IRC log for #asterisk on 20100221

00:02.38*** join/#asterisk nickaugust (~anonymous@32.172.57.154)
00:03.05*** join/#asterisk jksM (jks@193.189.93.254)
00:05.55*** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file)
00:05.56*** mode/#asterisk [+o file] by ChanServ
00:20.15*** join/#asterisk moy (~moy@74.12.129.100)
00:22.28*** join/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au)
00:23.26thazzahey all. I have a problem with asterisk 1.6.1.12 and realtime with IAX setup.. My iax requires the requirecalltoken=no option. however I can't work out how to add to the mysql database
00:29.17Katty:>
00:29.21Kattyi have chili
00:30.04Kattyhomemade chili at that
00:30.52ChannelZand it burns burns burns
00:30.53thazzaNo takers on RealTime setup with IAX?
00:30.56ChannelZthe ring of fire
00:31.22KattyChannelZ: the chili is HOT
00:32.00Kattyi put 4 oz ogreen chilis into it
00:32.16*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
00:33.01jayteedid you use cumin? chipotle pepper?
00:33.27Kattycumin, chili powder, garlic salt, onion powder, seasoned salt, black pepper
00:33.57Kattyground turkey, 2 cans of chili ready tomatos, and a can of tomato sauce
00:34.03Kattyand chopped onion
00:34.32Kattyoh, and minced garlic....and cheese on the top at the end
00:35.10jayteegarlic salt? how bourgeois
00:35.24Kattybourgeois does not parse.
00:35.49jayteeah, the minced garlic on top is good but I'd have substituted the garlic salt with the minced IN the chili
00:35.59Kattynono
00:36.06Kattyyou saute the meat with garlic and onion
00:36.11jayteeok
00:36.23Kattythen add the cans of tomatos and tomato sauce
00:36.28Kattyadd in seasonings.
00:36.33Kattysimmer for 30 minutes
00:36.36jayteesounds like quicky mix
00:36.58Kattyit doesn't take any time at all to throw together
00:37.23Katty1 C of chili is 145 calories
00:38.15Kattythe original recipe calls for kidney or chili beans, but ryan doesn't like the beans...it also calls for 1/2 C brown sugar, but ryan doesn't like his chili sweet
00:38.41jayteemine takes a bit longer. I buy the ground beef ground coarse for chili and lightly brown it then add it to the chili stock. the stock is chili powder, tomato paste, crushed tomatoes, pinto beans, cumin, dash of cayenne and a tsp of ground chipotle pepper.
00:39.53Kattyi don't think i've ever seen chipotle pepper anywhere
00:40.14jayteeMcCormick spices sells it
00:40.32jayteeand ground Ancho too
00:40.35*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
00:41.06jayteeright now I've got a chicken marinading in a mango/chipotle marinade
00:41.45jayteeit's a liquid marinade but there's a company that makes a mango/chipotle spice rub mix that's pretty good too
00:46.14*** join/#asterisk twanny796 (~twanny@85.232.221.180)
00:46.52twanny796is there a web based software that manages asterisk installed on another computer?
00:47.10p3nguin_ssh
00:47.47twanny796p3nguin_: :)
00:47.55p3nguin_That's how normal people do it.
00:50.08*** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361)
00:56.26*** part/#asterisk korihor (~korihor@190.205.251.97)
00:57.00BadHorsieWhat's the meaning of "FMPR" and "FMGL" on a follow me?
00:57.11hipitihopI sometimes get "pbx.c:4390 __ast_pbx_run: Timeout, but no rule 't' in context 'phones' .. can someone point me at what this means
00:59.42Chainsawhipitihop: A timeout occured, and your dial plan has no extension for it to jump to. It can't handle the error as a result.
01:00.11*** part/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au)
01:03.10*** join/#asterisk moy (~moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
01:07.11hipitihopChainsaw, sorry what timed out ?
01:07.42Chainsawhipitihop: Likely a call in the phones context. Hard to say really, you're not giving me much to work with.
01:08.30p3nguin_Here's what I put it a lot of contexts:
01:08.32p3nguin_exten => t,1,Playback(vm-goodbye)
01:08.33p3nguin_exten => t,n,Hangup()
01:08.58p3nguin_Now if there is a timeout, the system says goodbye and hangs up.
01:09.05*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
01:09.27hipitihopp3nguin_, is that aimed at me ?
01:09.42p3nguin_Pretty much.
01:10.07*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
01:10.08hipitihopok thanks, will check my phones context add
01:10.10p3nguin_It wasn't a solution to your problem, since you haven't given any helpful details, but it is related to your timeout not having anything to do.
01:16.22hipitihopChainsaw, p3nguin_sorry about the sparse info... I'm ver new to * so I'm not sure what is relavant .. just wokring my way through the online book.
01:18.57*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
01:23.12*** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com)
01:26.43p3nguin_hipitihop: Typically, verbose output leading up to the questionable lines would be useful to see why such lines have been produced.  For SIP calling, sip debug output it usually requested.
01:27.36*** join/#asterisk capitan (~captain@76.91.206.32)
01:29.01*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
01:31.06hipitihopp3nguin_, understand... actually watching the verbose output actually tells me now that is is actually related to me trying to make an outgoing call
01:31.28capitanhello everyone!
01:31.31p3nguin_Nevertheless, a timeout occurred on that call.
01:31.39capitanso... simple question... the asterisk documentation on voip-info.org says macros can only have the s extension...
01:31.48p3nguin_are you sure?
01:32.08capitanis this still true? or do newer asterisk versions support other extensions on macros?
01:32.27p3nguin_I think I've seen some macros with other extensions.
01:33.11capitanthanks p3nguin_
01:33.19capitanmaybe i can explain my problem...
01:33.26capitani'm trying to use freepbx with my voip phone
01:33.50capitanbut the macro-user-callerid executes at h,1 as though it'
01:33.57capitans,22
01:34.13capitanand hangs up
01:34.25capitani'm curious whether i'm using a freepbx version that's too new for my asterisk version
01:34.55p3nguin_No clue.  We don't do FreePBX here.
01:35.00p3nguin_~freepbx
01:35.01infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
01:35.14capitanright :S sorry...
01:35.30capitani figured i would ask generic asterisk questions here to help me with my (clearly freepbx) issue...
01:35.31p3nguin_As far as Asterisk is concerned, I think the macro is usually s, but can be something else if you explicity choose to use something else.
01:36.19p3nguin_It's a FreePBX issue, because Asterisk doesn't have the [macro-user-callerid] context until FreePBX adds it.
01:36.47*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
01:37.05p3nguin_I believe I have exactly one macro context in my extensions.conf, and it does use the s extension.
01:37.23capitanhehe
01:38.19capitanright... my question is too generic of a dialplan question to ask there, and too specific of a freepbx question to ask here :(
01:38.51capitani guess i could write my own macro outside of freepbx that exhibits this behavior and turn it into an asterisk specific question :P
01:39.34p3nguin_Macros are being deprecated in favor of Goto, if I understood the conversations correctly.
01:40.08capitanreally?
01:40.37capitanbut then you have to manage your own "call stack" so to speak, no?
01:41.25p3nguin_I'm not sure what you mean.
01:41.50capitani mean... how do you get back to where you were?
01:42.10p3nguin_Oh, maybe it was a GoSub and Return that it was being deprecated in favor of.
01:42.16capitanah
01:42.31p3nguin_I believe I misspoke, and I'm glad you asked what you did to make me realize it.
01:42.33capitanmakes sense... never understood the respective pros/cons of the two
01:42.56p3nguin_<@leifmadsen> GoSub() is preferred over Macro()
01:43.14capitanthat name looks very familiar :P
01:43.33p3nguin_<bmoraca_work> Orbixx, in 1.4, you're looking for Macros...in 1.6, check out "gosub"
01:43.33capitani'll take that as an authoritative answer ;)
01:43.44p3nguin_<ManxPower-work> dlynes_laptop: use a gosub instead of a macro if using 1.6
01:43.55capitanhehe... p3nguin_... what are you an archive? :P
01:44.09p3nguin_Yep.
01:44.59capitanhmmmm... still all quiet in #freepbx :(
01:45.04p3nguin_I use 1.4, and I still use GoSub()/Return().
01:45.14capitanis there a #freepbx on other irc nets?
01:45.21p3nguin_I doubt it.
01:45.33capitanya looks like everyone's on freenode these days
01:45.40capitanit's the "in" place to be ;)
01:46.08p3nguin_It's certainly _a_ place to be... don't know much more than that.
01:50.52*** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir)
01:53.49capitandarn... close... but not close enough :( https://issues.asterisk.org/view.php?id=14122
01:58.50*** join/#asterisk fakhir__ (fakhir@unaffiliated/fakhir)
02:02.06*** join/#asterisk d-k-t (~D@112.202.232.46)
02:06.52*** join/#asterisk dkirker-openmo-1 (~dkirker@openmobl/ceo/dkirker)
02:09.44*** join/#asterisk fakhir__ (fakhir@unaffiliated/fakhir)
02:14.46*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
02:17.40*** join/#asterisk fakhir__ (fakhir@unaffiliated/fakhir)
02:19.38capitanp3nguin_, thanks for all your help... but do you know what happens if you define a macro twice?
02:53.15*** join/#asterisk ELBunce (~erik@kde/developer/bunce)
03:00.51*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
03:00.58*** part/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
03:08.14hipitihopin my sip.conf is it possible to have one entry for my sip provider account for both inbound and outbound calls ?
03:08.35p3nguin_Yes, and that's ordinarily how you would configure it.
03:08.50p3nguin_Rarely do you need to break it apart into in and out.
03:09.30hipitihopp3nguin_, so if it works for incoming and it is type=friend then it should work for outgoing
03:09.59p3nguin_Probably, but change it to type=peer.
03:10.30hipitihopdoesn't that mean I can only use it for incoming ?
03:10.34p3nguin_nope
03:10.49p3nguin_Only type=user is restricted to one-way calls.
03:14.33hipitihopok, and so what is the typical outgoing entry in extension.conf
03:15.32p3nguin_exten => NXXNXXXXXX,1,Dial(SIP/${EXTEN}@youritsppeer)
03:15.46p3nguin_That's obviously for a 10-digit outgoing number.
03:15.58p3nguin_NANP
03:17.17hipitihopso no credntials are specified appart from the provider domain
03:18.24p3nguin_That depends.
03:19.23p3nguin_If you are registering dynamically, you'll also be providing a username and secret in your peer definition.
03:19.42p3nguin_If you are statically registered, you might not be sending the credentials.
03:20.08p3nguin_It also depends on if their system requires authentication on each call.
03:20.11hipitihopI register via a register entry in my [general] section
03:20.25*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
03:21.04hipitihopdoes that mean I don't dynamically register ?
03:21.14p3nguin_You'll probably want to use a username and secret in your peer definition unless someone has specifically told you not to.
03:21.30p3nguin_Using a register statement is for dynamic registration.
03:22.07p3nguin_That is how you tell your ITSP where you are and where to send calls when it receives a call on your DID.
03:22.08hipitihopok.. I guess that makes sense and means my system regiters to take inbound calls.
03:22.49p3nguin_Without the register statement, they would not know where you are and how to get a call to you.
03:22.55hipitihopso that also means, I probably need credentials specified in the outgoing context
03:23.14p3nguin_No, you put the credentials in the sip peer definition.
03:23.27p3nguin_What ITSP are you using?
03:23.39hipitihopgotalk in Australia
03:26.46p3nguin_I'm not seeing Asterisk configuration samples on their site, so we'll have to do it a different way.
03:27.17hipitihopthis is what I get using your extension sample http://pastebin.ca/1804374
03:27.57p3nguin_Put it in pastebin.com if you would like for me to view it.
03:28.09p3nguin_I still can't figure out the fascination with .ca
03:28.59hipitihopsorry, only used it because someone here asked me to another time so was just assuming it was this channels preference :-)
03:30.04hipitihopok, here http://pastebin.com/m1b4ea87
03:30.14p3nguin_I can't figure out why my browsers won't like it unless I use its IP address.  It's not a DNS issue, because I can dig or host lookup the IP address... but that's just a bother to do all the time.
03:31.08p3nguin_It looks like gotalk didn't know how to call 101.
03:31.16p3nguin_I hope you understand the problem with that.
03:31.55p3nguin_Let's try it a little different way.  One moment while I write up some rules.
03:31.56hipitihopI get the same regardless of the number I try and call
03:33.10p3nguin_Give me an example of a phone number that you dial to reach someone when calling them on POTS/PSTN.
03:34.04*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
03:34.46p3nguin_03 7010 5678 ?
03:36.07hipitihopyes, although don't know if that is a valid number... also mobile calls might start with 0402xxxxxx
03:37.11hipitihopwould the dialplan string from my ATA which had no problems making callings prior to asterisk help ?
03:38.37p3nguin_http://pastebin.com/d37a08915
03:39.31p3nguin_This assumes your sip peer is listed as [gotalk]
03:40.02p3nguin_[gotalk] will contain type=peer along with your username and secret.
03:40.14hipitihopaah, maybe that's where I am going wrong... my peer in the sip conf is [mysipuserid]
03:40.46p3nguin_It can be an arbitrary string, but it should make sense to you or anyone who you want to understand it.
03:41.46hipitihopbut your point is that whatever string is in the exten => section .... @blah must be [blah] in the sip peer definition
03:42.22p3nguin_If you want the Dial() command to use that peer and the credentials configured within it, yes.
