compromise your point :)
02:15.27[TK]D-FenderDirk|sleep: And am planning on gettin my hands on a 320 or 330 soon.
02:15.41Dirk|sleepok
02:15.50Dirk|sleeptell me a model to order, 501 ?
02:16.02[TK]D-FenderDirk|sleep: If you aren't doing dynamic features, you should focus on quality.  So its a matter of choosing the RIGHT polycom now.
02:16.11hadsheh
02:16.43Dirk|sleepmeh, fine, a 501 will be on my desk wednesday next week
02:16.44hads[TK]D-Fender is the resident Polycom freak ;)
02:16.58Dirk|sleepI'll rip t to pieces from that point :)
02:17.08hrmphhwhy would it send vm to W-80-myexten?
02:17.09Dirk|sleepfor now, my pillow calls :)
02:18.53*** join/#asterisk zotz (n=zotz@24.244.163.157)
02:19.22zimdogI am playing around with an IAX trunk between 2 * servers. I am wondering if there is a way to pass the caller name along withth e number to the second server. Right now I see the phone number only. I am using Macro(dialout-trunk,6,${EXTEN},,). Anyway to do this?
02:22.35*** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)
02:22.52[TK]D-FenderDirk|sleep: Note this upon return : 501 will COST to go PoE afterwards... make sure thats in your game plan.
02:23.04[TK]D-FenderDirk|sleep: The only real downside.
02:23.21[TK]D-Fenderok, BBIAB.. heading home.. worked late..
02:23.24*** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177815820.dsl.bell.ca)
02:23.28hadsLater
02:23.35[TK]D-Fenderhads: 20 min
02:23.36[TK]D-Fender:)
02:25.55fab5freddyI keep getting WARNING[3842]: pbx.c:1720 pbx_extension_help: No application
02:26.05fab5freddyCan somebody shed some light as to how to fix this?
02:26.50fab5freddyAlso getting chan_sip.c:3654 process_sdp: Unknown SDP media type in offer
02:29.18*** join/#asterisk tvietduc (n=chatzill@58.186.172.21)
02:30.05fab5freddyClearly I can communicate with Asterisk via my softphone software but part of the error is No application ' Dial ' for extension (home, 101, 1)
02:30.44fab5freddyWhen I use an extension that doesn't exist my softphone tells me user not found
02:31.02*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
02:34.03*** join/#asterisk hansin321 (n=hansin32@c-67-174-180-72.hsd1.co.comcast.net)
02:34.13*** join/#asterisk ecam (n=ecam@bb121-6-58-13.singnet.com.sg)
02:34.57ecamhey people, anyone know how to manually fix the "Interval too short" problem without using the patch?
02:35.12ecamas in, manually set the interval in some config file or something
02:36.47ecamokay, no one's around again, please reply asap, thanks!
02:37.51fab5freddyecam: have you ever dealt with the error pbx.c:1720 pbx_extension_help: No application ' Dial ' for extension (home, 101, 1)
02:39.19wunderkinfab5freddy, well, looks like you have some extra spaces
02:40.03ecamwhat about my problem?
02:40.41*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
02:41.27[TK]D-Fenderand... WHEE!!!!!!
02:41.37wunderkinWwwwAAaaVeeeee
02:41.59ecamwah-ah?
02:42.21wunderkinhave you never done the wave?
02:42.49ecamokay, i get the point!
02:42.55wunderkinheh!
02:43.10ecambut what bout my problem?
02:45.32[TK]D-Fenderecam, www.drphil.com
02:46.26*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
02:47.56ecamah hah... so i can ask this guy any question!
02:49.06*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net)
02:51.27[TK]D-Fenderecam, ATM you can ask me questions for free....
02:54.31ecamwell, does my question qualify? i'm really kdesperate you know
03:03.06[TK]D-Fenderecam, Well I missed the question..
03:03.17ecamokay...
03:03.42ecami'm getting the 23 interval too brief error
03:03.45ecam*423
03:04.13ecamproblem is, i'm using asterisknow, which it aint easy to apply the patch to
03:04.23[TK]D-Fenderecam, Ok, can't help you there...
03:04.39ecamthe asterisknow part killed any hopes huh?
03:05.14ecamokay, but lets say i'm not using asterisknow and i don't want to use the patch, what can i do?
03:05.29ecamis there some config parameter i can set to change the default interval?
03:05.48hadsBut you are using AsteriskNOW... :)
03:06.43ecamwell, i can still change the config files
03:07.25fab5freddyquestion about extensions as i am having authentication failure
03:07.26ecamanyway, i'm using asterisknow because i'm sandbox-ing, wannna evaluate asterisk quick
03:07.36ecami'll switch to a real install later.
03:07.46Qwellecam: better than trixbox at least
03:07.52fab5freddy[2345] type=friend username=tempuser secret=password
03:08.05Qwellpeople can actually help you with asterisk-gui generated configs ;)
03:08.20ecamwhat's with trixbox?
03:08.25Qwell~trixbox
03:08.27jbotfrom memory, trixbox is junk - avoid.  It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY!
03:08.29fab5freddyfor user i use tempuser and for password i use password, is this correct?
03:08.42ecamwow, lol
03:09.26ecamso there's really no solution for this setinterval thing?
03:09.56Qwellecam: where is the error coming from?
03:10.00apturaQwell I am starting to see it has lots of problems.
03:10.16Qwellaptura: what, the gui?  Report bugs
03:10.20fab5freddyQWell: can you check some light, [2345] type=friend username=tempuser secret=password, in my softclient i am putting user: tempuser password: password, are i missing something?
03:10.21apturabugs
03:10.37ecamits coming from my attempt to connect to another voip service
03:11.19apturaexample. It seems it has features already installed but went to install the module that install all the features such as dnd or callforwarding and it disabled every signal extention and feature.
03:11.26ecamas in, logged in the console
03:12.11fab5freddyI keep getting Username/auth name mismatch
03:12.31Qwellfab5freddy: user: 2345
03:12.38apturaso I have been building up a dial plan and all the bits and pieces and make sure it looks like it will work before installing asterisk or something that is 99.9% stable.
03:13.54ecamokay, this is it, i'm switching to a real asterisk install
03:13.55ecamlol
03:13.56ecamwith patch
03:14.33fab5freddyQwell: now i get Wrong password  is the password not password?
03:14.50ecamtry using extension number as username
03:14.57ecamand password as the password
03:15.10fab5freddyecam: doing exactly that and getting wrong password
03:15.15Qwellhuh
03:16.15ecaminteresting
03:18.52ecamoh yeah, what can i do to forward incoming calls to all the extensions i have?
03:19.29ecamas in, when i receive an incomming call from the trunk, all my extensions should ring
03:19.46ecamand the first one that picks it up gets the call
03:20.09[TK]D-Fender~trixbox
03:20.12jbotTrixbos is a full linux distro that includes *, FreePBX, and other 3rd party add-ons.  It is these things on top of * which make it seriously painful to support and hence you will find litte help here for it.  Try asking in #freepbx , or their foruns at http://www.trixbox.org/modules/newbb/
03:20.46ecamhuh?
03:20.55ecamyou're recommending me that or what?
03:21.10[TK]D-Fenderecam, No, just training my dog...
03:21.20ecamokay, bots are fun
03:21.29ecamwaitt... were you being sacarstic?
03:21.39Qwell~areyouadog?
03:21.40jbotBark! Bark!
03:21.40*** join/#asterisk jarg (n=jarg@189.157.103.143)
03:21.45ecamokay
03:21.49ecamlol
03:21.59[TK]D-Fender:D
03:22.14wunderkin~gofetch
03:22.15ecamso what bout my question?
03:22.23*** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net)
03:22.27ecamhmm, it doesn't do fetch
03:23.18fab5freddyi am i missing something, secret=password means the password i am using is password right?
03:25.44_VoicemeUpDotComyes
03:25.48_VoicemeUpDotComsip ?
03:25.51_VoicemeUpDotComor iax
03:25.59_VoicemeUpDotComi think iax needs md5 crap too
03:26.28fab5freddy_VoicemeUpDotCom: sip, sip:2345@192.168.2.11 failed for '192.168.2.10' - Wrong password
03:27.06[TK]D-Fender192.168.2.10 is NOT a userid!
03:27.09*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:27.20wunderkino rly?
03:27.55mihinomenestbut I want it to be!
03:28.33[TK]D-Fenderwunderkin, It'd BETTER NOT BE!
03:29.01fab5freddy_VoicemeUpDotCom: that is nowhere in my soft client, that is the ip address i am connecting from
03:29.40*** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242371209.dsl.bell.ca)
03:29.50MrTelephoneis there include support in the asterisk cfgs?
03:29.56MrTelephoneto include an external file
03:30.03QwellMrTelephone: yes, #include filename
03:30.09MrTelephonethanks qwell
03:30.18MrTelephonewould you say its poor practise to reload asteirsk every hour?
03:30.22MrTelephonejust a cfg reload
03:30.32Qwelleh..kinda
03:30.36[TK]D-FenderMrTelephone, Why would you consider doing so?
03:30.50Qwellthere are some valid reasons for it though
03:30.54MrTelephonei have a perl script that I want to run and create the mgcp.conf/sip.conf/extensions.conf
03:31.06MrTelephoneso that changes in the mysql database are applied to asterisk
03:31.17[TK]D-Fenderqwell : yeah... I want to hear his.  The axe-man wants to make sure the head fits the block ;)
03:31.27Qwellwhy not just use #exec?
03:31.33*** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk)
03:31.35Qwellmake it to the parsing on reload
03:31.41Qwell#exec is awesome..
03:32.14MrTelephoneI guess I should check it out
03:32.27MrTelephoneqwell do you write a lot of code for asterisk?
03:32.39Qwellnot much lately
03:33.02MrTelephonei have a question for you later but I'm not ready to ask it yet :-/
03:33.24Qwellsure, $125/hour, 2 hour minimum
03:33.34Qwell(I'm cheap)
03:33.49MrTelephoneI'm cheaper than that
03:33.52Qwell((and I'm also kidding, of course))
03:33.59MrTelephone0$
03:34.10MrTelephonethought you were serious for a second..
03:34.27mitchelocqwell: you don't write anything?
03:34.36MrTelephonewell I have to move disallow/allow to per endpoint instead of just under the [general]
03:35.03Qwellmitcheloc: sometimes, but not much lately
03:35.07MrTelephoneand I have to incorporate L/ft signal so that asterisk uses ulaw
03:35.14MrTelephoneoh well
03:35.23MrTelephoneso i can search #exec and find some information on it?
03:35.29QwellMrTelephone: yeah
03:35.35Qwell#exec rocks
03:36.26MrTelephoneare you talking about agi?
03:36.29Qwellno
03:36.43MrTelephonemy dhcp is running once an hour and dhcp reloads
03:38.07MrTelephonedo you have a link on some documentation on that?
03:39.13CrashHDhow do you specify wav49 format for the mixmonitor application?
03:43.41fab5freddyok i got it working, am registered and was able to dial 500 on the digium server.. now what's next?  where do i go from here?
03:46.08[TK]D-Fenderfab5freddy, to the moon Alice... to the moon!
03:46.24[TK]D-Fender</carbondating>
03:46.52fab5freddy[TK]D-Fender: what's a cool thing to do now that my pbx is up and running
03:47.03hadsPhone someone?
03:47.14[TK]D-Fenderfab5freddy, Learn how to play guitar and go on a world tour!
03:47.23Qwelldtmf guitar?
03:47.40fab5freddyhads: can i use a regular land line to make the outgoing calls?  i am connected through and internal network
03:47.50[TK]D-Fenderqwell : You still think DDR is going to get you on "Dancing With The Stars", don't you? ;)
03:48.03Qwellddr?  pfft
03:48.06brianwhat is a good company that does toll free origination (preferrably unlimited channels)
03:48.10[TK]D-Fenderfab5freddy, Sure... if you have a piece of hardware that allows it.
03:48.43fab5freddy[TK]D-Fender: all i have is a usb headset and a high speed modem/router
03:49.13[TK]D-Fenderfab5freddy, Then the short answer is nothing on YOUR premisis to do with that.
03:49.42fab5freddy[TK]D-Fender: what's the long answer?
03:49.44[TK]D-Fenderfab5freddy, You'll need a gateway device or PCI card of a type supported by *.  Or you can sign up with a SIP ITSP.
03:50.23fab5freddy[TK]D-Fender: do you have a link of where i can purchase such a PCI card or gateway device?
03:50.33hadshttp://www.google.com
03:50.48[TK]D-Fenderfab5freddy, What do you really want to do?
03:51.07fab5freddy[TK]D-Fender: i want to start by making outgoing calls
03:51.25fab5freddy[TK]D-Fender: than i want to receive incoming calls
03:51.35[TK]D-Fenderfab5freddy, Do you particularly wish to do so with an analog line you already have?
03:51.53fab5freddy[TK]D-Fender: no eventually i want to phase out the analog line
03:52.06[TK]D-Fenderfab5freddy, Then perhaps try signing up with www.unlimitel.ca
03:52.16[TK]D-Fenderfab5freddy, They have local DID's for you.
03:52.28[TK]D-Fenderfab5freddy, At pretty decent prices, and excellent quality
03:53.17*** join/#asterisk bmg505 (n=leon@196.209.178.209)
03:53.43fab5freddy[TK]D-Fender: i am looking at their hardware devices now.. what is a DID though?
03:54.07[TK]D-Fenderfab5freddy, Wan't talking HARDWARE there, jsut the service to PSTN
03:54.22[TK]D-Fenderfab5freddy, you can get a "phone line" through them over your net connection.
03:54.44*** join/#asterisk klasstek (n=nunyobiz@c-67-190-165-254.hsd1.co.comcast.net)
03:55.00[TK]D-Fenderfab5freddy, DID is a "phone number" that you can receive calls against.
03:55.09[TK]D-Fender(Direct Inward Dial)
03:55.09*** join/#asterisk rubber_chicken[] (n=blitzrag@CPE000fea3dbc27-CM0012c9db3d2e.cpe.net.cable.rogers.com)
03:55.26rubber_chicken[]yay Good Friday
03:55.39Qwellis it good?
03:55.49rubber_chicken[]not so far, no
03:55.53rubber_chicken[]but I have wine, so that helps
03:55.59[TK]D-Fenderfab5freddy, If you want to buy hardware I typically suggest you consider your wiring scenario and budget.
03:56.09rubber_chicken[]Qwell: where's your []? Not good enough for it anymore?
03:56.20Qwellrubber_chicken[]: I leave it at work :p
03:56.31[TK]D-Fender..... no coment....
03:56.32blitzrageQwell: makes sense :)
03:56.52blitzrageI don't want to know your name
03:57.00fab5freddy[TK]D-Fender: actually i wanted to use the least amount of hardware as possible, i want to start out with the low expenses till i get more familiar with the technology
03:57.01[TK]D-FenderI just want....
03:57.11blitzrage! ! !
03:57.13klasstekHas anyone had trouble with the TC400B and SIP deadlocks that produce "sipsock_read: We could NOT get the channel lock " messages?
03:57.24blitzragefab5freddy: Asterisk + laptop + headset + idefisk
03:57.24apturaI will have some wine.
03:57.24[TK]D-Fenderfab5freddy, Then you've got it already... soft-phone + headset.
03:57.47apturaTK you said trix was bugy right?
03:57.52[TK]D-Fenderblitzrage, z0mg!
03:57.56blitzrageaptura: I have a nice 2001 Montecillo with your name on it
03:58.12blitzragetrixbox is what it is -- but it runs Asterisk
03:58.18[TK]D-Fenderaptura, No, I said that its a canned POS and if you don't like it....  TFB :)
03:58.25klasstekWe just put up a new box with the transcoders, and it failed miserably. Took the transcoders out and put a codec_g729 in place and it didn't get locked up.
03:59.01[TK]D-Fenderaptura, If you want a better answer, ask a more specific question :)
03:59.29blitzrageklasstek: yah... sounds like an issue for Digium support probably -- I'm lucky in that I get to avoid using hardware
03:59.55fab5freddy[TK]D-Fender: so now i just need a phone line from unlimitel? $2.5/month and 1.1c per minute is certainly more than affordable
04:00.11klasstekblitzrage, avoiding how?
04:00.27[TK]D-Fenderfab5freddy, For nominal use it may be the best choice economically speaking...
04:00.27blitzrageklasstek: by not requiring hardware for what I do
04:00.46*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
04:00.47[TK]D-Fenderfab5freddy, I myself have no analog line, just dry-loop DSL.
04:00.52blitzragesame here
04:00.56blitzrageand cable too
04:01.04fab5freddy[TK]D-Fender: is your line with unlimitel?
04:01.09blitzrageI really need to spend some time load balancing those connections
04:01.17[TK]D-Fenderfab5freddy, No, my employer, and all of my clients :)
04:01.28blitzrageI get free phone calls because I use Asterisk!
04:01.31CuriosCatload-balancing between different IP ranges is annoying and unreliable.
04:01.33blitzrageAnd I can spoof my CID!
04:01.46fab5freddya cisco asa 55 series device can load balance for you
04:02.15blitzrageCuriosCat: it's mostly be for a backup -- always use cable unless traffic == bittorrent || cable != available
04:02.17*** join/#asterisk tengulre (n=tengulre@222.90.66.10)
04:02.24[TK]D-FenderFodr F350 load balances much heavier volumes...
04:02.35CuriosCatblitzrage: I wish I could get decent speeds with a single connection
04:02.37fab5freddy[TK]D-Fender: so i can connect this unlimitel line to connect to my asterisk system so it can sound like i am a bigger operation than i am?
04:02.51[TK]D-Fenderfab5freddy, Yup.
04:02.52blitzragefab5freddy: it'll let you have an IVR, yes
04:03.17[TK]D-Fenderfab5freddy, their pay-as-you-go feature allows up to 5 simultaneous channels.
04:03.18fab5freddyblitzrage: what is an ivr?
04:03.22blitzrage~ivr
04:03.24jboti guess ivr is Interactive Voice Response
04:03.27CuriosCatblitzrage: Where I live, BellSouth ("the new AT&T") sells DSL up to 6 mbit/sec (768 up). I'd be content with that, but since we're a third-party NSP, I only get to order up to 3mbit down/384k up
04:03.40klasstekIs anyone here actually using the TC400B and SIP in production?
04:03.54CuriosCat(the upside to that is that I'm not subject to silly bellsouth.net restrictions, and I get to assign myself static IPs and such)
04:03.57[TK]D-Fenderfab5freddy, Strike that IVR comment.  Unlimitel will jsut get calls in/out of your server. Sounding bigger than you are means doing more than jsut ringing whens omeone calls you :)
04:04.24blitzrageCuriosCat: ahhhh yah... the DSL loop is about the same as yours -- I have it for separate routes so I can test my servers in Florida from 2 separate locations
04:04.31fab5freddy[TK]D-Fender: no ringing, but answering the phone and saying press 1, etc..
04:04.36CuriosCatblitzrage: Where in Florida?
04:04.45blitzrageCuriosCat: Tampa and Miami colos
04:04.50[TK]D-Fenderfab5freddy, That is not Unlimitel's job, that is *'s, and yes, you can do all that and more.
04:04.52blitzrageI live in Toronto (downtown)
04:04.54CuriosCatwhere's the Miami one? NAP?
04:05.02CuriosCatI live in Palm Beach.
04:05.06blitzrageSagoNetworks
04:05.12CuriosCatHrm. Never heard of them
04:05.26CuriosCatI should probably pay more attention -- if they sell colo in Miami, that makes them a competitor :P
04:05.31blitzrageyah -- I prefer the colo in Miami
04:05.44fab5freddy[TK]D-Fender:  sweet, tomorrow i am going to sign up a line with them, what personal information do they require?
04:05.46blitzrageSago is in Tampa
04:05.59blitzragethe name for the colo in Miami escapes me... :S
04:06.00CuriosCatwho are the people in Miami?
04:06.03[TK]D-Fenderfab5freddy, Go check out their application form for all the details on their site.
04:06.04CuriosCatheh
04:06.16blitzrageyou work for one?
04:06.21CuriosCatYeah. Host.net.
04:06.27[TK]D-Fenderfab5freddy, I believe you have to make a minimum deposit of like 50$ to start, IIRC...
04:06.29blitzrageahhh... not that one I'm with :)
04:06.41CuriosCatshame. My network rocks :)
04:06.53blitzragehrmmmm
04:07.02CuriosCat(not that I'm biased or anything)
04:07.04blitzragewhat kind of bandwidth? :)
04:07.08[TK]D-Fenderfab5freddy, and you should register to AMUG qhile you're at it...
04:07.10blitzrageand with which links?
04:07.12fab5freddy[TK]D-Fender: they are charging $4/month for voice mail, isn't this a complete ripoff considering * can do this for me?
04:07.21fab5freddy~amug
04:07.34blitzragefab5freddy: yah -- just use your Asterisk box to do that
04:07.39[TK]D-Fenderfab5freddy, news to me... you don't want their VM.  only good if your server dies.
04:07.48blitzrageUnlimitel can sell hosted features, but you don't have to use any of them
04:07.49CuriosCatblitzrage: 3 gigs each to AT&T and Level3 for transit (Miami, DC, Chicago), as well as peering with some 200 networks throughout the US
04:08.03blitzrageLevel3 eh
04:08.12blitzragethat's who we just lost at the colo in Tampa
04:08.20[TK]D-Fenderfab5freddy, Where do you see a charge for VM?
04:08.21CuriosCatLevel3 is pretty good transit
04:08.29blitzrageRight now I'm sending most of my traffic over globX
04:08.34fab5freddy[TK]D-Fender: on their application form
04:08.37CuriosCatthey're the highest-quality tier 1 network as far as I'm concerned.
04:08.48blitzrageyah -- need good transits for being an ITSP
04:08.51fab5freddyCuriosCat: are you an IT manager?
04:08.56CuriosCatfab5freddy: Yes
04:09.03blitzrageyep -- that's been my experience, although I've run into a few routing issues upstream with them before
04:09.08CuriosCatalthough I still get my fingers dirty
04:09.17fab5freddyCuriosCat: do yuo purchase IT equipment on a regular basis?
04:09.31CuriosCatblitzrage: I can't think of a provider on the planet where you're not gonna ever run into routing issues. That's pretty much just the nature of the Internet.
04:09.34blitzrageCuriosCat: might have to look into your colo there -- we might be moving some boxes from Tampa
04:09.36CuriosCatfab5freddy: Yes.
04:09.45[TK]D-Fenderfab5freddy, Doesn't match their site.  Perhaps you are looking at the wrong service or the form is old.  Call them direct first.
04:09.47CuriosCatblitzrage: Let me know
04:09.49blitzrageCuriosCat: oh this I know :)
04:10.09blitzrageI've done plenty of routing
04:10.12blitzrageBGP isn't easy :)
04:10.19CuriosCatBGP is like chess.
04:10.23blitzragetotally
04:10.23CuriosCatLearning it is easy. Mastering it is hard.
04:10.25[TK]D-Fenderfab5freddy, I do see it, and no, you don't want it.
04:10.31blitzragethe concepts are simple
04:10.35blitzrageif you understand routing concepts in general
04:11.34blitzragewhich many people don't :)
04:11.37CuriosCateven when you do, you run into inexplicable weirdness.
04:11.46blitzrageevery single day
04:11.57*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
04:12.10blitzragesometimes my SIP/RTP does weird things for no real reason :)
04:12.19*** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net)
04:13.25CuriosCatmy biggest problem with VOIP is "I'm behind three NAT gateways, six firewalls and dual VPNs. Why won't my phone register?"
04:13.32fab5freddy[TK]D-Fender: don't like this so much                                                         The account will be
04:13.35fab5freddyre-charged automatically unless you inform us otherwise
04:13.40blitzrageCuriosCat: exactly
04:13.53blitzrageI think I got most of that figured out now -- I specifically test for dual-NAT setups
04:14.00MrTelephonethe internet is turning out to be 99% overhead
04:14.05blitzragemake sure it works with no modifications to the equipment
04:14.28CuriosCatblitzrage: I specifically don't use NAT anywhere. Saves me from countless headaches :)
04:14.45blitzrageCuriosCat: none of my servers run behind NAT, but many of my customers do
04:14.56blitzrageso I make sure my network can support other peoples shitty networks :)
04:14.56CuriosCatyeah, so do mine
04:15.13CuriosCatI also get some real nutcases.
04:15.19blitzrageI love the people who tell me I need STUN :)
04:15.25CuriosCatI got an urgent escalation the other day that my network was broken
04:15.34CuriosCatthe customer was trying to traceroute out from his colo, and all he got was * * *
04:15.39blitzrageheh
04:15.41[TK]D-Fenderfab5freddy, There is also les.net , and a handful of other local places to look.  Unlimitel is pretty good.. you just need to think about your intended usage
04:15.45CuriosCat....he had indiscriminately blocked all UDP and ICMP traffic on his firewall
04:15.54blitzrageI GET NOTHING ON THE FIRST HOP! YOUR NETWORK IS BUSTED!
04:15.59blitzragelol
04:16.13[TK]D-Fenderblitzrage, I still want to know how to set my laser printers at work to KILL.....
04:16.16CuriosCatwhen I explained to him that there's a reason those protocols exist, he basically told me "don't get smart with me. I'm tech-savvy, I understand how this works."
04:16.20blitzrage[TK]D-Fender: lol
04:16.24CuriosCat*that* put me in total BOFH mode
04:16.37blitzrageCuriosCat: those people are just fun
04:16.39blitzrageI love meeting them
04:16.51CuriosCat>clickety-click< int g8/35 <enter> shutdown <enter>
04:16.59CuriosCatSir, yes, I do see your link is down.
04:17.12CuriosCatNow please pack your computer up and return it to the store. You're too 'tech-savvy' to own one.
04:17.12blitzragebecause you totally don't need to be obviously smart -- just answering them truthfully for about 10 minutes usually makes them understand how little they know :)
04:17.56blitzrageCuriosCat: when people say something like they are "tech-savvy", I usually tell them that term was only used in 1998 and he's about 9 years behind now
04:18.35blitzrageCuriosCat: btw -- how are you associated with host.net?
