IRC log for #asterisk on 20100815

00:05.25*** join/#asterisk joobie (~joobie@CPE-121-219-40-191.lnse1.lon.bigpond.net.au)
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01:40.58Beirdo!wx ksea
01:41.09Beirdoargh, wrong channel :)
01:44.05Kobazdo de do
01:44.16Kobazwhat's the deadline to register for astricon as a speaker?
01:44.49MaliutaI would hope their PA has enough speakers ;P
01:44.58Kobazheh
01:45.05Kobazi see there's still open slots in the speaker schedule
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01:51.58Kobazawwww
01:52.02KobazThe cutoff date for entering data here is June 30, 2010.
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02:54.12nightwalk[208084.003233] wctdm24xxp 0000:01:07.0: Missed interrupt. Increasing latency to 22 ms in order to compensate
02:55.04nightwalkCan anyone tell me if that's a symptom of a symptom of a bad motherboard or a bad card? Or maybe the a card/motherboard incompatibility of some sort?
02:56.44nightwalkTotal IRQ misses as reported by /proc/dahdi/1 is 202, so this seems to be happening fairly frequently. I'm guessing it's the cause of the random disconnects since supposedly 30ms is the max before it cuts off.
02:58.07nightwalk...and as the line above demonstrates, it's already over 20.
02:59.19*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
03:02.17nightwalkMy money is on it being a motherboard problem, since this motherboard doesn't play all that nicely with linux, either. Just wondered if it was possible to establish that as fact given the above error.
03:05.19*** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net)
03:06.21ChannelZhard to say
03:07.09ChannelZTried moving the card into a different slot?  It could be sharing resources with something else
03:08.07p3nguinHow can a main board care what operating system is employed?
03:08.39nightwalkOops, guess they changed the max. Just got this:
03:08.42nightwalkwctdm24xxp 0000:01:07.0: ERROR: Unable to service card within 25 ms and unable to further increase latency
03:08.49ChannelZit might have a wonky chipset that doesn't play well
03:09.13nightwalkChannelZ: The board only has one usable PCI slot, unfortunately
03:09.26ChannelZhmph.
03:10.04nightwalkp3nguin: It's not about the board caring what the OS is, it's about the mainboard not being standards-compliant
03:10.20drmessanoLOL
03:10.37nightwalkhates non-standards-compliant vendors...grrrr
03:11.15drmessanoThere's no such thing as "standards" when it comes to hardware
03:11.52nightwalkOh, really? I must've dreamed acpi and even x86 then :P
03:12.26drmessanoReally, do you know how many vendor specific implementations of ACPI are in the kernel?
03:12.36drmessanoACPI != ACPI
03:13.03nightwalkYes, but that just means there are lots of vendors not following the standard
03:13.11drmessanoREALLY NOW?!?!
03:13.37p3nguinlike SIP, perhaps?
03:13.42nightwalkExtensions are one thing, but core functionality should NOT deviate from the standards unless there's an extremely good reason.
03:14.20drmessano"should NOT" is wonderful and almost never applies in the real world
03:15.26drmessanoWhich is why every "standard" is succeeded by yet another "standard" that's determined to not go wrong where the previous went wrong
03:15.40drmessanoWhich is, most of the time, implementation
03:17.11drmessanoSo your motherboard not working because it's not "standards compliant" is highly unlikely.. Chances are if it DID stick to some standard, it wouldn't work, because a non-patched or non-amended implementation of the "standard" probably hasn't been tested due to lack of real world examples
03:18.12drmessanop3nguin, Could you imagine if someone implemented a truly standards compliant SIP stack?  NOTHING would work with it..
03:20.36nightwalkAnd yet, there are tons of boards made by manufacturers that carefully draw inside the lines whose boards I have no problem with. Just boards from companies that think they know best and do it their own way (Apple), or companies that decide being able to list more "features" is better than bothering to conform properly to the standards
03:21.51nightwalkActually, it's not even the fact that they don't conform to standards so much as it's that they don't conform to the standards *and* they fail to make any sort of meaningful contribution to the oss driver base for it.
03:22.10p3nguinIf everyone made their hardware exactly the same, where would the differentiation of the brands be?
03:22.17nightwalkprice
03:22.24*** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com)
03:23.22p3nguinIf everyone made their hardware exactly the same, the cost would be the same and the profit margin between cost and retail price _should_ be the same.  In the case where one brand has a higher retail price, you could safely by the other brand (since the hardware is exactly the same).
03:23.28DogBoywell it would be like with slogans printed on them
03:23.33DogBoylike with t-shirts
03:23.44p3nguinbuy, that is.
03:25.03nightwalkAh, but some would be more efficient, and thus be able to make the boards more cheaply.
03:29.07nightwalkAnyway, none of you have any idea whether or not the latency thing can be traced back to a faulty/non-standards-compliant motherboard, huh?
03:30.57nightwalkThis problem seems to come up a lot, but so far I haven't found any pages that explain what the root problem might be.
03:59.14ChannelZput it in another computer and see what happens
03:59.25ChannelZthen you'll have your answer
04:12.05drmessanoWhich "Standard" would this motherboard be in violation of to be causing these errors?
04:13.05drmessanoLets get down to some science here, rather than a Google search for the glossary to the A+ Study Guide
04:13.32*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
04:14.06drmessanohttp://zhulizhong.blogspot.com/2010/04/troubleshooting-of-bri-cards.html
04:14.15drmessanoCheck out the "IRQ miss" question
04:14.29coppiceA common problem is BIOSes chasing benchmark results
04:15.28*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
04:19.50coppiceThis page says far more about PCI issues http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
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04:53.02gamednaevening all.
04:57.22*** join/#asterisk nova911 (~Adium@115.118.225.212)
05:12.04Alton35Ok ChannelZ or others, still trying to use the local channel,
05:12.15Alton35my problem is that it places the call twice, and neither works, not helpful,
05:12.26Alton35however, I have learned how to use pastebin  :-)   Here:  http://pastebin.com/ZEMvtjM1
05:48.04gamednaanyone know if its possible to run multiple instances of asterisk on the same machine?
05:50.06joobiegamedna, why do u want tod o that?
05:50.17gamednadevelopment, testing, etc...
05:50.29gamednaexperimentation
05:50.42gamednadont want the overhead of a VM
05:51.20joobiei duno man
05:51.25joobieit's just weird what you're suggesting tho
05:51.41joobierunning a dev and production build onthe same machine
05:51.45gamednano no
05:51.46joobieusually u'd seperate these out
05:52.08gamednathis is not production
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05:52.21JerJerxen / openvz
05:52.33gamednaJerJer: trying to stay away from VM
05:52.39JerJergood luck then
05:52.53joobiewhat's wrong wiht 1 dev instance then?
05:53.11JerJeryou might be able to tell asterisk to use all different config files / sockets / etc
05:53.13p3nguinWhat good would it do to have a second asterisk running?
05:53.40Alton35virtualbox is good
05:53.45gamednawell, i dont have to run VMs for testing interaction between two asterisk boxes
05:54.15p3nguinalton35: The problem with your call file is that you are calling the phone number twice.
05:54.17gamednapacket sniffing on local interface, instead of going on the ethernet
05:54.32gamednaum... many other advantages
05:54.37p3nguinalton35: What are you trying to accomplish?
05:54.54Alton35using the local channel, to be able to make the call from within my agi
05:55.05Alton35I'm still stuck.
05:55.25p3nguinalton35: Your call file says to call 9565815577, and after it answers, call 9565815577, then bridge them.
05:56.09joobiegamedna> packet sniffing on local interface, instead of going on the ethernet
05:56.15Alton35let me see
05:56.17joobiewtf..
05:56.17p3nguinalton35: Using a local channel is for when you need to call another location in the dialplan.
05:56.43Alton35well, it was suggested last night and I kinda though it was a good idea, since I thought you had to have a channel up to invoke an AGI.
05:56.46gamednajoobie: yea why not?   say im on my laptop not connected to a net
05:57.09p3nguinalton35: Do you simply want to call a phone number and after it answers play the sound file?
05:57.22Alton35no, there are database things to do in the agi
05:57.54p3nguinI can't see any reason to call the same number twice.  Especially if you intend to bridge the two calls together.
05:58.13joobiegamedna, dood you are a tripper
05:58.15Alton35yeah, I just don't see how I'm doing it, been trying to get rid of that.
05:58.16joobie.. use VM's
05:58.42Alton35gamedna, yeah, I believe asterisk was not written to be run twice on the same machine, might not be a good idea.
05:58.57p3nguinalton35: You're calling the number via local channel, then calling it via context/extension.
05:59.05p3nguinPick one.
06:00.00Alton35I didn't think the local channel called it.  I would prefer that it didn't, but if I don't call from there, then how to get my AGI invoked?
06:00.02gamednajoobie: i do use VM's...  but what about running 4+ instances on the same laptop... not really "efficient"
06:00.28joobiegamedna, why the heck do you want to run 4 instances of asterisk on the same box?
06:00.52Alton35maybe a philosophical aversion to VMs  :-)
06:00.56gamednajoobie: application development
06:01.10joobiegamedna, that's great - but why do you need 4 instances of asterisk on the same box?
06:01.11gamednaAlton35: not at all
06:01.15gamednaAlton35: love VM's
06:01.26gamednajust finding them rather inconvenient for what i want to do
06:01.48JerJerI run 3 guests on my macbook  - usually 2 linux and one winblows
06:02.28Alton35p3nguin: I have never found out how to dial out from within an AGI under asterisk, which doesn't seem to be such an unreasonable thing to do.
06:02.30p3nguinalton35: If you have to call the number via local channel to execute that part of the dialplan, don't include Context: and Extension: as well as Channel: in the call file.
06:02.38gamednajoobie: working on something that deals w/ asterisk boxes talking to each other
06:02.39gamednacant say more than that ATM
06:02.42Alton35ok, let me see here
06:03.15gamednaJerJer: i usually have 2  running... Windows and Linux.
06:03.30joobiegamedna, then go somewhere else for help if you dont want to explain why
06:03.38JerJergamedna:  if you look into the so called HA solutions they all use some form of virtualization technology
06:03.54joobiegamedna, what you're suggesting to do sounds stupid.. and you cant explain why you need the 4 instances, so bugger off
06:04.25Alton35I have seen too much server sprawl with virtualization, but if it's not too much trouble to get going, it can be useful here and there.