03:42.35hipitihopseems to obvious ;-)
03:46.49p3nguin_http://pastebin.com/d278bf445
03:48.36p3nguin_That incoming line sends calls to a SIP device with the name of 100.  It's just an example, since your phone probably isn't 100.
03:49.59hipitihopyes I think I follow that... my exsisting one is an [internal] context and it dails my ATA defined as friend in sip.conf
03:50.27hipitihopexten => s,1,Dial(SIP/LinksysPAP)
03:50.59p3nguin_The ATA should also be type=peer.
03:51.46hipitihopeven though the handset off it will initiate most calls ?
03:51.55*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
03:51.58p3nguin_I don't see how that is relevant.
03:52.00hipitihopbut as you said, peer does allow both directions
03:53.45p3nguin_The difference from a call standpoint is how the phone is authenticated.  user/friend is by username, peer is by IP address and port.
03:54.21p3nguin_The different in a programming standpoint is that friend creates two items in memory.
03:57.35p3nguin_And you won't want to use the 's' extension because a phone number will be called.  The s extension is for calls without an explicit number being called.
03:58.09p3nguin_exten => 100,1,Dial(SIP/LinksysPAP)
03:58.31p3nguin_Now your extension is 100.  Your device is still LinksysPAP.
03:59.46p3nguin_This is a perfect example of how an "extension" is not a phone, like so many people try to say.
04:02.32*** join/#asterisk correcaminos (~luis.agui@201.201.46.106)
04:03.02*** join/#asterisk correcaminos (~laguilar@201.201.46.106)
04:05.44hipitihopp3nguin_, seems like we are getting close, no errors but I can't hear anything.
04:06.28p3nguin_Is your system behind NAT?
04:06.59hipitihopsure but had no problem earlier when incoming was working so I haven't changed anything
04:07.08p3nguin_Oh.
04:07.22p3nguin_So incoming calls worked earlier?
04:07.36hipitihopand router is setup to forward anything on 5060 back to my * box
04:07.43p3nguin_They had two-way audio, also?
04:08.00hipitihopyes incoming worked and two way audio was perfect.
04:08.16p3nguin_You also have to forward the RTP ports.  But if you had audio, I guess you did that already.
04:08.42p3nguin_What's the status now?  Incoming calls don't work?
04:08.48hipitihopI just took a call from my partner on my mobile and she said she can't call in .. so I have broken it :-)
04:09.05hipitihopno I had done nothing previously with the rtp ports.
04:09.13p3nguin_hmm
04:09.30hipitihopso I'm going to have to go back to basics and review the recent changes
04:09.44p3nguin_How could audio work if the RTP ports aren't forwarded?
04:10.01hipitihopI'm just going to flick the ATA back to talking direct with my provider
04:11.27hipitihopp3nguin_, agree with you, if forwarding the rtp ports, I assume you are talking about that 20000 range, then I have not done so and I swear the incoming test did work fine
04:12.03p3nguin_10000-20000 should be default.  You can reduce it down if you aren't going to have 5000 simultaneous calls.
04:12.46hipitihopare thes udp tcp or both ?
04:12.51p3nguin_all UDP
04:13.05p3nguin_Just forward the same ports as what you have configured in rtp.conf.
04:14.20p3nguin_Since you are behind NAT, you should have also configured nat= localnet= and externip= (or externhost=) in sip.conf.
04:16.40hipitihophmm ok, as I said, all broken at the moment I'll have to backtrack and then apply what you have tought me... I really do appreciate your time
04:17.15hipitihop..sorry but have to pop out for 30 min or so
04:17.45p3nguin_No problem.  Hopefully we'll be able to get it all worked out pretty soon.
04:18.09hipitihopwhat timezone are you in p3nguin_
04:18.17p3nguin_CST
04:18.33p3nguin_(USA)
04:18.50p3nguin_GMT -6
04:19.09hipitihopso getting late for you
04:19.18p3nguin_22:19
04:19.53hipitihopok back in 30
04:20.20*** join/#asterisk skrite (~skrite@69.55.26.25)
04:20.53skritehey all
04:21.28*** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk)
04:24.14ChannelZok I am simultaneously fascinated and frustrated by watching curling
04:24.37*** join/#asterisk smooth_penguin (~smoove@59.95.10.208)
04:24.55kc8pxyChannelZ:  frustrated?
04:26.07ChannelZI have no idea what is going on
04:26.47kc8pxyChannelZ:  iirc it's something between bowling and darts.
04:27.25p3nguin_I was thinking like bowling and shuffleboard, but I can kinda see darts.
04:27.26ChannelZLike a GB team just went and pretty much landed a rock right in the center.. but as they were doing it, one of the guys from the OTHER team (CAN) was brushing the ice right as it was hitting the center
04:27.37*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
04:28.01p3nguin_He was probably trying to polish the ice so it would keep going all the way through the circle.
04:29.00ChannelZso the other team is allowed to interfere?
04:29.12p3nguin_To an extent, I guess.
04:30.58ChannelZI see now
04:31.03ChannelZis reading a bit on wikipedia
04:31.21ChannelZThe other team can only jump in once the stone crosses the line through the middle
04:32.07ChannelZI also thought the sweeping was to rough up the ice and slow the thing down but apparently it's the opposite
04:32.11p3nguin_And the sending team cannot cross that line, right?
04:32.34ChannelZWell it says once it crosses the line only one person can brush, not both
04:35.37p3nguin_Both guys brushed that one the whole way down.
04:38.54p3nguin_It's pretty neat how they make it curl and change its path.
04:39.10*** join/#asterisk toddejohnson (~toddejohn@ppp-70-226-192-134.dsl.spfdil.ameritech.net)
04:43.03Kattywanders in
04:44.14jayteehi
04:44.37Kattyhi jaytee
04:45.30eppigyHELLO
04:45.32eppigyI AM DAVE
04:45.40jayteeyes you are!
04:45.48jaytee:-)
04:45.49Kattyare you sure
04:45.56Kattyare you sure you're not... Bob
04:45.59Kattyor Mary
04:48.22p3nguin_or a name less common, like Waldo?
04:48.40*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rhrfdrafohqnlaug)
04:49.54p3nguin_hipitihop: Time's up!  If you aren't here, you're late.
04:50.04hipitihop:-) just arrived
04:50.20p3nguin_Damn, you said 30 minutes and you meant it.
04:51.58hipitihopok testing inbound I get this trace NOTICE[2373]: chan_sip.c:19546 handle_request_invite: Call from '09xxxxx' to extension 's' rejected because extension not found.
04:52.15p3nguin_Okay, your ITSP doesn't know to send your phone number to you.
04:52.30p3nguin_You can probably remedy that by adding it to the register statement.
04:52.56p3nguin_register => username:secret@sip.gotalk.com/yourphonenumber
04:53.09hipitihoptrying...
04:53.43p3nguin_That's how I solved it when sipgate used to send no phone number to me.
04:53.45hipitihopalthough not sure to include area codes and or country prefix
04:54.05p3nguin_You don't know your DID number?
04:54.31hipitihopnot sure what DID is .. I know my userid and I know my phone number.
04:54.58p3nguin_Your DID is the phone number that someone would dial from the PSTN to reach you.
04:55.24p3nguin_It typically includes the area code, but not the country code.
04:55.41hipitihopok, however should I include country prefix also e..fg Aus = 61
04:55.53hipitihopsorry, ok
04:56.24p3nguin_It annoys me when ITSPs don't know what phone number they should be sending.
04:58.12hipitihopok that has changed somehting.. now I get NOTICE[2373]: chan_sip.c:17269 handle_response_invite: Failed to authenticate on INVITE to '"04xxxx"
04:58.12*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
04:58.42p3nguin_Okay, uncomment that line in your peer that says insecure=
04:58.47*** join/#asterisk moy (~moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
04:59.04hipitihopalready uncommented, has incesure=invite
04:59.13Kattyholy friggin snickderdoodles batman, it's 11
04:59.46p3nguin_It is already insecure=invite and still produced that message?
05:00.38hipitihopp3nguin_, yes if we are talking about sip.conf and my gotalk pper section
05:02.58hipitihopp3nguin_, although I have not followed your pastebin to the letter, still tweaking with my original so I hope that is causing confusion
05:03.08hipitihop^ not
05:05.51p3nguin_While I did provide the paste as a guideline, it should work if you used it copy/paste into your conf.
05:09.11*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
05:12.07hipitihopp3nguin_, my current one http://pastebin.com/m2fd79cae
05:13.15p3nguin_How do you plan to call "LinksysPAP" from a phone?
05:13.24p3nguin_I mean, how do you intend to dial it?
05:14.30hipitihopnot sure what you mean... I want in general, any call coming in from gotalk to be routed to the ATA so it rings and we pickup
05:14.46p3nguin_http://pastebin.com/d5c46dfe8
05:15.10hipitihopsorry if I'm being a complete newbie, I'm sure the penny will drop soon
05:16.10p3nguin_You cannot pick up your phone and dial "LinksysPAP" on it.
05:16.19p3nguin_So don't make an extension called "LinksysPAP".
05:16.58p3nguin_Dial(SIP/LinksysPAP)  makes the call go to the ATA.
05:16.59hipitihopis that what the internal thing does ?
05:17.17p3nguin_See my post.  I corrected yours.
05:18.25p3nguin_At this point, the internal context is irrelevant.  But I want you to understand that no one can pick up their phone and dial  L i n k s y s P A P  on it.
05:19.30p3nguin_They could press those keys, but it would come out as 5465797727.
05:19.34hipitihopyou mean from a logicl pov ? iow, if I was using a softphone, could I not treat the ATA with a ahndset on it as a normal phone/extension ?
05:20.09p3nguin_So it would need to be exten => 5465797727,1,Dial(SIP/LinksysPAP) if that's what your intention was.
05:20.32p3nguin_phones are not extensions.
05:21.00p3nguin_Extensions are RULES on what to do when Asterisk receives the phone numbers or letters that are being sent for a call.
05:21.57p3nguin_exten => 5465797727,1,Dial(SIP/LinksysPAP)   <-- This says if you dial 5465797727 on a phone, it will place a call to the SIP device by the name of LinksysPAP.
05:22.07p3nguin_The extension is 5465797727.
05:22.14p3nguin_The device is LinksysPAP.
05:22.16hipitihopindeed... I just because I have these things in my config does not mean it is an intention :-) these configs are a combination of samples, parts from THE book etc .. my only intention at the moment is to be able to take and make calls from my ATA via my sip account
05:23.00p3nguin_If you don't understand this basic facts, you won't be making and receiving calls.
05:23.08p3nguin_s/this/these/
05:23.37p3nguin_I have given you the dialplan, but for some reason you choose to ignore it.
05:24.26jayteekinda reminds me of jmcdowell
05:24.45p3nguin_Heh, yeah.  Did you know he put tk on ignore?
05:25.02hipitihopp3nguin_, I'm not ignoring it and I don't intend to offend either.. I'm jsut trying to understand as I go instead of blindly pasting
05:25.04jayteehahaha, nothing like cutting off your nose to spite your face
05:28.28p3nguin_hipitihop: Maybe I can go over it one more time to help you understand it.  LinksysPAP is your device's name, as configured in sip.conf (at least that's what you told me).  That is not your extension, it is your device.  Extensions are not devices.  exten => 5465797727,1,Dial(SIP/LinksysPAP)   creates extension 5465797727 and makes it dial your device if 5465797727 is called.
05:29.22p3nguin_I doubt that you really want someone to have to press in 5465797727 from another phone on your system, though, so I choose to create extension 100 to dial your device instead.
05:29.31*** join/#asterisk uqlev (~yuriy@91.184.221.31)
05:29.49p3nguin_http://pastebin.com/d5c46dfe8
05:30.05jayteehipitihop, make sure you carefully and fully read Chapter 5 in the book if you want to understand your dialplan, contexts, etc.
05:30.48jayteeit will help you avoid errors moving forward when you want to do more advanced call processing
05:31.10p3nguin_Extensions are just rules to say what to do when numbers (or letters) are received.
05:32.32p3nguin_Asterisk receives a call to 2123, it knows to call my phone because of this rule:  exten => 2123,1,Dial(SIP/mydevice,48)
05:32.45p3nguin_My extension is 2123, my device is "mydevice".
05:33.17p3nguin_And it is using chan_sip to reach it, as designated by the "SIP/" part of the Dial() command.
05:33.35hipitihopyup that makes sense now
05:34.06*** join/#asterisk graphix (gfx@38.111.17.107)
05:34.42p3nguin_exten => _X.,1,Dial()   <-- this extension is a pattern match, matching ANY two or more numbers.
05:34.48hipitihopok dial plan 100% as per yours... still getting invite error though
05:34.59p3nguin_Of course, that is solved in sip.conf.
05:35.17p3nguin_You still have insecure=port,invite ?
05:35.44hipitihopno have insecure=invite
05:35.54p3nguin_Try port,invite and see what happens.
05:36.08p3nguin_I also included that in my paste earlier.
05:37.34hipitihopahh my bad totally missed the sip.conf part of that post , standby, matching
05:45.42hipitihopp3nguin_, ok some progress
05:46.23hipitihopp3nguin_, I can call out and I get audio, at least the remote end..getting to aother number's voice mail.
05:46.58p3nguin_Your outgoing call works and has two-way audio?
05:47.21hipitihopp3nguin_, I'll paste the results of the incoming which seemed to now answer but my ATA did not ring and I couldn't hear anything
05:48.13p3nguin_I'm going to need to see your sip.conf, your extensions.conf, and the debug output if you want me ot solve the problem.