04:18.39CuriosCatblitzrage: I prefer playing along and waiting for the change of attitude that occurs when he realizes that he was wrong AND that he'll have to admit it and apologize before I'll reenable his connection ;)
04:18.40CuriosCatbwahahahaha
04:18.48CuriosCatblitzrage: I'm the Director of Engineering.
04:19.16blitzrageCuriosCat: and apologize -- classic.
04:19.38blitzrageCuriosCat: nice -- good to know if I can convince the owner to give you a shot :)
04:19.42[TK]D-FenderCuriosCat, DOE.... come look at the head-lights ;)
04:19.51blitzrageif the prices fit into what we're paying now, I'm sure we can try it out
04:20.48CuriosCatblitzrage: Shoot me an email if you want me to get a quote together. stian@host.net
04:21.19blitzrageCuriosCat: I just emailed the man with the money -- will pass along your email
04:21.38CuriosCatcool
04:21.39coppiceblitzrage: tech-savvy == I know how to get some sucker to do the hard stuff
04:21.50CuriosCatcoppice: Good definition :D
04:21.59blitzragecoppice: getting some sucker to do the hard stuff is just good work practice!
04:22.25QwellCuriosCat: fire your web developer :P
04:22.27[TK]D-Fendercoppice, tek-savvy = Just enough knowledge (to be TRULY dangerous)
04:22.39blitzrageits a bit 2002 :)
04:22.47coppicein the same way as plagiarism and other cheating should score high points on an MBA course
04:22.48blitzrageQwell: fire the web developer of www.asterisk.org
04:22.56Qwellblitzrage: "web developer"...
04:22.57blitzragecoppice: :)
04:23.00CuriosCatQwell: I have.
04:23.04QwellCuriosCat: heh
04:23.11blitzrage"web developer" is right
04:23.13QwellCuriosCat: looks pretty bad in firefox
04:23.22[hC]:)
04:23.25CuriosCatQwell: The web site was done before I got hired. It's ColdFusion, it's broken in pretty much any browser except IE for Windows, and I'm not happy about it.
04:23.27blitzrageother than the header bar, everything is fine here
04:23.57Qwellthat big grey box is in front of the stuff on the right
04:24.04blitzragemy website looks like crap in IE but looks great in firefox :)
04:24.11[TK]D-Fender[hC], Bumper sticker : "Help save California... take a native with you as you leave!"
04:25.55CuriosCatQwell: I would love to get the site redesigned in a way that passes w3c validation, works in any browser and still allows sales&marketing to go into the nice little cms and make changes
04:26.15CuriosCatMost importantly, I disclaim ALL responsibility :0
04:26.29[hC][TK]D-Fender: haha.
04:26.43blitzragemine is w3c compliant, and still doesn't work in IE :)
04:26.56Qwellblitzrage: BONUS!
04:26.57CuriosCatblitzrage: Tried it in IE7?
04:27.11CuriosCatIE7 is, how do I put this
04:27.19CuriosCatLESS retarded about CSS rendering than IE6
04:27.30apturawho here uses vitelity
04:27.57[hC][TK]D-Fender: you're a polycom guy yeah? Do you have any 601's with 2 or more sidecar expansion modules on em, using hints?
04:27.59CuriosCatThe way I figure it, once my page is valid HTML, displaying it correctly is the browser's job. If it can't, get a better browser :0
04:28.04blitzrageCuriosCat: havent tried in IE7 no
04:28.11dougunlimited local dids?
04:28.12doughm.
04:28.18[hC][TK]D-Fender: I have a couple installs with 2+ expansion modules that use hints and they like to 'randomly reboot' a couple times a day on people and i cannot figure out why.
04:28.30coppicedoes IE7 do SVG?
04:28.39Qwelldoes IE7 do PNG?
04:28.45Qwell(transparent PNG)
04:28.48blitzragedoes IE7 do your mom?
04:28.53[hC]Heh, I was going to specify that, but you beat me
04:28.54dougi think it does pong
04:29.14Qwelldoug: can it do it in a gify?
04:29.48CuriosCatI run Windows Vista on my work laptop, Fedora Core 6 on my work desktop, OS X on my primary home desktop, and various versions of Linux, Solaris, FreeBSD and OpenBSD on my servers :p
04:30.10CuriosCatthat way, I'm sure to experience the quirks of a whole slew of operating system, rather than a mere one OS
04:30.14Qwellgod I'm tired - I read Solaris as Slackware
04:30.30coppiceCuriosCat: no CP/M? :-\
04:30.30CuriosCatQwell: IE7 does transparent PNG. IE6 does travesty to PNG.
04:30.31blitzrageI run CentOS on my servers, FC6 on my laptop (main computer), and FC5 on my server at home
04:30.33doug> Our service allows SIP/IAX Termination to US48 and Canada with no minimum for only 1.39¢ per minute.
04:30.35blitzrageI like to keep things simple :)
04:30.41dougis it just me, or does that sound kinda pricy?
04:30.44CuriosCatcoppice: Couldn't find an IP stack for CP/M on my Commodore 128 :9
04:30.46Qwelldoug: just a bit
04:31.00hadsIE7 does do png. Finally.
04:31.12dougoh
04:31.13dougno
04:31.16doug1.39 *cents*
04:31.17blitzrageyah, thats a bit pricey
04:31.18dougnot so bad.
04:31.18Qwelldoug: :p
04:31.20blitzrageoh
04:31.20coppicedoes IE7 do SVG?
04:31.22QwellVerizon math
04:31.27blitzrageheh -- .0139
04:31.28blitzragethat's not bad
04:31.35CuriosCatcoppice: What do you think this is, FireFox?
04:31.35[TK]D-Fender[hC], I have an IP 601 at the office with 3,, fully loaded and running SIP 1.6.6 - 2.0.3.B had no problems
04:31.35dougyeah, i think they oughta say $.0139
04:31.41hadscoppice: Not sure on that one. I don't think so.
04:31.55blitzrageI was kinda disappointed with the number of sim channels I could get on my new servers
04:31.56[hC][TK]D-Fender: really.. and you have more than 8 hints?
04:32.03coppiceeven the SVG in Firefox has its limitations
04:32.05blitzragevery very low CPU, but high load average -- gotta figure out why that is
04:32.19JunK-Yblitzrage: so saturday is the gam!
04:32.26blitzrageJunK-Y: oh yes it is!!!!
04:32.32fetcherblitzrage: what kind of codecs & interfaces?
04:32.33dougi kinda wonder what a "virtual" PRI is...
04:32.42[hC][TK]D-Fender: i only recently figured out how to modify proftpd to allow file appending (for logs) -- I didnt realize it kept overwriting so i had no log of why the phone crashed... it just seems to do it randomly a couple times a day.. I cant figuer out whats doing it.
04:32.43JunK-Ydid ya win 2nite?
04:32.49[hC][TK]D-Fender: but it only happens to sites that have 2+ sidecars with hints on them.
04:32.50blitzragefetcher: SIP w/ ulaw, non-reinvite, no transcoding
04:32.57blitzrageJunK-Y: nope, both our teams lost
04:33.02[hC][TK]D-Fender: they're running 1.6.7 and 2.0.3
04:33.07[hC](the two sites that exhibit it)
04:33.07[TK]D-Fender[hC], 3 modules, fully loaded with hints
04:33.13coppicedoug: a virtual PRI is one that is only working properly in someone's mind
04:33.14JunK-Yso we're sure to go, no?
04:33.21JunK-Yeven if we lose, no?
04:33.42blitzrageJunK-Y: if you lose in OT or shootout, you are in 8th place
04:33.53blitzrageMTL only needs to get 1 pt to be guarenteed a playoff spot
04:33.55CuriosCatBrain<->PRI interface?
04:34.09dougyeah, that must be it.
04:34.37JunK-Yso saturday i will watch the game :)
04:34.47blitzrageactually, thats not quite true... I think MTL can lose in OT, and if the Islanders win their next 2 games, they get 8th place
04:35.18blitzragebut the Leafs will not want OT -- they need to win in regulation, so they will pull the goalie if tied near the end of the 3rd
04:35.37QwellGo Leifs!
04:35.41Qwell:D
04:35.41blitzrage!!!
04:35.42JunK-Ythat will be a great game.
04:35.52blitzrageQwell:  my buddy messages me that all the time :)
04:35.55JunK-YQwell: shut up, go montreal!
04:35.55Qwellheh
04:35.59JunK-Y:)
04:35.59[hC][TK]D-Fender: hmm. strange. i guess i'll have to wait for more logs.
04:36.02blitzrageand Leif is a highway, I wanna ride it, all night long
04:36.06[hC]Go vancouver!
04:36.08Qwellumm
04:36.15Qwellblitzrage: That one's a bit much
04:36.26Qwellsounds like something Tilghman would say :P
04:36.26blitzrageQwell: that buddy is a hot blond chick though
04:36.32Qwellahh, well then
04:36.35blitzrageQwell: I was thinking the same thing :)
04:36.36[TK]D-FenderIn Quebec the Stanley Cup actually comes around more often than Haley's Comet ;)
04:36.39blitzragegood thing he's not around
04:36.44blitzragelol
04:36.59blitzrageboooo Vancouver
04:36.59JunK-Ymontreal has won the greatest stanley cup in the whole nhl history.
04:37.09blitzrageactually, Vancouver is the most likely team in Canada to win this year
04:37.16blitzragealthough none of them will win this year....
04:37.38haroldpwhen I reload * I get this error: pbx.c:4796 ast_add_extension2: Unable to register extension '7753298144', priority 1 in 'default', already in use
04:37.40JunK-Yso wheres the cup is going this year?
04:37.56JunK-Yharoldp: cause ya already define that extension.
04:38.00blitzrageJunK-Y: I think Anaheim or SJ
04:38.17JunK-Yi will go with pittsburgh
04:38.24blitzragethat's my pick for the East
04:38.25haroldphere's the line: exten => 7753298144,1,Answer()
04:38.30QwellGo Ducks! :P
04:38.49blitzrageJunK-Y: actually, I think the Penguins will lose this year because of lack of experience, and they will win it next year
04:38.51haroldp...for my DID
04:39.07blitzrageharoldp: where do you think you are...? #asterisk or something?
04:39.12blitzragethis is #hockey!
04:39.12JunK-Yharoldp: verify, ya've 2 priority 1 for that did.
04:39.15haroldphehe
04:39.29blitzrageQwell: heh... you actually know the team in Anaheim's name :)
04:39.30haroldpoh. duh.
04:39.31Qwellharoldp: something else is already at that exten,priority
04:39.32JunK-YQwell: i c the little californian!
04:39.36Qwellblitzrage: indeed I do
04:39.38haroldpthank you.
04:39.42JunK-YQwell: go huntsville, DAHHH!
04:39.57blitzragei don't think I could live in Huntsville
04:40.05blitzrageI've gotta way too used to living in downtown Toronto
04:40.10blitzrageit kicks some pretty serious ass
04:40.25blitzrageso much to do!
04:40.26haroldpok, that error went away.  thanks.
04:40.34JunK-Yblitzrage: so when are ya coming to mtl ?
04:40.46JunK-Yso we can return in an after?
04:40.47blitzrageJunK-Y: not too sure.... the train is so pricey to get up there
04:41.00blitzrageJunK-Y: return in an after?
04:41.07JunK-Yafter-hours.
04:41.16blitzrageyou mean a pub? :)
04:41.20JunK-Ytake a plane? a boat? a car? an horse?
04:41.32*** part/#asterisk doug (i=doug@zaxxon.telerama.com)
04:41.37blitzrageJunK-Y: actually, if I get a motorcycle this summer, I will drive up there
04:41.47blitzrageif I get it this summer, maybe next summer
04:42.07blitzragedepends how fast I can get comfortable on the highways
04:42.11JunK-Yarent ya coming at end of may for the training?
04:42.15blitzrageknowning me, it won't take long
04:42.24blitzrageJunK-Y: no idea... haven't been told, but that is a long ways off
04:42.37blitzrageprobably get told at the beginning of May :)
04:42.42JunK-Yi will try to go at it360.
04:42.46JunK-Yare ya going?
04:42.50blitzrageI'm speaking!
04:43.02blitzrage~taug
04:43.14jboti heard taug is The Toronto Asterisk Users group. The website can be found at http://www.taug.ca
04:43.14JunK-Yi might speak with simon ditner too.
04:43.14JunK-Ytaug sucks, amug rocks :)
04:43.26JunK-Yall about mtl vs tro
04:43.37blitzrageJunK-Y: I'm hoping the Asterisk part of the show is a hit -- then I can justify having an AstriCon Toronto!
04:43.38JunK-Y~amug
04:44.13JunK-Yjbot, amug is Asterisk Montreal User Group, see http://forums.amug.ca .
04:44.15jbotokay, JunK-Y
04:44.17JunK-Y~amug
04:44.18jbotmethinks amug is Asterisk Montreal User Group, see http://forums.amug.ca .
04:44.25blitzragejbot: no, amug sucks
04:44.34blitzrage:)
04:44.40JunK-Ymouhaha
04:45.05blitzrageyou guys all speak in french?
04:45.16blitzragedo I need to listen to those Learn French tapes I got:
04:45.17blitzrage?
04:45.25JunK-Yso we all speak chinese, ya didnt notice?
04:45.35blitzragenah, all the same to me :)
04:45.48blitzrageI need to learn a language other than a programming language
04:45.55JunK-Yif ya want to get a french canadian, better learn some french basics.
04:46.01coppiceJunK-Y: 你好
04:46.12JunK-Ycoppice: exactly!
04:47.20blitzragethat shows up as 2 boxes for me :)
04:47.35JunK-Ycoppice: 什么?
04:47.51blitzrageits like you guys are talking in encrypted text :)
04:47.51tengulrecoppice: where are you come from? china?
04:47.54JunK-Yblitzrage: we call it UTF-8!
04:47.56coppiceA friend's Mandarin speaking brother moved to Toronto thinking he'd have to learn French and English. Turned out his first priority had to be Cantonese
04:48.09blitzragecoppice: amen
04:48.25tengulreJunK-Y: Where are you from? CHINA??
04:48.28haroldpok, now when I dial out I get an, "SIP/teliax-086d4000 is circuit-busy" error.  I'm sure it's my config, but that isn't pointing me in the right direction.
04:49.04JunK-Ytechie:
04:49.06JunK-Y我來自加拿大.
04:49.19tengulreJunK-Y: are you chinese?
04:49.29JunK-Ytengulre: nope, im canadian.
04:49.48tengulreJunK-Y: but you type chinese character?
04:50.01JunK-Yi dont even know a damn word in chinese, long life to google translate!
04:50.07JunK-Y:)
04:50.36JunK-Ytengulre: i went take some beers with my brother, to celebrates my end of coop and the future win of montreal vs toronto (well i hope)
04:50.42JunK-Yjust to get blitzrage frustrated.
04:50.42coppice加拿大 == canada
04:51.00tengulre??!
04:51.04JunK-Ycoppice: yep
04:51.18blitzrageJunK-Y: don't worry, I gave up the Leafs making the playoffs last time the Leafs lost to MTL -- I said the Leafs would make the playoffs if they won that night, and not make it if they lost -- they didn't win
04:51.19JunK-Ycoppice: u speak chinese ?
04:51.26tengulreI m chinese.
04:51.54blitzrageI think coppice live(s/d) in Hong Kong
04:52.04tengulreso don't type owner country language here
04:52.14blitzrageeh?
04:52.23blitzragewhats wrong with embracing other cultures?
04:52.25coppice我識少少廣æ±è©±
04:52.26JunK-Ycoppice: so you now have to learn french!
04:53.25blitzrageno one owns anything; they merely possess
04:53.29tengulre我晕ï¼
04:53.38coppicewell, I needed to find a native French speaker this week to correct someone's translation, but I don't think that will inspire me to learn a whole language
04:53.39JunK-Ycoppice: 後來你談談.
04:53.58haroldpI see the connection to my providor in 'sip show status', but I can't get anything routed through it.
04:54.02JunK-Yive to go, ttyl guys.
04:54.03tengulreJunK-Y:什么æ„æ€å‘€ï¼Ÿ
04:54.22blitzrageJunK-Y: lates
04:54.46haroldp'sip show peers', rather
04:54.50tengulre昨天载中央电视å°æœ‰ä¸ªèŠ‚ç›®å«ï¼šã€Šå¤–国人的æ‰è‰ºè¡¨æ¼”》éžå¸¸å¥½çœ‹ï¼å‘µå‘µ
04:55.11tengulre加拿大是个美丽的国家。
04:55.49tengulrehaha
04:55.52tengulre哈哈
04:56.18coppicetengulre: are you from mainland china?
04:57.09tengulre是
04:57.26tengulre我æ¥è‡ªè¥¿å®‰ï¼
04:57.32coppiceI thought so. you're using that difficult to read simplified chinese :-)
04:57.36tengulre陕西çœä½ å¬è¯´è¿‡æ²¡
04:58.02tengulrecoppice: hehe. do u know xi'an ?
04:58.03[hC]ok so, im seeing chinese
04:58.05[hC]am i that drunk?
04:58.21blitzrage[hC]: yes you are
04:58.27[hC]blitzrage: sweet, i'll keep going then!
04:58.27coppicei've never been to 西安
04:58.28[hC]:)
04:58.57[hC]blitzrage: just get a utf-8 compatible irc client. I use xchataqua on my mac, and it 'just works'
04:59.02coppiceblitzrage: software built in the current century
04:59.45tengulrecoppice: do u know 秦始皇?
04:59.51CunningPikeblitzrage: xhcat2 works, too
04:59.57blitzragexchat2 eh....
05:00.10CunningPikeNo, xhcat2 ;)
05:00.10blitzrageI'm on linux though
05:00.25CunningPikeblitzrage: xchat2 runs on Linux
05:00.33CunningPikeblitzrage: gtk+ app
05:00.42blitzrageoh yah, I'm using 2.6.6
05:00.53blitzrageI'm obviously missing a library somewhere
05:00.54coppicetengulre: That's on the coast north of 北京, right? I went there to a customer once
05:01.09CunningPikeblitzrage: I'm on 2.8
05:01.09tengulrecoppice: yes,
05:01.12[hC]blitzrage: this is the exact reason i moved to a mac! :)
05:01.22blitzrageugh... mac
05:01.27[hC]blitzrage: you're probably just missing the font
05:01.29CunningPike[hC]: I'm on a Mac - and using xchat2 ;)
05:01.36[hC]CunningPike: in X11?
05:01.37coppiceI remember a really nice seaood restaurant on the seafront
05:01.38blitzrage[hC]: more than likely
05:01.41CunningPike[hC]: Aye
05:01.44[hC]CunningPike: not xchataqua? howcome?
05:01.59CunningPike[hC]: I like X11 - grew up on it
05:02.19CunningPike[hC]: Right now, I'm building Evolution 2.10 on my PowerBook
05:02.29tengulrecoppice: 我的中文åå« å¼ è…¾çº¢ï¼Œ 很高兴认识你ï¼
05:02.54CunningPike[hC]: It's been a bit of an ordeal, but we use Exchange at work, so I'm hoping it will be worth it
05:03.00apturahttp://www.pastebin.ca/426483 i know trix sucks but giving it a chance. What would case this error? DID inbound rings line two but get a verbal error. ohh yea, cannot upload my vm greeting message because of a bug in upload button. Probebely the issue. Anyway here it is. http://www.pastebin.ca/426483
05:03.01[hC]CunningPike: I grew up on X11 too., but i drank the apple koolaid hard, and i just try to get native aqua apps now
05:03.06[hC]CunningPike: xchataqua works great though.
05:03.44CunningPike[hC]: Aye - I use a blend - some Aqua apps I love, others not so much
05:03.59CunningPike[hC]: I use bluefish for an editor, for example
05:04.11CunningPike[hC]: gFTP for an FTP client
05:04.21apturaEvening CunningPike
05:04.26CunningPikeHey, aptura
05:04.34[hC]CunningPike: I use xchataqua, mail.app, subethaedit or textpad for an editor, and transmit for ftp
05:04.46[hC]CunningPike: i just really like that osx aqua apps are always built generally the same way
05:04.48apturaCunningPike what version you running at work?
05:04.52tengulrecoppice: do u translating it?
05:04.54CunningPikeaptura: Of?
05:05.02[hC]CunningPike: and i really like how everything ties together properly
05:05.05apturathe production system of 75 seats
05:05.10coppice我的中文åå«æ®·å¾·è¡›. 我是英國人
05:05.19CunningPikeaptura: 1.2.1, shortly to be 1.4.2
05:05.33apturaso 1.4.2 has tested stable then.
05:05.35CunningPikeaptura: We're scheduling an upgrade shortly
05:05.39tengulrecoppice: hehe, OK!
05:05.40CunningPikeaptura: So far so good
05:06.05CunningPikeaptura: 1.4.0 sucked, 1.4.1 was good, 1.4.2 passed all our tests
05:06.05apturahow do you really know it will be stable unless you put it online with customers?
05:06.23coppicetengulre: translating? :-\
05:06.28aptura1.4.2 is 99% stable. thats what I want to hear.
05:06.30[hC]I really need to start moving to 1.4, im on 1.2.17
05:06.36*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)
05:06.43[hC]Im really looking forward to using asterisk-gui
05:06.56CunningPikeaptura: We've been using it with a test group since it was released, plus we have a good test script
05:07.01[hC]and some of the new stuff that made it into 1.4 and never got backported. like the channel independent jitter vuffers
05:07.03[hC]buffers even
05:07.04CunningPikeaptura: Loadwise, who knows? :)
05:07.07coppice99% stable == down for 15 minutes a day
05:07.14apturahaha
05:07.16[hC]CunningPike: test script? what does it do?
05:07.23apturadont scare me now coppice ;)
05:07.29[hC]CunningPike: i really wish you'd leave your job and come work for me instead :P
05:07.34[hC]CunningPike: i need some asterisk nerds
05:07.38[hC]they're hard to find in vancouver
05:07.43apturaDo not want a client shouting at me when it goes down :)
05:08.13russellbasterisk is perfect
05:08.15CunningPike[hC]: 'It' is two of our people that run through a set of tests that test all the functionality that we use - not rocket science, sorry! :)
05:08.17russellbi don't know what you're talking about
05:08.28apturahc you mean somone who dived into the dial plan
05:08.36CunningPikeHey, russellb!
05:08.40russellbgreetings
05:08.45blitzragerussellb: !!!
05:08.56blitzrage1.4 doesn't crash on me except when I'm testing new things :)
05:09.28russellbblitzrage: yeah, but you're the guinea pig for a lot of stuff
05:09.36blitzragetotally agreed
05:09.36apturaso CunningPike what is it about 1.4 you want that 1.2 does not do for you?
05:09.45blitzrageI do a lot of things most people never touch
05:09.45russellbblitzrage: it's cool :)
05:09.49blitzragerussellb: heck ya!
05:10.04blitzrageI get to find all the bugs for people so it's perfect when they go to use it :)
05:10.15russellbblitzrage: Dwayne expressed a lot of interest in that Dundi project we talked about
05:10.41russellbblitzrage: so, he might work on it soon
05:11.07blitzragerussellb: the CUT(), or the multi?
05:11.13russellbmulti
05:11.20blitzragerussellb: great news!!
05:11.27russellbyup
05:11.56CunningPikeaptura: Couple of big things - IMAP voicemail, asterisk-gui and AEL2 - and a couple of small things - multiple extens for Pickup(), and a patch that provides called party ID that is specific to 1.4
05:12.00blitzragefound a bug in Transfer() today -- had someone look at it, and it was apparently a 2 line fix
05:12.17CunningPikeblitzrage: What were the symptoms?
05:13.03russellbblitzrage: I don't remember seeing a commit on that today
05:13.07blitzrageCunningPike: symptons were continuing on in the dialplan after a 302 redirect, then sending a 603 followed by retransmitting the 302 redirect after an ACK
05:13.21CunningPikeblitzrage: Ugh
05:13.22blitzragerussellb: patch hasn't been put on the bug tracker yep
05:13.24blitzrageyet*
05:13.28russellbah
05:13.29blitzragebut I tested it today, and it's all fixed
05:13.39russellbcool
05:13.43russellbi feel like coding.
05:13.45blitzrageso waiting for the patch creator to post it to the tracker
05:13.52blitzragerussellb: coding what?
05:13.56blitzragesomething new?
05:13.58russellbi don't know...
05:14.04blitzragesomething old?
05:14.06blitzragesomething blue? :)
05:14.14Qwellyou can borrow chan_skinny
05:14.16russellbsomething random and new
05:14.27blitzragerussellb: what was that other idea I had the other day....
05:14.30russellbi guess i should be working on my event stuff.
05:14.51russellbblitzrage: hrm... i don't remember
05:14.56blitzragegrrrr
05:15.01blitzrageI gotta start writing this stuff down
05:15.08russellbyes, you do!
05:15.18russellbi'll look at IAX2 for event processing ...
05:15.20russellbyessss ...
05:15.22*** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk)
05:16.11*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
05:16.21Qwellmy ankle isn't cracking, and it's really annoying me
05:16.23blitzragewow, not being able to remember is really starting to bug me
05:16.39russellbblitzrage: I may have written it down, I don't know
05:16.48Qwell...just thought you all might want to know that
05:16.50blitzragerussellb: good reason to keep logs I guess
05:16.58blitzrageQwell: thx for the update!
05:17.22russellbblitzrage: are you talking about choosing multiple cdr posting locations?  like primary, secondary ...
05:17.25*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
05:17.31CunningPikeBetter, dear?
05:17.42blitzragerussellb: hrmmm.... that might have been one.... but that doesn't feel like the one
05:17.48russellbblitzrage: yeah ...