06:04.48gamednajoobie:  why such hostility?  just wanted to know if anyone knew how to do it?
06:04.58p3nguinalton35: Or if you find that the call file doesn't work without those other fields being present, at least use a context/extension that isn't the same as the number you've already called by a local channel.  It's nonsense to call the same number twice at the same time.
06:04.58gamednajoobie: so bugger off yourself.
06:04.59xheliox"cant say more ATM" -- translation "I couldn't find a clue with two hands and a flash light"
06:05.12gamednasheesh
06:05.15joobiegamedna, sorry we don't support retarded configurations
06:05.56gamednajoobie: too bad you are so shortsighted.
06:06.05Alton35Hah, gamedna, like I say, asterisk not particularly written to be run more than once on the same machine, so it's not good practice to try doing it.
06:06.24xhelioxdistracts gamedna with a shiney metal object
06:06.32gamednaJerJer: like i said before, this is just for convenience in development.  Not for production...
06:06.55p3nguinI'm still waiting to hear what good it would do to have a second asterisk running.
06:07.05gamednaAlton35: have you tried it?
06:07.07p3nguinThere's nothing it can do.
06:07.11joobiehe cant tell you.. it's a secret
06:07.15Alton35No, because I know it's not written for that.
06:07.17p3nguinThe ports will already be in use by the first one.
06:07.27ChannelZbind it to another IP
06:07.29Alton35Maybe try one asterisk with 4 clearly-separated contexts or something.
06:07.39gamednalooking at the source and the conf files, looks like it can be done.
06:07.40*** join/#asterisk thansen (~thansen@S0106001c1092cd20.cg.shawcable.net)
06:07.47JerJerduno then
06:07.53gamednajust needs ssome major configuration
06:07.56xhelioxYou could probably setup 3 chroot environments and bind it bind to a different IP for each instance, I'm clueless as to why you'd do that.
06:08.01Alton35still not a good idea, you have a lot of people here telling the same thing
06:08.22gamednaAlton35: yea, but nobody saying, i tried it at it does not work.
06:08.36joobiebecause no one is retarded enough to go down that path
06:08.36Alton35that's because we know better than to even try it  :-)
06:08.38gamednaxheliox: good idea.
06:08.45xhelioxgamedna: Because shockingly, no one is dumb enough to try.
06:08.46gamednahehe
06:09.03joobieit's like loading multiple instances of apache to run 2 sites.. when you can just use virtual hosts
06:09.06gamednagamedna: hahhaha...  only dumb until it works.
06:09.06xhelioxWhich is saying a lot if you spend a couple of minutes watching this channel.
06:09.22Alton35part of learning about computers in general is knowing which path to take.  don't take the cursed and dark path.
06:09.22xhelioxUm, nope.. even if it works, it's still dumb.
06:09.47joobie.. or the path that is prone to failure and carries a large management overhead
06:09.49gamednajoobie: people have done that before.
06:10.01joobiegamedna, like i said, we dont support retarded configurations in here
06:10.11gamednaxheliox: yea but if it sames me time, then its not
06:10.19ChannelZthat's what #freepbx is for
06:10.22b14ckHey guys. I'm currently using dahdi ONLY for dahdi_dummy support. Is there a way I can completely remove the chan_dahdi.conf file without having Asterisk complain? Right now it's just an empty file, seems kinda pointless to have it.
06:10.26joobieheh
06:10.41Alton35they're saying that if you need to do it, then you're probably something something else wrong in the first place.  and they're offering to help you with the whole thing, but you won't say anything more.
06:10.57gamednaAlton35: i understand taht...
06:11.12gamednaAlton35: im not trying to be a prick, i just cant say anymore....
06:11.15xhelioxgamedna: In the time you've had this discussion, you could have had 3 VMs running. And even then I'm not sure what the point is.
06:11.21Alton35haha
06:11.22Alton35true
06:11.35gamednaxheliox: i cant run 3 vms.. im already resource bound on this machine
06:11.43Alton35well, what they're also saying is, if there's so much money at stake, then go somewhere else and pay someone.
06:12.02Alton35oh come on, VMs will run with 64mb of ram each.
06:12.12Alton35install slackware or debian 5 or something.
06:12.17gamednatrying to work around a CPU / memory resource limitation on my computer, so that i can test mutlpe asterisk system interactions.
06:12.30p3nguinb14ck: Is that file causing some problems to exist?  Why remove it if there is no problem?
06:12.49b14ckp3nguin, the only thing that bothers me about removing it is: asterisk complains in the full logfile when restarting.
06:12.58b14ckIt has no other effects.
06:13.01p3nguinleave it alone, then.
06:13.10b14ckAm I missing something?
06:13.19b14ckSeems like it wouldn't be complaining unless I did something bad =p
06:13.20p3nguinAn empty file surely doesn't take up much disk space.
06:13.28b14ckYah, it's the thought though.
06:13.34b14ckIt's a matter of pride, or something.
06:14.04xhelioxAhh.. I love the smell of stupid in the morning.
06:14.14b14ck...
06:14.24b14ckI just want to remove the file and not have asterisk complain.
06:14.43b14ckIncase I'm missing something, and it is actually causing a problem that I haven't yet caught.
06:14.45b14ckHow is that stupid.
06:15.07p3nguinMore nonsense.
06:15.08xhelioxHey guys.. I have this file on my Linux box /boot/vmlinuz-2.6.23 and when it boots I get an error, can I remove it????
06:15.49p3nguinIt's not taking up space to have it, asterisk complains if it's gone... the answer should be clear: LEAVE IT.
06:18.23b14ckSigh.
06:18.30b14ckThe file has been empty up until this point.
06:18.51b14ckSo, I'm assuming that the error I'm getting is pointing to some functionality I've broken.
06:18.58b14ckIt doesn't make me happy to just leave an empty file there.
06:19.05b14ckI want to know what I'm breaking.
06:19.21ChannelZJust get on with your life
06:19.30b14ckMight as well just read the source.
06:19.32b14ckbrb
06:19.37xhelioxlol
06:20.00b14ckDon't know why everyone has a problem answering a question. Might as well just say "I don't know".
06:20.36ChannelZyou're loading a module which has a config file.  You want to delete the config file.  It complains.  You are freaked out that it complains.  What answer is it you want?
06:21.09b14ckI'd like to know if by providing either an empty config file, or no config file at all, if it impacts the ability of dahdi_dummy to provide timing to my environment.
06:21.24ChannelZHere's a thought, try it.
06:21.38b14ckI have (as I mentioned).
06:21.46b14ckBut I'm not sure whether there is some other breakage happening behind the scenes.
06:21.48ChannelZIt takes 2 seconds to make a dialplan with MeetMe() or something.  Or presumably you already have something setup using it or else you wouldn't be needing dahdi_dummy
06:21.55b14ckAlready done that.
06:22.09xheliox"I don't want to live on this planet anymore.." -Professor Farnsworth
06:22.14ChannelZlol
06:22.17ChannelZblasts off
06:22.26xhelioxTake me with you!
06:32.50*** join/#asterisk boobsbr (~george@201008039169.user.veloxzone.com.br)
06:32.54boobsbrhowdy
06:33.27boobsbrIs there a way to test if the softphone I'm using is sending the DTMF to Asterisk?
06:34.37xhelioxWhat DTMF mode are you using?
06:35.45*** join/#asterisk JerJer_ (~PhatJ@asterisk/original-h323-guy/JerJer)
06:36.05xhelioxIf you're using rfc2833, you should see the DTMF packets in rtp debug. If you're using inband, I'm not entirely sure.
06:38.09boobsbrxheliox, inband with ulaw
06:38.42boobsbrxheliox, I'll give rfc2833 a try
06:39.17xhelioxafaik, that's the more reliable way to go.
06:44.28*** join/#asterisk ChannelZ (~bobm@burner.com)
06:45.03xhelioxChannelZ: wb - just had to reboot into Windows 98 and steal my idea, eh? :)
06:45.11ChannelZyes
06:45.16xhelioxBastid.
06:45.22xhelioxI want royalities.
06:48.21ChannelZI can offer you a cookie.
06:48.36xhelioxWhat kind of cookie?
06:48.51boobsbrxheliox, I managed a little test with ReadDigits() and SayDigits() and the DTMF is working. Thanks.
06:49.02ChannelZChocolate chip.
06:49.06xhelioxOk - that seems fair.
06:49.13xhelioxboobsbr: Excellent.
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07:29.22boobsbrxheliox: I can't get Asterisk to say the POUND digit with my little test routine, STAR and the numbers plays OK though.
07:30.02xhelioxAh.. the classic # bug..
07:30.12xhelioxlol - that's quite strange.
07:30.26*** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com)
07:31.26xhelioxboobsbr: I haven't a clue why that might be, but if you're using rfc2833 now.. you should see the dtmf packets in rtp debug, verify it's actually sending it.
07:31.42xhelioxOther than that, I've had way too many drinks to offer much more assistance for this evening. :)
07:31.59boobsbryes, the packets are going through with rfc2833, a bunch of them
07:32.21xhelioxWhich soft phone are you using?
07:32.41boobsbrTwinkle, the only one I could get working on Ubuntu
07:34.30boobsbrand X-Lite on a Windows VM, but the sound is horrible
07:34.46ChannelZI think it defaults to gsm being first on the list
07:34.47xhelioxyou're experiencing the same thing with both clients?
07:35.24boobsbrno, X-Lite doesn't seem to send the DTMF packets
07:35.47boobsbrand I changed the codecs on Twinkle to alaw and ulaw only
07:36.09xhelioxif you're using rfc2833, the codec matters none.
07:37.07Kyoshboobsbr: are you running asterisk on a vm or x-lite or both?
07:37.48boobsbrasterisk on a debian vm and xlite on a windows vm, both on virtualbox
07:38.37Kyoshhmm
07:38.46Kyoshim curious why x-lite on a vm?
07:38.52Kyoshor "in" a vm
07:39.43Kyoshrfc2833 works fine for x-lite, thats for sure.
07:39.53boobsbrbecause it failed to work in ubuntu, even though I installed the damned 32 bit libstdc++.so.5 library it required. i think it's an alsa/oss/pusleaudio problem
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07:40.40Kyoshwell you have to narrow down if the problem is on the host or the client (for ast of course, not virtualbox)
07:40.53Kyoshthen determine if virtualbox is playing a role
07:41.14Kyoshanyone will telly ou that you need to narrow down the problem and once you do, then you can begin troubleshooting it
07:41.29gamednai second that..