05:50.36hipitihopp3nguin_,  hold on just trying to get to an outside number wher eI can confirm I have both way.. so far onyl getting mailboxes so can confirm I can hear that end
05:53.14hipitihopp3nguin_, confirmed, two way at least to a mobile/cell works and good clear audio both ways.
05:54.04p3nguin_Okay, that's good.  So you're using exten => _X.,1,Dial(SIP/${EXTEN}@gotalk) for outgoing?
05:54.28hipitihopyes
05:54.56p3nguin_Depending on the different ways you can dial and depending on what gotalk will accept, you WILL have to refine that later.
05:55.09hipitihopsure
05:55.35hipitihopnow, incoming exten => _X.,1,Dial(SIP/LinksysPAP)
05:55.49p3nguin_I have at least three different outgoing extens for 7, 10, and 11-digit dialing for outgoing calls.
05:56.48*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
05:56.56p3nguin_such as  exten => NXXXXXX,1,Dial(SIP/1312${EXTEN}@voipms)
05:56.59hipitihopnow tried incoming again and same, I get a trace in the cli with ringing and answered ... but as mentioned it doesn't ring the ATA so stanby and prep another pastebin
05:57.22p3nguin_and  exten => NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
05:57.43*** part/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
05:57.43p3nguin_err, forgot my underscores in there.
05:58.27p3nguin_I didn't forget them in the pastes I made for you, though.
05:59.55*** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net)
06:00.23hipitihopsorry I'm so slow.. working with my remote * box so only have nano over ssh at the moment so copy pasting is a bit of a multi-part thing
06:01.31p3nguin_You could always use pastebinit.
06:02.02p3nguin_It's tricky when you want to hide your passwords, though.  You have to make sure you hide them before pasting.
06:02.55hipitihopthanks I'll make a note to look at that.. obviously I don't want to just paste real sip.conf becauxse all secrets are there
06:03.19p3nguin_That's a pastebin tool for the command line, if you didn't know.
06:07.11hipitihopp3nguin_, http://pastebin.com/m18847c6b
06:07.25ChannelZwe need an asterisk-aware pastebin that will automatically greek passwords and IPs
06:07.49p3nguin_Not IPs, that makes troubleshooting a network-related issue difficult.
06:08.07ChannelZyeah but sometimes people are pussies about it and want to change them
06:08.09hipitihopso from what I understand, I suspect my problem is related to the context defined in LinksysPAP, the context is "phones" however the internal is currently commented out in phones.
06:08.43p3nguin_hipitihop: You aren't making any call from the device to other internal devices, so it is irrelevant.
06:09.43p3nguin_Where's your gotalk peer definition?
06:09.59*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
06:10.00hipitihopleft it out as it is as per your spec
06:10.20*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
06:10.23p3nguin_And if you're natted like you told me, where's your nat, localnet, and externip or externhost?
06:10.55hipitihop<shrug> I need that in [general] in sip.conf ?
06:11.04*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
06:11.14p3nguin_Every time you paste me incomplete configs, it's wasting more time that we could be using to get this fixed.
06:11.43p3nguin_It's after 00:00 now, so someone else will have to take over soon if you don't hurry up.
06:11.54hipitihopp3nguin_, sorry, tried to save time as the peer laso had many secrets, standby.
06:12.35p3nguin_This is seriously a 10 minute job.  We've been discussing it for hours so far.
06:13.43hipitihophttp://pastebin.com/m4df4436f
06:14.05p3nguin_context=incoming-calls
06:14.08p3nguin_[incoming_calls]
06:14.16p3nguin_These do not coincide.
06:14.33p3nguin_They are supposed to.
06:15.02hipitihopeek, indeed, typo
06:15.03p3nguin_And you still don't have any nat info.
06:15.24p3nguin_If you are natted, you HAVE TO HAVE the nat stuff configured.
06:15.27p3nguin_~sipnat
06:15.27infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
06:16.03p3nguin_nat=  localnet=  externip= (or externhost= )
06:16.30p3nguin_They go in the general section.
06:21.09p3nguin_Actually, there are a few other settings I would configure, as well.
06:23.20hipitihopp3nguin_, http://pastebin.com/m58d5a972
06:25.25p3nguin_That outgoing call there at the bottom.  It was successful?
06:25.28*** join/#asterisk CrashHD (~CrashHD@c-76-114-27-96.hsd1.ca.comcast.net)
06:25.48hipitihopthat's an incoming attempt from my mobile to my sip number
06:26.16hipitihopsame number as we appended to the registration
06:26.17p3nguin_You still haven't corrected the context.
06:26.22p3nguin_context=incoming-calls
06:26.27p3nguin_IT is wrong.
06:26.40hipitihopsorry, didn't repaste the new one.. yes corrected to use _
06:27.31p3nguin_"sip reload"  and  "dialplan reload"  and then show me the entire failed incoming call.
06:29.03p3nguin_So far, that piece of a call was not an incoming call.
06:30.18hipitihopsame as before.
06:30.32hipitihopare you saying this looks like an outgoing call ?
06:30.37p3nguin_I don't know what that means.  Show me something useful.
06:30.49p3nguin_<PROTECTED>
06:30.52p3nguin_<PROTECTED>
06:30.56p3nguin_That's an outgoing call.
06:31.05idespinnerhipitihop, i looked at the pastebin , looks like an outgoing call to me aswell
06:31.08p3nguin_Show me something useful for a failed incmoing call.
06:32.15*** join/#asterisk albasheers (~basheer@89.148.43.51)
06:32.22*** part/#asterisk albasheers (~basheer@89.148.43.51)
06:32.56p3nguin_If what you put in that pastebin contained your entire [general] section of sip.conf, I would be surprised if anything works.
06:33.00hipitihophttp://pastebin.com/m14a1446
06:33.25p3nguin_That's still an incoming call.
06:33.42p3nguin_Maybe you don't know how to enable sip debuggin?
06:33.54p3nguin_Err, still not an incoming call.
06:34.20hipitihopp3nguin_, are we talking cross purposed ? ... I am trying to call my asterisk from a mobile phone, so I would expect it to come up as an incoming call
06:34.22p3nguin_sip set debug  or  sip set debug on  (depending on asterisk version).
06:34.35hipitihopI'm on 1.6
06:34.49p3nguin_That's what I'm talking about.  Incoming... from the PSTN to your DID number.
06:35.02p3nguin_What you showed me is an outgoing call.
06:35.38hipitihopp3nguin_, it is what comes up in teh cli when I attempt to call in from my mobile.... standby enabling debug
06:35.42idespinnerhipitihop, could you do a 'sip show peer gotalk' ?
06:36.36p3nguin_I'm still wondering if sip reload wasn't issued after making changes.
06:37.38idespinnerp3nguin_, thats what i'm thinking aswell, a sip show peer might unconver this...
06:37.42p3nguin_yep
06:37.43hipitihopidespinner, http://pastebin.com/m2e78a1e5
06:37.57p3nguin_<PROTECTED>
06:38.07hipitihopp3nguin_, I reloaded sip and dialplan
06:38.09p3nguin_So that little tidbit wasn't an incoming call.
06:39.33idespinnerhipitihop, looks good there, could you also do a 'sip show peer LinksysPAP' ?
06:39.45hipitihopis feeling very guilty, after wasting p3nguin's time and now has two people engaged on an trivial excercise
06:40.09p3nguin_I'm still waiting on a sip debug from a failed incoming call.
06:41.15hipitihopidespinner, http://pastebin.com/m6142c13d
06:41.21hipitihopp3nguin_, getting to that now
06:43.38idespinnerhipitihop, yea, that second sip show peer looks good aswell. definatley need a full sip debug of you calling into your gotalk number with your cellphone
06:44.08hipitihopyup got the debug, pages of it, sanitising now
06:48.20hipitihophttp://pastebin.com/m53291615 and I hope it's clean enough
06:49.14p3nguin_Looking for 07xxxxxxxx in phones (domain 203.xxx.xxx.xxx)
06:49.21p3nguin_I don't get it.
06:49.47idespinneryea...
06:49.54p3nguin_I want you to show me your ENTIRE sip.conf.
06:50.06p3nguin_Stop leaving pieces out.  The whole thing is needed.
06:50.23idespinnerother context's could be conflicting
06:50.31p3nguin_By the way, this is the last time I will be asking for it.
06:50.43p3nguin_I think a dozen times is plenty.
06:51.50hipitihopI thought I had done that in previous pastebins but standby... I understand it's late and I'm frustrating you.
06:52.22p3nguin_I have five ITSPs configured on my system, and I don't have this same issue with any of them.
06:55.57hipitihophttp://pastebin.com/m485d1aa4
06:56.10*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
06:58.11idespinnerthe only thing I can think of is that the IP address of gotalk is being registered as 2 seperate peers, and one of them has the context of [phones]...
06:58.38idespinneryou're not registered to gotalk twice are you?
06:59.00p3nguin_sip show registry?
06:59.18idespinnermaybe... thats a good thought...
06:59.41hipitihopsip.gotalk.com:5060            N      myuserid           105 Registered           Sun, 21 Feb 2010 16:57:53
07:00.01idespinnersip show peers?
07:00.11hipitihopjust checking my laptop to make sure nothing is logged in either
07:02.06hipitihopp3nguin_, as I mentioned quite some time back, I had no problem with incoming, the ATA would ring and I had audie for both ends... I started today trying to solve outgoing. now we seem to have reversed the situation
07:04.07hipitihopbut I'm no where near comfortable enough with all this to know which changes have had what effect... certainly appending my did to the registration stopped the need for 's' and only other main things is we switched a couple of things from friends to peers
07:04.42idespinnerhipitihop, basicaly the asterisk cli logs you post for incoming is not what I would expect as the incoming call is hitting the [phones] or [go-talk] context instead of the [incoming-calls] context....
07:05.39idespinnerso from what you posted here: http://pastebin.com/m58d5a972 it looks like SIP/myuserid-dc0f6198 is dialing SIP/07xxxxxxxx@gotalk
07:05.47p3nguin_yep
07:05.49p3nguin_exactly
07:06.06idespinneris  SIP/myuserid-dc0f6198 your PAP2 or is it gotalk or is it anything you can identify?
07:06.32idespinnerim assuming you scrubbed 'myuserid' in there
07:06.44hipitihopyes, that is my real gotalk userid
07:07.13hipitihopso wherever you see myuserid it is a replacment for the real gotalk one
07:07.49idespinnerok, so for whatever reason, incoming calls from gotalk are hitting you outgoing context(phones or gotalk), despite that your context set in sip.conf for gotalk is set to incoming-calls
07:08.11hipitihopif you look on my scrubbed sip.conf the registration ... has the same number but I called it 'userid' in the registration
07:09.57*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
07:10.35hipitihopalthough note the context has been fixed in http://pastebin.com/m485d1aa4
07:11.20idespinnerhipitihop, if we removed the denis context completley and did a sip reload...
07:11.30idespinnerim wondering if that resolves the issue...
07:11.39*** join/#asterisk albasheers (~basheer@89.148.43.51)
07:11.56idespinnerits a long shot, but i dont have to much else to build on
07:13.46hipitihopnope
07:14.33idespinneris the asterisk cli different?
07:15.17*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
07:15.46hipitihoptrying to find original debug to compare standby
07:15.48*** join/#asterisk oej (~olle@ns.webway.se)
07:16.18*** part/#asterisk albasheers (~basheer@89.148.43.51)
07:17.53p3nguin_Here's a working sip.conf with one ITSP and one phone, behind NAT: http://pastebin.com/d53d2f344
07:18.25*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)
07:22.36*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
07:22.56hipitihopp3nguin_, do you want to match that as best possisble ? or do you want me to take it as is and substitute my ip's, userid etc.
07:27.25hipitihopto my untrained eye the debug and end result certainly the same
07:35.23hipitihopok p3nguin_and idespinner, sorry to take up so much of your time.. can't thank you enough at your attempts to help
07:36.04hipitihopI'll switch my ATA back to direct sip provider for now so I can have normal calls again
07:39.25hipitihopis there a general asterisk cli command to stop all sip related activity so it's not trying to register or receive calls at the same time as my ATA
07:40.43p3nguin_module unload chan_sip
07:44.24hipitihopok, thanks again for help, have to run out to pickup from airport ... sorry to have been so much of a pita
07:45.02p3nguin_Here's a working sip.conf with two ITSPs and one phone, behind NAT: http://pastebin.com/d4bcd86c1
07:45.53*** join/#asterisk ELBunce (~erik@kde/developer/bunce)
07:48.01hipitihopyes I looked at that p3nguin thanks.... when I return I will try and apply as much from that as possisble obviously with my own ip's and userids etc
08:03.18*** join/#asterisk techie (~root@unaffiliated/techie)
08:04.15*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
08:08.44*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
08:39.52*** join/#asterisk ELBunce (~erik@kde/developer/bunce)
09:10.44*** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com)
09:18.17*** join/#asterisk Ad-Hoc (~nimbus@62.1.231.190.dsl.dyn.forthnet.gr)
09:22.18*** join/#asterisk ChannelZ (channelz@burner.com)
09:22.22*** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl)
09:22.45*** join/#asterisk ChannelZ (channelz@burner.com)
09:41.23FSB_1How can I make asterisk play a dial tone?
09:41.48FSB_1I am trying to make it so that if I press a double 0 I will be able to make outbound calls.