05:17.54blitzrageit must have been a dialplan function or something I was thinking of
05:18.00*** join/#asterisk Matrix9 (i=MiniMe@s142-179-197-109.ab.hsia.telus.net)
05:18.01blitzrageOH!
05:18.01russellbmaybe
05:18.04russellbo.O
05:18.28blitzragefeels like it had something to do with VM
05:19.01russellbsomething with that SIP NOTIFY weirdness?
05:19.12blitzrageyah.... something to do with hasvoicemail I think
05:19.27blitzrageand it sending something for some reason
05:19.37*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
05:20.03blitzragegrrrr
05:20.04russellbi give up
05:20.36russellbhow about i just make real distributed voicemail!
05:21.03russellbnot that it solves your problem.
05:21.11blitzrageyah... fuck I wish I could remember :)
05:21.18blitzrageit seemed like a good idea at the time
05:21.20coppicewhat is hard about real distributed voice mail?
05:21.31[hC]real distributed voice mail would be nice.
05:21.35blitzrageI just put it all in the DB
05:21.37[hC]right now i always have to pick a pbx to do voicemail on.
05:21.42*** join/#asterisk sharp (i=sharp@gateway/tor/x-5630b7e6c7904f48)
05:21.44[hC]otherwise its just too much of a pain.
05:21.49russellbcoppice: just getting message waiting indication on a different box from where your mailbox is located
05:21.51blitzrageODBC VM works fine for me (now)
05:21.54[hC]depending on the application, i suppose. but.
05:22.00blitzragerussellb: that is my bitch right now
05:22.10russellbblitzrage: heh, that's cool
05:22.22[hC]thats the bigget thing, yeah. i have an externnotify script do it with some scp hackery right now
05:22.39blitzrageyah... I'm so close to just having it working without a hacked together script
05:23.54russellbcoppice: it's not that it's "hard", really, it's just not done :)
05:27.52coppicerussellb: its a lot easier now. when we did it 10 years ago the internet was so bad that trying to get things in the right place at the right time required quite a bit of juggling
05:28.10Qwellblitzrage: it cracked
05:30.29russellbQwell: thanks for the update.
05:31.38blitzrageQwell: eh?
05:31.52blitzrageQwell: oh -- your ankle :)
05:35.43mitcheloc(your welcome)
05:35.50Qwellrussellb: maybe I'm getting stuff crossed, but are CDRs gonna be part of the event system also?
05:36.21russellbQwell: I had proposed it, but I don't know if Steve liked it or not
05:36.32QwellI assume that was how the conversation on the list started
05:36.37russellbyeah
05:41.48CrashHDmake the bad man stop
05:43.01blitzrageok, I'm going to sleep, night all
05:43.58CrashHDnight
06:02.50haroldpwooh, go outgoing working.
06:08.00*** join/#asterisk lineD (i=lineD@c-68-63-33-240.hsd1.al.comcast.net)
06:12.51lineDAnybody prof. skillxin know how readily somebody that your registered/iax2 with can corrupt your mysql database by asterisk api or otherwise if no "insecure" variable is set?
06:15.47CrashHDwhen using 1.4.2 sip jb forced I hear pops and clicks
06:18.04lineDThough I think even being a professor it would still be hard to proclaim knowing alot with seemingly so much more to discover
06:18.56lineDIve made a complaint about some shady billing and ref a call log and a system I made no changes to has its DB corrupt
06:22.26*** join/#asterisk jnc (n=jnc@205.234.240.46)
06:24.05*** part/#asterisk jnc (n=jnc@205.234.240.46)
06:40.30*** join/#asterisk zeeesh (i=zeeesh@202.38.55.125)
06:40.31zeeeshhi
06:45.55*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
06:45.57*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi)
06:45.59*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
06:51.40*** join/#asterisk littleball (n=littleba@bb220-255-154-211.singnet.com.sg)
06:52.07littleballhello, how to define the access control (deny/permist) in the iax.conf file so that only a specific subnetwork can access?
06:52.32littleballexample, i only want 202.192.283.240 subnet hosts can access this asterisk server
06:52.51littleballand deny other all
07:07.54*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
07:09.47*** join/#asterisk tvietduc (n=chatzill@58.186.171.64)
07:11.27*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
07:27.45*** join/#asterisk _lanlv_ (n=chatzill@58.186.171.64)
07:28.37*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
07:31.49*** join/#asterisk Thib_G (n=thib@abo-25-238-68.guy.modulonet.fr)
07:31.59Thib_Ghello
07:33.01*** join/#asterisk LasaK (n=mypain@203.117.213.88)
07:33.18Thib_GI would like trying Asterisk for RTC/SIP interface, but I don't know if it'll work with my onboard modem
07:33.33LasaKi had problem
07:34.18LasaKall client get unreachable status but there is no problem with the network
07:34.48LasaKdo you all ever had the some problem with me ?
07:35.19Thib_GMy modem isn't handled by ALSA or OSS, but can be used with linmodem
07:51.42tzafrir_laptopThib_G, what do you want to use the modem for?
07:52.07Thib_Gtzafrir_laptop: RTC interface
07:53.06tzafrir_laptopyou get the rtc from the modem or from the board's clock? Does the modem come with its own internal clock?
07:53.20tzafrir_laptop(RTC == Real Time Clock?)
07:53.28Thib_G( make call from Ekiga to "real" phone, by using Asterisk's interface )
07:53.55Thib_Gerr... RTC is a French sigle, sorry
07:57.33Thib_GIn fact, I want be able to phone from any of my computers to a real phone ( not VoIP ), using Asterisk. But I don't know if it'll work with my modem
08:00.10*** join/#asterisk AzianFlu (n=Yamabush@arcane/supporter/yamabushi)
08:03.17tengulreThib_G: the asterisk doesn't support modem.
08:06.14Thib_Gtengulre: On my laptop, my modem is handled by ALSA. So, I think it should work. But it isn't the case of the computer I want to use :(
08:07.11tzafrir_laptopThib_G, even if you wanted: a modem has the hardware to emulate a phone(FXO), but not to talk to a phone (FXS). Being FXS requires e.g. a power source
08:07.44tzafrir_laptopSo even f Asterisk has supported your modem, it would not have allowed you to connect a phone
08:08.14tzafrir_laptopWhat do you mean by "handled by ALSA"? As a sound card?
08:08.26Thib_Gyes, tzafrir_laptop
08:09.09Thib_GAnd, I want Asterisk to work like a phone, on the modem port, not to handle phones
08:09.41tzafrir_laptopAFAIK this generally doesn't work, except a few, very specific modems.
08:12.46Thib_GEven if the modem behaves like a sound card ?
08:13.59tzafrir_laptopI figure you could use the sound card capabilities as an extra phone (with chan_oss / chan_alsa), but not to originate calls to the PSTN
08:14.22Thib_G( I remind calling from a computer to a phone, using the modem, on windows, on a very old computer )
08:14.53tzafrir_laptopThis may be an interesting project to write some drivers for Asterisk using zaptel , unicall , chan_modem, or whatever. But it won't "just work".
08:14.55pfnhrm, how do I make my 7960 just use SIP as a peer and not as a user... on an appearance
08:15.22tzafrir_laptoppfn, s/type=friend/type=peer/
08:15.33tzafrir_laptoppfn, but why would you want that?
08:15.36pfntzafrir_laptop, but the 7960 continues to attempt to register
08:15.58pfnbecause I want the 7960 to just be able to dial out on a particular appearance, since there is no registration capability from the provider
08:16.02tzafrir_laptophave you played with the sip setings on the phone?
08:16.32pfnhaven't seen a "don't register appearance" type of an option
08:16.56tzafrir_laptoppfn, I still don't understand why you want such settings. To connect to Asterisk or to connect the 7960 directly to some provider?
08:17.16pfnwell, in this case, just directly to some provider
08:17.22pfnsince I don't feel like setting up my dialplan in asterisk right now
08:21.06*** join/#asterisk svenna_ (n=svenna@p548d3bea.dip0.t-ipconnect.de)
08:24.55*** part/#asterisk Thib_G (n=thib@abo-25-238-68.guy.modulonet.fr)
08:25.38LasaKdo you all ever had the some problem with me ?
08:26.06LasaKall client get unreachable status but there is no problem with the network
08:34.40CrashHDanyone notice safe_asterisk does not work for 1.4?
08:35.21CrashHDnm
08:51.21*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
09:06.14*** join/#asterisk tengulre (n=tengulre@222.90.66.10)
09:17.54*** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au)
09:19.38*** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net)
09:33.21*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
09:36.36*** join/#asterisk smurfix (n=smurf@debian/developer/smurf)
09:46.29*** join/#asterisk saftsack (n=oliver@p54a7d8a6.dip.t-dialin.net)
09:54.55*** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)
10:02.35*** join/#asterisk saftsack (n=oliver@p54a7d3ee.dip.t-dialin.net)
10:06.41*** join/#asterisk ecam (n=ecam@bb121-6-58-13.singnet.com.sg)
10:14.09*** join/#asterisk EnErGy[CSDX] (n=energy@bdpu.org)
10:14.39*** part/#asterisk EnErGy[CSDX] (n=energy@bdpu.org)
10:28.55*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
10:38.16*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
10:43.14*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
10:55.33thevokeanyone here an idea why i try writing to a file in an perl AGI script it doesnt write anything?
11:00.40mvanbaakdid you check permissions ?
11:03.19thevokejup
11:03.32thevokewhen i run it from commandline it works
11:04.28thevokehttp://doos.realroute.net/~michiel/testje.agi
11:06.19*** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it)
11:10.16*** join/#asterisk moranil (n=moranil@122.162.67.94)
11:15.02macTijnthevoke: hey ;)
11:15.20thevokehey ;>
11:15.36thevokeread above ;>
11:22.48*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
11:22.55*** join/#asterisk Fieldy (i=tkYNe2wM@gentoo/contributor/Fieldy)
11:23.30*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
11:26.07*** join/#asterisk saftsack (n=oliver@p54A7E0D3.dip.t-dialin.net)
11:31.22*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
11:35.45irulehi
11:39.46*** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de)
11:56.17*** join/#asterisk misk0 (n=misk0@62.48.116.68)
11:58.29misk0anyone compiled succesfully asterisk with srtp?
12:00.04*** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-89.dsl.peoril.sbcglobal.net)
12:01.12MACscrim using my asterisk box as a linux router as well. Recommendation on a script that i can use to work as the firewall that is sip friend and can provide QOS?
12:01.48*** join/#asterisk FreezeS (n=bla@82.208.157.125)
12:02.00FreezeShello
12:02.22FreezeSis it possible to transmit variables when dialing IAX on a remote server ?
12:04.21FreezeSI have 3 servers and only one has a PRA card. I want to be able to dial through the PRA but keeping the CLI whatever the registration server is
12:06.37*** join/#asterisk zotz (n=zotz@24.244.163.157)
12:10.44*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
12:15.38*** join/#asterisk ManxPower (n=manxpowe@7.sub-70-223-10.myvzw.com)
12:17.11*** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net)
12:17.15MrWupanyone used the aastra 9133i with XML? is it possible to change the display name?
12:21.44*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
12:23.09ManxPowerMrWup: Display name?
12:24.40MrWupManxPower, well yeah. im doing a DND php script. so that the phone user can press a hard key and it calls a script which sets them as DND on the database. the dialplan can then handle them as DND. the only problem is: there is no way to signal to the phone user that they are on DND mode. it doesn't seem like i can set an LED lamp, or set the main display name to show "Blah <xxx> - DND" or anything
12:25.40MrWupall that i seem to be able to do is show a text screen saying "DND Activated" when someone presses the hardkey, but then that message is not persistent and as soon as its gone people dont know if theyre still on DND or not, unless they check by pressing the button and unsetting DND and resetting it to display the XML screen text message again
12:27.25MACscris g729 prefered over g723.1?
12:27.34ManxPowerMrWup: Welcomet to the world of SIP
12:28.10MrWupManxPower, it would be so simple if the phone supported setting an LED via an XML command or setting the phone's setup via XML
12:28.16ManxPowerMACscr: Yes.  Since you can get G729 support for Asterisk ($10/channel), but you cannot get G723.1 support for Asterisk (the patent holders refuse to license it for a reasonable fee)
12:28.21MrWupi cant see why thats been left out (if indeed it has)
12:28.23*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
12:28.33*** join/#asterisk vtatian (n=vtatian@ns1.druzhba.lviv.ua)
12:28.43MACscrManxPower : can i test g729 before i purchase it?
12:28.48ManxPowerMrWup: the best you can hope for is to use hints in the dialplan to light up a BLF key.
12:28.56ManxPowerMACscr: no.
12:29.25MrWupyeah if only i could get BLF to work. theres another nightmare
12:29.26ManxPowerMACscr: you would only use G729 if you have not other choice in codecs
12:29.34MrWupstupid phones just wont access it
12:30.32vtatianCan any one help with change abcd bites at 16 ts with CAS signaling
12:30.52vtatianneed change idle code
12:31.46*** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de)
12:33.12misk0anyone compiled succesfully asterisk with srtp?
12:33.27ManxPowervtatian: in zconfig.h in the zaptel source is an option regarding the idle CAS bits
12:33.48ManxPowerI think it is zconfig.h
12:34.04coppiceyou set the idle bits in the zaptel.conf file
12:34.52coppicee.g.
12:34.53coppicecas=1-15:1101
12:38.05*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu)
12:38.53ManxPowercoppice: is that documented?  If so I must have missed it.
12:39.28coppiceI think its described in the example zaptel.conf
12:40.08MACscranyone know of a website to do a test voip quality that isnt java based?
12:42.12vtatianthanks
12:44.45*** join/#asterisk voltagex (n=voltagex@124-254-124-143-dsl.ispone.net.au)
12:45.25voltagexis there a way I can see what details are being sent in a register request to Asterisk? I'm sure I've got my username/password matching what's in sip.conf but I'm getting "401 Unauthorised"
12:46.07MACscrasterisk cli
12:46.22voltagexyeah, all I see is an MD5 hash
12:47.55ManxPowervoltagex: what non-debug message are you getting?  That will tell us most of what we need
12:48.17voltagex? the ATA just isn't able to register
12:49.21*** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)
12:49.21ManxPowervoltagex: and you should be getting a messge saying something like Rejected registration from bob <123@ip.address>  knowing the exact message will help
12:49.27voltagexhmm
12:50.49voltagexhang on, rebooting ata
12:52.06d00gsterguys, I have a sip client  (eyebeam) overseas connecting to my asterisk. since they have high latency, I dud a qualify=5000 in one instance and =no in another. I also forced the client to register every 300 seconds (eyebeam option). the client can pickup the line and call me anytime  of the day. when I call the client, they don't see the call come in and I go to vm. I asked the  client to dial 7777 and dial his extension and tha
12:52.25voltagexManxPower: different message now
12:52.31voltagexManxPower: SIP/2.0 407 Proxy Authentication Required
12:53.12MACscrManxPower:  im using my asterisk box as a linux router as well. Recommendation on a script that i can use to work as the firewall that is sip friendly and can provide QOS?
12:53.35frigidzephyrd00gster: did you verify via the CLI that your call is hitting the correct extension and its ringing that SIP peer?
12:55.06d00gsterI'll check
12:56.15*** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de)
12:56.28ManxPowervoltagex: Yes, that is normal, the ATA will then make the request again with password
12:56.43voltagexManxPower: well, it doesn't
12:57.03voltagexManxPower: I just get a busy signal
12:57.13ManxPowervoltagex: that would usually be an ATA problem then.
12:57.19*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net)
12:57.30ManxPowerBut since you are not getting any registration rejected messages I really don't have any more ideas.
12:57.49voltagexManxPower: yes, it is now registering correctly :S
12:58.12voltagex:/ switched to the other ATA port and it works
12:58.39voltagexManxPower: I have two built in to a Broadcom based router...repair is going to be annoying.
13:00.55irulehow can I exten => i,1,Playback(pbx-invalid)exten => i,n,Goto(s,restart) when caller dials invalid number?
13:02.16ManxPowerirule: have you tried it?
13:02.43iruleyes, 531464145654632456 makes * stay quiet
13:02.53*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
13:03.04ManxPowerirule: exten => i is run when an invalid option / extension is dialed during IVR type of stuff -- background, waitexten, etc.
13:03.29ManxPowerirule: are you dialing on a SIP device or a Zap port?
13:04.00iruleon a sip, but I have a couple x100p too
13:04.14iruleand an ata sipura 2000
13:04.23ManxPowerremember SIP devices wait for the full number to be dialed before sending the call to the server.
13:04.27*** join/#asterisk psk (n=psk@golia.caltanet.it)
13:04.34iruleoh ok thank
13:04.36ManxPoweralso, as I said, "i" is only for IVRs.
13:05.02ManxPowerIf you want do to that in a non-ivr situation you need an exten line that acts like a wildcard.
13:05.05iruleok, now I see 51324435321432 gives me a busy signal
13:05.27*** join/#asterisk Geert (i=geert@irssi/staff/geert)
13:05.28GeertHmmz
13:05.33GeertI have
13:05.35Geertexten => 1000,hint,SIP/1000
13:05.36ManxPowerAssume you can dial extens 2000-2999, then you can put in an exten => _XXXX,1,Whatever to catch anything that does not match an actual 4 digit extensions
13:05.36iruleId like that to be that cute voice telling me -please dial a real number
13:05.38Geertexten => 1001,hint,SIP/1001
13:05.40Geertand so on
13:05.43Geertcan I make it short?
13:05.51Geertlike exten => XXXX,hint,SIP/${EXTEN}
13:05.55blitzrageGeert: nope -- there is no pattern matching for hints
13:06.04Geertcrap, then I need a lot of hints :p
13:06.14Geertokay, thanks blitzrage :)
13:06.24Geertor have you got any other suggestions?
13:06.31ManxPowerGeert: you will find that using the extension as the SIP ID does not scale and will cause problems in anything but the simplest dialplans
13:06.34blitzrageGeert: yep :)  There is a patch on the bug tracker -- #7767 I think. Should make it to Asterisk 1.6
13:06.35irulewhat is a hint?
13:06.53*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
13:07.04ManxPowerGeert: I suggest you stop being lazy and accept the fact that dialplans are complicated.
13:07.07blitzrageGeert: create a script which generates the hints, then you can use.... #execute I think
13:07.47ManxPowerdialplans would be easy if there were no users.
13:07.55irulehehe
13:08.21frigidzephyrManxPower: lolz so true
13:08.47ManxPowermost users have needs that are different from all other users.
13:09.01ManxPowerOur method is set channel variables, then call a macro that handles the dialing
13:09.21blitzrageI use the [username#vpbx] format for my users, then associate an extension number with them in the database
13:09.54*** join/#asterisk nextime (n=nextime@unaffiliated/nextime)
13:10.06iruleyes variables are the best
13:10.26blitzrageexten => s,n,Set(HASH(user_info)=${ODBC_GET_USER_INFO(${USER_ID})})
13:10.51irulecan you send a different dial tone? everyone is mad at me because they have no idea if it is a real call or am I still playing with the phones lol
13:10.54nextimeHi. in users.conf, for a zapata trunk, in zapchan= option, can i put more than one channel like 1,2,3 or better a channel interval like 1-3? ( read as "is zapchan in users.conf an "alias" for channels= in zapata.conf?" )
13:11.06ManxPowerblitzrage: *nod*  My way is designed so the variables can be set by any method, database, included.
13:11.15blitzrageirule: you shouldn't be "playing" and integrating at the samet time
13:11.25blitzrageManxPower: yep, everyone has their own methods :)
13:11.29irulebut it is fun!
13:11.31irulelol
13:11.44blitzrageManxPower: asterisk is great like that -- so many ways to solve the same issue :)
13:13.07*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:13.51d00gsterfrigidzephyr > when the client first registered (the had to reboot) I managed to get through to the extension, not problems, 5 min later I tried it again and after ringing I got the  voicemail. so I can safely say * is  dialing the  right extension
13:14.31blitzragesounds like the client is behind NAT, and you need qualify=yes, and to have them register more often
13:15.07iruleis there a "press something to hear options" sound?
13:15.28blitzrageirule: I don't know... is there? check the sounds.txt file
13:15.56d00gsterblitzrage, I'll try qualify=5000 and they register every 300sec
13:16.08d00gsteranything else I need to do?
13:16.25blitzrageI usually do 'ls /var/lib/asterisk/sounds/*word_I_want_to_find*'
13:16.39frigidzephyrd00gster: what does the CLI say when it rings that extension? , you might turn on sip debug and see what response you get or if you just cant reach them
13:16.47blitzraged00gster: quite possibly -- NAT is a bitch... gonna take some time to understand the issues, and learning how to read a sip debug
13:17.28d00gsterok so qualify=5000 now
13:17.37d00gsterI'll versbos 9 and call them
13:18.15blitzrageverbose isn't going to tell you anything -- you need 'sip debug'
13:18.21*** join/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg)
13:18.44frigidzephyrd00gster: you might turn debug on for the console also in logger.conf
13:18.47yidiyuehanhi, any one knows whether there is any zaptel channel?
13:19.28frigidzephyryidiyuehan: you'll have to be more specific with your question
13:19.55yidiyuehani mean, any channel like #zaptel that i can ask question refer to the zaptel card.
13:20.00*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
13:20.57frigidzephyryidiyuehan: i dont know of any, i would think you can ask in here. Also if its a Digium card, Digium has free phone and email install support for the card
13:22.16yidiyuehanreally?. could you tell me the contact info? or never mind, i wll search it
13:22.30yidiyuehanhttp://www.pastebin.ca/426852, my problems posted here
13:22.30d00gstersomething not registering with this client. I turned sip debug and before I made a call I got this http://www.pastebin.ca/426855
13:22.42d00gsterlots of retransmits
13:23.05yidiyuehanas the zaptel card could not detect the hang up if i use two pots lines makiing call.
13:24.29frigidzephyryeah, send an email to support@digium.com and they will help you, install support is included with the purchase of their cards
13:24.46tzafrir_laptopyidiyuehan, a word of advice: put a space or whatever separator after the URL . Otherwise it becomes part of it. For instance: try http://asterisk.org/,
13:26.04tzafrir_laptopyidiyuehan, look into busydetect . IIRC the telco in Singapure doesn't have decent disconnect supervision
13:26.12*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
13:26.12*** mode/#asterisk [+o mog] by ChanServ
13:26.17d00gsterand now with an actual call http://www.pastebin.ca/426862
13:26.18tzafrir_laptopyidiyuehan, busydetect=yes in zapata.conf
13:26.36yidiyuehantzafrir_laptop. yes i have busydetect=yes
13:26.40yidiyuehanand busycount=5
13:28.04yidiyuehanfor single call in or call out it's fine via PSTN line, but once it's the case like this, i call in using cell phone, and the extension does not pick up the call,and the call is forwarded to another cell phone with another PSTN line, and again the remote ppl does not answer, then i hang up, then the two lines do not hang up...
13:28.18yidiyuehanis that a possible bug with zaptel 1.2.16 ?
13:35.57Dirk|sleepAnyone have experience using Cepstral voices with asterisk? Is the 8khz optimised version of the voice the one to go for or does it make no difference?
13:36.20Dirk|sleepConsidering they are the same price, I'd rather buy the better version unless theres a good reason not to
13:41.00ber_if i have a call coming in on a DID and I want to send it to some sip destination
13:41.09ber_is there a way to do it without answering the call
13:41.22ber_and only considering the call answerd in the CDR if the sip destination answers?
13:41.43ber_right now I have the DID answering and then executing a dial command
13:42.03ber_which is fine except that the CDR shows answered even if the destination from the dial command does not answer
13:42.07frigidzephyrber_: you may not need the Answer()
13:42.21ber_ah so if i do a dial
13:42.26frigidzephyrber_: does it work without?
13:42.27ber_it knows to connect the incoming call to the Dial?
13:42.32ber_i will try it now!
13:42.36frigidzephyrber_: i would think it should
13:42.55d00gsterfrigidzephyr, any idea's?
13:43.37frigidzephyrd00gster: i looked at the debug, i cant tell much from it,   just looks like it can't reach the destination,  if its behind NAT it will be difficult to find the issue
13:44.06frigidzephyrd00gster: do you have nat=yes on that peer?
13:44.44d00gsteryes
13:45.15frigidzephyrd00gster: im not sure =[
13:45.29d00gsterok thanks
13:45.46ber_yes it works without Answer
13:45.49ber_thanks
13:47.19frigidzephyrber_: no problemo
13:48.26*** join/#asterisk mDuff (n=ccd@user-387ocuv.cable.mindspring.com)
13:49.16ber_asterisk doesnt like it when you dont have a step 1 in a dialplan
13:49.19ber_:)
13:50.16frigidzephyr=D
13:50.53mDuffI'm trying to record calls from a queue to speex-format files, but this results in "ast_writefile: No such format 'speex'". The funny thing, though, is that there really is a /usr/lib/64/asterisk/modules/codec_speex.so (which isn't set as noload in modules.conf), and speex shows up in "core show translation". What am I missing here?
13:51.29ber_zephyr, do you know of a way to play a wav or mp3 file in asterisk from an offset time value?
13:51.36*** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu)
13:51.45b11dallo
13:52.42frigidzephyrber_: do an exten with a Wait(x)    to wait a few seconds then do an exten with a Playback, is that what you mean?
13:52.52thevokeanyone a code snippet here for WAIT FOR DIGIT ?
13:52.54frigidzephyrber_: not sure i understood
13:53.30frigidzephyrthevoke: exten => 100,1,WaitExten(x)     x is time to wait for you to press a digit
13:54.11frigidzephyrmDuff: check permissions on codec_speex.so?
13:54.13thevokefrigidzephyr: yeah, but this is in an agi
13:54.20thevokeusing perl
13:54.25thevoke<PROTECTED>
13:54.25thevoke<PROTECTED>
13:54.29thevokethis is what i do now
13:54.36frigidzephyrthevoke: ah, no idea then
13:56.44mDufffrigidzephyr: permissions are fine. (I'm also lazy, and haven't yet gotten * running non-root).