07:41.36gamednastart in one direction and work your way to the other side
07:41.37Kyoshcurrently your problem has too broad implicating factors for problems
07:41.43Kyoshbingo
07:41.50gamednaand dont make any assumptions
07:42.02Kyoshgotta rule out, not assume
07:42.04Kyoshsup game
07:42.30gamednahey kyosh..
07:42.54boobsbrwell, I'll try removing virtualbox outta the equation once I get a new machine next week.
07:42.55Kyoshnada, working on some iax2 integration
07:43.19Kyoshso far i got some nice fax over ip working with ulaw
07:43.49boobsbrso, can I use the * key to make blindxfers?
07:43.51Kyoshmy partner is workign on the hylafax tests
07:43.56*** join/#asterisk nova911 (~Adium@115.118.229.201)
07:44.11Kyoshboobs: if thats how your features.conf is configured
07:45.06Kyoshmsn mobile on my bberry bold killed my battery from 30% to 0 in 30 mins
07:45.13Kyoshso stupid
07:45.16boobsbrso, I only need to change the blindxfer option, to, let's say, *0 ?
07:46.03gamednain band fax or using T.38?
07:46.25Kyoshinband
07:46.34Kyoshsip+ulaw
07:46.44Kyoshspeed sucks, between 4800 and 9600
07:47.00Kyoshhowever we;ve tested hylafax and it uses iax at 14400
07:47.11Kyoshso im gonna make an integration
07:47.41Kyoshsadly hylaxfax has no database support except for log files
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07:47.47Kyoshgonna have to change that
07:47.57shido6where do you get your linksys srtp private key from?
07:48.09shido6voxilla's generator is broken
07:48.32Kyoshmy who?
07:50.31gamednaKyosh: nice work.
07:52.27Kyoshwhat work/?
07:52.33Kyoshwhat did i break now?
07:52.47gamednano, the fax stuff
07:53.12gamednathe hylafax iax @ 14,400
07:54.20Kyoshoh i dunno.  it just happened :-p
07:54.23WIMPyFax at 64000 might be more interesting nowadays.
07:54.43Kyoshoh man id be happy with that
07:55.16WIMPyThing is that it should be a lot easier than the old stuff.
07:55.29Kyoshshould be
07:58.19WIMPyTrouble is that you coldn't even have it on the same number with Asterisk.
07:58.55WIMPyDamn. The more I get into it again the more I think that there's still a very long way to go :-(
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08:10.06boobsbralright, I got Asterisk to understand that the pound sign is used for transfering
08:10.49boobsbrbut now it says the number i'm dialing is not a valid extension
08:10.52boobsbr=(
08:12.11boobsbris there a way to find out what number Asterisk thought I was dialing?
08:16.56boobsbrwow, it worked
08:17.08boobsbri don't know what i did, but it worked
08:18.11boobsbrdamn, this stuff is pretty cool
08:18.24Kyoshi gotta make a new ivr for a customer
08:18.28Kyoshi hate doing that
08:20.11WIMPyDoes anyone have a good idea how to patch BC based routing into Asterisk? Maybe by defining different contexts per BC in the channel config?
08:20.20WIMPyAny comments?
08:20.39WIMPy(probably the wrong day to ask)
08:20.44boobsbrsorry, what is a BC?
08:21.06WIMPyBearer Capability
08:21.18WIMPyi.e. the type of service.
08:21.37boobsbroh.
08:21.55boobsbrI'm pretty new to this, so, sorry again.
08:22.12boobsbrKyosh: do you have to record the messages too?
08:24.28Kyoshsadly yea.  my voice is too raspy and nasaly
08:24.47Kyoshmaybe i can get a voice emulator
08:31.39boobsbrIs there a way to test call quality?
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08:32.24boobsbrLike a standard sound file to be played at one end and recorded at the other, then compared and analyzed.
08:33.06Kyoshumm
08:33.11Kyoshstandard?
08:33.41Kyoshhmm
08:33.48Kyoshfor a single call?
08:33.52boobsbryeah
08:33.56Kyoshyou can perform an echo test
08:34.34boobsbrbut how can the input and output be analyzed?
08:35.06boobsbri tried the echo test, and it works perfectly, the calls also work
08:35.41boobsbrI need to write on a paper that the call quality is analog or superior to a POTS call
08:36.02boobsbrand my professor said I need a way to quantify this quality
08:36.19boobsbrjust saying it's good is not proof enough
08:36.39boobsbrgot the idea?
08:36.54Kyoshyea dude
08:36.56Kyoshumm
08:37.04Kyoshhmm
08:37.18Kyoshthe inbound call quality or the outbound call quality?
08:37.47boobsbrnow you got me
08:37.50Kyoshok
08:37.55Kyoshhave asterisk record the test call
08:38.06boobsbrthat's an interesting idea
08:38.10Kyoshthen grab the file from the /sounds dir and run it thru a spectrum analyzer
08:38.31Kyoshalso record the same phrase into a wav file and run it thru the analyzer
08:38.58Kyoshmake sure the wav file is 8bit, same bitrate
08:39.01boobsbrI thought about that
08:39.13Kyoshprolly the only way i can think of
08:39.18boobsbrbut I didn't know about the record function
08:39.18Kyoshi aint that smart
08:39.24Kyoshya
08:40.35Kyoshin sip.conf, under the extension you are calling, set record_out=Always and record_in=Always
08:40.54Kyoshor fromt he ext you are using
08:40.55Kyosheither way
08:41.04boobsbrgot it
08:41.09boobsbrthanks for the help
08:41.14Kyoshhope it does help
08:41.23Kyoshusing asterisk for class project?
08:41.29Kyoshwhat kinda class is that?
08:41.46boobsbrit's a graduation project
08:41.54Kyoshwhat kinda class is that?
08:42.05boobsbrelectrical engineering , telecom
08:42.19Kyoshno dude
08:42.21Kyoshno way
08:42.23Kyoshno fukin way
08:42.28boobsbrwhat?
08:42.35Kyoshgraduation projects for my bsee was way different
08:42.57Kyoshwaaaay diff
08:42.57boobsbrharder?
08:43.06Kyoshwe didnt get to play with shit like that
08:43.17Kyoshthen again, this is going back a bit
08:43.22boobsbrwhen?
08:43.40Kyoshoh umm
08:43.48Kyoshyou may have been out of diapers :-p
08:44.26boobsbr1983?
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08:47.26boobsbrhere goes another question, has anyone used pjsua? http://www.pjsip.org/pjsua.htm
08:47.59Kyoshnah dude i was a freshman in hs then
08:49.32Kyoshwhat is that link?
08:49.48boobsbra command line implementation of a sip client
08:50.46Kyoshoh
08:50.52Kyoshfor the purpose of?
08:51.05Kyoshits command line
08:51.36Kyoshlast smoke before bed
08:52.16boobsbrplacing and receiving calls from a remote location, using a command line shell. saves bandwidth
08:52.46boobsbrfor testing purposes, it's not practical for regular use.
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09:01.55boobsbrwell, gotta go to bed too, 6am already. thanks everyone for the help!
09:02.06boobsbr\q
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09:30.23gamednahave multiple asterisk instances running on the same machine w/o vm... no problems
09:30.31gamednadont need SUID
09:30.36gamednadont need chroot
09:31.06gamednaand no virtual macines..
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12:10.58rethusi have solvved all problems to compile app_cconference for asterisk 1.62. But i have still little problem to install it
12:11.42tzafrir_laptoprethus, just drop the module into the modules directory, right?
12:11.48rethusif i tyüe make install, i got "install -m 755 app_conference.so /usr/lib/asterisk/modules", but ffor make i have added the Parm: "ASTERISK_INCLUDE_DIR=/usr/lib64/asterisk/"
12:12.18tzafrir_laptoprethus, do you have a .so file?
12:12.32tzafrir_laptopIf so: just copy it to /usr/lib/asterisk/modules/
12:12.43tzafrir_laptopA bit simpler than fighting makefiles
12:12.43rethustzafrir_laptop: jep
12:13.00rethuson my system its lib64/asterisk...
12:13.05tzafrir_laptopDid it need patching? (if so, please submit fixes)
12:13.05rethusthats right.
12:13.26rethusyes, the patch is still createt on sorurceforge.
12:13.31rethusi get it from there
12:13.45rethusnow the make-precess runs like a charm
12:14.06rethusisn't there a parm for the makefile, where i can change the install-dir?
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12:57.14rethusk fund it. and works well
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13:56.53yidiyuehananybody knows how to kick out some people in a conference from auto attendant?
13:57.41EmleyMoorThe SIP VoIP mode on my N97 has stopped working :(
13:58.07yidiyuehanI know there are commands available in CLI for meetme kick, but how to control if I remotely dial into my * box?
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14:06.28*** join/#asterisk Benwa (~Benwa@unaffiliated/benwa)
14:07.14rethusi have installed Conference as additional applikation. in cli i have now the command conference.
14:07.27rethusdid anyone know, whit which command i could start a conference?
14:07.39rethuscore show help conference gives no output for this module
14:09.03Guggewhere did you get the application?
14:09.55*** join/#asterisk guilhermebr (~Guilherme@189.63.75.25)
14:11.29[TK]D-Fenderrethus: app_conference.so gives you "Conference"
14:11.58[TK]D-Fenderrethus: "core show applications" <- go check your list
14:12.12rethusGugge: i got it on sourceforge.
14:12.12rethus[TK]D-Fender: it is into this list
14:12.28[TK]D-Fenderrethus: Then that's what you call.
14:12.36Guggehttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference <- so maybe this gives you an idea rethus
14:13.02rethusif i say, help output nothing, means nothing usable... on section syntax is only "Not available"
14:13.34Gugge"core show application Conference", like all other apps
14:13.54rethusi have even try this: http://help.cloudvox.com/faqs/reference/reference-for-php#manageconf
14:14.09rethusdidn't work. jump directly from the agi-script back to the dial-plan
14:14.34rethusGugge: like i sayed above... there is no usable output
14:15.16Guggepaste the output
14:15.25Guggeon pastebin
14:15.27[TK]D-Fender[10:13]<rethus>i have even try this: http://help.cloudvox.com/faqs/reference/reference-for-php#manageconf <- this has nothing to do with app_conference
14:15.47rethushttp://pastebin.com/4fRSbPpw
14:16.21Guggeif that is the output you get from "core show application Conference" you have a seriously fucked up install :P
14:16.21rethus[TK]D-Fender: this is the same command... paste over phpagi
14:16.41[TK]D-Fenderrethus: So you have a broken AGI. What does this have to dowith app_conference?