09:42.04ChannelZcore show application PlayTones
09:42.14ChannelZoh
09:42.18ChannelZwell it doesn't really work like that
09:42.48*** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk)
09:43.02ChannelZI suppose you could set extension 00 to do PlayTones to give you a phony dialtone, and then WaitExten for the number
09:43.45FSB_1Am I thinking right if I use extension 00 to goto another context which is allowed to make outbound calls?
09:43.48ChannelZbut ideally you'd want to stop the dialtone after someone presses the first digit
09:44.16ChannelZWell why do you want to make people do this 00 business anyway?
09:44.40*** join/#asterisk Ad-Hoc (~nimbus@62.1.231.141.dsl.dyn.forthnet.gr)
09:45.33FSB_1It's a common thing to do in sweden.
09:47.16*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
09:47.16*** mode/#asterisk [+o putnopvut] by ChanServ
09:47.43*** join/#asterisk outtolunc (~richard@c-98-248-96-110.hsd1.ca.comcast.net)
09:48.55ChannelZare these sip phones or analog phones?
09:49.54FSB_1Right now I'm just playing around, but it's a common thing to have the telephone switches configured like this.
09:50.45ChannelZwe have sort of a similar thing in the US for business PBXs, to dial 9 for an outside line
09:51.12ChannelZon SIP phones though you can usually program the phone's dialplan to do this.. so that when you pick up and dial 9, the phone it's self makes a new dialtone
09:56.10*** join/#asterisk Igramul (~black@p5497CD30.dip.t-dialin.net)
09:59.14*** join/#asterisk vk4akp (~Ken@c114-77-251-186.ipswc3.qld.optusnet.com.au)
09:59.23vk4akpHi guys.
09:59.31vk4akpAnyone around?
10:00.03vk4akpI have a really different / strange question realating to Asterisk.
10:03.22*** join/#asterisk smooth_penguin (~smoove@59.95.12.115)
10:13.00vk4akpAll these names and no one here?
10:13.46coppiceyou might find asking a question is more productive
10:16.18tzafrirFSB_1, this is a rather simple thing to do: basically a context with WaitExten
10:16.33tzafrirTechnically it is a sort of an IVR with Asterisk
10:16.56tzafrir(same implementation: basic dialplan)
10:17.06vk4akpAh
10:17.08vk4akpPeople.
10:17.10vk4akp:)
10:17.22tzafririnfobot, tell vk4akp about ask
10:17.30tzafrir(bots as well)
10:18.07vk4akpOK. We are trying to get a Asterisk MeetMe conference running on an install that is on a virtual server provided by a hosting company.
10:18.32vk4akpThe issue is Ztdummy.
10:18.41vk4akpIs there a way to do the timing?
10:18.48tzafrirHave you tried using DAHDI instead? Specifically: latest DAHDI?
10:18.54vk4akpBecause I'm not sure if the normal kernal way will work on a virtual server.
10:19.07tzafrir(Enable core timing)
10:19.15vk4akpI haven't tried anything yet. Looking for advice on the best way to proceede.
10:19.27tzafrirWhat version of Asterisk is it?
10:19.35vk4akpOK. So I will looki into dahdi.
10:19.41vk4akpHang ten I'll look for the version.
10:20.33vk4akp1.2.27
10:22.09vk4akpLOL. INfobot says you are all here against your will. LOL
10:22.16vk4akpSomeone should proof read their bot. hahahah.
10:23.10*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
10:29.54*** join/#asterisk Tim_Toady (~moi@77.49.178.169.dsl.dyn.forthnet.gr)
10:33.39*** join/#asterisk smooth_penguin (~smoove@59.95.29.6)
10:34.38vk4akpOK. So more info needed.
10:34.51vk4akpwill Dahdi run on the 1.2.27 release?
10:35.19vk4akpIs ther ea link or direction that explains where to get and how to install Dahdi dummy? (Name?)
10:37.50vk4akpHumm. Looks like it might be called Dahdi_dummy ???
10:43.34*** join/#asterisk oej (~olle@ns.webway.se)
10:44.21*** join/#asterisk oej (~olle@ns.webway.se)
10:44.47vk4akpHey guys any answer?
10:45.05vk4akpNeed to get some info so I can request to move forward with this.
10:46.33tzafrirvk4akp, no. That version only works with Zaptel. But why do you use it?
10:46.51vk4akpI don't.
10:46.53vk4akpI run 1.4
10:47.02vk4akpBut this is another box on the other side of the world.
10:47.16vk4akpIt has an old install of Gentoo and Asterisk on it.
10:47.21vk4akpNot my box basically.
10:48.13vk4akpBut I have to add a conference on it for DonkeyBallz
10:48.13vk4akpSo the answer is that I have to request to update to Asterisk 1.4.22 or higher yes?
10:49.20tzafrirvk4akp, if you want dahdi
10:49.28vk4akpOK
10:49.36tzafrirbut before you do that: do you have a problem with building kernel modules?
10:50.02vk4akpAnd do you know for sure that Dahdi_Dummy in this (enable core timing) mode will work on a virtually hosted box?
10:50.24vk4akpHumm. Good question. This is my first time dealing with a virtually hosted box.
10:50.59tzafrirvk4akp, do you currently have dahdi running?
10:51.09vk4akpno
10:51.21vk4akpit's 1.2.27 remember.
10:51.36tzafriryou can try building (but not installing) dahdi-linux and dahdi-tools
10:52.09vk4akpwhat will that do?
10:52.49vk4akpWhat if I emerge them?
10:52.56tzafrirIf you enable core timing in include/dahdi/dahdi_config.h , you'll only need to insmod dahdi.ko itself . Otherwise, you'll also need to insmod dahdi_dummy
10:53.20vk4akpOK Thats good info thanks.
10:53.24tzafrira very basic test that it works, even without using (or building) dahdi-tools:
10:53.25vk4akpLet me doco that b4 I move on.
10:54.23tzafrirtime head -c8000 /dev/dahdi/pseudo >/dev/null
10:54.37vk4akpIs Dahdi 2.2.02 new enough?
10:54.38tzafrirThis should give you no errors, and should take ~1 second
10:55.04tzafririf it gives an error or takes a time that is much different: something is wrong with the timing
10:55.20tzafrirI would prefer 2.2.1,
10:55.32tzafrirany special reason for not using it?
10:56.00vk4akpOK. I can wget and hand compile a newer one but 2.2.0.2 is available now in portage. If that is likely to work it's very quick for me to install it.
10:56.12tzafrir(note that what I suggest to you is to download the source tarball, build it, but *not* install anything from it)
10:56.38vk4akpOK. 2.2.1 it is then. Umm. LInk for SVN or somethign please?
10:57.15tzafrirhttp://downloads.asterisk.org/pub/telephony/dahdi-linux/
10:57.18tzafriror svn:
10:57.45tzafrirsvn co http://svn.asterisk.org/svn/dahdi/linux/branches/2.2/
10:57.51vk4akptnx
11:10.35vk4akp<PROTECTED>
11:10.43vk4akpIs this th eline I uncomment before doing a make?
11:12.36*** join/#asterisk pawz (~pawz@ppp118-208-171-146.lns20.bne4.internode.on.net)
11:18.04*** join/#asterisk pawz (~pawz@ppp118-208-171-146.lns20.bne4.internode.on.net)
11:18.50tzafrirvk4akp, yes, just uncomment that
11:19.10tzafrir(I hope to eventually make that enabled by default and do away with dahdi dummy)
11:19.23vk4akpOK
11:19.29vk4akpYep done that and a make.
11:19.49vk4akpNOw I am a bit queezy about doing a insmod on a virtually hosted box. This won't kill anythign will it?
11:22.41vk4akpinsmod: error inserting 'dahdi.ko': -1 Invalid module format
11:25.41vk4akpSo do I need to do a make install now instead and then insmod dahdi_dummy instead?
11:26.37*** part/#asterisk war9407 (war@liquidswords.org)
11:32.05*** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001)
11:34.11*** join/#asterisk smooth_penguin (~smoove@59.95.51.58)
11:42.09tzafrirvk4akp, what is the error you actually see in the kernel logs?
11:42.38vk4akpOMG. I wouldn't even know where to look for that.
11:42.50tzafrirdmesg | tail
11:42.59vk4akpcan't I just insmod dahdi_dummy ?
11:43.00tzafriror the logs in /var/log
11:43.01vk4akpOK Looking.
11:43.15*** join/#asterisk pawz (~pawz@ppp118-208-171-146.lns20.bne4.internode.on.net)
11:43.18tzafrirno. If you insmod, you have to resolve dependencies yourself
11:44.24vk4akpdahdi: disagrees about version of symbol struct_module
11:51.16*** join/#asterisk moy (~moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
11:53.18*** join/#asterisk sun28 (~light@sun28.ipfw.su)
11:54.03*** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de)
11:54.24*** join/#asterisk `Sauron (sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
11:55.11*** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it)
12:00.35*** join/#asterisk puzzled (~patrick@535335AA.cable.casema.nl)
12:01.17vk4akpAh. Damn I know. The Kernel source revision doesn't match teh installed revision. :(
12:01.19vk4akpGrrr.
12:04.11*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
12:05.54*** join/#asterisk angryuser (~angryuser@34-156-167-83.reverse.alphalink.fr)
12:05.58*** join/#asterisk pa (~pa@unaffiliated/pa)
12:06.31*** join/#asterisk fnordus (~dnall@70.70.0.215)
12:06.39*** join/#asterisk pawz (~pawz@ppp118-208-171-146.lns20.bne4.internode.on.net)
12:08.00vk4akpThe kernel version is 2.6.16.29.xs3.1.0.289.2650
12:08.15vk4akpAnd I think that's a virtual kernel provided by the hosting company.
12:13.14*** join/#asterisk StuZZZs (~stuart@rabbit.dbplc.com)
12:14.04*** join/#asterisk darkskiez (~mhb@cpc4-broo7-2-0-cust263.know.cable.virginmedia.com)
12:15.53*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
12:16.39tzafrirvk4akp, hmm... so you can't run your own kernel?
12:17.00vk4akpMM. That is somethign I will have to look into and ask.
12:17.22vk4akpBecasue of the time difference the people who pay for the box are asleep right now.
12:17.27*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
12:17.35vk4akpBut I have given them all the information and what part we are up to.
12:17.52vk4akpI googled that kernel revision number and find it is a vmware kernel.
12:17.59vk4akpSo the plot thickens. :)
12:18.09vk4akpIt will be interesting to see if ther eis a work around for this.
12:18.23vk4akpI guess it is of reasonable iportance to Asterisk as well.
12:18.37vk4akpAs it could mean that virtually hosted box's can't run a aconference. :(
12:21.19*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)
12:23.04*** join/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au)
12:23.19thazzaHey All.
12:36.41Amorsenvk4akp: You're unlikely to get anywhere with 1.2.27 + vmware + meetme
12:37.09vk4akpIs there any way to create a timing source for meetme conference with out having access to the kernel source?
12:37.26vk4akpI'm fairly sure they will let me update teh asterisk to 1.4
12:37.54vk4akpBut I think the kernel source could be an issue.
12:38.05AmorsenMeetme is dependent on dahdi_dummy, not just for timing
12:38.44vk4akpOK so the stumbling block for now is the VM kernel + source etc.
12:39.40AmorsenWell I'll be impressed if you get good enough timing from a vmware guest that it will work decently, but I suppose it isn't completely impossible
12:39.41*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
12:40.36AmorsenYou could try a test with 1.6.2 and app_conference, that would remove the need for dahdi
12:41.11tzafrirvk4akp, rather: either use app_conference (an external app) or upgrade to 1.6.2 and use app_confbridge
12:41.20AmorsenSorry, app_confbridge
12:41.40vk4akpOK. Interesting.
12:41.46vk4akpI will put that to them also. TNX>
12:42.24*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
12:44.14vk4akpOK. Thanks for your help all tonight.
12:44.21vk4akpI think this ends my options for this evening.
12:44.29vk4akpAll the Best!
12:55.11*** join/#asterisk Akiraa (~Akiraaaa@79.112.14.141)
12:59.08*** join/#asterisk oej (~olle@ns.webway.se)
13:07.09*** join/#asterisk nightrid3r (kvirc@41.214.225.88)
13:13.25*** join/#asterisk tuxx- (~tuxx@nightshade.nl)
13:13.31tuxx-o-hai.
13:16.33*** join/#asterisk nightrid3r (kvirc@41.214.225.88)
13:17.03*** join/#asterisk nightrid3r (kvirc@41.214.225.88)
13:22.58*** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net)
13:26.25*** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk)
13:27.39Kobazhaha
13:27.40KobazKatty:
13:27.51KobazKatty: chatroulette got a story on slashdot
13:28.06KobazKatty: via the new york times no less
13:39.08*** join/#asterisk albasheers (~basheer@188.116.235.226)
13:40.55*** join/#asterisk |AnToS| (~31749@93-44-103-101.ip96.fastwebnet.it)
14:09.31*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
14:10.13*** part/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au)
14:10.43*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
14:25.58*** part/#asterisk albasheers (~basheer@188.116.235.226)
14:42.52*** join/#asterisk catojo (~catojo@189.24.48.139)
14:43.29*** join/#asterisk rommer-droid (~rommer-dr@kiev.netrom.com)
14:58.00*** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net)
15:02.42*** join/#asterisk nightrid3r (kvirc@41.214.204.88)
15:03.16*** join/#asterisk nightrid3r (kvirc@41.214.204.88)
15:05.37*** join/#asterisk jaytee (~jforde@unaffiliated/jaytee)
15:11.23*** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk)
15:15.55*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
15:24.10*** join/#asterisk slima (slima@unaffiliated/slima)
15:40.16*** join/#asterisk [netman] (~netman@40.Red-88-17-244.dynamicIP.rima-tde.net)
15:42.21*** join/#asterisk ProperPHB (~4858eb78@gateway/web/freenode/x-mbgsklkykvbmhkjo)
15:42.27ProperPHB~savemoney
15:42.28infobot<Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards.