13:57.55frigidzephyrmDuff: not sure what could be wrong there =]
13:58.20mDuffHmm. I suppose sometime when the business is down I can restart Asterisk with full logging and see if we're having any trouble loading the module.
13:58.57frigidzephyrmDuff: show modules like codec_  to see if its loaded
13:59.12frigidzephyrmDuff: then just type  unload codec_speex.so    then   load codec_speex.so
13:59.22frigidzephyrmDuff: to see if there is an issue
13:59.46mDuffIt's loaded...and unload/reload doesn't show any errors. Let me see if it works in practice now...
14:00.49frigidzephyrI am not sure that you can even record in a speex format, whether you have the codec or not
14:00.49*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
14:00.56mDuffahh; that may be it.
14:01.04frigidzephyrhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record
14:01.18frigidzephyrdoesnt mention speex as being a supported recording format there
14:01.25mDuffIndeed.
14:01.27mDuffWell, phooey.
14:01.30frigidzephyr=D
14:01.41mDuffThank you for the help!
14:01.46frigidzephyrNo problem!
14:02.28ber_zephyr, say i have a 10 minute wav file
14:02.44frigidzephyrber_: "You have a 10 minute wav file"
14:02.48ber_hehe
14:02.54frigidzephyrber_: =D
14:02.58ber_i stop listening to it 5 minutes in
14:03.12ber_by pressing a IVR digit saying 'pause'
14:03.19ber_and my ANI is logged etc
14:03.25ber_i would like to re-access the IVR from the same ANI
14:03.30ber_and resume playing in that location
14:03.57ber_5 minutes into the wav file
14:04.08frigidzephyrber_: not sure how you would do that, that sounds pretty cool tho
14:04.13ber_i was thinking of adding a fast forward or reverse option
14:04.16ber_well i can clip the wav file
14:04.22ber_and delete the first 5 mins
14:04.25*** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de)
14:04.29ber_but that would only allow forward progress
14:04.44ber_and doesnt solve a fastforward problem
14:04.49frigidzephyrright
14:05.01ber_i can probably code a command to handle it
14:05.12frigidzephyrwould be nice if you could playback at a specific time point in the sound file
14:05.17ber_but i didnt know if there was something existing
14:05.40frigidzephyryeah not that i know of
14:05.41ber_i dont know anything about the wav file format
14:05.52ber_but i think everything on the asterisk box is normalized to 8000 samples/sec
14:06.11*** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com)
14:09.35*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM]
14:09.35*** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) [NETSPLIT VICTIM]
14:09.35*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) [NETSPLIT VICTIM]
14:09.35*** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk hal2k (i=am@chello084112159217.16.11.vie.surfer.at) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk errr (n=errr@fedora/errr) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk znoG (n=gs@24.232.95.132) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk hads (n=hads@reef80.anchor.net.au) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk olds (i=olds@antitech.xmission.com) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk Vec (n=Vec@dsl-244-208-173.telkomadsl.co.za) [NETSPLIT VICTIM]
14:09.36*** join/#asterisk Kapsel (i=kapsel@62.242.240.33) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk djflux (n=djflux@mm.shermfin.com) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk jaycampbell (n=jay@spcxn.got.net) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk grinsbalu (i=grinsbal@homer.netvanced.info) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk ilan (i=ilan@69.60.110.251) [NETSPLIT VICTIM]
14:09.37*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM]
14:09.38*** join/#asterisk robwi (n=rowi@slave.rixport80.se) [NETSPLIT VICTIM]
14:09.38*** join/#asterisk svanlund (n=dasv@slave.rixport80.se) [NETSPLIT VICTIM]
14:09.38*** join/#asterisk bulle (n=bulle@c-db2971d5.015-48-626c671.cust.bredbandsbolaget.se) [NETSPLIT VICTIM]
14:09.38*** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) [NETSPLIT VICTIM]
14:09.38*** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM]
14:09.38*** join/#asterisk mcab (n=mb@66.195.193.151) [NETSPLIT VICTIM]
14:09.38*** mode/#asterisk [+o angler] by irc.freenode.net
14:09.38frigidzephyrwhoa
14:10.39polerinit is just not natural to drink coffee through a straw :/
14:11.12frigidzephyrits dangerous
14:12.24b11dnice
14:12.25iruleall incorrect numbers atm are getting a busy signal back to me, how can I cache them to tell the caller that number is incorrect and ask to retry?
14:12.30*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
14:12.41irulecofeeeeeeeeeeee  hmmmmmmmm
14:13.37b11dI call it "heated bean juice"
14:13.43b11d"coffee" though eh.. neat.
14:13.44b11d:)
14:14.01ber_http://search.cpan.org/~jdb/libwin32-0.27/Sound/Sound.pm
14:14.08ber_there is a perl module that can deal with wav from offset
14:14.20ber_just need to link that in to however asterisk plays wavs
14:14.38frigidzephyrneat
14:16.05frigidzephyrirule: you could use check the dialstatus variable and do something based on that,  http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
14:18.15ber_http://www.voip-info.org/wiki/view/stream+file
14:18.16*** join/#asterisk ppyy (i=ppyy@218.93.156.131)
14:18.27ber_stream_file AGI command has an offset built into it
14:18.41ber_done and done
14:20.15*** join/#asterisk AF-Slash (n=AF-Slash@216-161-216-130.hlna.qwest.net)
14:21.19ber_http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ControlPlayback
14:21.21ber_that is even better!
14:21.27ber_asterisk is so far ahead of me :)
14:22.08aydiosmioas far as SIP/RTP is oncerned T.30 is pretty much a standard phone call, correct? No out of band data or special encoding is required.
14:22.17MrWupguys im having strife with the aastra 9133i
14:22.35aydiosmioI keep confusing T.30 and T.38
14:22.45MrWupive finally found an XML object AastraIPPhoneStatus which i can push to the 9133i and it sets a message on the display
14:22.47MrWupand it works
14:23.02*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
14:23.23MrWupproblem is... before it sets the display message it always says "page load error" on the phone. and then when u clear that error, the message is set as it should be
14:27.53ManxPoweraydiosmio: each device does the T.38 standard a little different.
14:28.07*** join/#asterisk yassine (n=yassine@dsl.voicint.com)
14:28.17ManxPoweraydiosmio: and yes, the encoding would be TIFF data, rather than modem audio
14:29.32aydiosmioManxPower: would be tiff for T.38, audio for T.30?
14:29.53*** join/#asterisk ixela (i=ixela@nat/digium/x-eb86fb15e783b03b)
14:30.51*** join/#asterisk ixela (i=ixela@nat/digium/x-6c14c1252aac2604)
14:34.54*** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-0723e5ab64919614)
14:36.29rudholmis there a ring cadence ("rX") that is no ringing at all?  (or very brief ringing)
14:37.05rudholmI tried 0-9 and the closest I came to one was r3, which is three quick rings
14:39.03*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
14:39.03*** mode/#asterisk [+o anthm] by ChanServ
14:44.36*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
14:50.44irulethanks frigidzephyr, what is the path to asterisk sounds?
14:50.55*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
14:51.19frigidzephyrirule: var/lib/asterisk/sounds    i think, thats what googleis for
14:51.27frigidzephyrirule:  =D
14:51.46polerinirule: /var/lib/asterisk/sounds/  I think is the standard, though debian puts it in /usr/something or other (grain of salt i'm a newb.  locate tt-weasles.gsm
14:51.56*** join/#asterisk l2cache (n=ghansen@64.128.254.98)
14:52.21sulexon deb I think is /usr/lib/asterisk/sounds
14:52.21polerinthat'll tell you where it is on your system
14:52.31l2cachewhat steps do i need to take to roll back from asterisk 1.4.0 to 1.2.13 ?
14:53.20*** join/#asterisk monstertruck (n=monstert@c-66-176-203-204.hsd1.fl.comcast.net)
14:54.36[TK]D-Fenderl2cache, wipe out your modules folder, and compile your nw *
14:54.47polerinI'm only in...err. 9 channels right now but it goes up to ~20 sometimes
14:54.58monstertruckhi, any idea what can cause asterisk to complain like this:
14:55.00monstertruckHuh? An ilbc frame that isn't a multiple of 50 bytes long from IAX2 (20)?
14:55.21l2cacheThats what i thought i had to do...after i compiled the old ver it started....OK .. then i did a ps - ef | grep asterisk and it was never running
14:55.31polerinoh .. woops
14:55.33polerinwrong channel
14:55.58*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
14:58.28*** join/#asterisk Fieldy (i=5o7doF1o@gentoo/contributor/Fieldy)
14:58.29drfreezeI've been researching T1 cards from Digium and Sangoma
14:58.49drfreezeAny experiences out there that can help me decide. So far, it seem Sangoma has the edge.
15:01.27frigidzephyrDigiums install support rulez
15:01.33irulewhat are the different call transfer methods?
15:01.57b11dsangoma cards are the shit
15:02.03b11di've got three a104d and they rock
15:02.08drfreezefrigidzephyr: yes it does. I assumed that sangoma also had some type of support.
15:02.17b11dyeah their support is great too
15:02.20b11dplus, they're canadian
15:02.27irulethanks frigidzephyr and polerin, btw I just installed 1.4.0 on debian and it is in /var/lib/asterisk/sounds/
15:02.40frigidzephyrirule: no prob, =D
15:03.22d00gsterCan someone suggest a canadian provider with good availability and cheap /min rates?
15:03.43irulewhat is the name of the method of answering a call from phone a when phone b is ringing and nobody us sitting in phone b?
15:04.15aydiosmioit's called "employee b is fired"
15:04.22irulelol
15:04.37irule"get the phone moron"!
15:05.22mvanbaakirule: call pickup
15:06.35polerinactually, i'm sorta interested in that, I'm goign to have one softphone line and one line that rings everything, I was curious how to pick it up the softphone from the other line
15:06.53mvanbaakby default it's *8
15:07.11iruleanyone care to enlighten us with an example? ;D
15:07.18mvanbaakthat is, if the devices are in the same callgroup/pickupgroup
15:07.47mvanbaakyou dont have to put it in your extensions.conf
15:08.11mvanbaakjust make sure both phones/softphones/whatever have the same callgroup= and pickupgroup= config lines
15:08.17*** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net)
15:08.21iruleoh cool, what other features don't need any confguration? is there a list online or something_
15:08.35*** join/#asterisk marv[work] (n=timr@24.214.206.254)
15:08.47mvanbaakhttp://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
15:08.49*** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com)
15:09.03irule*8 does not work in my setup :(\
15:09.04*** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com)
15:09.28ManxPowerjust remember chan_iax2 does not support call and pickup groups
15:10.15*** join/#asterisk MikHell (n=michel@c-71-56-231-165.hsd1.co.comcast.net)
15:10.18ManxPowerirule: make sure your SIP device is actually allowing you to dial *8.  May of them trap * codes in their default dialplan
15:11.14*** join/#asterisk hedge77 (n=ambray@209.42.192.214)
15:12.22*** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com)
15:12.39iruleI tried with a siemens euroset 3200 trying to get a call from an exten to an alcatel cheap analog phone
15:13.32mvanbaakHA !
15:13.46mvanbaaknice story !
15:13.57irulehuh?
15:13.58mvanbaakone customer has a really bad dsl line
15:14.12mvanbaakso they called the provider
15:14.28ManxPowerI wish I could get HPEC working on my system
15:14.40mvanbaakthey said: "it's not our line. we can see your voip provider uses demo software. tell them to buy a license and your trouble will go away"
15:14.51mvanbaakgheh, we are using asterisk demo version now
15:14.52mvanbaak;)
15:15.42frigidzephyrManxPower: what problem you have with HPEC?
15:16.15frigidzephyrManxPower: if its with Digium hardware, call them and they will fix it for you, or help you fix it
15:16.17MrWupin asterisk how do i check is variable ${REMOTE_STATUS} is 1 or 2 or 3 or 4... i.e. if its 1 then go to voicemail, if its 2 then hangup etc
15:16.19MrWup?
15:16.21frigidzephyrManxPower: I might be able to help though
15:17.20mvanbaakMrWup: ael ?
15:17.27MrWupael?
15:17.48b11dael?
15:18.15MrWupasterisk extension language
15:18.17hedge77~ael
15:18.18jboti heard ael is Asterisk Extension Language - a dialplan language with 'c like' syntax?
15:18.18MrWupi dont think i need that though
15:18.40MrWupsurely its possible to just check if a variable is 1 or 2 or 3 or 4 and do different things based on which one?
15:18.47hedge77in ael it is
15:18.48MrWupwithout ael
15:18.59mvanbaakin ael it's really simple
15:19.30mvanbaakin normal extensions.conf it's a bit messy but can be done
15:19.30polerinthat sounds awesome
15:19.31poleringot a link for it?
15:19.35[TK]D-FenderMrWup, "show application gotoif"
15:19.35d00gsteranyone here can help me troubleshoot a possible nat issue? http://www.pastebin.ca/426862
15:19.37MrWupah
15:19.44*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-230.buff.east.verizon.net)
15:19.48mvanbaakexten => bla,n,GotoIf("${VARNAME}" == "1"?someexten)
15:19.52MrWupthanks [TK]D-Fender
15:19.59hedge77polerin: http://www.voip-info.org/wiki/view/Asterisk+AEL2
15:20.00Corydon76-homenot double-equals, but yes
15:20.08mvanbaakehm, sorry
15:20.09mvanbaaklol
15:20.09polerinhedge77: awesomeness
15:20.27[TK]D-Fendermvanbaak, far worse than  double "=" in there ;)  Thats NASTY sloppy .....
15:20.52Corydon76-homeActually:  GotoIf($[0${VAR} = 1]?somepri)
15:21.15mvanbaaknow I know why I always did everything in an agi
15:21.16mvanbaak:)
15:21.49MikHellCould somebody help me setup my dialout ?
15:21.50hedge77if(${VARNAME}=1) { goto somecontext|someexten|1; } is mucho mas superior
15:22.10MikHellAll seem to work on asterisk side except that it does not connect
15:22.12mvanbaakbut with ael I completely dropped the agi scripts
15:22.24wunderkinfrigidzephyr, i have an hpec problem but keep getting blown off, what is your position with digium? first time i've seen you here
15:22.51*** join/#asterisk Goodjoke (n=chatzill@mail.theenergynetwork.com)
15:23.05mvanbaakswitch(${VARNAME})
15:23.06frigidzephyrwunderkin: i know them =D whats your HPEC issue?
15:23.12Corydon76-homehedge77: yes, but it all gets compiled into the same logic anyway
15:23.38SuPrSluGi had someone ask about doing  video presentations, not conferencing, but would like it to be interactive. Anyone try streaming video w/ asterisk as a conference room. Possible?
15:24.16mvanbaakyeah, thanks to aelparse ;)
15:24.22Corydon76-homeAll AEL is is an abstraction layer above extensions.conf.  What you're relying on is for the experts to know neat tricks to convert your dialplan into
15:24.40wunderkinfrigidzephyr, i have an internal ticket, it needs an hpec developer person
15:24.52MikHellAnyone using les.net?
15:26.15*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
15:26.26frigidzephyrwunderkin, im sure they will get to it soon, what was the particular issue with HPEC?
15:27.40hedge77Corydon76-home: yeah but it is much cleaner to write.  Looking at the dump from aelparse -w is pretty funny though.
15:27.42wunderkinright, i have trouble even getting acknowledgements from them... license keys being held and kernel panics when using hpec and pri
15:28.10Corydon76-homehedge77: why not learn to code efficiently and bypass the abstraction?  ;-)
15:28.15*** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com)
15:29.27Corydon76-home(That was tongue-in-cheek; I don't expect everybody to bypass AEL)
15:31.22mvanbaaklol
15:32.19Corydon76-home<drevil>No, Mr. Powers, I expect them to DIE!</drevil>
15:34.35Goodjokei am a relative newb... any recommendations on how to config my digium tdm808b card?
15:35.09*** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com)
15:35.24mDuffOne of my users (on a Sipura SPA-841s) are complaining that "in the middle of a conversation... it sounds like someone just holds down a number button". I'm using dtmfmode rfc822; is there anything I should look into, other than swapping out the phone?
15:38.48*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
15:40.20*** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net)
15:41.56mvanbaaktell them to stop hitting buttons while on a call
15:42.14MrWupin a macro can i do waitexten and wait for an extension to be dialled?
15:42.17*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
15:42.30*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
15:42.55mDuffmvanbaak: I'm pretty darned sure she isn't doing that. The SPA-822 isn't anything like a candybar where one can hit buttons by mistake.
15:43.54MrWupcause inside i macro i have various s,1 s,2 priorities, then when i do waitexten and the user dials 1 (and 1,1,Dofunc) is in the macro, asterisk expects that 1 has to be valid in the context which called the macro
15:43.58drfreezeAnyone here ever used Yate?
15:44.00MrWuprather than valid in the macro itself
15:45.37mvanbaakhhmm
15:46.00mvanbaaknice thread with Theo and Michael about the broadcom wireless drivers
15:46.22b11dTheo is the man
15:46.37*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-c704bb8527fa35eb)
15:49.13mvanbaakbrb, fixing food
15:54.36MrWupIM IN A FIX
15:54.37MrWupoops
15:54.39MrWupcaps
15:55.24MrWupcontext1 calls macro1. we then jump out of macro1 and into context2. how can I make context2 dial the ${MACRO_EXTEN} of macro1
15:55.24MrWup?
15:55.44MrWupi.e. i need context2 to dial the original extension which the user tried to dial in context1 before we went into a macro
15:55.49*** join/#asterisk Waverly360 (n=irc@209.12.249.243)
15:56.10Waverly360Anyone here ever messed with the Asterisk CDR Areski GUI?
15:56.13*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
15:56.28Zeeekwhat do you asterisk for?
15:57.33*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
15:57.40ZeeekCuriosCat ? What happened to FlatCat?
15:57.52Zeeekand ScaredyCat?
16:00.33mvanbaakMrWup: Goto(exten2|${MACROEXTEN}|1)
16:01.00MrWupand that passes that variables which can be used in exten2?
16:01.54mvanbaakI have no idea what you want to do
16:02.22MrWupok the macro gets a variable from a php script
16:02.32MrWupthen we jump out of the macro into context2
16:02.38MrWupcontext2 need to be able to access that variable
16:03.57*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
16:04.10mvanbaakMrWup: did you read http://www.voip-info.org/wiki-Asterisk+variables
16:04.22MrWupi can set a channel variable right?
16:05.03mvanbaakyup
16:05.39MrWupah cool
16:06.51ZeeekIf you wanna talk about asterisk there's a conference at 12:30 PM EDT, i.e. in about 20 minutes
16:07.04ZeeekSIP access to conference
16:07.04hedge77gak pickup() why won't you find my channels ;_;
16:07.40Zeeekhttp://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
16:07.53ZeeekSIP conference
16:08.54[TK]D-FenderZeeek, and whats the URI to call in?
16:09.18ZeeekSee http://x2z.eu cause I don't remember any of it :)
16:09.34ZeeekAnyone can join but you need a PIN
16:09.59ZeeekI can give you guest PIN if you don't want to sign up
16:10.08Zeeek(If I can find the list somwhere)
16:10.22[TK]D-FenderZeeek, Sure...
16:10.49*** join/#asterisk ming_zym (n=ming_zym@124.254.55.212)
16:10.52ZeeekIf I give the PINs here whoever uses it fiorst will have it and it won't be good
16:11.17ZeeekI just got back off the road, I'm totally wasted so the conf should be uhhhhhh
16:11.25ZeeekI NEED SOMEONE TO TALK !
16:12.05Zeeekso, you can get your PIN here: http://www.talkshoe.com/talkshoe/web/tscmd/signup/1
16:12.40ZeeekJeeze,, I must still be featured on twitter
16:12.45Zeeekwait this isn't twitter
16:13.02*** part/#asterisk l2cache (n=ghansen@64.128.254.98)
16:13.03Zeeekjoinj asterisk-twitter
16:13.23[TK]D-Fendermaybe I'll just call in LD... that's only $1.80
16:13.53ZeeekIf you like - else try one of these: 2007 2007 20
16:13.57Zeeek2007 2007 00
16:14.03Zeeek2007 2007 22
16:14.14*** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net)
16:14.21Zeeekbut there's no danger in signing up, free, they won't spam you it just takes a second
16:14.56thinwiresHey guys I was wondering if someone could help with a Polycom phone setup, I keep getting "Url call is disabled"
16:15.05*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)
16:15.45[TK]D-Fenderdamn, you need a PIN no matter what.
16:15.53Zeeeksee above
16:16.08Zeeekyou also need the program id - read the page: http://x2z.eu
16:16.17Qwell[]umm
16:16.53Qwell[]why not just...setup a meetme?
16:16.53ZeeekThe program id is 22622
16:16.53[TK]D-FenderI'm in :)
16:16.53Zeeekbecause this will handle 1000's of calls
16:16.56Qwell[]and you have how many users currently?  5?  10?
16:16.59ZeeekAND live streams AND recording AND RSS all automatic
16:17.00b11dlol
16:17.04[TK]D-FenderIs this actually a CONFERENCE, or just a broadcast where no one can really talk?
16:17.13ZeeekI myself alm not called in so I don't know
16:17.21ZeeekEveryone can talk
16:17.24Zeeekand anyone
16:17.34ZeeekBut if no host, no one so I better get mioving
16:17.37*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
16:18.08*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
16:18.39aydiosmiosweet, my first T.30 fax through an ATA & *
16:19.03Zeeekyou can listen to the stream without logging in
16:19.11Zeeekhttp://x2z.eu for details
16:20.19[TK]D-Fenderhow many in?
16:21.13thinwiresso would anyone be able to help me out with a polycom ip650?
16:22.44Mahmoudare country codes always 5 digits?
16:23.12Goodjokei just did an install... how do i configure my digium tdm808b card?
16:23.25d00gsterI thought they were 3 Mahmoud
16:23.28*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
16:23.34ManxPowerGoodjoke: pretty much the same as the TDM400P
16:23.39hedge77are the linksys spa-941 phones any good?
16:23.54Mahmoudd00gster, http://www.lincmad.com/countrycodes.html
16:24.21Mahmoudd00gster, i think they just removed the 0 padding.. my country appears 971, while i use 00971
16:25.09Mahmoudi'm creating dial plans for long distance calls, and want to know how numbers look like
16:25.16Mahmoudusing SPA 3102
16:25.42Waverly360So no one has ever used the Asterisk CDR Areski GUI?
16:26.03hedge77cool kids don't need guis
16:26.04b11dit doesnt exist
16:26.09b11dhedge77, you're SO right.
16:26.24Waverly360It's for call billing
16:26.27b11di dont even have a monitor actually..
16:26.37Waverly360Management likes graphs and pretty charts
16:26.37b11dim just typing, and reading the results off the flashing led's on my switch
16:26.45Waverly360not txt files
16:26.47hedge77look for asterisk-stat
16:26.56b11di wrote a billing package called fat-pelt once..
16:27.03b11dthen i pretty much gave up on it :)
16:27.04*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
16:27.05Waverly360asterisk-stat is what I'm using.
16:27.21Waverly360on the voip-info pages it's called the CDR Areski GUI
16:27.27hedge77oh
16:27.37b11dvoip-info is all propaganda.. didnt you know?
16:27.44b11dits run by the "big telcos"
16:27.48b11dto discredit us
16:28.01thinwireshey guys does this "NOTICE[2767]: chan_sip.c:14354 handle_request_register: Registration from '<sip:10.1.1.114@10.1.1.202>' failed for '10.1.1.114' - No matching peer found" mean that the phone is trying to use "sip:10.1.1.114@10.1.1.202" as a username?
16:28.16b11dit means 10.1.1.114 coudlnt register as a sip peer
16:28.27b11dand yes..
16:28.37b11dits using 10.1.1.114 as its account name
16:28.56*** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net)
16:28.57hedge77~phones
16:28.58jboti heard phones is http://bani.anime.net/phones/.  SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom
16:28.58wunderkinit is broken, you must send it to me
16:29.10b11d~hedge77
16:29.15b11dnothing??
16:29.20b11d~b11d
16:29.21jbotb11d is a constant source of misinformation...
16:29.23b11d:)
16:29.23hedge77derp derp derp
16:29.41thinwireslol you want me to send you my brand new IP 650... I'm going to have to say no ;-)
16:29.42Waverly360hmm...who else has a comment?
16:29.51Waverly360~jbot
16:29.53thinwiresthanks for the infor b11d
16:29.56wunderkinhedge77, marklar!
16:30.06sevardthinwires: what about your old one?
16:30.11b11dany time thinwires :)
16:30.13wunderkini mean.. marklar, marklar..?
16:30.24b11dmarklar is marklar but only when marklar supercedes other marklar
16:30.32[TK]D-Fenderthinwires, the field labeled "address" in NOT THE IP ADDRESS OF THE SERVER.  it is your account name (the [whatever] in sip.conf)
16:30.39hedge77i totally marklared that marklar right in the marklar
16:30.44[TK]D-Fenderthinwires, thats the problem
16:30.46b11dmarklar.. totally f*ing marklar
16:31.11wunderkin[TK]D-Fender, deja vu, oui?
16:31.39thinwiresD-Fender: yes, I just got that one out, totally a boob on the side of polycom imo :-/
16:31.56Zeeekjoin us on Asterisk Conference - see http://x2z.eu
16:32.20*** join/#asterisk fbffff (n=fbffff@35.38.124.24.cm.sunflower.com)
16:32.35wunderkinpolycoms are easy once they boot and as long as they dont keep rebooting :D
16:32.46Qwell[]Zeeek: that's hardly an "asterisk conference", if it isn't running...*ON* asterisk
16:32.56hedge77anybody know when the polycom 320's will actually be available?
16:32.57Zeeekit is!
16:33.06wunderkinaren't they now?
16:33.06ZeeekTalkshoe uses asterisk on the stream
16:33.09thinwiresI've read about those problems, but this phone has been running like a champ
16:33.11b11d320s??