14:16.41rethusno, thats the output if i try to dial in
14:16.44rethus;)
14:17.10Guggewhy dont you just make a dialplan with Conference(123)
14:17.11[TK]D-Fenderrethus: You have not called it in there
14:17.23[TK]D-Fenderrethus: FIX YOUR AGI
14:17.35rethushere the output of help for the module: http://pastebin.com/wx83MdCd
14:17.35[TK]D-Fender]reththis is not a problem with app_ference
14:18.07[TK]D-Fenderrethus: [10:12]<Gugge>http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference <- so maybe this gives you an idea rethus <--- he GAVE you a link for this
14:18.38Guggeactually, that link is the first google hit on "asterisk conference" :P
14:18.48Guggeno sorry, the second hit :)
14:18.51Guggethe first hit is meetme
14:18.59rethusthanxs, i have seen this now. Only want to show, that there is no syntax-explenation on the module.
14:19.15Guggeim gonna bet there is a readme
14:19.17[TK]D-Fenderrethus: WE KNOW.  Big deal.  They didn't put the instructions in there.
14:19.22[TK]D-Fenderrethus: DEAL WITH IT
14:19.29rethusso i now the parms now, but not the exact syntax.
14:19.40rethusthats what i'm searching for.
14:19.48[TK]D-Fenderrethus: Your problem (well, just one of them anyway) is that you aren't even CALLING the application.
14:19.51Guggewhat dont you get from the wiki link?
14:19.53rethusbut now i first look for my agi
14:20.21rethushow could u see, that i not called in into my agi ?
14:20.29rethus[TK]D-Fender
14:20.51[TK]D-Fenderrethus: You see when dialplan apps get called in CLI.  it ISN'T
14:21.05[TK]D-Fenderrethus: I can see it isn't called... because it ISN'T
14:21.08rethusah, ok
14:21.29[TK]D-Fenderrethus: We see the RX & TX and there ISN'T anything.
14:22.11Guggerethus: try opening the README file from the source, and search for "Using app_conference"
14:22.21Guggeit actually is documented in the file you downloaded .....
14:23.24rethusGugge: ahh, k. Thanks. I've found it
14:23.36[TK]D-FenderGugge: For reference he was working on just installing it for WEEKS
14:24.09Guggeimpressive ...
14:29.17rethusso, if i logged in on the agi-interface, i should see this on CLI with agi-debug ?!
14:29.56rethusisn't that "<SIP/dev-00000006>AGI Tx >> agi_request: auth_congregation.php" the indication, that this script send a request to agi ?
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14:30.37[TK]D-Fenderno, that is a refernce to some PHP file
14:30.42[TK]D-Fenderrethus: ^
14:31.53rethusthis means, my connection data (username and password) for connecting to agi... may be wrong?
14:35.29[TK]D-Fenderrethus: AGI doesn't HAVE passswords.  What the fuck are you talking about?
14:36.13[TK]D-Fenderrethus: rethus You dont' seem to have the slightest clue about ANYTHING you are working on.  I suggest you go hire a consultant.
14:37.02rethusif u want to use agi, u have to set webenabled true in manager.conf and there u can give a "secret" means password?!
14:37.39[TK]D-Fenderrethus: that is ***AMI**, not AGI
14:37.52[TK]D-Fenderrethus: And what does AMI have to do with your use of app_conference?
14:37.54GuggeAMI/AGI ... who can tell the difference :P
14:38.10[TK]D-FenderGugge: Everything on the outside looks the same!
14:38.16Gugge:)
14:40.40[TK]D-Fenderrethus: And if you are using AMI via PHP_AGI as called by the AGI dialplan app... you aren't even ACCESSING it via the "web".
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14:44.40[TK]D-Fenderrethus: And you also don't call dialplan apps via AMI <---
14:44.54[TK]D-Fenderrethus: Stop pouring windshield washer fluid in your GAS TANK
14:48.55rethus<PROTECTED>
14:49.32[TK]D-Fenderrethus: You are talking about things in the same sentence that have nothing to do with each other.
14:49.40rethus"In everything, do to others as you would have them do to you"
14:49.42[TK]D-Fenderrethus: You can't call dialplan apps via **AMI**
14:49.56[TK]D-Fenderrethus: AGI doesn't have passwords.
14:50.02[TK]D-Fenderrethus: Nothing you say makes any sense
14:50.17[TK]D-Fender[10:36]<[TK]D-Fender>rethus: rethus You dont' seem to have the slightest clue about ANYTHING you are working on. I suggest you go hire a consultant.
14:51.06rethusis that reason enough ffor you to disrespect me... cause i asking for help
14:51.33[TK]D-Fenderrethus: We can't even tell what you are actually needing help with.  The pieces don't even add up
14:51.47[TK]D-Fenderrethus: And you never stated your goals.
14:52.07[TK]D-Fenderrethus: No clue.  No goals.  No CODE.  No debug.  NOTHING
14:52.18[TK]D-Fenderrethus: You have done nothing to help yourself in this process
14:52.38[TK]D-Fenderrethus: What little you have every shown or asked has made no sense
14:52.50[TK]D-Fenderrethus: How can anyone help you?
14:53.19[TK]D-FenderWell I've wasted enough time on this... I'm off for a while
14:53.25rethusthats not a reason do disrepect or kidding me. if you don't wan't to help me... only ignore me, but stop kidding me.
14:53.43jamkowhat is the reason for using openser if you can use dundi?
14:54.10[TK]D-Fenderrethus: None of the pieces you have mentioned belong in the same SENTENCE.  How can we "help" that?  You are talking about things that don't belong together
14:54.23Guggejamko: what does openser and dundi have in common?
14:54.29[TK]D-Fenderjamko: OpenSER has nothing to do with DUNDi
14:54.38jamkoI understand that
14:54.49[TK]D-FenderHOLY FUCKING SHIT THE CRAZIES ARE OUT IN FORCE
14:55.05jamkoMy question is, if you are not having issues with load balancing, or nat, why would one use openser?
14:55.08Guggemaybe we are just having a nightmare [TK]D-Fender :)
14:55.23jamkoHey fender go fuck yourself.  Anyone have an intelligent answer?
14:55.30[TK]D-Fenderjamko: LOTS of reasons.
14:55.40jamkolets hear one to start you asshole
14:56.20Guggeactually, i dont see any reason to run openser
14:56.29Guggeas it no longer exists :P
14:56.36jamkoopensips then
14:56.56[TK]D-Fenderjamko: To proxy *'s connection to multiple providers, to route out multiple public interfaces since * + multi-homed = PITA, to support LARGE installs where * can't handle setup of so many calls or clients at all.  So you can use * as a BACKEND since SER does RADIUS billing so much better.
14:57.05[TK]D-Fenderjamko: HUNDREDS of reasons.
14:57.17[TK]D-Fenderjamko: TOTALLY different scale.
14:57.39[TK]D-Fenderjamko: And SER is a proxy, * is NOT.  Proxies can do things * cannot.
14:57.42GuggeBut of cause, if asterisk does what you need, there is never a reason to mix in another app
14:57.59[TK]D-Fenderjamko: Yes they can relate to each other and be used together benificially.
14:58.09[TK]D-Fenderjmakbut they are VERY different
14:59.17[TK]D-Fenderjamko: And starting out of the gate with relating SER to DUNDi is a bad start.  DUNDi isn't even really a load balancing tool.
14:59.56[TK]D-Fenderjamko: It is a "misc peering".  There is no weighting, prioritizing,, logging, etc.
15:00.47[TK]D-FenderDUNDi = nearly worthless.  Also a poor hack at trying to be a "general E.164"
15:01.34[TK]D-Fenderis out for a while.
15:01.35WIMPywoot?
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15:01.54WIMPydundi is great for a set of decentral servers.
15:05.44jamkook.. So in a ser + * scneario, * would be used for pbx function, and SER for signalling.. but does ser stay in the middle after the call is setup, or does it reinvite the endpoints together?
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15:09.56bougymanjamko: that's up to you.
15:09.56p3nguin<jamko> Hey fender go fuck yourself.  <--- I wonder if this guy thinks he's the first one to think of that remark.
15:10.31bougymanin that scenario, * kind of is the endpoint.
15:10.46jamkop3nguin:  I wonder if you haver ever seen a vagina.  Go fuck yourself.
15:11.06bougymanjamko: this is completely uncalled for
15:11.17jamkoIs it necessary to be condescending assholes in a chat room where people come for help?  NO, it's not.. so again go fuck yourself.
15:11.33bougymantake your own advice and ignore the people who you aren't comfortable with.
15:11.38p3nguinOh, a real wise guy, trying to complicate human anatomy with telephony.
15:12.01jamkooh a real wise guy... Think of that yourself?
15:12.17p3nguinNo, I read it on a blog.
15:12.35bougymanjamko: what led you to this question, may I ask?  you are talking a scenario that would usually only come up in a high-traffic or complex routing needs situation.
15:12.40jamkothanks for the tip bougyman, but I se here and watch these guys act like pricks over and over to other people.  It is completely uncalled for.
15:12.53bougymanjamko: agreed, but so is what you are doing.
15:13.00jamkotrue.
15:13.07bougymanespecially when someone is spending their time trying to understand your problem so I can help.
15:13.17jamkolol .. thanks..
15:14.54jamkoI'm just trying to visualize the big picture.  Right now I have maybe 25 phones registered to a single asterisk box, but starting thinking bigger, and thought of bandwidth limitations, failover, load limitiations on a single asterisk server, etc.
15:15.06bougymanah.
15:15.21bougymanyes, to scale asterisk, you have to go horizontal.
15:15.27jamkoand I like to setup labs and see how stuff works.
15:15.40bougymanthe asterisk side is the easy part.
15:16.06bougymanusing kamalio or ser (both the same sip-router code) takes a pretty solid knowledge of the stack.