15:42.51ProperPHBI'm proud of my money saving prowess. ;)
15:43.18ProperPHBSorry I haven't been here in a while. Our debt collection call center got burned down by some arsons.
15:45.49*** join/#asterisk ELBunce (~erik@kde/developer/bunce)
15:48.50*** join/#asterisk GNUtoo (~GNUtoo@host49-13-dynamic.54-79-r.retail.telecomitalia.it)
15:50.16*** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com)
15:52.47ProperPHBBy the way, I'm Gremlin.
15:53.10ProperPHBI would be here as Gremlin, but the post it note on my monitor with my passwords was lost in the fire.
15:54.14*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
15:54.51Pan3Dsorry about the fire, I need to make marshmallows
15:55.25tzafrir_laptopProperPHB, arson? Are you sure it's not your soldering iron?
15:57.16ProperPHBWell, we're working out of my boss's apartment now doing boiler room cold calling.
15:57.34tzafrir_laptopProperPHB, anyway, can't you ask the sysops to mail you a new password or something?
15:57.46ProperPHBThe SIP trunk is doing not so well on residential DSL. :)
15:57.58ProperPHBPerhaps.
15:58.46*** join/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net)
16:00.09nightrid3rProperPHB: /nickserv help setpass
16:02.30nightrid3rhmmm can't find out how to mail key
16:04.49voipmonkhe said sysops
16:05.13voipmonkyour age is showing
16:05.15voipmonk:)
16:06.22Kobazheh
16:07.05*** part/#asterisk GNUtoo (~GNUtoo@host49-13-dynamic.54-79-r.retail.telecomitalia.it)
16:07.09*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:07.09*** mode/#asterisk [+o leifmadsen] by ChanServ
16:07.17jayteemorning leif
16:07.24leifmadsenmorning
16:15.20*** join/#asterisk nix8n82 (~AndChat@63.162.27.14)
16:17.47*** join/#asterisk atis_work (~atis_work@193.238.212.171)
16:18.36*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
16:18.59*** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com)
16:25.25*** join/#asterisk iq (~chatzilla@unaffiliated/iq)
16:25.27iqHi
16:26.26*** join/#asterisk d-k-t-2 (~D@112.202.232.46)
16:39.41*** join/#asterisk oej (~olle@ns.webway.se)
16:43.37*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
16:45.01*** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com)
16:45.17*** join/#asterisk korihor (~korihor@190.205.251.97)
16:48.12*** join/#asterisk Alagar (~Administr@122.164.36.202)
16:55.28*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
17:00.06*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
17:03.33p3nguin_fsb_1: You want DISA for the double 0 and an getting dialtone.  (in case no one told you.)
17:05.06*** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org)
17:06.29*** join/#asterisk brezular (~brezular@adsl-dyn181.78-99-155.t-com.sk)
17:07.40*** join/#asterisk bn-7bc (bjarne@2001:16d8:ee6c:0:219:e3ff:fe3b:d34a)
17:10.53*** part/#asterisk bn-7bc (bjarne@2001:16d8:ee6c:0:219:e3ff:fe3b:d34a)
17:11.57*** join/#asterisk bn-7bc (bjarne@2001:16d8:ee6c:0:219:e3ff:fe3b:d34a)
17:12.25*** join/#asterisk d-k-t-3 (~D@112.202.232.46)
17:15.46*** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net)
17:16.01*** join/#asterisk ariel_ (~ariel_@c-24-127-196-248.hsd1.fl.comcast.net)
17:16.47*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)
17:18.15ariel_hello everyone
17:20.49*** join/#asterisk oej (~olle@ns.webway.se)
17:24.51*** join/#asterisk smooth_penguin (~smoove@59.95.23.181)
17:24.53*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
17:27.53FSB_1p3nguin_: Thanks,
17:28.13FSB_1p3nguin_: But when trying out DISA I get a busytone as soon as I try to dial anything.
17:28.50FSB_1Do I need WaitExten()?
17:29.07tzafrir_laptopFSB_1, you need WaitExten . You don't really need DISA
17:29.43p3nguin_fsb_1: Optionally, use pattern matching for your outgoing context, where the extension is something like this:  exten => _00XXXXXXXX,1,Dial(SIP/${EXTEN:2}@itsppeer)
17:29.49tzafrir_laptopJust send the user over to a different contenxt where WaitExten is used
17:30.17p3nguin_You really shouldn't be trying to construct some illogical dialplan mechanism with WaitExten().
17:31.11Kobazheh
17:31.35p3nguin_You don't really need DISA if you use the pattern matching method, but DISA will provide that second "false" dialtone.
17:33.52ChainsawOn the subject of tones, is there no way to set the cadence for the ringing tone?
17:34.09Chainsaw(My FXS gateway does it correctly, Asterisk always seems to produce an american single ring)
17:34.34p3nguin_indications.conf
17:34.44*** join/#asterisk oej (~olle@ns.webway.se)
17:35.16carrarYou need to conform to the American ring tone
17:35.24carrarotherwise we will invade your country
17:35.43p3nguin_We might do it anyway.
17:35.50Chainsawp3nguin_: That has ringcadence for the physical bell (callee), not the confirmation tone (caller).
17:37.04p3nguin_ring won't set what you want to set?
17:37.28p3nguin_for example:  ring = 413+438/400,0/200,413+438/400,0/2000
17:37.45Chainsawp3nguin_: No, the physical ring is correct but the confirmation tone (*please* note the difference here, it is quite important) is continuous, american-style.
17:38.11p3nguin_ringcadence is the physical bell, ring is the RINGING SOUND.
17:38.24p3nguin_Ringing()
17:38.28p3nguin_PlayTones(ring)
17:38.35FSB_1Ahhh
17:38.36FSB_1Fuck it
17:38.45FSB_1I'll just put my self in a special context.
17:38.57carrarYou are special
17:39.07carrar[shortbus]
17:39.18FSB_1waits for the trolling to commence.
17:40.00p3nguin_I almost spit tea on the monitor over "special context" being [shortbus].
17:41.09p3nguin_chainsaw: Which locale of ringer sound do you want?  (I don't know where you are)
17:41.11carrarWhat kind of tea!
17:41.16Chainsawp3nguin_: uk
17:41.36Chainsawp3nguin_: Which is a double short ring.
17:41.47Chainsawp3nguin_: Not a continuous ring.
17:41.48p3nguin_Plain ole Lipton black tea (probably with orange pekoe).
17:41.49*** join/#asterisk smooth_penguin (~smoove@59.95.58.54)
17:42.48p3nguin_chainsaw: Did you set country=uk in the general section of indications.conf?
17:42.54Chainsawp3nguin_: Yes, I did.
17:43.29AmorsenIf you're using dahdi, you also need to set it somewhere in its config
17:43.40AmorsenIIRC
17:43.41ChainsawAmorsen: This is through SIP, but thanks.
17:43.43p3nguin_ring = 400+450/400,0/200,400+450/400,0/2000  does it for me.
17:44.06Chainsawp3nguin_: Ah, I think I know what happened here.
17:44.20Chainsawp3nguin_: I think I have an old indications.conf on a newer Asterisk. (Because I don't see that ring setting anywhere)
17:44.26p3nguin_oh
17:44.33Chainsawp3nguin_: That would do it, wouldn't it. Sorry about that.
17:44.48*** join/#asterisk rocksfrow (~kyle@pool-71-179-183-143.bltmmd.fios.verizon.net)
17:45.12p3nguin_I bet it confused people calling your system and getting a US ring tone.
17:46.37rocksfrowdoes anybody use the rhino failover cards?
17:46.49rocksfrowor anything comparable,
17:47.01rocksfrowor maybe the fonebridges?
17:49.18*** join/#asterisk Rajmohan (~raj@122.165.25.171)
17:51.03Kobazrocksfrow: i use WTI a/b switches
17:51.36rocksfrow??redundancy
17:51.39rocksfrownotta? lol
17:51.41rocksfrowhrm..
17:51.46p3nguin_What is a "notta?"
17:51.57rocksfrow??redundancy didn't return anything
17:51.57rocksfrowlol
17:52.01rocksfrownotta
17:52.07p3nguin_I don't know what a notta is.
17:52.09rocksfrownot a thing
17:52.15p3nguin_hmm
17:52.21Kobazhttp://www.urbandictionary.com/define.php?term=notta
17:52.36rocksfrowhey there you go, look at definition 1
17:52.37rocksfrowll
17:52.43rocksfrow"not a thing"
17:52.58Kattyhi
17:53.02rocksfrowanyway...
17:53.07p3nguin_Maybe someone got confused with the Spanish word "nada," meaning nothing.
17:53.28Kattybmoraca_work: ping
17:53.34rocksfrowp3nguin_, it is what it is
17:53.57Kobazhttp://www.bicomsystems.com/products/C/P/797/255_2797/  who makes that server they use
17:54.18Kobazlooks like NEC is stamped on the bottom right
17:54.24smooth_penguinhi Katty
17:54.29Kattyhi smooth_penguin
17:54.47p3nguin_It does say NEC.
17:55.06smooth_penguinwell NEC just assembles
17:55.10smooth_penguinand resells
17:55.15rocksfrowKobaz, do you use the single card?
17:55.18p3nguin_http://www.nec.com/global/prod/express/
17:55.26Kobazhttp://www.necam.com/Servers/FT/
17:55.39Kobazlooks cool
17:55.57rocksfrowKobaz, http://www.wti.com/AFS-Series/AFS-RJ45-Channel-Card.html ?
17:55.57Kobazrocksfrow: two servers, one t1 card in each server
17:56.23Kobazrocksfrow: that's not for a pc
17:56.31Kobazrocksfrow: that's for their blade-style a/b switches
17:56.39rocksfrowoh so you have the entire encolusure?
17:56.56rocksfrowyeah..
17:57.03Kobazhmm
17:57.13Kobazthey make a three a/b that's 1u
17:57.16Kobazwhere did it go
17:57.27rocksfrowah..
17:57.31rocksfrowyeah that'd be sweet
17:57.37rocksfrowdiscontinued?
17:57.44p3nguin_WTF... Banquet TV dinners are barely even a snack, now.  How lame.
17:58.01Kobazhopefully not, it's an amazing product
17:58.04Kobaztelnet interface and everything
17:58.05Nuggettelnet is eeeeeeevil!
17:58.06Kattyrecommends hormeal completes
17:58.13Kobazauto switchover via voltage feed
17:58.37Kobazhttp://www.wti.com/AB-Data-Switches/PLS-345-Physical-Layer-Switch.html
17:58.42Kobazmaybe they stopped making it?
17:58.59rocksfrowkobaz, have you ever checked out the rhinos?
17:59.06Kobazrocksfrow: i've looked
17:59.12Kobazi've had such bad experiences with rhino
17:59.17Kobazthat i haven't considered buying one yet
17:59.19rocksfrowoh really?
17:59.26Kobazthey make really shitty t1 cards
17:59.30rocksfrowhrm..i'm also considering the fonebridge2
17:59.34Kobazbad firmware
17:59.47rocksfrowwhat t1 cards do you use?
17:59.49Kobazspent 5 months trying to get a dual span working
17:59.51rocksfrowi have dig
17:59.53Kobazsangoma
18:00.36Kobazafter i switched to sangoma cards, about 2 months later i get an email saying they found the problem and it was bad dsp code... "here try this firmware"
18:00.41Kobazno thanks... sangoma works out of the box
18:01.01*** join/#asterisk mog (~mog@c-71-228-185-24.hsd1.al.comcast.net)
18:01.01*** mode/#asterisk [+o mog] by ChanServ
18:01.14Kobazthey were good about buying back all my extra rhino hardware though
18:01.28Kobazhad about $15k in rhino boards
18:01.42rocksfrowdamn
18:02.04Kobazthey were working fine for a while... and then i bought a new card for a project... ran out of dual spans
18:02.26Kobazi've had other problems with them too... bad dtmf recognitions, dialtone problems with analog lines, pickup problems with analog lines
18:02.48rocksfrownow this is with their t1 cards right, not the actual failover unit
18:02.57Kobazand then since it took that long to debug the problem with the dual span, it's like... wtf is gonna happen if a different card has problems
18:03.06rocksfrowright
18:03.08Kobazyeah, i haven't used the failover unit
18:03.12Kobazit probably works fine
18:03.30Kobazso far i've had zero problems with sangoma cards...they are amazing
18:03.35rocksfrowso how does the failover unit interface
18:03.39Kobazi thought i had a problem with one, but it turned out to be asterisk
18:03.45rocksfrowit looks like the rhino is dependent on the main server
18:03.49Kobazit's just a simple relay
18:03.59Kobazit takes power from the host computer
18:04.06rocksfrowoh only power
18:04.06rocksfrow?
18:04.11Kobazbut if it loses power, it switches to the failover
18:04.24rocksfrowokay that makes sense
18:04.40rocksfrowthe rhino also has a 2pin attachment to auto-restart the master
18:04.43rocksfrowon failure..i like that
18:04.49*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
18:06.04Kobazthe rhino board is also much cheaper than the wti switch... which is $700
18:06.46rocksfrowyyyyyyyyyyeah.