16:33.27hedge77http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-54293451520.htm
16:33.42b11dohh sweeet
16:33.53*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
16:33.54type0I think I'm going to win the award for the biggest piece of shit voip hack in the world, that works
16:34.26hedge77i dunno man i have to poll the manager iface every 2 minutes to make sure queue members are still valid
16:34.55thinwiresok now my phone just has a busy tone when I pickup, does that sound like a problem in my sip.conf?
16:35.16type0Transparent T-1 (via a t1 tdm over ip device) through a 500 mile microwave link, into a router, over to 2 ATA's out to a 450mhz wireless dialtone radio, 15 miles into a camp
16:35.46hedge77wow you win
16:36.23b11dtyp0.. how dare you bring technology that far into "gods country"
16:36.33type0i thought so
16:36.33b11dyou are the kind of person who RUINS WILDERNESS
16:36.34b11d:)
16:36.52type0nah, the construction guys ruin it.. i just give them interweb and dialtone
16:37.23b11dgive them a porn mag and a 6 pack and they'll be happy
16:37.27b11dthey dont need no "web" or "tones"
16:37.33type0but thats a good idea right?
16:37.52type0thats after work
16:37.56b11d:)
16:37.57type0people have to be able to dial the office
16:38.08type0the only other option is going on top of the mountain (5000ft) to make a call
16:38.13b11darent satellite phones cheap enough to use for that kind of thing yet?
16:38.14type0thats after driving 15 miles
16:38.16b11dwhat are they going for these days
16:38.16hedge77i thought that's why you got jobs in the wilderness, avoiding phones
16:38.18type0nah
16:38.20[TK]D-Fender~gs
16:38.21jbotmethinks gs is South Georgia and the South Sandwich islands, or ghostscript.  GrandSuck phones are cheap junk which should be avoided with extreme prejudice
16:38.23type0satellite is still pretty expensive
16:38.24[TK]D-Fender---^^
16:38.35b11dhedge77.. its why i'd work out there.
16:38.44b11dtk?
16:39.03hedge77guess the BLAST_GRANDSTREAM counter ran out
16:39.15type0i was thinking about getting some 2ghz microwave shit and shooting another t-1 down to the camp
16:40.10[TK]D-Fender?
16:40.29type0so i didnt have to use the ATA's
16:43.35pigpenQuestion:  when I have by g729 codec  registered, to activate it, do I need to restart asterisk?
16:43.59Qwell[]pigpen: You may be able to just load the module actually
16:44.17pigpenthanks...it as been awhile since I messed with it.
16:45.01pigpenerr..and if it isn't listed?
16:47.05apturaTK what is a ip phone that can take the elements?
16:47.40thinwiresok so asterisk is reporting that my SIP is registered and the phone is telling me I have a new voicemail, which is correct, but when I pickup the phone it goes straight to a dial tone... any idea's?
16:48.03wunderkin...
16:48.19apturaand it should
16:48.40*** join/#asterisk darken_darken (n=marco@21.140.76.83.cust.bluewin.ch)
16:48.43thinwiressorry, hahah straight to busy tone
16:49.20thinwiresso, I just called the line and it came through on the phone and worked, full duplex, but when I try to make an outbound call it goes straight to a busy tone
16:50.00[TK]D-Fenderaptura,  what "elements"?
16:50.30wunderkinh and o?
16:51.32*** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com)
16:52.02wunderkinso is it just picking up the phone you hear a busy tone (which ive never seen on a poly) or after you make a call, you get a busy, there is a difference
16:53.08thinwirespicking up I go straight to a busy
16:53.20*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)
16:53.39hedge77you mean ass soon as you lift the handset or right when you try and dial a number?
16:53.53thinwiresI just had someone call in from the outside and the conversation worked full duplex, soon as I lift the handset/speakerphone
16:54.11hedge77freaky
16:54.41thinwiresisn't it? it has to be some sort of problem with creating a call, it can recieve them just fine, maybe friend/peer/user issue?
16:54.42*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
16:54.52polerinhmm.. thoughts about AEL?  I'm used to C/PHP syntax so I can read it a bit better, but has anyone encountered problems with it?
16:55.14*** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net)
16:55.16*** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net)
16:55.24hedge77uhh if it's happening as soon as you take it off-hook the problem is probably with the phone
16:55.43hedge77since if you haven't dialed anything yet it shouldn't even be talking to *
16:55.53thinwiresif I take the phone off the hook would that show anything in the asterisk console?
16:56.00thinwireshm
16:56.34hedge77you can try sip set debug peer <extno> and see if anything comes through
16:57.05*** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)
16:57.45*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
16:58.08*** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)
16:58.36thinwireswait, I think i got it
16:59.10*** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net)
16:59.16thinwireslol ok now I have the problem where it goes to a busy after I dial the number
16:59.21*** join/#asterisk nybble (n=nybble@about/apple/performa/nybble)
17:00.37hedge77well then the asterisk CLI should give you a reason for it
17:00.51Waverly360Ok, in researching CDR and all of it's glory, I'm finding two conflicting schemas for the CDR database.
17:01.19thinwireser, how do I turn the debug off?
17:01.28hedge77sip no debug i think
17:01.31Waverly360One schema includes start, and end times for the calls
17:01.34Waverly360the other doesn't
17:01.39bkruseset set debug off
17:01.41Waverly360There's also a calldate field
17:01.42*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:01.43bkrusein 1.4
17:01.53Waverly360What I don't understand, is which field I'm supposed to grab the date from
17:02.12bkruse~pb
17:02.19jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
17:02.26b11d~bkruse
17:02.35b11dwow.. nothing again..
17:02.40bkrusehaha
17:02.42bkruse~lart b11d
17:02.42jbotburns b11d to a crisp with a laser
17:02.46b11d:)
17:02.49bkruse:D
17:02.51b11dsweet.. crispy b11d
17:02.57Waverly360Do most people consider calldate the date in which the call started, or the date when the call ended?
17:02.58bkruseumm, yummy
17:03.01polerinheh
17:03.01b11dhaha
17:03.29b11dcalldate has to be the origination time, i'd wager
17:03.32b11dit wouldnt make sense otherwise
17:03.33*** join/#asterisk nybble (n=nybble@about/apple/performa/nybble)
17:04.17Waverly360that's what I was thinking..though in a billing sense, if you talked on the phone throughout a night for example, and your 3 hour call ended at 2 AM.  Would they want to bill that to the next day?
17:04.58b11di'd bill it to the previous day
17:05.04b11dor split it up :)
17:05.12Waverly360let's not get crazy now :)
17:05.16*** join/#asterisk zavoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
17:05.24zavoidhey all
17:05.32nybblehelloo
17:07.09zavoidanyone ever have trouble with bridging g723 and g729?
17:08.10Qwell[]zavoid: You can't bridge different codecs..
17:08.11hedge77do polycoms only support [ua]law and g729?
17:08.13zavoidi can bridge g.711 to g.729 but not g.723 to g.729
17:08.28zavoidwell not brige but bring one codec into * and out the other end with a different codec
17:08.29*** join/#asterisk Fieldy (i=8RyPR4lV@gentoo/contributor/Fieldy)
17:08.38hedge77transcode you mean
17:08.42zavoidyes sorry
17:08.51Qwell[]asterisk can't transcode g723
17:09.01Qwell[]not without hardware to do so
17:09.21apturaso that takes the load off the cpu ?
17:09.31apturaI figure it would be.
17:10.09zavoidbut it can transcode g711?
17:10.18thinwiresok so this is the output I get from the debug http://pastebin.ca/427133 , if I have the debug off the console doesn't show any information when I try to make a call...
17:11.40mvanbaakI'm off
17:11.41mvanbaaklatero
17:12.38codefreezeWaverly360: which database are you using? it may not matter; the "start" date is what the backends throw into the date field in the db's.
17:13.31Qwell[]zavoid: please don't message people
17:13.39zavoidsorry qwell
17:14.53zavoidso qwell why can i can transcode g711 but not g723? is there anyway i can do it other then hardware?
17:15.18Qwell[]no
17:15.37Qwell[]not unless you want to spend a couple hundred thousand dollars for licenses
17:17.11zavoidfor g723 license?
17:18.40*** join/#asterisk Matrix9 (i=MiniMe@s142-179-197-109.ab.hsia.telus.net)
17:19.13Qwell[]yes
17:21.56apturaare there any current blue tooth ear pieces that can interface with asterisk? I have a company that sounds interested if the interfacing would work.
17:22.19Qwell[]aptura: there is chan_cellphone on the bug tracker that can do headsets
17:22.26apturaoverhead paging is difficult to listen to with power equipment running.
17:22.48apturaso thay would need a cell phone with blue tooth for it to work.
17:22.57Qwell[]no, it can use headsets
17:23.02apturaI see
17:23.16apturahow about the ear piece?
17:23.22Qwell[]same thing
17:23.33polerinQwell[]: what's the interface for dialing like on that?
17:23.42Qwell[]polerin: what, bluetooth?
17:23.53apturaand how good is asterisk with voice recognition ie, "call john" so it would call his extention
17:23.53polerinyeah, for an "answer only" headset
17:23.56Qwell[]same as zap, basically
17:24.10Qwell[]aptura: asterisk doesn't do voice recognition
17:24.15apturaokay
17:24.18apturaat least not yet
17:24.22Qwell[]there are third-party addons, but they aren't recognition
17:24.24*** join/#asterisk saftsack (n=oliver@p54a7cc8f.dip.t-dialin.net)
17:24.26*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
17:24.32Qwell[]well, ignore that
17:24.47Qwell[]but it's not gonna know every word you say
17:25.11polerinaptura: tbh have you looked at having a softphone on the computer with bt?  that's an easier way to do it
17:25.13iruleif I type wrong nuimbers I get a busy signal, how can I change it to exten => i,1,Playback(pbx-invalid)exten => i,n,Goto(s,restart)
17:25.23apturaBC translink has the best voice recognition technoligy I have seen so far. You can say and street or ave from destination and then give you the buss info.
17:25.23polerin(I think ... still definatly a newb so..)
17:25.51hedge77oh wow is pickup actually that broken in 1.4
17:26.17[hC]aptura: ah, another vancouver guy i see.
17:26.18apturabut Idealy want all there employees and and managers on a hands free blue tooth and skill the overhead intercom.
17:26.27apturayup
17:26.28hedge77you have to specify the *calling* context to get it to work.
17:26.34apturaskill=skip
17:27.11hedge77aptura: but then they'll look dumb
17:27.30hedge77"take out the earpiece, you're not uhura"
17:28.26apturathay dont care
17:28.27*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
17:28.41polerinthat would need the BT base antenna in a good location
17:28.48apturawaisting time running back and forth for the wall phone is taking away from there bottom line.
17:28.59polerincan you have BT be cellular correctly?  Haven't looked
17:29.35hedge77bt's range isn't that great if the building is that big it's probably not what you want
17:29.35apturaso which version includes chan_cellphone
17:29.45apturait is small
17:30.15apturanow can blue tooth mesh antennas be installed
17:30.49[hC]aptura: do you work for a voip company in vancouver, or are e ou doing this on your own?
17:33.05*** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net)
17:34.45*** join/#asterisk mrichmanM (n=richmanm@70.89.184.1)
17:34.58thinwireson a polycom phone in the SIP Config -> Server ->, should I use TCP Prefered for Transport?
17:35.42[TK]D-Fenderthinwires, no
17:36.15thinwireswhich should I use? I had it on DNSNaptr and I couldn't recieve calls
17:36.25*** part/#asterisk ming_zym (n=ming_zym@124.254.55.212)
17:36.30[TK]D-Fenderthinwires, thats what t should be.
17:39.45thinwiresD-Fender: any ideas why that would cause me to not get the calls but it shows "missed call" immediately, even shows the correct number for the caller id
17:42.35Hmmhesaysyet another friday
17:47.19*** join/#asterisk friedrich| (n=friedric@e177246045.adsl.alicedsl.de)
17:48.12[TK]D-Fenderhttp://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/
17:49.18[TK]D-Fender~docs
17:49.19jbotmethinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
17:49.21[TK]D-Fender~book
17:49.23jbot[book] a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:49.30[TK]D-Fenderthinwires, hold on a sec...
17:50.34mrichmanMIs their a way to connect a web interface or gui of some sort to configure asterisk?
17:51.18zavoidso is the DIGIUM TC400B the only card then can do g.723 to g.729 transcoding?
17:51.22*** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
17:51.36Qwell[]zavoid: I don't think it can do g723 to g729 directly.
17:51.45irulemrichmanM http://www.voip-info.org/wiki-Asterisk+GUI
17:51.48zavoidit says it can
17:51.56Qwell[]not to each other
17:52.03Qwell[]it can indirectly though, I think
17:52.39zavoidbut also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible.
17:52.41zavoidhmmm
17:53.02zavoidi miss real media gw's lol
17:53.14*** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com)
17:53.53*** join/#asterisk goldsmurf (n=rgoldber@h216.70.9.114.superiorbroadband.com)
17:55.54goldsmurftruly apologize for the noobish question, but all my incoming sip invites are being responded to with 404s.  I've got the simplest config you can imagine, and I'm not able to come up with the right terms to google for a answer.
17:56.33b11dgargamel is probably up to his old tricks
17:57.35goldsmurfI appear to be properly registered (user and peer) with my sip provider (outbound calls are fine)
17:59.04goldsmurfit would appear to my asterisk-untrained eye that * is not even trying to drop the call into the context
17:59.49d00gsterany nat experts around?
18:00.43*** join/#asterisk nickpiter (i=nickpite@ppp85-140-70-122.pppoe.mtu-net.ru)
18:01.03mrichmanMirule: Thank you as i read them they are complete solutions I was hoping to find something i can put on top of an existing install
18:01.10nickpiterwho have working callback configurations ?
18:01.51nickpiterCalling process is ok for both sides,
18:01.51nickpiterbut when 1st call-leg established, script is executing hangup
18:01.51nickpiterbut call is not hanging phisically, after this asterisk normally calling to 2nd side and establish bridging.
18:01.51nickpiterBut i not need this hangup, it is cancelling billing process for 1st leg.
18:02.45*** join/#asterisk Maghteridon (i=ValleDiL@88-149-166-76.f5.ngi.it)
18:02.46Maghteridonhi...
18:02.49Maghteridonagain =)
18:03.07HmmhesaysI've done callback apps in the past
18:03.24Hmmhesayscron is your friend
18:03.56MaghteridonI have a couple of questions, quite easy I suppose... First one is... I'd like to enable the possibility to make a blind/attended transfer on  an extensions that also have a voicemail timeout
18:04.13Maghteridonexten => 7199,3,Dial(SIP/7199,10||tT) <-- doesn't seem to work, since I lose the possibility to make the transfer
18:04.18Maghteridonwhich is the correct sintax?
18:04.36Hmmhesayswhy would you lose the possibility to make a transfer there?
18:04.46nybblei'm thinking of writing a call back script... takes email request from my cell phone, and calls me back and connects my number... that way i take advantage of free incoming calls on cell phone... any opinions/observations on this matter?
18:04.51polerinHmmhesays: have you done ~wardialing type stuff?  (actually an appoint reminder, but yahknow)
18:04.58MaghteridonI have no clue, but if I write exten => 7199,3,Dial(SIP/7199||tT), the transfer works, while if I add the ",10" it doesn't
18:05.38Hmmhesaysnybble: if you have some free termination service to call your cell phone I don't see any problems
18:05.52nybble:D
18:06.00HmmhesaysI wrote a little app in perl that did something similar about a year ago
18:06.01*** join/#asterisk CrossRoad (n=SilentVa@209.172.67.146)
18:06.28nybblei'm thinking php mail grabber, running with cron every say five minutes... that way i'll get the 'call' within a few minutes of the request
18:06.51Hmmhesaysyeah php would be super simple also
18:06.54Maghteridon(the second question is... what is the default key to accept a transfer, in the attended transfer?)
18:07.03nybblenot to mention, i'm familiar with php
18:07.13CrossRoadany suggestion on good outbound only provider
18:07.13bkrusei LOVE php
18:07.24bkrusei need to open source the php stuff ive done for asterisk
18:07.25goldsmurfok so perhaps I do not understand "s".  now that I specify the DID as the extension, the call comes in.  I thought I could use just "s" in my incoming context to grab everything.  IT that not the case - does it have something to do with the way the invite is formed?
18:07.32MaghteridonCrossRoad such as voipcheap?
18:07.38bkrusemainly just alot of routines and classes that are easy to use for agi and web based integration
18:08.21CrossRoadMsghteridon: Yes.. we are basically looking for a secondary outbound provider
18:08.35Hmmhesayss will grab something without a destination number
18:09.26goldsmurfok - so obviously if the invite specifies a number, s gets skipped over.  further, I suppose all incoming sip invites have a number.
18:09.55nybblehmmm... i need outbound that allows me to specify CID settings
18:10.00nybblecurrent one doesnt
18:10.01goldsmurfis that the case (I realize I need to do more learning about sip - I will)
18:10.08mDuffDoes MixMonitor not support MONITOR_EXEC?
18:10.39CrossRoadnybble: same issue for us.. our current outbound provider does'nt support CID
18:10.39Hmmhesayssome devices won't send a number
18:10.54Hmmhesaysusually s is reserved for use in macros though
18:11.29nybbleyea, CrossRoad. all i was looking for was setting CID Name, dont care about modding the number... suggestions anyone?
18:11.37goldsmurfHmmhesays, ok good to know.  I obviously currently lack experience with asterisk and voip in general.  Thanks for the answer.
18:11.46nybblecanadian DiD
18:11.56Hmmhesaysfeel free to donate to me via paypal, lol
18:12.09goldsmurf$.25 is cool?
18:12.38Hmmhesayssure
18:12.38Hmmhesays<--gmail
18:12.50Hmmhesaysthat way I can tip the girl at the club tonight
18:13.06Hmmhesays3 second lap dance woooo
18:13.21*** join/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg)
18:13.48nybblelol
18:13.48[TK]D-FenderHmmhesays, I say G3 live on wednesday :D
18:13.48[TK]D-Fendersaw*
18:13.48Hmmhesaysbah really?
18:13.48Maghteridon[TK]D-Fenderdo you have any clou about what I said? :P
18:13.48[TK]D-FenderHmmhesays, floor seats :D
18:13.49HmmhesaysI bet that was awesome
18:14.19[TK]D-FenderHmmhesays, Indeed.  Joe Satriani, John Petrucci, and Paul Gilmour.
18:15.04HmmhesaysI really only like satch out of the 3
18:16.11[TK]D-FenderHmmhesays, I liked Petrucci with Dream theater / Liquid Tension Experiment, and only knew Gilmour from Mr. Big.  I knew that he was a great technical player and you might be surprised byt he show.
18:16.47[TK]D-FenderMaghteridon, What was your question?
18:16.53Hmmhesaysi've only seen Gilmour on youtube
18:17.05Hmmhesaysplaying some ungodly fast riff
18:17.40[TK]D-FenderHmmhesays, he was previously from Racer-X, so go figure ;)
18:17.56Hmmhesayshaha yeah
18:19.24[TK]D-Fenderoops.. gilbert
18:19.25Cybertoyhttp://zfoneproject.com/partners.html ... anyone know what happened to zfone and asterisk? according to their page it should be available by the end of March...
18:19.34Cybertoy... allthough it does not mention what year... :)
18:20.31Hmmhesayshhaha
18:21.02Hmmhesaysheh Mr. PGP
18:21.26CrossRoadHi all.. any suggestion on (business class) outbound call termination providers with CID support?
18:23.25HmmhesaysI dunno, I use voipjet quite a bit
18:23.32CrossRoadare they good?
18:24.07Hmmhesayssome have had bad experiences, some good
18:24.17HmmhesaysI use vitelity as a backup
18:24.18*** join/#asterisk Fieldy (i=m41kheaz@gentoo/contributor/Fieldy)
18:24.33Hmmhesaysbetween those two I have 0 complaints
18:24.38syzygyBSDhow can I manually specify the tftp boot server for my polycom 501?  it can't be given from my dhcp server
18:24.40CrossRoadHmmhesays: do they support CID?
18:25.25Hmmhesayscallerid?
18:25.34CrossRoadyes
18:25.40Hmmhesaysyes
18:25.45Hmmhesaysyou can set your outbound cid
18:25.47CrossRoadoh cool thanks
18:25.57frigidzephyrhmmhesays: are they only per min?
18:25.59[TK]D-FendersyzygyBSD, enter the IP in your bootrom
18:26.09syzygyBSDthanks
18:26.17Hmmhesaysfrigidzephyr: yes
18:26.19syzygyBSDI was trying to find it through the web configuration
18:26.29nybblehmm... whatsa good way to accept input on a menu while still having the playback playing (so you dont have to neccessesarily wait for playback to finish before keying selection) <- yes i know, bad spelling
18:27.12[TK]D-FendersyzygyBSD, its not in there.  thats SIP app config, not boot IIRC
18:27.39[TK]D-FendersyzygyBSD, reboot the phone and go into setup
18:27.39syzygyBSDthanks
18:27.51[TK]D-FendersyzygyBSD, And what in the name of God are you doing in the web config?  Don't make me come over there and hurt you! ;)
18:28.04hedge77[5~/away
18:28.13nybbleperhaps waitexten
18:28.14syzygyBSDwell, before today I only managed 1 or 5 polycoms
18:28.21nybblenope i stupid
18:28.49Qwell[]1 or 5?
18:29.01nybblelol!
18:29.15[TK]D-Fenderqwell : Illogical operators ;)
18:29.30*** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net)
18:29.43Hmmhesaysi'm really suprised at how well cmd mysql works
18:29.44*** join/#asterisk Fieldy (i=YawCysV3@gentoo/contributor/Fieldy)
18:29.56Hmmhesaysi have had zero problems with it on a production system for 6 months
18:31.16syzygyBSDprobably closer to 10, but they are all at different offices
18:31.29nybblewow. i've been playing the Command and Conquer soundtrack on repeat for the last three hours
18:31.38syzygyBSDwell some of them are, 8 are at one office, 4 at another... more somewhere else
18:32.01nybbleback in a bit
18:32.06hedge77voipsupply is claiming mid-april for the polycom IP320's to be available.  Do you think that's accurate?
18:32.38Hmmhesaysnybble thats a little odd
18:32.43syzygyBSDI am wondering when I am getting my 7960 from voipinfo...
18:33.05HmmhesaysI used to use a 7960 extensively
18:33.46Hmmhesaysgreat speaker phone
18:33.51syzygyBSDI never have, but I have to support a ton of them now, so I have to have one to test
18:34.00syzygyBSDwill be my 5th phone on my desk
18:34.24*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
18:34.30*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:34.35hedge77everybody post your phone counts time for e-peen stroking
18:34.57b11di've got one phone
18:35.10hedge77you fail at e-peen
18:35.22syzygyBSDhedge77: your count?
18:35.41Hmmhesays7960's are cake to administer a large amount
18:35.42syzygyBSDI should go grab a stack of them just so I can through out a number like 30
18:35.46b11dactually looking around, i've got four.
18:35.57b11donly one connected
18:36.08b11dno wait.. i've got a box of six 430's under my desk..
18:36.10Hmmhesayshow much you want for one b11d?
18:36.12b11dso like.. i've got 10
18:36.13syzygyBSDHmmhesays: ya, they are all setup already, but if there are any problems (like today) I have to deal with it
18:36.14hedge772 7960's, 2 analog into a PAP2, 2 grandsucks.  Plus soft phones
18:36.24syzygyBSDb11d: on your desk...
18:36.52HmmhesaysI need a 7960 again
18:36.58hedge77i think all the power adapters might be a fire hazard
18:37.15b11di can move the box up to my desk
18:37.44hedge77branch offices :argh:
18:37.50red9012t.38 fax, anyone using it in asterisk?
18:38.09Hmmhesaysi use the fork to handle my faxes
18:38.26syzygyBSDwho faxes anymore?
18:38.27b11dyou got that all straightened out then Hmmhesays?
18:38.33syzygyBSDemail
18:38.43hedge77somehow we are set up with a fax going into a pap2 and it works.  I have no idea how and I'm afraid if I touch it it will stop working
18:38.44Hmmhesaysb11d: its ugly, but it seems to be working for now
18:38.47b11dcool
18:39.02Hmmhesaysi use app_txfax and app_rxfax
18:39.06red90121.4 seems to be suppporting it.
18:39.26Hmmhesaysyou can compile the apps into 1.2 also
18:39.32Hmmhesaysbut if you want bleeding edge fax, look elseware
18:39.33b11dyou need spandsp for those, right?
18:39.36syzygyBSDyes
18:39.37Hmmhesaysyeah
18:39.46gambolputtyThe Internet at night:  http://upload.wikimedia.org/wikipedia/en/d/d2/Internet_map_1024.jpg
18:39.56syzygyBSDlol, bleeding edge fax.. isn't that an oxymoron?
18:40.07b11dneato
18:40.20HmmhesayssyzygyBSD: yeah kind of
18:40.32[TK]D-FendersyzygyBSD, No, he still uses the old thermal paper roll models and let the cutter rust a little ;)
18:40.36Hmmhesaysugh do I dare implement func_odbc on a production system
18:40.47Hmmhesaysbwhaha
18:40.48b11dI triple dog dare you
18:40.54red9012hmmhesays  -- whats the main problem if you've had with fax support?
18:41.11Hmmhesaysdifferent hardware talking t.38 to each other
18:42.30Hmmhesaysthats what I had to set a t.38 gateway in the middle
18:43.23Hmmhesaysi'm not so sure about asterisk 1.4 yet
18:43.43Hmmhesaysmost of the functionality that is added I've found elseware
18:43.56Hmmhesaysexcept the 3rd party module chanskype
18:45.16Hmmhesaysand the room goes silent
18:45.31b11dlol
18:45.57Hmmhesayschanskype actually works pretty well, giving outside users the ability to call into your system via skype is a good thing
18:47.27iruleI notice that exten => _.,1 does not work in 1.4 folowing these instructions http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension what gives_
18:47.49Hmmhesayswhy would you want to use _. ?