15:16.29bougymani'd be hanging out in #kamalio to glean that special knowledge.
15:16.40jamkoWell when dealing with NAT, you can't reinvite a peer and have good results.  But does keeping ser in the middle defeat the purpose?
15:17.07bougymanjamko: that's where a proxy comes in.
15:17.14jamkoand can it even stay in the middle?  I was under the assumption it could not pass rtp traffic.
15:17.29bougymanit does not deal with rtp at all
15:17.39bougymanrtpproxy or a similar tool hanles the rtp
15:18.02jamkowow.. the rabbit hole...
15:18.13bougymani'm _not_ trying to be an ass, but once again this is a kamalio/ser/sip-router problem, not an asterisk one.
15:18.20bougymanmaybe that's why the * guys in here went off?
15:19.12jamkono, they went off because I asked the question openser v. dundi, and because they are elitist know it alls, decided to be pricks.
15:19.28jamkodundi is part of asterisk, no?
15:19.54bougymanok maybe some of both, then.. dundi is part of * but your problem isn't an asterisk one.
15:20.31jamkomaybe.. but my question boils down to asterisk clustering with failover.
15:20.42jamkoand I believe asterisk has something to do with that, no?
15:20.47bougymanthere are as many ways to do that as there are people who have done it.
15:20.54jamkook and..
15:21.10jamkomy theory is (as naive as it might be)... is to
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15:21.32cuscohi
15:21.41bougymanopenvz live migration? check.  sip-router handling external load balancing? check.  SBC facing endpoints? check.
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15:21.46cuscocan we use case in AEL, or only if else ??
15:21.47bougymanthere's just a ton of choices, jamko
15:21.49[sr]howdy people
15:22.21WIMPy[sr]!
15:22.23jamkohave a bunch of asterisk boxes, and when the amount of phones surpasses the supported load in asterisk, you register the next group of phones to another asterisk box, and use dundi to "find out" about the others.
15:22.25Guggecusco: http://www.voip-info.org/wiki/view/Asterisk+AEL2 case is described there
15:22.31[sr]WIMPy: how r u?
15:23.09WIMPyDebugging...
15:23.17cuscothanks Gugge
15:23.55WIMPyBut I had to debug the debugger first.
15:24.03[sr]WIMPy: thats the best time for it, on a sunday afternoon (here)
15:24.34_GuhitI'm running version 1.6.2 and have the problem, when a second call connects to the asterisk box things get laggy, key presses are missed, and the sound in confbridge is broken and choppy.  The asterisk box is only using about 1% CPU.  I'm using iax or sip channels, there is no dahdi hardware enabled on the box.  The asterisk box is a co-lo, so there is plenty of bandwidth.
15:24.40WIMPyIt's a test setup, so doesn't matter.
15:25.11Gugge_Guhit: what kind of hardware do you run it on?
15:25.47_GuhitGugge: It's an old 1.33ghz athlon, 2GB RAM
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15:26.26Gugge_Guhit: and no virtualization on it?
15:27.05Alton35Maybe "core show translation" and see if any of the numbers are too large.
15:27.11[sr]WIMPy: i see
15:27.39*** join/#asterisk fr00d (~andi@unaffiliated/fr00d)
15:27.39_GuhitGugge: I'm running in a FreeBSD jail, but there is nothing else running on the box.
15:27.44fr00dHello!
15:28.19Gugge_Guhit: strange: i have no idea then
15:29.35fr00dIs that possible to use asterisk with KabelDeutschland? I have an account where the two wires at the cable modem are diabled and phone normally just works with the shipped Fritz!Box 7270 via TAE, S0 or DECT. What I want is to install asterisk on my OpenWRT and connect a USB ISDN interface to use the s0 bus. Is that possible?
15:29.59fr00dOh just strip one "is that possible" ;)
15:30.23WIMPyfr00d: Just get the config out of the FritzBox and put them into sip.conf.
15:30.43WIMPyGoogle will give you howtos for that.
15:31.31_GuhitAlton35: what should I be looking for in that output?
15:32.01_GuhitHow large?
15:32.44fr00dHmm, damn. My Fritz!Box which I just got yesterday broke this morning and I thought I can get faster a USB ISDN interface to get telephony work again.
15:32.51Alton35_Guhit: let me see
15:33.40WIMPyfr00d: If you can't get at it's configuration, you're screwed.
15:34.07*** part/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)
15:34.09WIMPyOtherwise the USB thing is possible.
15:35.19Alton35_Guhit: Do you know which codecs you are using?  If you are using different ones, the computer has to translate between them.
15:35.28fr00dOk, then I'll search for a howto on that and try to setup the configuration so I have a basic undertanding and hofully I just need to insert my credentials to get this work when I got a working Fritz!Box from KD.
15:35.40_GuhitAlton35: gsm and ulaw
15:36.05Alton35And the number(s) in your table where those intersect?
15:36.08_GuhitAlton35: My numbers are in the 5000 range for that
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15:36.28fr00dWIMPy: Could you advice me a usb isdn interface which would work?
15:36.33WIMPyfr00d: Mind you, for me KDGs telephony has been pretty horrible.
15:37.05Alton35_Guhit: Golly, that shouldn't kill you.  Trying to think of what else it could be.
15:37.24fr00dWe will see. :D
15:37.24WIMPyfr00d: It needs to be based on a CCD HFC chip.
15:37.55fr00dI saw a howto with a conceptronics c128u device.
15:38.16WIMPyfr00d: Haven't heard of that one.
15:38.27fr00dDon't know if there's a really great variety of devices.
15:39.18*** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)
15:39.26WIMPyfr00d: I've got Trust and a X-Tensions mouse-shaped ones that work.
15:40.12_GuhitAlton35:  How much latency on the SIP trunks would be too much, I'm getting between 85-180ms.
15:40.51fr00dWIMPy: Where do you come from and which provider do you use for that?
15:41.35WIMPyfr00d: de and I tried KDG, but had to cancel that early as it was not usable.
15:41.45Alton35_Guhit: That's high.  I can't think of how it would slow down the computer or asterisk though.
15:42.43WIMPyfr00d: Actually using sipgate worked much better than KDGs own telephony service.
15:42.48WIMPy(at least for me)
15:43.25fr00dYes I know, I've got a sipgate account, too.
15:43.37fr00dBut never tried that with asterisk.
15:43.50WIMPyfr00d: Ok, then you do have a working phone connection at least.
15:44.10fr00dHehe
15:44.25fr00dBut I don't have any credits on my sipgate account, just the free one.
15:44.46fr00dI'd really like to use my flat at KDG.
15:44.48WIMPySo do I.
15:46.09*** join/#asterisk pif (~ldm@zenon.apartia.fr)
15:46.42fr00dHmm, this would add 8,90 € per month.
15:46.50fr00dBut it would work...
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15:52.34*** part/#asterisk rethus (~suther@p5087A9D5.dip.t-dialin.net)
15:55.35p3nguin_guhit: 5000?  Holy cow!  Between 1 and 5 might be a normal number for the translation between gsm and ulaw.
15:57.01_Guhitp3nguin: I just switched everything to gsm and it seems to be working better, any ideas those high number, or anything I can do to diagnose?
15:57.28Alton35p3nguin: just verifying that you've seen my message.
15:57.54p3nguinhuh?
15:58.11Alton35I sent you 2 messages last night and one a while ago.
15:58.22Alton35like so
15:58.49p3nguinYeah, I don't receive unsolicited private messages.
15:58.55Alton35aha
15:59.26Alton35basically offering to stuff your paypal full of money for some help.  ok, not quite, but something.
15:59.34Alton35can't afford leif's $200/hour
16:00.07_GuhitOn my desktop I'm seeing about 1000/400 for that translation...the numbers are in microseconds.
16:00.09p3nguinYou want to discuss it privately?  I'll open it up for you.
16:00.13Alton35ok
16:00.34p3nguinTry now.
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16:26.17stevehi all
16:27.15steveI'm trying to figure out which IP a specific provider sends inbound SIP connections from, so I can add them to my firewall... I know where they go to, but can't seem to crack where they're coming from - can anyone point me to a specific portion of the log that would reveal this please?
16:30.10Alton35should be the same IP.
16:30.19Alton35it would be unusual for it to be different
16:30.28steveit's definitely not the same IP
16:30.46Alton35or turn off your firewall and do:    sip set debug peer xxxxx
16:30.54steveah, good idea
16:31.11stevewill that put info into asterisk/full?
16:31.23Alton35I don't know, but it sure will put it on the console.
16:31.46Alton35You'll have to scroll back to read it all.
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16:35.36Alton35steve: I'm not sure how you'd see which IP to use, maybe look in your firewall log?  or ask the guys at the provider.  seems like the right way to go about things.
16:38.39jamkosteve: you should be able to see this from the cli when a call comes in.  The only way you would not is if  you don't have 5060 open in your firewall.  Even if the call is rejected, you will see the ip address in teh console.  You should not need to turn on debugging to see this.
16:39.28jamkojust make sure your core verbosity is at least 3
16:40.21steveyeah I actually found the problem not where I expected it to be
16:40.30stevefirewall wasn't permitting loopback it would seem
16:40.38steveasterisk wasn't pleased :P
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17:00.44cuscohi..
17:01.47cuscowhen I am performing an outbound call...
17:02.31cuscoafter the Answer, it is still ringing on the other end... is there any flag on Dial() that allows me to identify for how long it actually rang,
17:02.46cuscoor how long I actually connected?
17:03.09jamkoyes.
17:04.01jamkobut is the other end a sip device?
17:04.10jamkosounds like you have a nat issue.
17:04.47cuscono
17:04.59cuscothe other end is PSTN trough dahdi
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17:05.24cuscowhat is the flag i am looking for? or the cdr var?
17:06.45jamkoDial(SIP/4025,15)
17:06.53jamkothe 15 would be your timeout
17:07.07cuscooh no, Im not looking for a timeout
17:07.19jamkooh i see.
17:07.23jamkoi misread what you said.
17:07.24cuscoIm trying to count the time on-line
17:08.09cuscowe insert info on sql database, when dialing out and when hang up
17:08.54cuscoand the time between dial and hangup would be ok, unles the called party does not answer or answers after 5 secs ringing
17:09.06cuscoso.. how can I count only the connected time?