18:06.51rocksfrowdamn key stick
18:09.23Kattygrumps
18:11.43rocksfrowi'm interested to hear if anybody has success using the fonebridge2
18:14.25*** join/#asterisk correcaminos (~laguilar@201.201.46.106)
18:17.13*** join/#asterisk atis_work (~atis_work@193.238.212.171)
18:18.30Fulg0reis it possible to do faxes over asterisk/voip too?
18:18.35p3nguin_yes
18:18.50Fulg0reis it hard to setup?
18:19.01eppigyTRANAJO
18:19.03p3nguin_It is not recommended to do fax over voice over IP, but it does work most of the time.
18:19.31p3nguin_It is no more difficult to set up than any other configuration.
18:20.00Fulg0reit is better just to get a analog line?
18:20.11p3nguin_For reliablity purposes, yes.
18:20.34p3nguin_If you don't care about that and just want to experiment, go ahead and set up faxing on Asterisk.
18:20.44p3nguin_Make sure you use ulaw for your codec.
18:21.08Fulg0rei'd have to go to my local telco to get the analog line right? can't cut them out?
18:21.19p3nguin_or alaw if you aren't in North America.
18:26.59*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)
18:27.41*** join/#asterisk dobry (~d@95.111.7.95)
18:33.28Fulg0reif i forward a voip number internationally....would there be international charges?
18:35.00p3nguin_I guess that depends on exactly what you mean.
18:39.40Fulg0rebasically its so someone living in a different country can have a local north american number
18:39.53Fulg0rebut he wants it on his cell phone on not tied down to a internet line
18:40.01p3nguin_Yes, but you did not define "forward."
18:40.27Fulg0reoh, well im not sure the best way to do that so i said "forward" :)
18:41.29*** join/#asterisk ahall (~dentist@shell.ev6.net)
18:41.43p3nguin_If you have a US DID on your Asterisk system in the US, and calls to his DID Dial() to his cell phone number in another country, then you will pay for international calling termination rates.
18:42.57*** join/#asterisk iq (~iq@unaffiliated/iq)
18:43.02p3nguin_The only free calling would be by SIP URI.
18:44.24Fulg0resip uri?
18:45.18Fulg0reand would those call to his did to his cell phone in his country be over voip or regular ptsn lines?
18:45.19*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)
18:45.33p3nguin_Depends on how you configure things.
18:46.07Fulg0rewhat would be a good way to configure it to avoid long distance charges?
18:46.35p3nguin_SIP URI dialing is like  Dial(SIP/1234@your.domain.com)  where the call is only SIP and goes directly to a host by the name of your.domain.com.
18:46.58Fulg0reohhhh
18:48.35p3nguin_You could configure a VoIP gateway at his house... it would register to the ITSP in the US with his US DID (or to your Asterisk system), and calls to his device would route out his PSTN connection.
18:48.35Fulg0rebut that couldnt go to a cell phone?
18:49.00p3nguin_That configuration would be like if he picks up his home phone and dials his cell phone number.
18:49.46Fulg0rebut if he did that and called his local cell would it be a international call anyway?
18:50.09p3nguin_Is his house in the same location as his cell company?
18:50.28Fulg0reyes
18:50.37p3nguin_Then it would be a local call.
18:50.42Fulg0rehe currently has something like that...
18:51.18Fulg0rehe's got a vonage box with a us number...but if he calls his local cell number which is in india, using the vonage box its a international call
18:51.22p3nguin_Call to the US DID comes to the device.  The device calls from his house phone line to his cell phone.
18:51.44Fulg0reooooh, so the box would have to have another connection to his local telco?
18:51.53p3nguin_he doesn't have POTS/PSTN phone service?
18:52.20Fulg0reyes he has that too...but they are different lines
18:52.27p3nguin_The device would use his local telco to call his cell phone.
18:52.55p3nguin_The same as if he picked up his phone and dialed his cell phone.
18:52.58Fulg0rethe vonage box goes over his internet connection, and the pots line goes to his telco, but those 2 are not connected
18:53.13p3nguin_Forget the Vonage ATA.
18:53.24Fulg0rek
18:53.35p3nguin_Let's use a Linksys SPA-3102 as an example.
18:54.12Fulg0rek, bought up the product page
18:54.14p3nguin_Connect the 3102 between his telephone handset and the wall jack.  Also connect it to the internet.
18:55.03p3nguin_Configure the 3102 accordingly (more on that later).  Now if he picks up his phone and dials a number locally, he uses his local telco to connect the call.
18:55.33p3nguin_If someone calls his home phone number, he can also answer the phone when it rings in.
18:56.05p3nguin_With me so far?
18:56.30Fulg0reyep
18:56.50p3nguin_Now send a call to his device over the internet.
18:57.01p3nguin_The 3102 takes the VoIP call.
18:57.43p3nguin_The 3102 can be configured to ring the handset, as well as dial out to another phone number.
18:58.36Fulg0reand the other phone number can be his local cell
18:58.43p3nguin_exactly
18:59.12*** join/#asterisk ChrisWi (~admin@mx2.wwserver.net)
18:59.17p3nguin_So it would be the same as if he picked up his home phone handset and dialed his cell number via his local telco.
18:59.40Fulg0reso he would need internet connection, this spa3102, and local phone line for it to all work
18:59.49p3nguin_yes
19:00.15p3nguin_He already has two of the three.
19:01.00*** join/#asterisk Tim_Toady (~moi@77.49.236.7.dsl.dyn.forthnet.gr)
19:01.28Fulg0rejust talked to him, he doesnt have a home phone line
19:01.47p3nguin_Then you're going to end up paying international termination rates.
19:01.50Fulg0relet me see if he can get that set up though
19:02.06Fulg0reare international termination rates high?
19:02.29p3nguin_Depends.  What are the first 5 digits of his cell phone number?
19:02.51p3nguin_If I wanted to pick up my phone in the US and call him... the first 5 digits.
19:02.52Fulg0re01 91 98
19:02.57*** join/#asterisk uqlev (~yuriy@91.184.221.31)
19:03.04Fulg0rethan it would be 011 91 98
19:03.10p3nguin_One moment while I look it up.
19:04.43p3nguin_India Mobile, 9198 prefix, $0.03030000 USD per minute
19:04.57p3nguin_That's what it costs me using VoIP.ms services.
19:05.06p3nguin_3 cents per minute
19:05.18*** join/#asterisk Kamel (klo_028@c-76-123-106-217.hsd1.fl.comcast.net)
19:05.22Fulg0rethat doesnt sound bad
19:05.31Fulg0rei think
19:05.47p3nguin_You can call for a lot of minutes before you justify a local phone line at his house.
19:05.51Fulg0reso it would be the u.s. did cost, plus 3 cents a minute on top of that
19:05.56Kameli'm looking for a good american SIP provider, primary concern is price. any suggestions? (sorry, i've looked for an extended period of time and am coming up with nothing)
19:06.04Kamelunlimited preferably, only one line required
19:06.47p3nguin_fulg0re: Correct.  You can get an unlimited DID (actually limited to around 3000 minutes) for $6.95 per month.
19:07.21p3nguin_kamel: Check Flowroute, VoIP.ms, and CallCentric... in that order.
19:07.26Kameli find it difficult to spend $20/month when i can use skype for $60/yr, using a SIP proxy for skype is very unappealing to me :(
19:07.41Kamelp3nguin_: thanks a ton for your suggestions, means a lot to me
19:07.44p3nguin_I haven't spend $20 on my phone services in the past 5 months.
19:07.57p3nguin_together.
19:09.02Fulg0reis there some other configuration required to get the international termination and setup to get the call routed there?
19:09.04p3nguin_kamel: Do you need a phone number in a specific area code?
19:09.43p3nguin_fulg0re: Nope.  I just have to make sure international calling is enabled in my control panel, otherwise I got a circuit busy error.
19:10.54Kamelp3nguin_: well, i'd prefer to get one in my local area, but it's generally not difficult to come by. if necessary, i could always use a google number to route my calls to have a local area code
19:11.31p3nguin_kamel: You can get free incoming calls (a free DID) with three different companies that I know of, but you don't get to choose the area code.
19:12.12Kamelp3nguin_: does it still charge per minute?
19:12.31p3nguin_kamel: IPkall will give you a phone number in WA.  IPcomms will give you one in RI.  And sipgate does have a small selection, but you aren't likely to find one local to you.
19:12.45p3nguin_kamel: Those are free DIDs, with free inbound minutes.
19:13.00p3nguin_kamel: You can't make phone calls outbound with those free services.
19:13.32p3nguin_kamel: For outbound calling, you need to get termination services from either flowroute, voip.ms, or callcentric (other companies cost more).
19:14.23p3nguin_kamel: Flowroute charges $0.0098 per minute for outbound calls to US numbers.  VoIP.ms changes $0.0105 per minute for calls to US numbers.
19:14.27*** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net)
19:14.33p3nguin_charges
19:14.57p3nguin_CallCentric has an unlimited outbound calling package for around $20 per month.
19:16.08p3nguin_kamel: Do you primarily take incoming calls or make outgoing calls?
19:17.05Kamelhonestly a mix between the 2, but just slightly more outgoing i'd say
19:17.26Kameli don't use a lot though
19:17.26*** join/#asterisk atis_work (~atis_work@193.238.212.171)
19:18.13raden_workKamel, you can use skypw with asterisk if you need cheap :)
19:18.36Kameli think what i'm going to do is get a local landline with no features at all i think can be had for $13/monthish, then use voip for long distance
19:18.45p3nguin_I would probably sign up for a free DID phone number with IPcomms, then sign up with Flowroute for outgoing calling.  With this configuration, you'll spend under 1 cent per minute for outgoing calls and incoming is free.
19:19.07raden_workKamel, why not just get a unlimited inbound line for $6 a month VOIP ?
19:19.10Kamelraden_work: i was under the belief that you had to use a skype proxy which required to have a computer running the full skype client for it right?
19:19.32raden_workKamel, you using asterisk ?
19:19.37ChannelZno you can use chan_skype - $60
19:19.51raden_workChannelZ, thats $60 per channel correct  ?
19:19.58raden_workChannelZ, 1 time fee ?
19:20.03ChannelZyes
19:20.15Kamelraden_work: not yet, in the design phase right now, currently i only a skype ATA usb box and i'm hating it
19:20.16raden_workChannelZ, have you been using it ?
19:20.23ChannelZA little yes
19:20.31raden_workhow it been working ?
19:20.39raden_workKamel, what are your needs ?
19:20.57Kamelraden_work: not well if i'm honest, i hate the skype client for a large amount of reasons
19:21.04ChannelZWorks fine though I don't use Skype Out
19:21.36raden_workChannelZ, Skype out works great :)
19:21.59raden_workKamel, asterisk has a module for $60 per channel so you no longer need to use skype client
19:22.11p3nguin_kamel: In the configuration that I presented, you'll be able to make 1300 minutes of outbound calls for the same price as the phone line you're wanting to buy.
19:22.35Kamelraden_work: i'm currently just looking for a good home setup, but once i get everything set up am considering using it for some business needs, but overall very light even with the business requirements. faxing is important, and i'd love to be able to use 56k dialup as a backup internet service in the event that my ISP is down, but that doesn't really change my phone plans
19:22.50*** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net)
19:22.52Kameljust that i will probably end up with atleast a most basic incoming POTS phone line
19:22.56*** join/#asterisk TimeRider (~steve@78.32.26.1)
19:23.28raden_workKamel, # of lines , budget ? how many min a month
19:23.57raden_workand you should follow p3nguin_  's suggestions
19:24.05p3nguin_Suit yourself.  I'm trying to give you the best option for the least price.  If you make more than 1300 minutes of outbound calls, then your $13 phone line would be justified.
19:24.06raden_workand a 56k for backup ? whats the point ?
19:24.12raden_workfind a local wifi provider cheaper
19:24.44Kamel2 lines, preferably less than $6/month but probably an optimistic goal (i have no specific limitation on spending, just prefer not spend more), and i would say the minutes even on the busiest month will barely hit 1000 (inc+out combined)
19:24.50raden_workDSL or Cable as main | WIFI for failover asterisk SIP with p3guins suggestions or chanskype if you need cheap
19:25.25raden_workKamel, I have 14 channels / Lines  $22 a month and 4 channels are unlimited inbound
19:25.32Kamelp3nguin_: not arguing, just stating i'm going to probably be getting an additional basic POTS line, and only use it for faxing and dialin
19:25.49KingDavidNYCHi everybody, has anybody programmed in trixbox?, is it true that I can just write my code on the extensions_override.conf file and trixbox will use my code instead of the one on extensions.con?
19:25.55raden_workKamel, use email to fax service $5 a month
19:26.16raden_workKingDavidNYC, #trixbox
19:26.32KingDavidNYCraden_work: good idea
19:27.04raden_workKamel, what you are wanting todo is not difficult
19:29.43Kamelraden_work: yea, i'm aware, just looking for the best/cheapest option, did not know about the skype client for asterisk
19:29.59raden_workits a module not a client
19:30.11raden_workand its $60 per channel
19:30.54Kamelone time fee or annually or?
19:31.22p3nguin_For $60, I can have a year of regular non-skype calls.