18:48.27apturato be lazy
18:48.29aptura:)
18:48.51*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
18:49.45[TK]D-Fenderirule, pastebin your entire context and the device calling into it.
18:49.47[TK]D-Fender~pb
18:49.51jbothmm... pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
18:50.04*** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net)
18:50.07iruleI want to catch all undefined numbers and kindly tell user to dial correctly
18:50.13iruleok
18:50.29[TK]D-Fenderirule, its not that we don't trust you... its just the we don't trust you ;)
18:50.48BSD_Techso you monitor all dialed number
18:50.54syzygyBSDthe we don't trust you?
18:51.09BSD_Techand you want to let your client know you monitor the numbers they dial
18:51.51syzygyBSDumm, isn't that like the phone company giving you the message, "the number you dialed is no longer in service"
18:52.06syzygyBSDONO THEY MONITOR ME!
18:53.20irulehttp://www.pastebin.ca/427324 lol here it is
18:53.37no-i-rulesorry
18:53.45no-i-rulei wish i could use commas
18:53.51MaghteridonCan anyne tell me the right syntax to have an extension with timeout & capable to make blind/attended transfers?
18:54.06[TK]D-Fender"Punctuate!", he exclaimed.
18:54.20no-i-rule:)
18:54.20[TK]D-Fenderirule, Ok, where should I be looking in there?
18:54.21Maghteridonexten => 7199,3,Dial(SIP/7199,10||tT) doesn't work... It doesn't allow me to do the transfer
18:54.48[TK]D-FenderMaghteridon, read the instructions again, your parameters order is WRONG. "show application dial"
18:55.40[TK]D-FenderMaghteridon, And while you're at it, pick a delimiter and stick with it...
18:56.02Maghteridon[TK]D-Fender, actually if I remove the ",10" it works..
18:56.38*** join/#asterisk shinux__ (n=shinux@86.62.8.178)
18:56.50[TK]D-FenderMaghteridon, Indeed.  thatsbecause you are putting your options in the 4th paramter where they do not belong.
18:57.06_VoicemeUpDotComhehe coman and pipe
18:57.11[TK]D-FenderMaghteridon, go read the INSTRUCTIONS again.
18:57.28[TK]D-Fender_VoicemeUpDotCom, Yup, thats what I said...
18:57.38_VoicemeUpDotComDial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):
18:57.50MaghteridonI read it in the voip... info site O_o exten => _XXX,2,Dial(SIP/${EXTEN},5,mTt)
18:57.51_VoicemeUpDotComso |10|tR
18:57.53_VoicemeUpDotComor whatever
18:58.08_VoicemeUpDotComyou have .. ,10, SPACE, options
18:58.09_VoicemeUpDotComthat not rigt
18:58.19_VoicemeUpDotComyour passing your options as a url
18:58.30_VoicemeUpDotCommove all left one bit
18:59.00_VoicemeUpDotComexten => 7199,3,Dial(SIP/7199,10,tT)
18:59.04_VoicemeUpDotComfor reading impared
18:59.32[TK]D-Fender_VoicemeUpDotCom, Reading is probably fine.. its the post-processing I'm worried about ;)
18:59.35Maghteridon_VoicemeUpDotCom last string you wrote was my first attempt and it didn't work..
18:59.40_VoicemeUpDotComoh.. a code 18
18:59.55Hmmhesayshave any of you guys ever gotten --- (0 headers 1 lines)---09  asterisk when trying to sip debug a peer?
18:59.57_VoicemeUpDotComhappens alot.. like the client who called me nd said our service was down.. ( she had a power outtage and using xlite)
19:00.15demlakhi
19:00.22[TK]D-Fenderirule, well?
19:00.29iruleplease excuse my delay, this is supposed to be the one that simply catches nonexisting unmbers that begin with 7 & 8 exten => _[7-8],1,Playback(pbx-invalid)exten => i,n,Goto(s,restart)
19:00.43_VoicemeUpDotComadd a dot
19:00.48Qwell[]remove a -
19:00.52_VoicemeUpDotCom<PROTECTED>
19:00.53_VoicemeUpDotCommaybe
19:01.01iruleoh yes I tried with a dot and it did not work
19:01.12iruleIll try  _[78]. thanks
19:01.16_VoicemeUpDotComi think you need either dot.. for anything else after. . or fill the full space with N XX's
19:01.33_VoicemeUpDotComso either _X.,1, or _XXXXXXXXXXX,1,
19:01.35_VoicemeUpDotComlike
19:02.00_VoicemeUpDotComQwell,  hehe didnt know about the dash part thanks..
19:02.10_VoicemeUpDotComas brian would say .. NEXT !
19:03.08Hmmhesaysbah these people have a lot of sip phones already running, but x-lite doesn't work
19:03.44_VoicemeUpDotComyeah.. xlite .. uses the display as the from sip user...
19:03.52iruleI get this [Apr  6 08:00:28] WARNING[1346]: file.c:553 ast_openstream_full: File pbx-invalid)exten => i does not exist in any format
19:04.16_VoicemeUpDotComshow dialplan _[tab]
19:04.19_VoicemeUpDotCompastebin
19:04.46irulewho! me? :s
19:04.53_VoicemeUpDotComyes
19:04.57_VoicemeUpDotComirule,  ;()
19:05.02irulewhat is that? lol
19:05.37*** part/#asterisk Cresl1n (i=matt@nat/digium/x-7b5136890c000103)
19:05.51iruleI put my complete dialplan on http://www.pastebin.ca/427324
19:05.52hedge77irule: wouldn't it actually be like exten=>i,1,Playback(pbx-invalid);  NEWLINE exten => i,n,Goto(s,restart); ?
19:06.19irulebeats me
19:06.28iruleIll try that lol
19:06.29hedge77i do this in ael so i have no clue
19:06.32_VoicemeUpDotComyou still need to remove dashes
19:06.46_VoicemeUpDotComexten => _[7-8],1,Playback(pbx-invalid)exten => i,n,Goto(s,restart)
19:06.46irulewhich dashes?
19:06.54irulethanks
19:06.55_VoicemeUpDotComyou mising a enter.. and
19:07.11_VoicemeUpDotComstart there.. repaste new extensions for default
19:07.26_VoicemeUpDotComcomment exten => 700,1,Goto(s,1)
19:07.29_VoicemeUpDotComfor now
19:07.33_VoicemeUpDotComalso
19:07.39_VoicemeUpDotComif you gonan use 700 as well..
19:07.58_VoicemeUpDotComyou need X to order i think..  _[78]X.
19:08.13iruleok
19:08.18_VoicemeUpDotComso 700 is parsed before/after the catch all..
19:08.37_VoicemeUpDotComand i still see no dots... do waht we said..repaste..prey
19:08.53*** join/#asterisk J4k3 (n=jsuter@openwrt.us)
19:08.58_VoicemeUpDotComde donde estas ?
19:09.04hedge77so what is the point of doing exten => 600,1,Macro(stdexten,600,SIP/sip600) a bunch of times instead of exten => _6XX,1,Macro(stdexten,${EXTEN},SIP/sip${EXTEN}); one time?  assuming the extension numbers match the sip users that is.
19:09.07iruleGuadalajara
19:09.14*** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
19:09.17_VoicemeUpDotComporque tienes coment en espanol
19:09.31_VoicemeUpDotComah .. bien.. me fui ahi 2-3 veces
19:09.51_VoicemeUpDotComtengo una chica ahi lol
19:09.58irulejejeje
19:10.04hedge77kekeke ^_^
19:10.10[TK]D-Fenderhedge77, When 640 doesn't exist things dont crap out.
19:10.17_VoicemeUpDotCommas que una ..pero.. shht ..
19:10.20irulechicks rule
19:10.31[TK]D-Fenderhedge77, And it lets the phone get a 404 to know its not valid.
19:10.49irule_VoicemeUpDotCom I cant even handle one babe propperly lol
19:10.56_VoicemeUpDotComhhe
19:11.01_VoicemeUpDotComviagra man
19:11.27iruleof a fist up there to start her grin
19:11.48nybbleback
19:12.23*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
19:12.42J4k3yo guys, plz don't laugh too hard... but I need to emulate a POTS line for modem testing purposes...  my ILEC-provided landline is so noisy/worthless that I can't get anything to work across it except my best frying eggs impression.
19:13.15J4k3can one get away with pushing data through the pap2
19:13.39_VoicemeUpDotComdepends
19:13.51_VoicemeUpDotComlower baud rates to 9600 and prey
19:13.56hedge77depends on what dark magic you use
19:14.02_VoicemeUpDotComalots of preying needed for analgo crap + modem + data
19:14.24apturaJ4k3 did you call them to ask that it be troubelshooted? could be bad crimp connection or grounds on there end.
19:14.43_VoicemeUpDotComactually i found best result with #1 sacrifying a virgin chicken on a full moon while dancing on your left leg only surrounded by 21 virgins
19:15.12_VoicemeUpDotCom#2 drink the blood of the reult of #1.. and #3.. see a doctor
19:15.28J4k3aptura: yeah, but its like a weekly re-occurance...  its also like $47/month for the second POTS line :o
19:15.30nybblehmmm
19:15.37nybbleold p3 laptop = great personal asterisk box
19:15.38J4k3(after taxes and whatnot)
19:15.45_VoicemeUpDotComJ4k3,  wahts your 20 ?
19:16.01J4k3nybble: You can find P-M and C-M dell laptop 'bases' (stripped down lower ends) for under $200
19:16.03_VoicemeUpDotCom47 a pots sounds like .. really foreign
19:16.22J4k3Texas, Windstream, Contel tarrif.
19:16.26nybblei'm using an old Compaq Armada m700
19:16.31_VoicemeUpDotComah
19:16.32Hmmhesaysi used to run asterisk with gnugk on a pII 233
19:16.38syzygyBSDhuh, I never knew there were virtual IP addresses...
19:16.41_VoicemeUpDotCom2nd biggest state of usa
19:16.43nybbleits nice to mount it on the ol' wall
19:16.43hedge77what's the contel tarrif?  an extra $30 charge?
19:16.53Qwell[]syzygyBSD: well, there aren't physical IP addresses
19:17.04J4k3well, this is a business POTS line.
19:17.05syzygyBSDwell, I meant with vrrp
19:17.07J4k3not residential
19:17.16_VoicemeUpDotComa telco here charges long distance from on nxx to another nxx.. they in same building same provider lol
19:17.18_VoicemeUpDotComtelus
19:17.28hedge77O_o
19:17.31J4k3hahah daaaamn
19:17.32J4k3thats balls
19:17.33_VoicemeUpDotComoh .. same switch also
19:17.35syzygyBSDtexas is the second biggest state?
19:17.41syzygyBSDthought it was montana...
19:17.48_VoicemeUpDotCom=MTRLPQ50FMD
19:17.57_VoicemeUpDotComtexas i think.. alaska the #1
19:18.03Qwell[]syzygyBSD: I thought it was the biggest...
19:18.05_VoicemeUpDotComfor now..
19:18.09Qwell[]stupid geography
19:18.13syzygyBSDalaska
19:18.16apturaAlaska would probebly be the first
19:18.18b11dalaska by far
19:18.24syzygyBSDya, it is 1/3 of the US
19:18.24_VoicemeUpDotComwith all the metling... maybe alaska wil be the smallest some day soon
19:18.25b11dyou can fit half the USA into alaska
19:18.29b11dpretty much
19:18.33Qwell[]apparently I didn't pay attention in elementary school either
19:18.36b11dcontentental USA that is
19:18.42_VoicemeUpDotComhehe learned it on ..think you can beat a 5th grade lat night
19:18.44J4k3you could put most of the american population in cold storage up there.
19:18.49J4k3humans, cows, and everything else.
19:18.53Qwell[]_VoicemeUpDotCom: oh, I could beat a 5th grader
19:18.58_VoicemeUpDotComcause i never remember when canada GAVE alalska to usa
19:19.00Qwell[]however, that isn't the title of the show
19:19.06_VoicemeUpDotComthats lame..
19:19.14apturaRussia owned Alaska
19:19.20syzygyBSDlol Qwell
19:19.22b11dyeah
19:19.28apturaand US purchaced it for pennies on the dollar.
19:19.36Qwell[]pennies on the million
19:19.42hedge77so how do they go about picking contests for that?  Do they intentionally look for dumb people or ask tricky questions?
19:19.49_VoicemeUpDotComimho nothing is owend by anyone.. indians will get us all.. they still want gov financing and help but host 80% of online casinos gambling license
19:19.51syzygyBSDthat was before they knew there was oil up there
19:19.59hedge77yeah alaska was a great deal
19:20.02apturayea
19:20.03aptura:)
19:20.05syzygyBSDrussia already had enough frozen tundra
19:20.30apturaright now the next rush is Alberta Oil fields
19:20.37_VoicemeUpDotComhttp://www.kahnawake.com/gamingcommission/
19:21.14apturaproblem is its so cold that it takes alot of energy to extract the oil out of the -20F sands and that means burning alot of oil to do so.
19:21.16_VoicemeUpDotComyeah.. that going to be bad.. after calgary is pumped out youll get unemployment of 50%+
19:21.19b11dthe tar fields of alberta
19:21.29Hmmhesayslol
19:21.45_VoicemeUpDotComhouses went up 180% in last 5 years
19:21.47apturaVoicemeU so true. Alot of the processed oil is going to the States.
19:21.54_VoicemeUpDotComgood investments still for another 4
19:22.02*** join/#asterisk kje (n=kje@lime-gw16.one.at)
19:22.27_VoicemeUpDotComstupid paul martin .. made a deal with mini bush for electricity.. he tried to force hand ais selling to USA  .. BEFORE ourselves
19:22.44apturaI just think as a county and as a world we need to think that oil will start to come to a end. It took millions of years for the earth to create crude oil and mankind it going to burn it up in 200 years.
19:22.57_VoicemeUpDotComoh check this out hold on
19:23.00hedge77can't you get paid an allowance to go live up in the territories?  or is that just for teachers and such.
19:23.18apturato frigen cold up there
19:23.21hedge77forget oil, I look forward to the day we conquer canada for their fresh water
19:23.29hedge77mmmmm glaciers
19:23.43apturayou can make a great living in Fort Saint John but its very cold.
19:24.26apturahedge, I also predict that in the next 30-50 years that wars may be fought over water resources as a result of global warming.
19:24.31apturaif not sooner.
19:25.07polerinok i'm an idiot apparently.  http://eclexia.net/files/log.txt
19:25.26apturaI am one of the lucky few that is using renewable fuel for my vehicle.
19:25.45polerincompile log for zaptel.  debian sarge, 2.4 kernel, /usr/src/linux-2.4 exists
19:25.54*** part/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg)
19:25.55_VoicemeUpDotComhttp://www.youtube.com/watch?v=QmqpGZv0YT4
19:26.01_VoicemeUpDotComair car
19:26.04J4k3paul martin is a representative of what drives the weirdo jackass right-wing (bush/blair style)
19:26.14_VoicemeUpDotComworks perfectly.. 0 emmision 200 miles autnomy.. refills in 8 mintues.. with air
19:26.14J4k3whatever or whoever drives it.
19:26.28J4k3wtf air car?!
19:26.36_VoicemeUpDotComcompresed air
19:26.38_VoicemeUpDotComcheck http://www.youtube.com/watch?v=QmqpGZv0YT4
19:26.42apturaVoice you mean liquid CO?
19:26.45_VoicemeUpDotComnice liy done by the frenchies
19:26.54_VoicemeUpDotComno compressed air.. and new kind of motor
19:27.09apturaLiquid CO may make more sence.
19:27.17_VoicemeUpDotComnope
19:27.20Hmmhesaysliquid carbon monoxide?
19:27.26apturadioxide
19:27.26_VoicemeUpDotComits a reg tire compressor.. with more psi of course
19:27.27aptura:)
19:27.37HmmhesaysCO is carbn monoxide
19:27.54apturaCo2
19:27.56_VoicemeUpDotComman these guys hsould put open source on the net. like in that valkilmer film.. (the saint) that would fuck oil companies up.
19:28.11apturaI run biodiesel and make it at times.
19:28.17_VoicemeUpDotCombut greed makes all those new techs go away.. or a 22 cent bullet..
19:28.26Hmmhesayshard to do where I live
19:28.31_VoicemeUpDotComyou know house oil.. is diesel with a colorant..
19:28.39Hmmhesaysbiodesiel is the consistency of jello when it gets cold here
19:28.48apturahow cold
19:28.53_VoicemeUpDotComyou can use in car.. and now truck cops are checking the color in your truck tank to be sure you dont use.. since its 60% cheaper
19:28.57Hmmhesaysgets down to -30F in the winter
19:29.16_VoicemeUpDotCombtw if your car is diesel .. like the volks
19:29.19apturaHmmhesays a can of karo may help.
19:29.32Hmmhesayshaving a diesel engine to run it in might help too
19:29.45Hmmhesaysethanol is really popular up here though
19:30.01apturaThere are diesel heaters that can be plumbed into the heater line system.
19:30.07hedge77_VoicemeUpDotCom: truck cops checking color what?
19:30.13apturabut at 600 dollars pretty pricy.
19:30.31Hmmhesayswe have a large ethanol plant 200 miles away
19:30.50_VoicemeUpDotComof ythe gas in your diesel trucks
19:30.58_VoicemeUpDotComCAUSE you can use REG house tank oil
19:31.02_VoicemeUpDotComheating oil
19:31.16_VoicemeUpDotComis diesel with a colorant to stop you from using in your diesel car/truck
19:31.47_VoicemeUpDotCommy buddy does 1500 km a week and uses that since its cheaper heating oil (diesel with colorant) then diesle at the pump
19:31.49demlakwhen i want to connect to a asterisk not in my subnet (somewhere in the internet) with x-lite... do i have to forward ports on my router?
19:31.50_VoicemeUpDotComless taxes..
19:31.58_VoicemeUpDotComdemlak,  you should
19:32.01Hmmhesaysoffroad diesel also
19:33.05demlakso.. no chance when i am in network not controlled by myself? e.g. with my laptop in a internetcafe
19:34.47_VoicemeUpDotComah
19:34.53_VoicemeUpDotComwell.. no
19:34.54irulecan I send someone a pre-configured softphone to work with my server?
19:34.58_VoicemeUpDotComproblem is you can dial out..
19:35.20_VoicemeUpDotCombut toreceive a call.. that will hit router at port 5060...or anyport
19:35.27_VoicemeUpDotComand since no connection the router will drop it
19:35.33demlakhmpf
19:35.42_VoicemeUpDotComone way is to force xten ( to use 5060.. ) i guess.. and make a call , /register/
19:35.43irulehow about openvpn?
19:35.53_VoicemeUpDotComso router keeps conenciton open.. use subscription etc
19:36.05JunK-YHmmhesays: whats up dude?
19:36.09_VoicemeUpDotComany statefull /keep open connection should work
19:36.15b11dopenvpn owns
19:36.34polerindemlak: you really want to use voip on a netcafe connection?
19:36.35polerin:P
19:36.36hedge77openvpn casues headaches with vista
19:36.44[TK]D-Fenderirule, Yes, you can.
19:36.56demlaki tried openvpn.. but i can only ping the asterisk server.. but x-lite can´t connect.. and i don´t know why...
19:37.00[TK]D-Fenderirule, And no, you don't need VPN
19:37.16b11dvista causes headaches.
19:37.20_VoicemeUpDotCommaybe rport ?
19:37.38Hmmhesaysi kind of like the new idefisk
19:38.09demlakwell... laptop <-openvpn over the internet-> router <-LAN-> asterisk
19:38.15demlakthats what i tested...
19:38.21demlaklaptop can ping asterisk
19:38.28Hmmhesaysopenvpn rocks the casbah
19:38.35b11daye
19:38.40demlakbut x-lite on laptop cant conenct to asterisk
19:38.42JunK-YHmmhesays: any plan to go at cluecon this year?
19:38.58HmmhesaysJunK-Y: not sure yet, I got the reminder phone call a few days ago
19:39.00Hmmhesaysare you going?
19:39.25JunK-Yi will try. will have to check if isnt falling in my mid-term exams first.
19:39.39Hmmhesaysit is in june this time
19:39.43JunK-Yya.
19:40.27HmmhesaysI might be able to make it this year
19:40.40JunK-Ydont forget ur spikey hairs! :P
19:40.46Hmmhesaysahah of course not
19:40.56JunK-Y:)
19:40.59b11dJunK-Y.. how long have you had this crush on Hmmhesays?
19:41.18Hmmhesayshas it already been 2 years
19:41.27JunK-Ywhatcha mean?
19:42.04Hmmhesayshe was making a joke
19:42.26JunK-Yb11d: he wants me for 2 years ya :P
19:42.27polerinlikely the only con I'm going to get to go to this year is PN
19:42.32b11d:)
19:42.32_VoicemeUpDotComlol
19:43.39nybblei really _hate_ bell mobility
19:44.05Hmmhesaysdid they shoot your dog?
19:44.35*** join/#asterisk kje (n=kje@lime-gw16.one.at)
19:44.52JunK-Ynybble: what are ya talking about? everyones LOVES bell! :)
19:46.04polerinJunK-Y: **twich**
19:46.22_VoicemeUpDotComim glad jean lafleur in jail
19:46.27_VoicemeUpDotComhe frauded our gov for 30 million
19:46.30_VoicemeUpDotCom300
19:46.53_VoicemeUpDotComthey cought him coming back from belize.. lol he went to insure is retirement i guess
19:47.18nybblehmmhesays: they shot my blackberry
19:47.22JunK-Y_VoicemeUpDotCom: he will stay in for what? 1 year? 2 years?
19:47.28_VoicemeUpDotComno idea yet
19:47.36nybblelol JunK-Y , polerin
19:47.40_VoicemeUpDotComi would  do the chineese way.. 22 cents charged to family..
19:47.40JunK-Yhe doesnt care.
19:48.20nybblewell, have to head out.. family good frdiay things.... talk to you fine group of people later
19:49.33JunK-Yanyways, im back to work, Hmmhesays ttyl.
19:56.56*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:57.19HmmhesaysLater JunK-Y
19:57.36*** join/#asterisk nikko (n=nikko@69.85.203.178)
20:00.47Hmmhesayswell normally I wouldn't use iax2, but this guy farked up his network so bad I had no choice
20:06.36*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
20:13.41*** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl)
20:15.05*** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net)
20:15.19*** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net)
20:15.57*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
20:16.04*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
20:16.53*** join/#asterisk renzema (n=renzema@58.252.186.81.lund.res.dyn.perspektivbredband.net)
20:16.54*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
20:17.10HmmhesaysAirwolf, that show rocked
20:18.20b11dyes
20:18.23b11dit totally did
20:18.30b11dStolen from the US Gov't to FIGHT CRIME
20:18.31b11d:)
20:18.34*** join/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg)
20:18.52b11dyou can watch the 1 hour Airwolf movie on google video too
20:19.00b11dand by movie, i mean, 1 hour pilot episode
20:19.31yidiyuehanhi, guys, any one pls explain to me how asterisk invoke sendmail?
20:19.31irulecan I sent CLI> output to a log file?
20:19.44b11dlogging.conf
20:19.49b11dand syslog
20:19.51iruleok thanks
20:19.56b11dnp
20:20.00*** join/#asterisk [shodan] (n=shodan@ip214.96-113-216.pppoe1.joliette.intermonde.net)
20:20.04Hmmhesaysi can download the whole series courtesy of giganews too
20:20.20Hmmhesaysso b11d what do you think of this awesome weather we're having
20:20.24yidiyuehanas i have the sendmail with voicemail problem and the linux gusy there doubt that asterisk use the /etc/mail/sendmail.mc to do sendmail ?
20:20.55b11doh its so great
20:20.58b11di love this wind today
20:21.15b11dmy back yard was just beginning to dry out and grass was coming up..  now :|
20:21.25Hmmhesayslast year at this time I was riding my motorcycle in a t-shirt
20:21.51fluffyfluffyyidiyuehan: you trying to send mail to a local or remote account?
20:22.39b11dyeah it was nice..     oh well, it'll clear up.
20:23.04renzemaHi all.  I hae a bit of a problem.  I am using a vood 322 adaptor to reach my asterisk box.  The problem is (from my very limited understanding of asterisk) that the box is periodically sending notifcations to the vood that there are 0 messages waiting for it.  The vood is then interpreting this as a reason to ring the phone a few times.  This happens every few minutes and is most annoying.  The log extract is at http:
20:23.22yidiyuehanfluffyfluffy, i am trying to use asterisk voicemail function and send to my hotmail account
20:24.03fluffyfluffyyidiyuehan: from a console if you do, mail youraccount@hotmail.com does it get there?
20:24.11renzemaIs there any way to kill these notifications?
20:24.58[TK]D-Fenderrenzema, Sure, remove the "mailbox= line from your sip.conf entry
20:26.41renzemaHi Fender. If I do that, will that affect the voicemail for the extension? (sorry if this is a basic question)
20:26.54fluffyfluffyyidiyuehan: do you mean you are trying to send mail via the command voicemail in your extensions.conf or you are trying to send mail via your voicemail.conf file?
20:27.34yidiyuehanfluffyfluffy, in fact i use freepbx to create the extensions
20:27.43yidiyuehanand upon there i put the hotmail address for voicemail
20:28.55fluffyfluffysorry. I don't use freepbx so I don't know how it sets up extensions.conf. But first you should verify that you can send mail to your hotmail account by using the mail command in a terminal.
20:30.13yidiyuehanyes, i can the linux guys help me that, but with asterisk voicemail, i couldn't
20:30.48yidiyuehanbut if it's via extensions.conf or voicemail.conf, they both use sendmail under /etc/mail am i right?
20:31.53*** join/#asterisk grantm (n=grantm@kolob.wingateservices.com)
20:31.55fluffyfluffyyidiyuehan: hmmm... I'm not sure but I don't think so. asterisk should have no business messing around with anything in /etc/mail. It should be running /usr/sbin/sendmail
20:33.01*** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com)
20:34.06yidiyuehana? a u sure this point?