17:10.19jamkosorry no clue man.  pass
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17:11.16*** join/#asterisk SWFu (~SWFu@unaffiliated/swfu)
17:11.16Guggecusco: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
17:11.33Gugge"CDR Variables" on that page
17:12.17SWFuCould any give me some pointers about UK providers please?
17:12.54cuscothanks
17:12.54jamkoswfu, what kind of pointers?
17:13.11SWFuLike who to use
17:13.23madprhow can i escape a variable in dialplan application arguments? (coz it may contain a comma, which makes asterisk crazy)
17:13.30SWFuOf what I understand a SIP provider?
17:13.54SWFuOnly started looking into the idea this morning
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17:19.19jamkois this for calls in the UK, or you just want the provider to be located in the UK?
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17:50.40[TK]D-Fenderjamko: Somewhat belated but watch the anatomical vulgarity.
17:52.26[TK]D-Fender[11:19]<jamko>no, they went off because I asked the question openser v. dundi, and because they are elitist know it alls, decided to be pricks. [11:19]<jamko>dundi is part of asterisk, no? <- and no, we weren't fractionally as harsh as your retorts.
17:53.24[TK]D-Fenderjamko: And DUNCi has no relationship with SER, isn't a "load balancer", etc.  Some larger points of what SER can add to a hybrid install were already provided
17:53.30[TK]D-FenderDUNDi*
17:57.59jamkoright I understand it has no relation to SER.  I am just want a bunch of asterisk servers to be able to talk to each other, without overcomplicating the issue.  By "talk to each other"
17:58.27jamkoI mean, find a sip registration, or extension that is not on the local box.
17:58.44[TK]D-Fenderjamko: So far that doesn't say "load balance" either...
17:59.00[TK]D-Fenderjamko: Do you need to "search" to find where it is appropriate?
17:59.35[TK]D-Fenderjamko: Or is this a scenario where you know 2XX = PBX1, 3XXX =PBX2, etc?
17:59.45jamkoright but do I need load balancing, if I only put say 100 phones per box?
18:00.23jamkoright.
18:00.37[TK]D-Fenderjamko: A single * can handle hundreds of calls depending how you do it
18:00.55[TK]D-FenderjamThousands even.  I think the record is a bit over 10000 on a single box
18:01.22jamkowow.
18:02.02[TK]D-Fenderjamko: Sangoma sellsan 8 Port PRi card... that's 240 calls just to support that 1 card.  Of course * can handle more.
18:02.41jamkowell, then my problem gets to be bandwidth restraints on a single internet connection in front of asterisk, which is why I was thinking multiple boxes, behind multiple internet connections.
18:03.03jamkothen using dundi to link them.
18:03.42jamkoso say with g729 I max out at 25 simultaneous calls, but want 500 calls to act as if on 1 pbx.
18:04.44[TK]D-Fenderjamko: That gets a LOT messier...
18:04.45jamkoand I am pure sip, no need for cards.
18:04.52[TK]D-FenderjamBecause of *'s transcoding,e tc
18:06.18jamkoFender : would opensips help at all in sorting out that mess?
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18:07.29[TK]D-Fenderjamko: it might help.... it is a complicated beast though...  not sure I'd advise it yet.  I'd have to see a clearer map of your layout.
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18:16.04Kobazanyone know if you can still squeek into astricon as a speaker?
18:16.16Kobazon the site it says the deadline is june 30... but there's still empty slots in the schedule
18:19.01rethussomeone here who is using phpagi?
18:19.17rethustry to find a way to create a new conference via agi.
18:20.37ChannelZWell you can use 'exec' to call a dialplan app like ConfBridge or MeetMe
18:20.54[TK]D-Fenderrethus: You don't create conferences with AGI
18:21.23[TK]D-Fenderrethus: You call dialplan appas with AGI.
18:24.26rethusso this should work for creating a new conference on channel 1 as admin-user?
18:24.26rethus$agi->exec('meetme', '1', 'a');
18:25.22[TK]D-Fenderrethus: that is JOINING a conference.  This is not "creating" anything
18:25.31rethusk
18:25.34Alton35I use phpagi.  Not that I can help you much.
18:25.53Alton35Stick with these guys for now.
18:26.19bougymanonly one way to find out, Kobaz
18:26.23[TK]D-FenderAlton35: It isn't even an AGI program.
18:26.27bougymanthere are usually cancellations at these things, too.
18:29.02rethusmy problem is, that asterisk jumps into the phpagi-script... but doesn't matter what i do, it come back (while don't entering this conference) - say "AGI Script auth_congregation.php completed, returning 0" and switch back to dialplan.
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18:29.38rethusso i checked the asterisk-logs and start agi debug, but get no error which show me, why the conference is not enterd
18:29.54[TK]D-Fenderrethus: Your AGI is BROKEN.  We've rtond you thins many times.
18:30.00[TK]D-Fenderrethus: You're doing it wrong.
18:31.22rethusso how can i check it step by step to get it to work again?
18:31.53rethusand what do you mean with agi... my php-script?
18:32.04[TK]D-FenderYES
18:32.28rethusk, but it works on another machine (ubunbtu) like a charm (not here on SUSE)
18:32.28[TK]D-Fenderrethus: Your AGI script is BROKEN.  Please get this in your head.  BROKEN.  Bad code.  Errors.  FAIL.
18:33.02[TK]D-Fenderrethus: rethus We can't see anything so how are we supposed to know WHY it is failing?
18:39.31Kobazbrokes
18:39.44Kobazteh borken
18:41.26jamkorethus: maybe you want to stick with ubuntu?
18:41.50[TK]D-FenderBecause clearly his DISTRO is at fault.. yeah.. uh huh
18:43.02jamkoeither way, it is just logical to stick with what is working.
18:43.18jamkounless you can fix what is not, which apparently he can't/
18:44.04[TK]D-FenderNope.
18:44.05Kobazor he can learn about the system he's developing on
18:44.31[TK]D-FenderKobaz: this has been going on for weeks.
18:45.46Kobazhehe
18:47.39b14ckHey, is anyone using the new dialplan pattern matcher in production? extenpatternmatchnew=yes ?
18:48.44Kobazb14ck: what's it look like
18:49.13b14ckKobaz, what?
18:49.41Kobazthe patterns
18:49.44Kobazor is it just a new engine
18:49.57rethusjamko: i would, but is not my server
18:50.00b14ckIt's a new algorithm that you can (optionally) use.
18:50.06Kobazhow's it work
18:50.10b14ckI wonder if it's been well-tested or not.
18:50.14b14ckAnd if it is production safe.
18:50.18b14ckMight as well give it a try.
18:52.30b14ckNice, seems to work!
18:52.42b14ckWe've got like ~60,000 extensions =p
18:52.46b14ckSo that's a big speedup.
18:52.47b14ckheh
18:53.53Kobazoh... efficiency improvements?
18:54.14b14ckYah, apparently, with 10,000 extensions, the speedup is 374x
18:54.24b14ckWith 1000, it is 25x
18:54.35Kobazheh
18:57.51rethusso, i have found the error now. it was a problem with the used PATH in Asterisk. in php-script i had an include()... i didn't know why, but asterisk always seems to start from /tmp as basedir.
18:58.49[TK]D-Fenderrethus: * does what you tell it to.
18:59.59rethuscause no errors @ all was thrown, it was an error which is hard to find. And there ist one thing, that makes this problem more difficult. If u start asterisk with the -cv command out of another directory... this direcory is the base-dir.... so if u start asterisk as deamon via init.d, it has another basedir than starting it with -cv
19:01.22rethusif some phpagi-users here... how do you debug your scripts. is there a way to make asterisk more verbose on showing this kind of php-script-errors?
19:01.24[TK]D-Fenderrethus: Maybe you should look at what the STARTUP script is doing.  That is your job
19:01.40rethusyour right, good idea
19:03.19rethusmhh, no path-parms in the init-script.
19:06.46[TK]D-Fenderrethus: asterisk.conf <-
19:07.14*** join/#asterisk shapr (~shapr@c-76-29-246-212.hsd1.al.comcast.net)
19:07.37rethusso no parms in the config-files of asterisk, (on asterisk.conf its only set as record_cache_dir). so i realy wonder, why the handling on ubuntu is deifferent from suse
19:08.06[TK]D-Fenderrethus: Maybe you should look at BOTH system similarly.
19:08.20rethusis there an parm for asterisk, which set the basedir manualy from cli?
19:08.55[TK]D-Fenderrethus: Go look for yourself
19:09.12[TK]D-Fenderrethus: Takes less to find out than it did for you to ask.
19:11.34*** join/#asterisk Mark22 (~mark@unaffiliated/mark21)
19:14.06jamkomaybe you should keep your systems uniform and stay with ubunutu.  I think you would have saved a lot time, unless you simply are just doing this for the sake of doing it.
19:17.53shaprDoes dahdi-linux-complete put its bunch of kernel modules into the modules.conf?
19:21.23jamko1
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19:24.36rethusjamko: i would stay on ubuntu, but its not my server where my asterisk-webapp is installed... and so i must get this running on SUSE
19:25.53Mark22Hello, I am trying to get LCDial (http://www.voip-info.org/wiki/view/Application+LCDial#ApplicationLCDialLeastCostRoutingFailOve) to work (mainly because it gives some failover options). log information: http://yourpaste.net/5932/  dialplan (part related to this call): http://yourpaste.net/5933/  I did check if I could connect with the login details in lcdial.conf to mysql from the commandline and it did work without any problem
19:26.12Mark22what could be the solution?
19:26.34p3nguinshapr: modules.conf is deprecated and not present on current Linuxen.
19:27.08p3nguinshapr: Actually, disregard what I just said.
19:27.55p3nguinshapr: I was thinking of something else rather than /etc/asterisk/modules.conf.
19:28.11[TK]D-Fender"Couldn't connect to database server '127.0.0.1'." <- looks blatant enough to me
19:28.19[TK]D-FenderMark22: ^^
19:28.42Mark22[TK]D-Fender: it looks good, however why couldn't it connect is my problem (so I could find a solution)
19:29.04[TK]D-FenderMark22: It's wrong.  It says its wrong.  It isn't lying.