19:33.00raden_workp3nguin_, I was just looking at that me to there rates are actually high compared to my wholesale rates
19:33.06ChannelZit's a one-time $60 to buy a license for the channel driver, then monthly through Skype at whatever rates
19:33.14Kameli see, that sounds like what i'd want... i am ok with using skype, i just don't like their pc client so if i could get away from that it sounds great
19:33.16raden_workKamel, one time buy you still need to pay skype out feee
19:33.27KamelChannelZ: ah, i see, that's not bad
19:33.32raden_workKamel, use callcentric
19:33.38raden_workget the office unlimited plan
19:33.51raden_work$9 a month 3 inbound channels unlimited incoming
19:34.06raden_workor vitelity
19:34.27*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
19:34.38raden_workKamel just do pay per call !!! no month fees
19:36.09Kamellol thanks for your help guys, honestly i'm pretty new to all of this so sorry if i haven't been making much sense to you, i'm still learning
19:36.45drmessanoSkype for Asterisk has been decent.. I havent paid for any of their PSTN services, just client <> client
19:37.07KingDavidNYCthat #trixbox room is so empty, looks like a funeral
19:37.31drmessanoKingDavidNYC, we've been noticing a nasty smell coming from over there for 2 years
19:38.10KingDavidNYClol, I wish I would keep away from it(trixbox) myself
19:38.25drmessanoSo do it
19:38.50Kameli currently use their unlimited in/out and am satisfied with it, i just hate the skype client and my current ata is extremely bad (to make a call, i must dial ##0011__________* where the #'s in the middle is the areacode+number i'd like to call and it's horribly annoying
19:38.51KingDavidNYCcant, the customer has a trixbox box
19:39.12Kamelalso breaks many features on the physical phone, like calling a missed call back from caller id
19:39.13Kameletc
19:41.06Kamelthinking i'm going for a cisco SPA3102
19:41.32drmessanoDo you have an analog line?
19:41.40Kameli will
19:42.00drmessanoWhy would you want to do that?
19:42.16Kamelbut like i was saying earlier, wont be used heavily for voice i'll be using primarily VOIP
19:42.21raden_workwow 21 people
19:42.23raden_work31
19:42.32Kamelto do what? get an analog line?
19:42.36drmessanoYes
19:42.57raden_workdrmessano, he makes no sense
19:43.20Kamellmao
19:43.26Kamelit does make sense
19:43.36raden_workKamel we saturate our DSL connection with 8 - 10 voice calls all day long no issues so I dont know why you would need a pots line
19:44.21Kameli want an analog line for a way to connect to the internet during down time, it's just a bare basic pots line, no frills at all, it's only used for data or as a backup if the internet is down
19:44.35drmessanoLOL
19:44.48drmessanoWhere are you located?
19:44.53Kamelin florida
19:45.06drmessanoHow is dialup internet even a viable backup anymore for anything?
19:45.15Kamelit's better than nothing
19:45.29drmessanoSave your money on the pots line and get service from another carrier if you need it that bad
19:45.34drmessanoNobody uses dialup for backup
19:45.40Kamellol
19:45.42ChannelZMaybe you just shouldn't have a computer or a phone
19:46.34drmessanoBy the time you get service from an ITSP and pay for the analog line, you can get a cheap backup line from the other provider around (or one of)
19:46.42Kameli don't see what's wrong with my requirements, internet service providers go down, if i bundle my dsl service with a basic bare pots line it is basically free anyway
19:47.02drmessanoIf the ISP goes down why do you think the SAME LINE will work?
19:47.36drmessanoRarely is there a pure data outage on a DSL line.. unless your provider sucks
19:48.16Kamelwell, you have a point there, but still if you're getting it for free, i don't see what the disadvantage would be
19:49.15drmessanoSo you get the line and the service for free?
19:49.40Kamelmy local provider is AT&T, if i get dsl 6 meg service with a home phone line, it's $24.95 and they have other promotions etc, if i get it direct, it's $47.95/month
19:50.13Kamelthe land line is $13/month for the basic (very) no frills phone line, no caller id, no voicemail, no long distance, etc
19:50.35drmessanoand they give you dialup service?
19:50.43Kamelso that's $37.95 with a phone and dsl or $47.95 without
19:50.49Kamelyes, 20 hours per month free
19:51.00Kamelover that it will cost, but i wont be going over 20 hours
19:51.01Kamellol
19:51.09Kamelunless the ISP blows, then i'd switch back to cable
19:51.21drmessanoDo you know what the hell you're doing at all?
19:51.24Kamelbut on cable right now and am spending $60/month for just the cable and it's an introductory thing
19:51.48Kamelof course, why do you ask?
19:51.49drmessano$24.95 is an introductory with AT&T
19:51.56Kamelthis is also introductory but it's for a whole year
19:52.03Kameli only have 1 month left on this plan
19:52.12ChannelZand then after a year he will be back trying to figure out how to get something else cheaper
19:52.16raden_workAT&T has DSL FOR $19.95
19:52.19Kamelthe AT&T one is for 1 year, specifically
19:52.33KamelChannelZ: lmao, yea....
19:52.49drmessanoNot sure who your cable is with, but I pay $60 for Comcast Business 6meg service
19:52.53Kamelraden_work: but 768kbit sucks :(
19:53.17Kameldrmessano: it is comcast, maybe there's a better plan but i don't have cable tv in my home and no desire to
19:53.24Kamelmaybe without tv service business is better
19:53.25ChannelZI pay $60 for comcast business for 12mbit/2mbit
19:53.26Kameli'm on res
19:53.28drmessanoThats without TV
19:53.48Kamelmine is 12/2 right now
19:53.50drmessanoChannelZ, same plan, they havent doubled the speed here yet
19:53.52Kamelbut technically it's 6
19:54.04ChannelZdrmessano: bitches!
19:54.07drmessanoI get 20/2 with speedboost
19:54.18Kamelit's 6 with a possibility of 12 depending on how saturated the network is according to what the rep told me
19:54.20drmessanoBut thats short lived
19:54.56Kamelhonestly having a large download isn't as important to me as having good up and down, which is a disadvantage to AT&T over comcast
19:55.40Kameli was interested in the business 20 meg plan, but the upload was still just 2 megs which is kinda pointless, i'd take a 5/5 line over a 20/2 one any day
19:56.11ChannelZyeah me too actually
19:56.29ChannelZthough Comcast has docsis 3 out here, they have a 100/10 plan for $200/mo
19:57.08p3nguin_I've tried to explain to Charter that some people don't give a shit about the download speed when they are asking for business service.  They wanted to convince me that 5/1 was "good."
19:57.12Kamelthey have docsis 3 here too, but i am unaware of a 100mbit package, they have a 50mbit package i know, but it's still only 5mbit upload @_@
19:57.40Kameli mean 5mbit is nice, but for $135/month they quoted me
19:57.42Kameland res
19:57.50Kamelwhich means it would have the 200gb limit still
19:58.00drmessanoI take that back, I think we did get the speed increased now
19:58.12Kamelerr, well 200-250, the rep explained 200gb was the soft cap where they started watching
19:58.14ChannelZyay!
19:58.22drmessanoI ran an entire speedtest at 2.5meg down
19:58.40drmessanoUsually its first 1MB at boost, then throttled to 1
19:58.44drmessanoSo thats different
19:58.52p3nguin_curl http://speedtest2.eastlink.ca/superlarge.bin > /dev/null
19:58.59*** join/#asterisk ManxPower-work (~EWieling@216.186.151.147)
19:59.11ManxPower-work~answers
19:59.12infobotrumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
19:59.57p3nguin_Only getting 2060k today.
20:00.20drmessanoI dont have curl.. Guess I will have to boot into Windows and use my GNUWin package :( SAD FACE
20:00.43p3nguin_wget would be okay, too.
20:01.09drmessanoSo would an "lol"
20:01.10p3nguin_You'll just end up with the file and have to delete it, where curl doesn't output a file unless you tell it to.
20:01.37Kamelwell, you guys may think i'm crazy for ideas, and you may be right, but thanks for helping me, i learned a lot by our convo. who knows, maybe i'll check into comcast business service and see if they could give me a flat rate at the current speed. i'd be ok with that, comcast is a decent isp, i just don't much care for their monitoring excersizes etc
20:02.06Kameli'm also very much so the type of person who will switch isp's out of principle, more people should be like that, then comcast wouldn't play their bs games
20:02.38drmessanoI use what works.. Principle costs too much
20:02.53Kattywheeeeeeeeeeeeeeeeeeeeeee
20:03.19Kamelnot when they end up sending you a bill for overage or somethind stupid like that
20:03.25drmessanoI guess when you have time to piss, and don't care what works, that's fine.. But last thing I need is to switch ISPs every 6 months because Betanews says Comcast or AT&T is more evil
20:03.38Kamellmao
20:03.43Kamelthat's true
20:03.52TJNIIProblem is you have lots of ISPs, but you still have to pay Comcast or Qwest for the actual wire to the building.
20:04.00Kameldrmessano: very curious, does comcast business block port 80 incoming?
20:04.07ChannelZno
20:04.12drmessanoComcast BUSINESS
20:04.12ChannelZor 25
20:04.16ChannelZor anything.
20:04.28KamelChannelZ: ah, nice, didn't know was always curious
20:04.33*** join/#asterisk albertoandrade (~albertoan@201.22.33.3.dynamic.adsl.gvt.net.br)
20:04.36drmessanoThat means you run a work thingo off of one
20:04.38drmessanoNot a home thingo
20:04.45ChannelZhonestly, I'm not sure why people would get comcast home anymore since the business price came down
20:04.59drmessano$59.95 includes gateway
20:05.02drmessanoCant beat it
20:05.03Kameldrmessano: i understand, but most businesses don't use their business connection to host their websites anyway
20:05.17Kameli wouldn't
20:05.19drmessanoKamel, I guess you never heard of MS Exchange
20:05.19ManxPower-workBecause most people are idiots and can't be bothered to learn enough to get the best value.
20:05.20ChannelZI mean home is cheap if you have cable TV through them too but unless you're just a non-tech web surfing family, it's barely worth it
20:05.22drmessanoor Webmail
20:05.32carrarWhats MS Exchange
20:05.39carrarSounds like a FAIL
20:05.40drmessanoExactly
20:05.57ManxPower-workcarrar: It's when you should exchange Microsoft for Linux.
20:06.09carrarhaha
20:06.14drmessanoPorts 80 and 25 are a drop dead requirement for many businesses
20:06.17Kameldrmessano: i guess, i'm not very business savvy i guess =\
20:06.24ManxPower-workdrmessano: but not for home users.
20:06.39carrarSendmail, Pine & DaviCal!!
20:06.42*** join/#asterisk iq (~iq@unaffiliated/iq)
20:06.45ManxPower-workIf more ISPs blocked port 25 the world would be a better place.
20:06.46p3nguin_If I'm going to pay for BUSINESS service, I'm going to run my BUSINESS stuff on it.
20:06.48drmessanoManxPower-work, except we were discussing Comcast BUSINESS service, but I guess I didnt use enough CAPS
20:07.04p3nguin_helps drmessano with more caps
20:07.19ManxPower-workdrmessano: actually someone asked why someone would get home instead of business, that's what I was respoinding to
20:07.40Kameland btw i do know what ms exchange and webmail are, just when i think of businesses i think of places that have 5 computers connected to 1 cable line through a horrible router with some idiot they overpay to be the "IT guy" and basically isn't even used as much as many residential lines
20:07.58Kamelperhaps my idea of businesses is incorrect
20:08.14Kamelunless a large business, which wouldn't use comcast as their isp
20:08.16drmessanoNot the couple hundred customers I used to support
20:08.27drmessanoLots of SBS out there
20:08.40Kamelshrugs
20:08.54Kamelclearly just not an area i know much about, what businesses use that is
20:09.08ManxPower-workAt least you have options.
20:09.32Kameli know what i use, but i never do things the way most places operate
20:09.42ManxPower-workAt me cabin, the options are dialup, HughesNet, or Verizon EVDO/Cellular
20:09.49Kamelfor better or worse lol
20:10.00drmessanoKamel, you're a home user.. that doesn't apply
20:10.02p3nguin_And you chose which one?
20:10.29ManxPower-workp3nguin_: Me?  Verizon EVDO
20:10.30carrarPacket over Smoke
20:10.39Kameldrmessano: you're right, but i do a lot of things which most home users dont, hosting sites from my home pc would be one of those things
20:10.40carrarwifi!
20:10.48carrarmicrowave
20:10.55drmessanoOver a DSL line?
20:11.01ManxPower-workcarrar: I've investigated them all.
20:11.06drmessanoYoure kidding, yah?
20:11.15p3nguin_Yes, over a Digital Subscribe Line line.
20:11.16ManxPower-workCheapest I could find was $600/month for T-1 internet.
20:11.35carrarI have a T1 here and two bonded DSL's
20:11.42carrarMe loves T1
20:11.49Kameldrmessano: me? no, i'm not kidding, but the sites i've hosted have been small and for specific purposes, not commercial or anything like that
20:11.57carrarand I don't care if it's only 1.544
20:12.11carrarI control both ends
20:12.19drmessanoKamel, so you're a hobbyist then
20:12.21p3nguin_1.544 is good enough for a few people at once.
20:12.34Kameldrmessano: yes, that may describe me well
20:13.02carrarplus there is free wifi all around this city
20:13.07drmessanokamel, hobbyists annoy me.. Home user mentality, IBM aspirations
20:13.32ManxPower-workdrmessano: don't forget "the budget of a homless person"
20:13.37drmessanoHAHAHHA
20:13.42Kameldrmessano: well, i do have ambitions of doing some business, but on a small scale. my wife is a nail tech and would like to start a salon for example, which i may do something like create a simple website from my home connection for
20:14.03drmessanoKamel, or you could just have it hosted for THREE dollars a month
20:14.08Kamelmostly just to avoid paying for hosting services
20:14.39drmessanoI should have ducked out of this a long time ago.. Hour of my life I will never get back :(
20:14.46p3nguin_Who does it for $3/mo?