20:34.15BigCanOfTunaDoes asterisk maintain the last PSTN number it recieved, assuming you have caller id?
20:34.37Qwell[]BigCanOfTuna: no, but you can with like 1 line of dialplan
20:34.38[TK]D-FenderBigCanOfTuna, If you code it to, sure
20:34.48fluffyfluffyyidiyuehan: I'm not sure. But there should be no executables in /etc/mail. So it shouldn't be running anything in /etc/mail. But the mail config files should be there.
20:34.52[TK]D-FenderBigCanOfTuna, * doesn't do much of anything that you don't tell it to.
20:35.25BigCanOfTuna[TK]D-Fender: Gotcha...just looking at some example dialplans and it didn't appear to mention that. What command do I need to look at?
20:36.08fluffyfluffyyidiyuehan: I'm pretty shure the default command asterisk will run is /usr/sbin/sendmail -t.
20:36.11[TK]D-FenderBigCanOfTuna, You would probably use AstDb for this "show function DB" and the CALLERID function as well.
20:36.27BigCanOfTuna[TK]D-Fender: Thanks dude!
20:38.15*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
20:40.26*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
20:41.10yidiyuehanfluffyfluffy, the voicemail is running under voicemail.conf
20:41.33yidiyuehan103 => 103,IT,kaimei@hotmail.com,,callback=from-internal|attach=yes|saycid=yes|envelope=yes|delete=no
20:41.35renzemaHi [TK]D-Fender... that did the trick.  Now if I can just get sipcat to stop overwriting the sip.conf file, I'll be all set.  Many thanks.
20:42.09*** join/#asterisk snuffop (n=marty@c-67-163-68-68.hsd1.il.comcast.net)
20:42.12[TK]D-Fenderrenzema, Stop using GUI's and other junk to run your system for you.
20:42.21Hmmhesaysusually a good plan
20:42.55renzemayea, it is, but I am trying to start a company, not learn how to administer a pbx system.
20:43.40Hmmhesaystrixbox will get you going, but in the long term it is better to write your own dialplan
20:44.19Hmmhesaysfreepbx has gotten a lot better since its beginning
20:44.20snuffoprenzema:  the logic then show that you pay someone to do it for you or learn to do it yourself  (everything costs) ether $ or your Time
20:44.28fluffyfluffyyidiyuehan: grep kaimei /var/log/maillog and see if there are any errors.
20:44.31nikkothe asterisk-gui has some quirks, but builds a pretty clean dialplan imo. it a little goofy if you are trying to do an IVR but it works
20:44.32renzemaI'll probably go with a pure system in the future, but for now, sipcat works fine.
20:45.44renzemaI need to do an IVR, and the sipcat gui makes it really easy for a beginner.  but now that I see how it is modifying the files, it provides a stepping stone.
20:45.55Hmmhesayswhat kind of ivr?
20:46.22Hmmhesayspress one for sales, press 2 for tech support, press 3 for some hot action
20:46.28yidiyuehanfluffyfluffy, yes it does has errors!!!!!
20:47.43renzemavery basic for now. just a few menus.  Long term I will need to do something quite complex - outbound to one party who will press 1 to confirm that they are ready to answer the question, then outbound to another party who has a question (triggered by a web interface).  At that point I'll either have to really dig into the system or hire someone.
20:48.04yidiyuehanfluffyfluffy: Apr  7 03:59:46 pt3 sendmail[7401]: l36Jxkoc007401: to="customer" <laipeng@totalhearingcare.com.sg>, ctladdr=asterisk (501/501), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30995, relay=[127.0.0.1] [127.0.0.1], dsn=5.1.1, stat=User unknown
20:48.18fluffyfluffyyidiyuehan: ahhh :) now we are getting somewhere. Do you understand the errors and if not can you paste the log somewhere so we can look at it together (please don't paste into the channel).
20:48.27*** join/#asterisk wunderkin- (n=kev@dslstat-ppp-95.fastq.com)
20:48.38*** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)
20:48.46fluffyfluffyyidiyuehan: hmmm...
20:49.29*** join/#asterisk harleya (n=harleya@001-793-323.area1.spcsdns.net)
20:49.46thinwireshi guys, I was wondering if I could get a quick hand on this one last problem, for some reason when i call into my phone I get these error messages http://pastebin.ca/427540 ... I can call out no problems
20:50.02fluffyfluffyyidiyuehan: thats not the email address you gave earlier though. can you use "mail" to send to that address from a terminal?
20:50.34renzemaanyway, thanks again.
20:50.36BSD_Techcool asterisk and mythtv on the same box
20:50.46BSD_Techand it rocks on bsd
20:52.01yidiyuehanfluffyfluffy, i will try now and see how
20:58.56polerinmeh,  how do you stop asterisk-safe if it's bouncing?
20:59.06_VoicemeUpDotComkillall -9 procename
20:59.15_VoicemeUpDotComshould be a shell script
20:59.22_VoicemeUpDotCom/bin/sh/asterisk-fsafe crap
20:59.31_VoicemeUpDotComkill the shell kill the childs
20:59.40polerinerr safe_asterisk actually
20:59.49polerinkay
21:00.47_VoicemeUpDotComor edit asterisk safe..make exit 0; as first line..kill child
21:00.48*** join/#asterisk icel (n=dan@65.200.26.90)
21:00.52polerinarg.  okay didn't realize killall didn't expand asterisk to safe_asterisk.  probably good that it doesn't but It's that kind of day for me
21:01.05polerin`killall safe_asterisk` did it
21:01.10_VoicemeUpDotComyep
21:01.40icelGot a question- I just configured a Digium TE405P with a T1, what do I need to do to set up my DIDs?
21:01.55_VoicemeUpDotComnothing much
21:02.03[TK]D-Fenderthinwires, [Apr  6 16:46:46] NOTICE[6816]: chan_local.c:562 local_alloc: No such extension/context 24227542224133@numberplan-custom-1 creating local channel
21:02.10_VoicemeUpDotCompoint capata context to inbound.. make something in [inbound] in your extensiosn.. have fun
21:02.13[TK]D-Fenderthinwires, Pretty obvious error
21:02.23_VoicemeUpDotComnumberplan-custom-
21:02.34_VoicemeUpDotComdoesn exist .. or doesnt match for  24227542224133
21:02.54icelso just make an inbound context in extensions.conf and point it to an individual phone?
21:03.00_VoicemeUpDotComhence the use of NoOp("Getting call on XXX from XXXX on chan XXXX
21:03.06_VoicemeUpDotComyes
21:03.31icelhow to point context to inbound?
21:03.52_VoicemeUpDotComin zapata context
21:03.58_VoicemeUpDotComzapata.conf i gues
21:04.03icelThanks
21:06.53_VoicemeUpDotCom[channels]
21:06.53_VoicemeUpDotComlanguage=en
21:06.53_VoicemeUpDotCom;callerid=Private
21:06.53_VoicemeUpDotComcontext=inbound
21:07.54*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
21:09.02thinwiresD-Fender: it might be an obvious error, but I'm obviously confused on why it is trying to call 24227542224133
21:10.19*** join/#asterisk S2AnGeL (n=S2AnGeL@CPE0014bf103d31-CM000039529869.cpe.net.cable.rogers.com)
21:10.37S2AnGeLhow does one ip authenticate  with no md5
21:10.46S2AnGeLhow do I do the registration string?
21:11.11S2AnGeLor do I?
21:11.22S2AnGeLgrnvoip
21:11.29yidiyuehanfluffyfluffy,i can send out mails by using mail with some email address, but not all
21:11.30[TK]D-Fenderthinwires, -- Now forwarding IAX2/trunk_1-2 to 'Local/24227542224133@numberplan-custom-1' (thanks to SIP/600-08203a58)
21:11.43[TK]D-Fenderthinwires, Yuo mean this does point the finger clearly?
21:12.09S2AnGeLsays they support asterisk.. but the most they give me is "  exten => _123NXXNXXXXXX,1,Dial,SIP/64.243.115.76/${EXTEN:2}   "
21:12.45S2AnGeLthe 123 is the prefix (example)
21:13.09S2AnGeLand from that point on they say thats all they can do to help me through email..
21:13.13thinwiresD-Fender: This is why I'm here, it does not point the finger clearly because while i understand enough to know that something is wrong with the where it is forwarding this to, I'm unsure of what is causing the problem.
21:13.16[TK]D-FenderS2AnGeL, ${EXTEN:2} onlty strips off the first TWO digits....
21:13.32S2AnGeLI tell you once I figure this out I will write up a proper  thing for voip-info
21:13.37[TK]D-Fenderthinwires, It just told you right to your face that your PHONE SIP/600 forwarded the call there.
21:13.49[TK]D-Fenderthinwires, Go look at your phone!
21:14.16thinwiresyeah, thanks a bunch asshat.
21:14.24S2AnGeL[TK]D-Fender:  that is not helping me at all
21:14.44S2AnGeL[TK]D-Fender: but thanks for at least talking to me
21:15.43S2AnGeL[TK]D-Fender: have you ever setup a ip authentication only to asterisk??
21:15.44[TK]D-FenderS2AnGeL, Ok, your dial statement is otherwise fine looking.  That is an unauthenticated dial however.  if you need to auth I suggest creating a peer entry in sip.conf.
21:15.50S2AnGeL[TK]D-Fender: once again your not helping me
21:16.03S2AnGeL[TK]D-Fender: I have created a peer
21:16.18S2AnGeLwith out any user name and password (secret0  like they mention
21:16.26S2AnGeLits just ip authentication..
21:16.58[TK]D-FenderS2AnGeL, *sigh*.  Dial(SIP/user:pass@64.243.115.76/${EXTEN:3}).  Something like that.  but you should NOT be adding auth infor to extensions.conf
21:17.20[TK]D-FenderS2AnGeL, You should amke a PEER entry in sip.conf to includ the user, pass, host, etc.
21:17.37S2AnGeLthey tell me my email address is my user name.. but I think thats just for the web interface
21:18.10S2AnGeLI will try it but .. I dunno..
21:18.10[TK]D-FenderS2AnGeL, perhaps you could link us to the guide they provide so we can see if you're missing something.
21:18.34*** part/#asterisk yidiyuehan (n=yidiyueh@cm89.sigma116.maxonline.com.sg)
21:19.40*** join/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net)
21:20.19hedge77is there a way to make pickup() work in 1.4 if you don't necessarily know what the dialing context is?
21:20.31S2AnGeL-- Got SIP response 423 "Interval Too Brief" back from 64.243.115.76
21:20.33S2AnGeLsec
21:20.33stuntdoubleIs there a way to make a general voicemail?  I have extensions for SIP in 101 and 102 but I want a general voicemail for them (say 100).
21:20.44stuntdoubleIs adding SIP 100 the only way?
21:20.53hedge77you can make whatever voicemail boxes you want
21:21.07stuntdoubleIf I edit voicemails.conf it will get overwritten
21:21.18hedge77oh taht's what you get for using a gui then :P
21:21.27[TK]D-FenderS2AnGeL, I thnk I've seen that from providers that force you to register at high frequencies to auth yout IP.
21:21.27stuntdoubleYea, it's weird
21:21.33stuntdoubleI'm use to only using the conf files
21:22.00[TK]D-FenderS2AnGeL, for which they don't need auth on the CALL, they just ID your IP from the contant REGISTER's they expect you to throw out.
21:22.11*** join/#asterisk zotz (n=zotz@24.244.163.157)
21:22.36[TK]D-Fenderstuntdouble, Go check in your GUI's support channel
21:23.07S2AnGeLhttp://grnvoip.com/faq.htm  is all they have.. and when I ask support what does a registration string look like for asterisk.. he regurgerates that like I posted above.. the only info in the faq..
21:23.38S2AnGeL[TK]D-Fender:I figure I do not need a registration string? but still confused?
21:24.21_VoicemeUpDotComfxo = pstn right ?
21:24.23[TK]D-FenderS2AnGeL, You need to link us to some info on your provider.
21:24.25_VoicemeUpDotComsignaling from pstn
21:24.47[TK]D-Fender_VoicemeUpDotCom, yes
21:25.02S2AnGeLhttp://grnvoip.com/faq.htm   thats pretty much it.. you can scoure it all you want thats all there is..
21:26.01S2AnGeLeven in the account login it just has balance no documentation.. ...  just has cdr list. and numous links not working
21:26.10S2AnGeLgood prices for termination though
21:26.11*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
21:26.30[TK]D-FenderS2AnGeL, Sounds like they expect you to be on a fixed IP and don't need to register or auth at all...
21:27.15[TK]D-FenderPlease note that GRNVoIP.com will authenticate inbound calls based on IP address or SIP domain/realm only. GRNVoIP.com
21:27.27stuntdoubleopps, i realize i'm in the wrong channel.
21:27.35S2AnGeLyes well I am on a fixed ip.. how the heck do I make a call.. I guess I do not need a registration string then ?
21:27.54S2AnGeLgunna try with out sec
21:28.11_VoicemeUpDotCom<PROTECTED>
21:28.13_VoicemeUpDotComhmm
21:33.43*** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C)
21:34.00*** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3)
21:34.03DocHollidayanyone know of a faxback number so i can test my newly installed VoIP fax machine?
21:34.08*** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com)
21:34.55thinwiresyou could sign up at efax for a free acount
21:35.59DocHollidaythinwires, smart guy.. good call
21:36.03BigCanOfTunaIs there anyway I can look at the contents of DB from the console...I don't see anything about DB
21:36.20CrossRoadHas anyone has experience with the provider NuFone
21:36.37MRH2hi i'm getting an error compiling a recent revision of 1.2 - does this normally require a zaptel from about the same time frame (I'm stuck with an older one)?
21:36.53[TK]D-FenderBigCanOfTuna, "database show ...."
21:37.07[TK]D-FenderBigCanOfTuna, "help database"
21:37.21BigCanOfTuna[TK]D-Fender: thanks.
21:37.40[TK]D-FenderMRH2, Yes, you need to stick with a matching Zaptel.
21:38.53MRH2I am keeping them both on 1.12.x
21:39.02MRH2but assume it needs to be even closer
21:41.03[TK]D-FenderMRH2, what is 1.12.x?
21:41.20*** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net)
21:41.44MRH2it is my nig gingers trying to type 1.2.x
21:42.41S2AnGeLanyone have a good termination provider for canada calling $0.009
21:42.51*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
21:43.18S2AnGeLthat uses sip or iax2? that allows like 5 channels at a time
21:44.25S2AnGeLfaxback number  might just send you on a list for faxs for the rest of your life.. being resold over and over again..
21:44.41S2AnGeLmuch easier to fax or get a fax from someone you know..
21:44.51_VoicemeUpDotComus
21:45.08S2AnGeLDocHolliday: don't use a faxback number
21:45.24DocHollidaywhy?
21:45.36S2AnGeLDocHolliday:faxback number  might just send you on a list for faxs for the rest of your life.. being resold over and over again..
21:45.45S2AnGeLfax spam
21:45.57S2AnGeLits like impossible to get off
21:46.05DocHollidayS2AnGeL, i need a way of testing my fax machine...
21:46.52S2AnGeLDocHolliday:so you don;'t know anyone with a fax machine?
21:46.52S2AnGeLask for a quote to be faxed to you from some place for something
21:47.08S2AnGeLhey you can use one I am not stopping you just telling you they are bad
21:47.19*** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com)
21:47.27DocHollidayS2AnGeL, thanks :P
21:47.41DocHollidayanyone want to send me a fax? :P
21:49.21sevardI thought fax spam was illegal in most states
21:49.37thinwiresno, I get fax spam all the time, it's so annoying
21:49.42DocHollidayall i know is i need to test my fax machine :(
21:49.44thinwireslolz
21:49.54DocHollidayqwell, thats fine? do you mnd?
21:49.58DocHolliday*mind
21:49.58polerinlol
21:50.08polerinno no no. totally lemonparty
21:50.23Qwell[]no, that means I would have to view and print it
21:50.23Qwell[]I'm not willing to go that far
21:50.29_VoicemeUpDotComzttoll only shows ztummdy
21:50.33DocHollidaycan you fax me something?
21:50.46Qwell[]DocHolliday: That would require me getting up
21:50.47_VoicemeUpDotCombut l;smode shows wctdm                  40768  0
21:50.51_VoicemeUpDotComi need a modprobe it ?
21:50.56thinwiresdoc, pm me
21:51.01*** join/#asterisk pkempgen (n=pkempgen@ACAECA3D.ipt.aol.com)
21:51.44_VoicemeUpDotComi get
21:51.44_VoicemeUpDotComZT_CHANCONFIG failed on channel 1: No such device or address (6)
21:51.46thinwiresyes
21:51.56DocHollidayqwell never knew you were that damn lazy :P
21:51.59DocHollidaythinwires,  PM
21:52.18thinwireslol whats the command for that? I downloaded the most useless client ever
21:55.29polerinthinwires: what client?
21:55.29polerinlol
21:55.29polerin/msg normally
21:55.29polerin/msg <user> <msg>
21:55.30wunderkin<comic book guy voice>
21:55.30polerin<tired and wanting to go home voice>
21:55.30thinwiresyeah I found it, google ftw... It just slipped my mind, but I got babbel, I have a mac and my other client expired
21:55.31wunderkino.. mac.. yeah that sucks..
21:55.31wunderkinlol j/k
21:55.31thinwires>-|
21:55.52_VoicemeUpDotCom<PROTECTED>
21:55.54_VoicemeUpDotComany idea ?
21:56.05_VoicemeUpDotComtdm
21:57.01_VoicemeUpDotComhmm ahah
21:57.05_VoicemeUpDotComclient told me tdm400
21:57.06_VoicemeUpDotCom3:00.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card
21:57.09_VoicemeUpDotComsame ?
21:57.19*** join/#asterisk certron (n=atannir@pool69-59-255-17.kewr1.s.vonagenetworks.net)
21:59.20certrongreetings. I am trying to get asterisk to execute an outgoing ssh command to another host and put it in the background. it executes the command but then immediately exits.
22:00.03_VoicemeUpDotComseem it needs wanpipe
22:00.04_VoicemeUpDotComlol
22:00.05[TK]D-Fender_VoicemeUpDotCom, No, not at all a TDM400... equivalent intended puropose, but that is a Sangoma card for which you'll have to set up the wanpipe drivers,e tc
22:00.52certronhmm. maybe i could get away with a setuid bash script...
22:01.59DocHollidayanyone able to help me setup asterisk to send incoming fax requests to a SIP extension?
22:02.30DocHollidayAsterisk currently is setup and working for voice, just not sure how to setup for fax
22:02.42[TK]D-FenderDocHolliday, "exten => fax,1,Goto(somecontext,s,1)
22:03.03[TK]D-FenderDocHolliday, This is if you are running an IVR upon answering the call
22:03.12DocHollidayyup
22:04.11DocHolliday[TK]D-Fender, what do i put in the fax context?
22:04.25[TK]D-FenderDocHolliday, what "context"?
22:04.35DocHollidayfax
22:04.41[TK]D-FenderDocHolliday, This technically has nothing to do with "contexts"
22:04.53[TK]D-FenderDocHolliday, its an EXTEN that is applicable in an IVR
22:05.13[TK]D-FenderDocHolliday, jsut like i,t,s,h, and so on
22:05.27wunderkins,h,i,t
22:05.29DocHollidayright but i want asterisk to forward requests to a SIP extension, that extension is an ATA that connects to a fax machine
22:05.37certroninteresting. where is the fax detection actually done?
22:05.51DocHollidaythe ATA is already setup and communicating with Asterisk on extension 299
22:07.17DocHollidaywhat extra information do i need to pass asterisk for it to forward requests to the EXTEN?
22:08.00[TK]D-FenderDocHolliday, exten => fax,1,DialSIP/myfaxisusingthis)
22:08.16[TK]D-FenderDocHolliday, exten => fax,1,Dial(SIP/myfaxisusingthis)
22:08.43russellbwunderkin: it's actually ... o,s,h,i,t,a,fax
22:08.52[TK]D-FenderDocHolliday, you aren't "forwarding", there is no "magic", there is only DIAL <-
22:08.58[TK]D-Fenderrussellb, z0mg!
22:09.06russellb:)
22:09.42russellbthose are the special extensions ...
22:09.47russellbwell, I guess there is a 'T', as well
22:09.52Qwell[]wunderkin: I always yell at people when they run an app, and use args that could spell out a word...
22:10.00Qwell[]like `netstat -plant`
22:10.09Qwell[]if they don't use them in that exact order, it makes me angry :p
22:10.22DocHollidayheh
22:10.29DocHolliday[TK]D-Fender, would you be willing to send me a test fax? :P
22:10.43Qwell[]russellb: so...
22:10.49Qwell[]T,o,s,h,i,t,a,fax?
22:10.57russellbha
22:10.58Qwell[]too much?
22:11.04JuggieQwell, did you listen to the entire cd yet?
22:11.09Qwell[]Juggie: twice
22:11.29Juggiei'm on like #4 now :P
22:11.31DocHollidaynow i need a way to test my fax :(
22:12.13*** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com)
22:17.10*** part/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net)
22:17.14*** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177817390.dsl.bell.ca)
22:18.11fab5freddyAnybody here recommend a DID provider for Montreal?
22:18.49_VoicemeUpDotComvoicemeup
22:19.04fab5freddyCan I sign up a line right now?
22:19.26Qwell[]_VoicemeUpDotCom: Can you also voice down?
22:20.12_VoicemeUpDotComvoice down ?
22:20.13_VoicemeUpDotComlol
22:20.19_VoicemeUpDotComunfortuneatley no
22:21.13fab5freddy_VoicemeUpDotCom: their rates seem high compare to unlimited..
22:21.28_VoicemeUpDotComyou get waht you pay for
22:21.34Qwell[]cheap, reliable, good customer service
22:21.36_VoicemeUpDotComand 1.9 scaling to 0.09
22:21.36Qwell[]pick up to 2
22:22.00_VoicemeUpDotComtdm grade..
22:22.09Qwell[]sometimes you get a crapshoot, and get none, sometimes you'll get lucky and get 2
22:22.16_VoicemeUpDotCombest GUI in the market.. total control on your options.. 911 enabled
22:22.24Qwell[]best GUI in the market?
22:22.28_VoicemeUpDotComyep
22:22.32_VoicemeUpDotComcant beat us
22:22.36Qwell[]riiiight
22:22.38_VoicemeUpDotComsignup and see ;)
22:23.03_VoicemeUpDotCommaybe not the most options.. but does the job.. and hell of alot better then ANYthing iv seen..
22:23.06DocHolliday[TK]D-Fender, for some reason asterisk isnt detecting the calls are faxes
22:23.09Qwell[]isn't making a statement like that...illegal?
22:23.14_VoicemeUpDotComits noob proof like vonage.. but offer flexibility..
22:23.19DocHollidayanyone know why asterisk wouldn't detect faxes?
22:23.22_VoicemeUpDotComand our api's going out this month
22:23.32[TK]D-FenderDocHolliday, show us what you're doing (PASTEBIN)
22:23.55fab5freddy_VoicemeUpDotCom: do you accept paypal? and can i pay as i go?
22:23.57garreelDocHolliday: I had same problem
22:24.15_VoicemeUpDotComyes to both
22:25.19*** join/#asterisk jarg (n=jarg@189.157.103.143)
22:26.11fab5freddy_VoicemeUpDotCom: how much to get started with a phone line that can connect to *?
22:26.17DocHolliday[TK]D-Fender, http://www.pastebin.ca/427657
22:26.23_VoicemeUpDotCom10$
22:26.37_VoicemeUpDotCom15 if need a number.. i think
22:26.42*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
22:26.46[TK]D-FenderDocHolliday, that is NOT an IVR.
22:27.04DocHollidaywhat do you mean?
22:27.21fab5freddy_VoicemeUpDotCom: what are the monthly fees? how will the $15 be used?
22:27.40[TK]D-FenderDocHolliday, that entire syntax is broken, and "s" is not a PRIORITY, it is an EXTENSION.
22:28.22DocHolliday[TK]D-Fender, where do i start fixing things? :)
22:28.24garreelI had a fax detect problem on outbound calls... not incoming
22:28.29_VoicemeUpDotComas a deposit
22:28.38_VoicemeUpDotComno monthly unless you get a did in mtl or elsewhere
22:28.43_VoicemeUpDotComelse monthly is 4.95
22:28.54Qwell[]a deposit?  wtf
22:29.07DocHollidayfor the record the crappy syntax seems to work fine for voice, just not fax :P
22:29.57_VoicemeUpDotComyes.. its prepaid pay as you go
22:29.57InnatechQwell: its essentially a declining balance account, or so it sounds. So, you need a balance to charge against. VoicePulse connect works the same way.
22:29.58_VoicemeUpDotComwat you expect ? we open lines to afgna for free ?
22:30.00Qwell[]prepaid, pay as you go, with a monthly fee?
22:30.04fab5freddy_VoicemeUpDotCom: i believe i will need a did in montreal to connect to *, correct me if i am wrong
22:30.05Qwell[]bbl
22:30.12DocHolliday[TK]D-Fender, can i please have some help?
22:30.20_VoicemeUpDotComprepay +mothly is for dids
22:30.31_VoicemeUpDotComlike any service out there man.. where have you been ?
22:30.45_VoicemeUpDotComif you need a did yes, the service will be4.95 for the did..
22:31.00[TK]D-FenderDocHolliday, you should not be running an IVR off anything except "s".  Go fix the basics and then we'll see.
22:31.00_VoicemeUpDotComrest of deposit goes on your usage, you willg et low warning email etc
22:31.11fab5freddy_VoicemeUpDotCom: if i sign up now when will the line be ready to connect to *?