19:32.19Mark22if I do "mysql -h 127.0.0.1 -u asterisk -p asterisk" and after that enter the password (same user/database/password as listed in lcdial.conf) it works, why lcdial can't use it is my problem :S (passwords contains A-Za-z0-9)
19:32.33Mark22I don't say the log is lying, I only can't find the problem
19:34.59[TK]D-FenderMark22: We sill can't see enough to point it out to you either.
19:35.10[TK]D-FenderRock, hard place.  Hard place, rock.
19:35.16[TK]D-Fenderhas settled the introductions.
19:35.53Mark22do you know how I can get more information about it?
19:36.39[TK]D-FenderMark22: We don't see all configs, database dumps, etc
19:36.57[TK]D-Fendercomparative local logins, etc
19:38.29drmessanoSounds like either a RAM failure or maybe a bad PCI slot
19:38.51drmessanoThat's as good of a guess as any, with no data
19:39.15drmessanoMaybe check the LED's and the color of the power cord
19:41.06shaprp3nguin: So, dahdi modules to modprobe do show up in /etc/asterisk/modules.conf?
19:41.39rethushave anyone an idea, how i can make a benchmark while "converence_app" vs. "meetme" ?
19:41.42p3nguinshapr: I wouldn't expect them to, since /etc/asterisk/modules.conf is for asterisk modules and you're asking about kernel modules.
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19:51.34Godfather_jhi
19:54.08shaprhhi
19:54.44xhelioxjello
19:56.21[TK]D-Fenderbeats xheliox with a Pudding Pop
19:56.51xhelioxSeems like a perfectly good waste of a Pudding Pop.
19:59.19shaprOne or more of the dahdi kernel modules has make my system extremely flaky. I want to only load the wctdm module. Does anyone know how to keep the other modules from loading?
20:01.57[TK]D-Fendershapr: remove them
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20:22.57cuscowhat would be the most common cause for a call to disconnect while on hold for 30 secs
20:23.00cusco?
20:26.29jamkortpholdtimeout= can cause issues, not sure about the most common though.
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20:32.53tompawGood evening, everyone.
20:35.58ChannelZmaybe your MOH music is really terrible and people just hang up
20:37.09xhelioxChannelZ: Disco lives forever!
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20:40.10Kobazhmmm
20:40.16shaprI'm going through the asterisk book (but using asterisk 1.6-current). I've reached chapter 4, where it talks about setting up a dialplan for test calls...
20:40.31shaprHow do I know that my TDM400 is setup correctly?
20:40.39Kobazin 1.6.2... it seems if you don't answer the call... cell phones calling into a t1 won't hear ringing
20:41.11shaprAlso, I know the red card is an FXO, and the green card is an FXS, but I don't remember which slot on the card maps to which RJ11, does anyone know?
20:41.13tompawGuys, I am trying to test my a2b installation under full load. But before the cpu and ram are consumed, I'm getting a bunch of errors like these: http://pastebin.com/Hfq7Sewn
20:41.38GameGamer43|Mactompaw: a2billing is not supported here per the channel topic
20:41.55Kobazeven if i do Answer()... Ringing()... cell phones still wont hear ringing
20:41.58tompawGameGamer43|Mac: these are not a2b errors, it's related to * only.
20:42.07Kobazi have to play music on hold with a recorded wav of ring
20:42.28tompawmy ulimit is set to unlimited btw...
20:43.41tompawIt looks like there are some sorts of limits hidden somewhere in */agi configuration that I cannot find. Syslog is clean, there is free cpu and ram...
20:45.44tompawGoogling for "rtp.c: Unable to allocate RTP socket: Too many open files" takes me way back to 2006 and is mostly related to ulimit, which is not the case here...
20:46.14[TK]D-Fender[16:40]<shapr>How do I know that my TDM400 is setup correctly? <- USE IT
20:46.44shapr[TK]D-Fender: I would love to, but I don't yet know how to do that.
20:47.11[TK]D-Fendershapr: You don't know how to DIAL?  Very sad
20:47.26[TK]D-Fendertompaw: [Aug 15 22:15:33] ERROR[26266] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files <- FS lockout.  Somrhitng is opening too many files.
20:48.37*** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com)
20:48.44tompaw[TK]D-Fender: checked the ulimit, /proc/sys/fs/file-max, running * as root, no OS errors in syslog... maybe I should try disabling the cdrs?
20:49.16[TK]D-Fendertompaw: No, the lack of ability to open files killed your ability to open SOCKETS as well because they are effectively the same
20:49.42shapr[TK]D-Fender: I still have lots of basic questions like, should I hear a dialtone when I have a phone plugged into the FXS port of my TDM400? If not, should I get some reaction when I dial?
20:49.43[TK]D-Fendertompaw: [Aug 12 20:16:48] ERROR[5549] utils.c: write() returned error: Broken pipe <- also an error common to AGI/AMI screwups
20:49.55tompawcorrect. I only wonder on which "layer" is this limit enforced, sinced it's not the filesystem.
20:50.02[TK]D-Fendershapr: You should.  Did you conenct the molex to the card?
20:50.11shaprI did, yes.
20:51.37shapr[TK]D-Fender: I don't get a dialtone on the FXO port, but I do get power to the phone and am able to dial.
20:54.15*** join/#asterisk CoderForLife (~Miranda@cpe-174-101-150-41.cinci.res.rr.com)
20:54.38[TK]D-Fendershapr: What do you mean  no tone on the FX port?
20:54.42[TK]D-FenderFXO*
20:54.50tompawlol, someone is having a similar problem to mine and asked at a2b forum. the response was: http://forum.asterisk2billing.org/viewtopic.php?f=2&t=2661&p=9530&hilit=too+many+open+files#p9530
20:55.02tompaw"THIS IS NOT ASTERISK SUPPORT FORUM!!!"
20:55.03tompaw=)
20:55.19shapr[TK]D-Fender: I mean that, picking up the handset does not give dialtone, but the phone is powered.
20:58.40[TK]D-Fendershapr: Phones should plugged into FXS ports, not FXO
20:58.55shapr[TK]D-Fender: Ah right, sorry... I meant the FXO port.
20:59.23shaprI don't have access to the PSTN at the moment, so the FXO doesn't do anything for me.
20:59.36p3nguinheh
20:59.50shaprEr, I meant the FX*S* port...
20:59.56shaprgrumbles
21:00.54*** join/#asterisk Alagar (~Administr@122.164.34.24)
21:03.12shaprI'm just trying to get asterisk talking to my analog handset. Once I have a vaguely functional config, I can poke around in the system more productively.
21:05.09tompawwhoa
21:05.31tompaw[TK]D-Fender: you're actually right, looks like centos ignores my limits, just checked /proc/<*_uid>/limits... 1024 !
21:08.11[TK]D-Fenderimagine that....
21:10.10tompawfixed.
21:10.50*** join/#asterisk DennisG (~DennisG@84.30.136.208)
21:11.09tompawit went up to 190 calls :-) now I'm stuck with this broken pipe. If only asterisk reported which part causes the error...
21:25.09*** join/#asterisk Diffen2 (~diffen2@c-2875e555.042-17-73746f11.cust.bredbandsbolaget.se)
21:30.16*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
21:33.40*** part/#asterisk rethus (~suther@p5087A9D5.dip.t-dialin.net)
21:42.30*** join/#asterisk ruied (~ruied@95.69.57.109)
21:53.12*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
21:53.30Diffen2hello. i have two asterisk and i have created a user on the first one and then im trying to use that user as a siptrunk on the second asterisk. is that possible?
21:54.22*** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com)
21:57.50[TK]D-FenderDiffen2: Yes.
22:04.21Diffen2ok nice i think i have managed it to register as a peer but when i call in i get username mismatch from the second asterisk
22:04.41Diffen2guess i have missed something out in sip.conf?
22:05.21b14ckCan someone explain to me what allowoverlap does (in sip.conf)? What is 'overlap dialing support'? I googled it, but couldn't find any straightforward answers.
22:07.46*** join/#asterisk Pegasus_RPG (~chatzilla@p4FF90751.dip.t-dialin.net)
22:08.34Pegasus_RPGhello there. I'm having a problem with * using Broadvoice VoIP service. Inbound calls are going straight to the BV voice mail system and Asterisk's console doesn't ever see the calls
22:08.49Pegasus_RPGIt used to work fine a year ago
22:09.00Pegasus_RPGI've since dist-upgraded to the latest Debian Testing
22:09.21Pegasus_RPGAny ideas on what to try?
22:13.10shaprIs there a guide that translates the Asterisk book's 1.4 configuration directives into 1.6 directives?
22:13.14*** join/#asterisk fofware (fabian@190.225.15.129)
22:13.28jamkoPEGASUS: Sounds like the provider is sending calls to the wrong ip, or not at all.
22:13.29shaprFor example, there's no zapata.conf file in 1.6
22:15.47jamkoPEGASUS: Or you don't have the sip port specified in your sip.conf, open in your firewall.
22:16.01Pegasus_RPGchecks his firewall
22:16.35*** join/#asterisk Carp1 (~Carp1@cpe-24-92-37-23.nycap.res.rr.com)
22:18.15carrarshapr, that would be the all the text in ChangeLog since that release :)
22:21.09Pegasus_RPGjamko: My firewall looks good. I have 69, 5060-5063 and 10K-20K all UDP forwarded to my * box. Also if I have my SIP hard phone connect directly to Broadvoice, it works fine
22:22.43Pegasus_RPGwithout forwarding anything special in the firewall
22:23.26jamkowhat about your linux firewall?  is it off?
22:23.55*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
22:27.03Pegasus_RPGjamko: I never installed one
22:27.41Pegasus_RPGjamko: though iptables is installed...but there's no "iptables" process running
22:29.58jamkoare you setup for ip auth through your provider?
22:30.23Pegasus_RPGjamko: Don't know what that is
22:30.45carrariptables -n -L
22:31.34jamkohow do you authenticate your asterisk box to your origination provider?
22:34.54Pegasus_RPGcarrar: no rules are listed
22:35.10Pegasus_RPGjamko: using the steps given here: http://broadvoice.com/support_install_asterisk.html
22:36.06*** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com)
22:36.30[TK]D-FenderPegasus_RPG: enable SIP DEBUG, and show us  your REGISTRATION, and your CALL ATTEMPT.
22:36.31[TK]D-Fender~pb
22:36.32infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
22:36.34[TK]D-Fender^^^
22:40.17Pegasus_RPGhttp://pastebin.ca/1917743
22:41.22[TK]D-FenderPegasus_RPG: What router are you using?