22:31.13*** part/#asterisk certron (n=atannir@pool69-59-255-17.kewr1.s.vonagenetworks.net)
22:32.09_VoicemeUpDotComaround 4 mintues
22:32.16[TK]D-FenderDocHolliday, #
22:32.16[TK]D-Fenderexten => 8500,n,Goto(s,6)   <- there is also no exten "s" in this context, let alone a priority 6 for it
22:32.17_VoicemeUpDotComtime to cinfirm email,account and set up a peer in there
22:32.26DocHolliday[TK]D-Fender, can i change my existing n's into s's?
22:32.52DocHolliday[TK]D-Fender, heh i dont know how that one happened :P
22:32.53[TK]D-FenderDocHolliday, listen closely again. "s" is an EXTEN, not a PRIORITY.
22:33.04*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
22:34.55garreel[TK]D-Fender: if I have an outgoing extension to dial some number on a zap channel... and if dialing such extension (say from my sip phone) I get a fax answering... should dialplan jump to the fax extension?
22:34.57fab5freddy_VoicemeUpDotCom: so how do i go ahead with the signup given that i want to pay via paypal?
22:35.00DocHolliday[TK]D-Fender, is there anything wrong with the extensions or macros themselves?
22:35.21_VoicemeUpDotComfolow the signup its gonna send you to the right place after you confuirm email
22:35.51[TK]D-FenderDocHolliday, Yes there is.
22:36.28[TK]D-FenderDocHolliday, go read the "ivr tips" link on the IKI
22:36.31DocHolliday[TK]D-Fender, fixed the voicemail issue
22:36.31[TK]D-FenderWIKI
22:36.42[TK]D-FenderDocHolliday, Which one?
22:36.59*** join/#asterisk AlexCeli (n=alex@190.42.145.41)
22:37.00DocHollidaythe Goto(s,6)
22:37.21DocHollidayi just got rid of it, didnt even need it
22:38.06AlexCeliSomeone can help me with an AGI problem?
22:38.46InnatechHow much of a PITA would I be getting myself into if I were to re-architect my PBX such that I didided it over 3  asterisk boxes, one on a VPS, one at work, one at home, with the second two just connecting extenstions to the first? Any comments on using DISA for that, or generally re: * on a VPS ?
22:39.42AlexCeliEXEC COMMAND DON'T WORK ON * 1.4.2?
22:40.03*** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com)
22:41.42DocHolliday[TK]D-Fender, can you give me an example of where I can i use more "s's" in my dialplan?
22:44.38[TK]D-FenderDocHolliday, http://www.pastebin.ca/427673
22:45.38*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
22:45.42DocHolliday[TK]D-Fender, wow its complex
22:46.35AlexCelihi
22:46.42DocHolliday[TK]D-Fender, that doesnt seem to incorporate fax?
22:46.43garreelcan anyone explain what is silence_threshold parameter on AMD application?
22:47.16[TK]D-FenderDocHolliday, ..... just add exten => fax,1,Dial(SIP/myfaxonadumbatathatsboundtofail)
22:47.33garreelperhaps is it the background noise level?
22:47.49riddleboxhas the BLF groups changed in 1.4?
22:47.52AlexCeli[TK]D-Fender: Hi, i have a problem with AGI and PHP. In * 1.0.9
22:47.57AlexCeliecho "EXEC SetCallerID \"$row[nombre] <$row[telefono]>\"\n";
22:48.03*** join/#asterisk Fieldy (i=lWrQpoO9@gentoo/contributor/Fieldy)
22:48.03AlexCeliand works fine
22:48.18AlexCelibut with * 1.42 the command not work...
22:48.59[TK]D-FenderAlexCeli, SetCallerID was deprecated in 1.2 and removed ENTIRELY in 1.4  Go read the changelogs and realize that things are the same as 4 years ago
22:49.03AlexCeliI use the script for use an external database..
22:49.12[TK]D-Fenderaren't*
22:49.27AlexCeli[TK]D-Fender ok, let me see...
22:50.36*** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net)
22:50.58Innatech./bahamavention [TK]*
22:51.29AlexCeli[TK]D-Fender ok, but
22:51.30AlexCeli<PROTECTED>
22:51.37AlexCelidon't work...
22:51.38[TK]D-Fendero>O
22:52.53AlexCelii receive this error
22:52.54AlexCeli<PROTECTED>
22:52.54AlexCeli<PROTECTED>
22:52.54AlexCeli<PROTECTED>
22:52.54AlexCeli[Apr  6 17:30:21] WARNING[19863]: res_agi.c:1118 handle_exec: Could not find application (Set(CALLERID(all)="Deborah)
22:52.54AlexCeli<PROTECTED>
22:52.55DocHolliday[TK]D-Fender, i have to change your s's to our phone number or else it wont work :P
22:53.05DocHollidayright now its busying out
22:54.05*** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-222a502e7808aade)
22:55.29DocHolliday[TK]D-Fender?
22:57.03[TK]D-FenderDocHolliday, No, you DON'T.
22:57.19DocHollidayso how do i make it stop busying out?
22:57.48[TK]D-FenderDocHolliday, Yuo need to stop thinking 1-dimensionally like you have to cram everything into 1 bloody super-context.
22:58.23[TK]D-FenderDocHolliday, exten => 14166287102,1,Goto(mainmenu,s,1)
22:58.57[TK]D-FenderDocHolliday, You are allowed to LEAVE the context you start in... its not a prison
22:59.50[TK]D-FenderDocHolliday, Think that you can have an IVR that you can access froma  phone you set up that has NOTHING to do with any other provider or hardware...
23:01.31DocHolliday[TK]D-Fender, its still busying out
23:02.42rudholmso I want a phone to not ring, but there doesn't seem to be any ring cadence setting for that.  I've tried r0 through r9.  Anyone have any ideas?
23:02.58DocHollidayi have a feeling it needs to answer as that phone number?
23:03.59rudholmwell, the situation is my house has a lot of phones, all of them on their own line.  I don't want them all to ring when a call comes in from outside Asterisk, but I want to be able to answer from any of them.
23:04.04[TK]D-Fenderrudholm, How about not DIALING it? :)
23:04.17rudholmhaha, I knew someone would say that :)
23:04.24[TK]D-Fenderrudholm, well DUH!
23:04.26[TK]D-Fender:D
23:04.34rudholm:-p
23:04.50DocHolliday[TK]D-Fender, yeah it doesnt like this new configuration :)
23:05.21[TK]D-FenderDocHolliday, I never pasted my sample as a COMPLETE REPLACEMENT.  It was a fully functional SAMPLE.
23:05.30DocHollidayhaha
23:05.46[TK]D-FenderDocHolliday, so how about cleaning up your mess an pastebining it.
23:06.28DocHollidaywell i took your replacement and it doesnt seem to work unless i replace all the s's with the phone number else it fals
23:06.47InnatechRudholm: does the phone your using allow you to disable the ringer? Many/most do.
23:06.52[TK]D-FenderDocHolliday, What did I jsut tell you?!
23:07.09DocHollidaynow it works ;P
23:07.28[TK]D-Fenderrudholm, Forget cadences.... you are dialing a damn phone... EXPECT T TO RING.  thats its JOB!
23:07.33DocHolliday[TK]D-Fender, it was my fault
23:08.08rudholmInnatech: but then the phones wouldn't ring on any calls, including internal.
23:08.29*** join/#asterisk pkempgen (n=pkempgen@ACAECA3D.ipt.aol.com)
23:08.30Innatechoh, I didn't realize you wanted it to be selective. Heh.
23:08.37rudholmInnatech: (and I collect vintage phones, most of which don't do "silent")
23:08.48Innatechrudholm: aha. Bell ringers. Now I feel your pain.
23:09.21[TK]D-Fenderrudholm, so you're stuck with the realist solution of NOT DIALING THEM :0
23:09.32rudholmInnatech: yeah, my inbound calls come in via their own context, separate from internal calls, so it's easy to apply a different cadence for internal or external calls (which is fairly common in PBX tradition)
23:09.42[TK]D-FenderInnatech, Its a boring analog phone.. what do you expect? :)
23:10.05[TK]D-Fenderrudholm, Again, forget cadence, whay are dialing phones you don't want to ring?
23:10.12rudholm[TK]D-Fender: That's Microsoft's approach to problems for which they have no solutions:  "Don't want that"
23:10.15rudholm:-p
23:10.29DocHolliday[TK]D-Fender, for some reason when i enter a 4 digit extension it goes to the wrong phone :P
23:10.34DocHolliday(with this new config)
23:10.35rudholm[TK]D-Fender: because when calls come in from outside, I want to be able to answer on any phone in the house.
23:10.48[TK]D-Fenderrudholm, "Doctor, doctor.. it hurts when I raise my arm like this!"
23:10.49Innatechwithout a symphony of bells. Heh.
23:11.01InnatechDoctor: "Don't raise your arm like that."
23:11.19InnatechDoctor: "$75 co-pay, please."
23:11.19[TK]D-Fenderrudholm, Use PICKUP  or something then.
23:11.42rudholmthat requires user training
23:12.13DocHolliday[TK]D-Fender, does your config only handle 3 digit extensions?
23:12.16[TK]D-Fenderrudholm, We are talking about analog phones here, right?
23:12.23rudholmyep
23:12.34[TK]D-FenderDocHolliday, It handles whatever you give it.
23:12.49[TK]D-FenderDocHolliday, Where did you come up with that crazy hypothesis from?
23:13.10[TK]D-Fenderrudholm, sorry... dumb phones need smart people to sue :)
23:13.15[TK]D-Fenderuse*
23:13.23*** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com)
23:13.27rudholmI had an ISDN TA that could do what I want.
23:13.52DocHolliday[TK]D-Fender, dont ask :P
23:13.52[TK]D-Fenderrudholm, And ISDN phones aren't dumb analog :)
23:14.22sbingnerrudholm, just Dial() all your local lines at once?
23:14.24rudholmno, the TA did it.  I wasn't using ISDN sets
23:14.27[TK]D-FenderDocHolliday, you seriously need to stop trying so hard.  then STOP entirely.  and then...
23:14.31[TK]D-Fender~osmosis
23:14.33jbotrumour has it, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
23:14.57sbingneror am I missing something
23:15.00DocHolliday[TK]D-Fender, thanks :) I really appreciate when people can see the level of dedication I have towards my work.
23:15.11sbingneroooh
23:15.22sbingnerrudholm, you can define a custom cadnce I believe
23:15.24*** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net)
23:15.36rudholmsbingner: yeah, inbound calls from outside the PBX just Dial(room1&room2&room3) --etc
23:15.42rudholmsbingner: ooh, that'd be perfect
23:15.44[TK]D-Fendersbingner, has no impact on dumb analog phones....
23:16.03sbingner[TK]D-Fender, then how does distinctive ring work?
23:16.16sbingner[TK]D-Fender, can't just define one that does only the non-ringing half of the ring?
23:16.26[TK]D-Fendersbingner, tell me a phone that will IGNORE a distinctive ring pattern....
23:16.39sbingner[TK]D-Fender, you make a new distinctive ring pattern that is NO ring
23:16.48rudholmwhen a call comes in from my front door, I use r3 (three really short rings) so that people in the house know it's the door and not a regular phone call.
23:16.59DocHolliday[TK]D-Fender, okay i have finally modified it as needed, want to see?
23:17.21[TK]D-Fendersbingner, how would you specify it for some, but not all ?
23:17.37InnatechYou could try to define a pattern that has the shortest possible single ring and then a loooong pause and then another "chirp"...I've heard that before (although not on * ). I don't think you can define a completely silent ring cadence, can you?
23:17.38[TK]D-FenderDocHolliday, Does it work?
23:18.01DocHolliday[TK]D-Fender, yup!
23:18.13DocHollidaybut there is a difference between working and perfect :)
23:18.14sbingner[TK]D-Fender, that would be the tough part... but luckily this is open source so it's possible ;)
23:19.00rudholmInnatech: even a reduction would be good.  the sound of 10 Western Electric bell ringers is a bit much.
23:19.10*** join/#asterisk [shodan] (n=shodan@ip045.96-113-216.pppoe1.joliette.intermonde.net)
23:19.15Innatechhehe.
23:19.22Innatechstuff some cotton balls into the bells.
23:19.23sbingneractually, don't you dial as Zap/1r1? so it'd be "Zap/1&Zap/2r4" if 4 is no ring
23:19.28[TK]D-Fendersbingner, Sure... dangle hope in front of him and tell him to finish his Raw Cat Sigh Hence degree!
23:20.15rudholmsbingner: I use variables for my extensions
23:20.32InnatechActually, if you can make the chrip short enough, you might not even get a real "ring" given that its an electromechanical ringer. The afteroffice ring bells at work here are like that. The cadence is so short it just goes "clunk-clunk."
23:20.57rudholm[TK]D-Fender: I guess it's beyond you, I'll have to wait for Strom to get back :-P
23:20.57sbingneryea the distinctive ring pattern is set on a per-channel basis in the dial command.   Dial(Zap/2r1&Zap/3r2|15|t) would make 2 distinctive rings for a single dial command
23:21.10rudholmsbingner: yes
23:21.36sbingnerrudholm, I expect you'll have to find in the zaptel source where those are set and make a new one, then recompile zaptel
23:21.36DocHolliday[TK]D-Fender, yes it works...
23:21.50rudholmsbingner: when a call comes from the front door, it's Dial(${kitchen}r3&${bedroom}r3&${garage}r3...
23:23.17rudholmsbingner: yeah, I could hack the code (which I'm already doing for my coin deposit tone detection hack) but I was hoping it could be done without more hackery.
23:23.59[TK]D-FenderDocHolliday, Good... then no need to show me.  I only hope you retain some of what you should ahve learned from this.
23:24.01rudholmsbingner: if I modify zaptel I have to maintain a patch file (or never update zaptel :) )
23:24.39DocHolliday[TK]D-Fender, yeah but i still need to get the fax working :P
23:27.20*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)
23:27.49sbingnerrudholm, in chan_zap.c there's a structure with the default cadences, you could try adding one there
23:28.01type0who's a good voip reseller?
23:28.03type0voipstore?
23:28.14[TK]D-FenderDocHolliday, Why do you tell me it works one second and then tell me it doesn't the next?  Are you just completely retarded? :)
23:28.42DocHolliday[TK]D-Fender, screw off :) I was implying the voice now worked.. hadn't tested the fax component
23:29.12[TK]D-FenderDocHolliday, go ahead and pastebin where you have NOW gotten yourself to.
23:29.12[TK]D-FenderDocHolliday, You could have just said that :)
23:29.17[TK]D-FenderDocHolliday, but NOOOOO you had to get my hopes up!
23:29.32DocHollidayI dont know ahaha
23:29.45DocHollidaysec
23:30.22sbingnerrudholm, you can set it in zapata.conf -- see http://www.asterisk.org/doxygen/1.4/Config_zap.html
23:30.32*** join/#asterisk ManxPower (n=manxpowe@60.sub-70-223-142.myvzw.com)
23:30.46[TK]D-FenderDocHolliday, Ironic for an Asterisk user, you need some serious COMMUNICATION courses :)
23:30.49DocHollidayhttp://www.pastebin.ca/427727
23:30.54rudholmsbingner: ah, that's more like it
23:31.05DocHolliday[TK]D-Fender, funny!
23:31.13SplasPoodDoes anyone know of a call pickup app for the latest asterisk 1.2 that allows you to specify which *extension* you want picked up?
23:31.26ManxPowerBTW, does any one know the frequency range of a PSTN call?
23:31.42DocHolliday[TK]D-Fender, http://www.pastebin.ca/427727
23:31.49[TK]D-FenderDocHolliday,  You should leave your menu in its own context, and use GOTO to get there.
23:32.11[TK]D-FenderDocHolliday, It will work, but once your in your menu you can actually dial your # again!
23:32.36DocHolliday[TK]D-Fender, i actually forgot to paste a line
23:32.39ManxPowermany sites claim that 60Hz hum should not be heard on a phone since the phone's audio stuff does not go down that low.  Any hum would have to be a harmonic of 60Hz
23:33.02DocHollidaybetween 29 and 31: [local-internal]
23:33.19type0anyone know of a polycom reseller who would be in the office at this time today?
23:33.39rudholmtype0: pretty much anyone in Pacific Time would be.
23:33.57DocHolliday[TK]D-Fender, the fax failed again
23:34.00DocHollidayany ideas?
23:34.19sbingnerrudholm, but it appears to be inside an #ifdef ZAPATA_PRI block ;)
23:34.25ManxPowertype0: Maybe.  Do you have a credit card and is it shipping to a USA address?
23:34.46[TK]D-FenderDocHolliday, So you call in with a fax and it doesn't react?
23:34.48rudholmsbingner: heh
23:34.57DocHollidaydo you want to see the console output?
23:35.00DocHolliday(correct)
23:35.27rudholmsbingner: it looks like the custom cadences in zapata.conf are exactly what I was looking for.  thanks for the pointer.
23:35.31[TK]D-FenderDocHolliday, in zapata I know ther is an option for "faxdetect".  I don't recall one for SIP.
23:35.33ManxPowerDocHolliday: is the calling fax machine sending the fax tone?
23:35.39*** join/#asterisk simonkern (n=simonker@p54AA81A2.dip0.t-ipconnect.de)
23:35.53[TK]D-FenderDocHolliday, if it doesnt' do it natively, there used to be an app (mvfaxdetect I think)
23:36.06[TK]D-FenderDocHolliday, nvfaxdetect*
23:36.09garreelnvfaxdetect
23:36.11DocHolliday[TK]D-Fender, there is no reason why it shouldn't be picking up the faxes?
23:36.11ManxPower[TK]D-Fender: there is an app called NVFaxDetect.  We use that.  A link is on voip-info.org
23:36.13[TK]D-FenderDocHolliday, go look around
23:36.21DocHollidayhah :(
23:36.24*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
23:36.39DocHollidayManxPower, i need asterisk to basically direct the faxes to a SIP extension
23:36.43garreelbut it seems is no longer maintaned
23:36.52[TK]D-FenderDocHolliday, I was uncertain if the general channel driver would be smart enough to listen for fax tones.... it SHOULD.
23:36.53DocHollidaythe sip extension is basically an ATA with a fax machine hooked up to it
23:36.55sbingnerno it's not... I read the source wrong
23:36.57ManxPowerWe only run the NVFaxDetect on destinations that have an e-mail address configured.
23:36.59sbingnerit's RIGHT after the #endif
23:37.01type0ManxPower.. sure do
23:37.03rudholmsbingner: now I need to figure out if I can replicate this on a Sipura ATA, since a few of my FXS ports are on Sipuras.
23:37.13ManxPowerDocHolliday: Best of luck getting reliable fax over SIP
23:37.17[TK]D-FenderDocHolliday, this would be so much easier if you just bought another DID :)
23:37.26sbingnerrudholm, haha that's a completely different animal
23:37.30[TK]D-Fenderok, I've got to get ready to head out... back in a few....
23:37.34DocHolliday[TK]D-Fender, well i'll comment out all the other crap for now? hows that.
23:37.37sbingnerrudholm, I doubt it
23:37.41rudholmsbingner: yes I know, hence why I didn't even ask :)
23:37.47[TK]D-FenderDocHolliday, Crap?  that was the WORKING part :)
23:37.56DocHollidayLOL
23:38.12DocHollidaysorry to talk about your workmanship like that ;)
23:38.22rudholmsbingner: I plan on migrating to a TDM800, which will obviate my Sipuras, so it's not that big of a deal.  I can work around it in the meantime.
23:38.26DocHollidayis there anything that needs to be in the sip.conf?
23:38.39ManxPowertype0: Gaston Dureau, Avenue Computer Supplies, 504-523-7874, but you might have better luck with a west coast company
23:38.46ManxPowerTell them Eric sent you.
23:38.57simonkernHi, if i want to start asterisk + misdn this happens: http://www.ubuntuusers.de/paste/8974/ can somebody help me?
23:39.22DocHolliday[TK]D-Fender, i.e. is there something that has to be in the [general] section of sip.conf for it to work
23:39.26ManxPowerDocHolliday: How is the call arriving into Asterisk?
23:39.32garreelI think it shoud detect faxes without NVFaxDetect external application since sometimes I got : NOTICE[11959] chan_zap.c: Fax detected, but no fax extension
23:39.35DocHollidayover SIP
23:39.50sbingnerrudholm, did that work for you?
23:39.53ManxPowerDocHolliday: chan_sip does not do fax detection AFIK
23:40.05ManxPoweryou would have to use NVFaxDetect
23:40.05DocHollidayManxPower, ohhh
23:40.21DocHollidayand will that be able to send faxes to a SIP extension?
23:40.25ManxPowerWhy would it since fax doesn't generally work very well over IP?
23:40.50rudholmsbingner: I plan to migrate my POTS / FXO to a BRI interface and my TDM400 + Sipura FXS ports to a TDM800 (to eliminate the Sipuras)
23:41.05rudholmsbingner: gonna test in a moment...
23:41.09DocHollidayManxPower, if it doesnt work very well then i'll buy POTS, but why not try?
23:41.11ManxPowerDocHolliday: no.  It will detect the sending machine's fax tone and send the call to exten => fax if it detects it.  you would put a Goto to send the call to whatever exten handles your faxes
23:41.27DocHollidaygreat
23:41.29sbingnerApr  6 13:41:10 ERROR[3086202560]: chan_zap.c:10390 setup_zap: Ring or silence duration cannot be zero: 0,1000
23:41.41ManxPowerDocHolliday: We don't even run faxes over Asterisk.  We use dedicated POTS lines
23:41.47sbingnerlet's see what happens with 1
23:41.52DocHollidayManxPower, right
23:43.06DocHollidayManxPower, so i have to recompile asterisk?
23:43.48sbingnerusing 1 as the ring duration seems to have worked for my phones
23:44.06ManxPowerDocHolliday: of course
23:44.16Mahmoudhow to encrypt IAX2 communications?
23:44.23DocHollidayManxPower, the nvfaxdetect homepage seems to be down
23:44.37DocHollidaywhere can i download the source? :)
23:44.43sbingnerrudholm, I used this: cadence=1,-5000
23:45.38rudholmsbingner: I'm gonna try a ring duration of 0
23:46.46sbingnerrudholm, you can't... see my earlier error message
23:46.50Mahmoudany one knows how to encrypt IAX2 calls?, the configuration file doesn't say anything
23:47.01ManxPowerDocHolliday: no idea.
23:47.13DocHollidayheh well i'm out for today, thanks guys.
23:47.18sbingnerrudholm, unless you find out WHY it dosn't like a 0 duration and change it in the source ;)
23:47.20rudholmsbingner: ah, yes, and my test verifies that too :)
23:47.28rudholmsbingner: 1ms is short enough :)
23:47.34Mahmouddoes asterisk support SIP encryption?
23:47.36DocHolliday[TK]D-Fender, oh and you did a fantastic job with that IVR :) ... =P
23:47.37sbingnerrudholm, my phone didn't ring at all with 1
23:47.39rudholmsbingner: I doubt the mechanical ringers will respond to that :)
23:47.49sbingnerthat's not 1ms, that's like... 1 HZ or something
23:47.57ManxPowerMahmoud: No!
23:48.07MahmoudManxPower, this sucks..
23:48.19garreelDocHolliday: I have the source
23:48.23*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
23:48.45MahmoudManxPower, what about IAX2 encryption? still no?
23:48.45garreelbut didn't tried it
23:49.35rudholmsbingner: hmm, it doesn't seem to be taking the custom cadence
23:49.39Mahmoudasterisk will be the worst open source PBX once freeswitch is finished =P
23:49.46sbingnerrudholm, what did you put in?
23:49.56rudholmsbingner: cadence=1,1000,-5000
23:50.01sbingneryea that won't work
23:50.13sbingnerit has to be even... read the docs! lol... put what I put
23:50.25rudholmsbingner: I tried that too
23:50.31InnatechFrom scrollback -- isn't 60 cycle hum noise from the mains? Like a poorly grounded turntable?
23:50.58sbingnerrudholm, there would be an error message when the config was loaded if it doesn't like it... but 1,-5000 works fine for my phone
23:51.14ManxPowerBTW, the answer to my question is "plain old telephone service (POTS) has remained in the range of 300 Hz to 3.4 kHz, with a maximum signal-to-noise ratio of 30 dB"
23:51.42sbingnerrudholm, did you define all the default cadences also?
23:51.45rudholmsbingner: a "reload" on the command-line should be enough, right?
23:51.50rudholmyeah, I added all the default ones
23:52.00rudholmso my new one is r5
23:52.10sbingnerrudholm, no, you need to restart
23:52.16rudholmsbingner: oh :)
23:52.39rudholmthere we go
23:53.06*** join/#asterisk sysreq (n=sysreq@frank109.158.intermobilex.com)
23:53.17ManxPowerI came SO close to telling a customer "You're 75 years old, you should even be able to hear a hum at 400Hz!"
23:53.21rudholmsbingner: it works.  I just needed to do a full restart
23:53.39rudholmsbingner: thanks for taking your time to help me.
23:53.41sbingnerrudholm, your phones may not pick up callerid then tho
23:54.09rudholmsbingner: yeah, that may be true if they're looking for the ring voltage
23:54.11sbingnerrudholm, you could see if there's a particular value the phone recognizes AS a ring but doesn't ring so that it'll get callerid
23:55.02rudholmsbingner: yeah, it seems likely that a CallerID box accepts a wider range of "ring" signals than a mechanical 20Hz bell does
23:55.41rudholmsbingner: but I think a lot of CID boxes will just pick up and display any CID spill on the line, especially the ones that read the CID-on-call-waiting
23:56.27rudholmsbingner: I can test that right now...standby...
23:57.55sbingnerrudholm, 10 worked for me on my cordless phone
23:58.13rudholmthe CID box I'm using is apparently picky
23:58.32sbingnerrudholm, actually... my cordless phone worked at 1 -- lol
23:58.33rudholmit doesn't even like the default r3 (it's not a Call Waiting ID capable unit)
23:58.46rudholmsbingner: yeah, I suspect a lot of things will work with 1
23:58.46sbingnerrudholm, you DO have the -5000 in there right?
23:58.50rudholmyes

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.