22:41.40Pegasus_RPGLinksys WRT310N v1
22:41.50Pegasus_RPGFirmware Version: v1.0.09
22:42.23[TK]D-FenderPegasus_RPG: Ok, not seeing the problem, just yet.  Confirm that your WIN IP on your router is indeed : 79.249.7.81
22:42.26Pegasus_RPGI have it set to try to do QoS on the * IP
22:43.01[TK]D-FenderDon't screw with the traffic in any way.  Just ensure 5060, 10000-20000 all UDP is forwarded and that's all on the router side.  No SIP ALG transform, etc
22:43.01Pegasus_RPGyup 79.249.7.81
22:43.25Pegasus_RPGand the firewall doesn't have anything SIP-specific. Just QoS
22:43.34Pegasus_RPG(which doesn't work all that well, I might add)
22:43.48[TK]D-FenderPegasus_RPG: Ok, BTW, I just got to the bottom... ensure that your broadvoice peer has "insecure=port,invite"
22:43.59Pegasus_RPGahh, it's using insecure=very
22:45.14Pegasus_RPGok that helped...now it's just not matching the extension
22:45.32Pegasus_RPGwhere's a list of wildcard characters?
22:45.42[TK]D-Fender~book
22:45.43infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
22:45.46[TK]D-Fender^^^
22:45.46carrarheh
22:46.09Pegasus_RPGthanks
22:46.46Pegasus_RPGhttp://www.asteriskdocs.org/ doesn't seem to work, btw
22:46.53Pegasus_RPGdigs up his old PDF copy
22:47.14carrarhttp://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
22:47.26golikwid|macPegasus_RPG: works fine here
22:48.01Pegasus_RPGgaah, I hate it when ISP DNS servers try to be "helpful" with their host-not-found catch-alls
22:48.32golikwid|macyea i dont use my isp's dns server for that reason
22:48.35Pegasus_RPGah, so ! means what _ used to in v1.0.x?
22:49.33[TK]D-FenderNope
22:49.51[TK]D-FenderYou also shouldn't need a wildcard match
22:52.06Pegasus_RPGwheee it works now!
22:52.36Pegasus_RPG(Somehow my first exten =>_<number>   had n instead of 1)
22:52.46Pegasus_RPGthank you so much everyone
22:53.26[TK]D-FenderPegasus_RPG: you're welcome
22:53.36Pegasus_RPGI do also have a number reformatting question.
22:53.43[TK]D-FenderShoot
22:53.58Pegasus_RPGI live in Germany, but use BroadVoice for US calls
22:54.37[TK]D-FenderPegasus_RPG: Don't worry, we've already alerted the authorities....
22:54.41Pegasus_RPGI want * to take any dialed number starting with 0, strip off the 0, add 01149 and the rest of the remaining digits, and tell BV to call that
22:54.45Pegasus_RPGlol
22:55.43golikwid|mac(01149)+0|NXXXXXX
22:55.47golikwid|macsomething like that i think
22:59.05Pegasus_RPGalso, sicne I call a number of european countries with variable-length numbers, how can I have it match any length for int'l calls? _011X.  ?
22:59.32golikwid|macyea the .
23:00.25golikwid|macwhats the number pattern used un europe
23:00.28golikwid|mac?
23:01.20florzthere is non, essentially
23:01.22Pegasus_RPGthere is no defined rule
23:01.22florz+e
23:01.29[TK]D-Fender[18:54]<Pegasus_RPG>I want * to take any dialed number starting with 0, strip off the 0, add 01149 and the rest of the remaining digits, and tell BV to call that <--- exten => _0.,1,Dial(SIP/broadvoice/01149${EXTEN:1})
23:01.36Pegasus_RPGeven within Germany, the number lengths can vary
23:01.48golikwid|macweird
23:02.00golikwid|macsounds like more complicated dialplans than around here
23:02.31florzonly because asterisk is kindof not adapted to it
23:02.46golikwid|macim sure there is a way
23:02.55golikwid|macjust more lines of patterns to match
23:03.13florzthat doesn't really help you
23:03.30florzvariable length means you can't know when a number is complete
23:04.33Pegasus_RPG[TK]D-Fender: thanks, that works perfectly
23:04.45[TK]D-FenderPegasus_RPG: Or at least precisely as you asked
23:04.49Pegasus_RPGyes...
23:04.57[TK]D-FenderPegasus_RPG: Now go read up on viables & patterns
23:05.00Pegasus_RPGit mangles if I manually dial 011xx... of course
23:05.02Pegasus_RPGI will
23:06.47Pegasus_RPGthanks alot for your help and patience with me
23:08.30Pegasus_RPGuh, I can't find an explicit negative match notator
23:08.44Pegasus_RPGthat is a "don't match" operator
23:08.57Pegasus_RPGor "match everything but 123"
23:09.29[TK]D-FenderPegasus_RPG: No such thing
23:09.57Pegasus_RPGthat would explain why I can't find it. :)
23:10.03[TK]D-FenderPegasus_RPG: You have to have a "match ALL that takes precedence LAST
23:10.14Pegasus_RPGall
23:10.17Pegasus_RPGer ahh
23:10.39*** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf)
23:12.38xSmurfhey guys, I'm working with the manager api and using the Originate action to connect a sip extension and an outbound call. There are two things I cannot figure out. The first one being how to send a caller id to the sip extension (ex: Outbound <5551234567>) and the second is it possible to play back "please wait while I try to connect you" when the sip extension has answered and the outbound call is being dialed?
23:14.05xSmurffor a ref, this is my php script connecting to the manager api using sockets http://pastie.textmate.org/private/5p96sjkihuop6m8yyn91cq
23:14.55xSmurfnothing fancy
23:15.35[TK]D-FenderxSmurf: does it Originate?
23:15.46xSmurfyeah it works great :)
23:16.10[TK]D-FenderxSmurf: then that part is almost irrelevant
23:16.44xSmurfI know I know, I just posted it for reference, some people are more picky and like seeing the big picture ;)
23:17.25xSmurfand others might just enjoy the script, I'm just exploring after I wrote an Address Book.app plugin for Click to dial
23:17.30[TK]D-FenderxSmurf: I suppose...
23:19.00xSmurfback the the relevant part, is there anyway for to display CID information to the sip extension?
23:19.37*** join/#asterisk root52 (~root52@ip68-228-177-7.cl.ri.cox.net)
23:23.27root52hey all... http://pastebin.ca/1917791 So This problem just started yesterday. Seems that my dahdi channel goes into red alarm and then comes out a min or two later. Then when it is not in alarm and I make a test call I get a message I have not seen before and it never picks up the call like it says it does. Any Thoughts. The odd message is line 11 and 12 in the pastebin.
23:23.54[TK]D-FenderxSmurf: Which?  the Channel: or the EXTEn?
23:24.33xSmurfthe exten, basically I wanna display the number from the channel to the exten when originate does the local dial
23:24.52xSmurfCallerID doesn't seem to be display, and anyhow, I don't want that passed to the channel!
23:27.29[TK]D-FenderxSmurf: then set it in your dialplan
23:28.48xSmurfyeah mean create a custom extension that the call is directed to and that then connects the exten and the chan? yeah I was just thinking of that, though I'd like to avoid having to play with the dialplan (don't always have access to it)
23:30.14[TK]D-FenderxSmurf: And why not?
23:30.42[TK]D-FenderxSmurf: Next set a SIMPLE variable in your Originate and set the CID to that.
23:30.56xSmurfyeah I figured
23:30.57[TK]D-FenderxSmurf: Do not attempt to set it directly
23:31.04xSmurfas I said don't always access to the dialplan
23:31.56xSmurfwhat about the playback of some file to the exten while the call is connecting? same thing??
23:32.05[TK]D-FenderxSmurf: and I asked you why not... and why would you need continuos access to it?
23:32.30xSmurfnot continuous... access period
23:32.39[TK]D-Fender[19:31]<xSmurf>what about the playback of some file to the exten while the call is connecting? same thing?? <- this you clearly need control over the dialplan for.
23:33.00[TK]D-Fender[19:32]<xSmurf>not continuous... access period <- please answer this simple question in some meaningful way.
23:33.00xSmurfok
23:33.05xSmurfsay I can ask an admin to add a manager for an API, but not to add custom stuff in the dialplan
23:33.32xSmurfdoesn't matter anywayI pretty much have my answer, it that case that would be a no go
23:35.16[TK]D-FenderxSmurf: If you have access to the API you could add your own support dialplan regardless of them
23:35.33xSmurftrue
23:35.46xSmurfwell
23:35.58xSmurfdepending on the permissions of the user
23:37.12[TK]D-FenderxSmurf: If your environment is hostile, don't bother asking us "how" when you are being faought against.  These restraints are ARTIFICIAL.
23:37.32[TK]D-FenderxSmurf: ":p
23:37.57xSmurfso if someone is in a restricted environment they shouldn't attempt to do anything with asterisk? makes sense
23:38.04xSmurf;)
23:40.11root52hahah
23:40.35root52sorry wrong window ;-)
23:40.50*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
23:41.29[TK]D-FenderxSmurf: No, asking how and then saying. "yeah that'll work, but I'm not allowed" makes it sould like telling you how is a waste of time...
23:41.36[TK]D-FenderxSmurf: But you knew that already...
23:41.58xSmurfno  it's not a waist of time, at least I learnt how it's done
23:42.30xSmurfmentioning just allows me to express that although this is a solution, if there are others I'd be interested in knowing them
23:42.34[TK]D-Fenderoh well
23:43.31xSmurfindeed
23:50.01*** join/#asterisk cnu (~cnu@the.ultimate.lamer.la)
23:52.11root52FWIW the problem I mentioned above has cleared itself up. all is well after i left the line from the telco unplugged for about two min. Odd but i'll take it. Perhaps the telco's switch got "confused" about what state the line was in and it just needed a min or two to "reset".
23:53.04*** part/#asterisk Pegasus_RPG (~chatzilla@p4FF90751.dip.t-dialin.net)
23:55.42fenrusit's not unheard of that sdh over different kinds of sdsl etc takes a minute to synchronize
23:56.18fenrusand just pure sdh/pdh aswell, have been waiting for minutes for loops in the sdh to appear in my routers when the transmission people have set them.

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