20:14.48Kameldrmessano: i guess, i was just giving example
20:14.49ChannelZcarrar: how does the bonded dsl work?  I'm looking at switching telecom companies for my office because Qwest sucks, and the new company has a 25/2 bonded DSL product
20:15.09drmessanop3nguin_, I see ads in PC World all the time for $3 hosting places
20:15.37Chainsawcarrar: It's basically just two ADSL connections that are glued together on both the ISP end and your end.
20:15.46drmessanoGoDaddy gives you free hosting with your domain reg
20:15.49Chainsawcarrar: (So you'll probably have to get a different line delivered/installed for it)
20:15.55ManxPower-workChannelZ: In my experience "it works mostly well, except when the damn carrier can't keep all the pairs working"
20:16.08Chainsaws/carrar/ChannelZ/
20:16.13drmessanoLimited, but I think Kamel Toes Nail Salon may not need more than 20GB a month transfer..Just sayin
20:16.35Kamellmao
20:16.43*** join/#asterisk atis_work (~atis_work@193.238.212.171)
20:17.10Kamelexcept when i host up the latest beta of windows on it and post it on pirate sites
20:17.28Kameli'm only kidding...
20:18.01drmessanoOh, I am only going to be kidding when I recommend you leave Asterisk for the adults and buy a Skype phone from Wal Mart then
20:18.15Kamelbut meh whatever. i like the flexibility of handling my own equipment, there's certainly some advantages to hosting yourself
20:18.27drmessanoIs it LOL j/k :) or :) lol J/K ?
20:19.17drmessanoKamel, there's not, really.. running shit at home you can easily offload somewhere else for cheap is a waste of time.. having 5 servers at home and pretending to be Google gets old
20:19.31drmessanoThere's sunlight out there, really
20:19.48*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
20:19.55p3nguin_I heard that sunlight burns!
20:20.22Kameltbh i don't really care what you think of my ideas for doing the things i do, it's my decisions to make, i will be the one ultimately reaping the benefits or disaster that comes from them
20:21.48drmessanoAh.. the "I dont give a shit what you think" conversation begins.. Going to find that sunlight now.. CYA'S!
20:22.09Kamellater
20:23.37Kamelwell i do care about your input, it's valuable to me. i just don't care about what judgements people pass about what i should do and how i should do it... it just doesn't have any effect and deters from beneficial conversation
20:23.48Kamelon that note, thanks for your help and teaching me
20:24.07Kamelanyway, i'm out too, going to the store
20:29.20Kattywtb shrimp recipe
20:31.01nightrid3rmmmmm shrimp :D
20:31.47Kattyindeed
20:31.52Kattybut i need a tasty way to prepare it
20:40.35nightrid3rdon't ask me, i only eat the stuff
20:46.30*** join/#asterisk ttwhy (~tekkno@p4FECFA10.dip.t-dialin.net)
20:48.45*** join/#asterisk TimeRider (~steve@78.32.26.1)
20:53.10*** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com)
20:54.32KamelKatty: i just steam my shrimp, then include it with any normal meal *shrugs*
20:58.17p3nguin_If you aren't going to Return() at the end, is there any difference in Gosub() and Goto()?
21:01.07nightrid3rp3nguin_: you'll probably run out of stack space and crash
21:01.37p3nguin_In what condition?
21:01.49nightrid3rgosub without return
21:03.21nightrid3rin the old 8bit day's precompilers used to check on that cos of the stack limits
21:03.54nightrid3rdon't know how it works today but probably a similar system
21:04.16*** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net)
21:05.05nightrid3rbest is to avoid goto as much as possible to prevent your code from looking like spagheti
21:05.49*** join/#asterisk Liqdfire (~panda@65.34.91.10)
21:06.32LiqdfireI am having an issue with realtime and odbc, anyone able to help ?
21:06.57p3nguin_Well, I have had to use a Goto because I had a problem with more than one context having a 1 in it.  Such as phones included call-return and blacklist contexts, both which had an option 1.  So when I called the blacklist and pressed 1, it actually registered 1@call-return.
21:07.15p3nguin_The solution was to change the includes to Goto()s.
21:07.27p3nguin_Essentially creating IVRs out of those two contexts.
21:09.43nightrid3rsometimes its impossible to avoid goto's but try to limit them
21:11.19LiqdfireI have asterisk setup to read the configs using realtime over odbc from a SQL2008 server
21:11.41Liqdfirewhen I connect using a softphone it connects just fine
21:11.43p3nguin_Macros are being deprecated in favor of Gosub.  That was my reason for asking if a Gosub without a Return should instead be a Goto.
21:12.10Liqdfirehowever when registering my aastra phone, when it tries to update the sip config table it throws a sql exception
21:12.57nightrid3ryes, if you don't use return you have to use goto
21:12.59LiqdfireI noticed it was passing an empty string in for some of the columns that I had set to integers, so I changed them to varchars, but it is still crashing
21:16.56*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
21:17.38norrecdoes anyone know if you can add in ilbc support without recompiling?
21:18.03ManxPower-worknorrec: Why don't you want to recompile?
21:19.07norrecManxPower-work i installed though the repos =/
21:19.39Liqdfireis it a production box ?
21:19.53ManxPower-worknorrec: You begin to understand why we don't support packages.
21:20.04ManxPower-worknorrec: the answer is: No!.
21:20.15norrecyeah, i'm realiseing the mistake of that one
21:20.18ManxPower-workYou could rebuild the packages yourself.
21:20.51ManxPower-workjust don't install a compiled from source asterisk on top of a packaged installed Asterisk
21:20.57norrecwell, i'm thinking about just saving the config dir, and reinstalling from source
21:21.11Liqdfiredo it
21:21.25Liqdfireespecially if it is not a prod box yet
21:21.25norrecManxPower-work: yeah, i figured that would be a bad idea
21:21.31ManxPower-worknorrec: *nod* If you are going to be installing it on multiple servers, I would actually rebuild the package.
21:22.11*** join/#asterisk albasheers (~basheer@188.116.235.226)
21:22.12ManxPower-workYou might have a learning curve in package management, but it will serve you well in the future if you're going to be managing multiple servers.
21:22.16norrecwell, sounds like thats what i will do then, does asterisk save any configs outside /etc/asterisk?
21:22.37*** part/#asterisk albasheers (~basheer@188.116.235.226)
21:22.48ManxPower-worknorrec: no, but you also have voicemail messages, etc.
21:23.13ManxPower-workOne thing you could try is delete everything from /var/lib/asterisk/modules/*
21:23.34norrecManxPower-work: yeah, i dont really have any in there, just test shit, but my configs are all setup already
21:23.55norrecManxPower-work: what does that do?
21:24.04ManxPower-workfind a way to "uninstall" the asterisk pakage without deleteing files.  Then install Asterisk on top of that.  Should work OK as long as you stick to the same major version of Asterisk.
21:24.11*** join/#asterisk mnicholson (~mnicholso@nat/digium/x-kxrkcwlerladnloz)
21:24.12*** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler)
21:24.12*** mode/#asterisk [+o angler] by ChanServ
21:24.16*** join/#asterisk fish-bulb (~cstewart@nat/digium/x-ldltbgtkgwiqndiw)
21:24.19*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
21:24.20*** mode/#asterisk [+o putnopvut] by ChanServ
21:24.40ManxPower-workthe most likely issue you would have is with the Asterisk modules
21:24.46norrecah
21:25.05ManxPower-workIf you stick to the same major version everything else should work fine.
21:25.37norrecso, with the source, how would i upgrade between 1.6, and 1.8 when it comes out?
21:25.54norrecor between 1.6.0 and 1.6.2
21:26.00ManxPower-worknorrec: there should be upgrade notes that covers that.
21:26.08norrecah, alright
21:27.59norrecso for the souce install, does it try to use root or asterisk as the user by default?
21:28.40ManxPower-worknorrec: That's a config file thing, so your source install should be the same user as your existing install.
21:29.45ManxPower-workwhen you install asterisk with "make install" also a "make config" which installs the "config" startup scripts, usually via init.d
21:30.08norrecalright
21:30.10ManxPower-work"make samples" will install the default set of Asterisk config files, overwriting your existing.
21:32.06ManxPower-workthe init scripts are another potential point of problem, so reinstalling them is a good idea.
21:32.23norrecalright
21:34.13*** join/#asterisk catojo (~catojo@189.24.48.139)
21:34.37norrecso would i do the ilbc install instructions after i install asterisk or at the same time?
21:35.17Tim_Toadybefore you compile it
21:35.48Tim_Toadyyou get the source, copy it in codecs/ilbc/ and then build asterisk
21:36.24norreckk
21:36.42tzafrir_laptopIsn't there a script to download the source?
21:36.50tzafrir_laptop(of the ilbc codec)
21:37.10Liqdfirewhere is the realtime code for Sip channels? so i can see what it is doing whrn preparing the sql statements for odbc.
21:37.19*** join/#asterisk adnc (~numer@unaffiliated/adnc)
21:37.22bn-7bcwhy didi the devs strip ilbc out of the standart distribution?
21:39.49tzafrir_laptoplicensing issues
21:42.45bn-7bcok
21:43.48*** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com)
21:50.30*** join/#asterisk smooth_penguin (~smoove@59.95.4.127)
21:51.36*** join/#asterisk adnc (~numer@unaffiliated/adnc)
21:55.16Liqdfire[Feb 21 16:55:07] WARNING[3699]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect...
21:55.16Liqdfire*** glibc detected *** asterisk: corrupted double-linked list: 0x00002aaaac4c09f0 ***
21:55.35Liqdfirethat is what I am getting when any sip client tries to register
21:57.08norrecso, for ilbc, i wanted to have some of my peers use ilbc but my sip trunks for outbound/inbound are g711u, will asterisk auto convert, or what do i need to set up to convert between the two?
21:58.02*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
21:58.02*** mode/#asterisk [+o leifmadsen] by ChanServ
21:58.44voipmonkasterisk will convert them for you norrec
21:59.15norrecoh good, thanks =D
22:01.38*** join/#asterisk MindTheGap (~MindTheGa@189.59.199.118)
22:08.06*** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
22:08.25*** join/#asterisk pokui (~pokui@mbp-pjo.imul.com)
22:10.58*** join/#asterisk iq (~iq@unaffiliated/iq)
22:33.14*** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net)
22:35.57norrecis there a way to get more detailed information about a channel, like the codec being used and bandwidth info and such?
22:36.04skritehey all, what is the CLI command that lets me see what channels there are available ( not the core show channels ) i need to know if my card is set up right
22:36.29skritewait sorry, just found the help command :)
22:38.00*** join/#asterisk etnos (~etnos@c-75-74-66-161.hsd1.fl.comcast.net)
22:39.00p3nguin_core show channeltypes
22:39.01*** join/#asterisk vk2dgy (~rossw@ali-syd-3.albury.net.au)
22:39.49skritep3nguin_, hello again, you were the one helping me Friday.  i did the show channeltypes and dahdi did not show up, should it have?
22:39.59vk2dgyhi good folks. Anyone here using Future-Nine as a SIP provider with Asterisk 1.4.18?
22:42.12*** join/#asterisk mykhyggz (~col@evolone.org)
22:43.30p3nguin_skrite: http://pastebin.com/d706b8a08
22:43.52p3nguin_skrite: If you have the dahdi channel driver loaded, it should show up.
22:47.00*** join/#asterisk capitan (~captain@76.91.206.32)
22:47.13skritethanks p3nguin_ here is what i get http://pastebin.com/d1b7177b2   the correct drivers seem loaded when i do an lsmod
22:49.02p3nguin_skrite: kernel modules are not the same as asterisk modules.  Asterisk modules are the channel drivers.
22:49.43p3nguin_module load chan_dahdi
22:49.58p3nguin_That should load the dahdi channel driver if there is no problem.
22:51.58skritedamn missing the /usr/lib/asterisk/modules/chan_dahdi.so
22:53.21ManxPower-workskrite: damn install dahdi before Asterisk
22:53.38skritei did
22:53.47skritebut i think i did screw some things up
22:53.56skritewill check some apt info
22:54.41ManxPower-workUm, how do you screw up "make install"?
22:55.23ManxPower-workapt info?
22:55.26nickaugust[TK]D-Fender: ping!
22:55.55ManxPower-workYou're wasting our time asking about packaged Asterisks?
22:59.08skriteManxPower-work, wasting your time? sorry, channel did not look too busy.  i have done a couple of installs not only of asterisk, but was trying to replace zaptel, think that is where it went wrong
22:59.34skritestill troubleshooting though
23:01.35*** part/#asterisk pokui (~pokui@mbp-pjo.imul.com)
23:02.53ManxPower-workuninstall the packages, install zaptel and asterisk from source.
23:04.44skriteManxPower-work, gotcha, thanks
23:05.05*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net)
23:07.39*** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net)
23:35.32vk2dgyso, do I presume nobody here uses future-nine for their sip provider? Does their reputation precede them or something?
23:36.26vk2dgymy drama is that I just can't seem to get my asterisk 1.4.18 to talk to them "properly" - it registers, but whenever I try to place a call, I get:
23:36.31vk2dgysip_call: No audio format found to offer. Cancelling call to (xxxxx)
23:40.25*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
23:42.21*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
23:42.46voipmonkwhat codecs do u have set for that provider vk2dgy ?
23:45.06vk2dgyI have the following (commented one or the other for testing)
23:45.23vk2dgydisallow=all
23:45.24vk2dgy;allow=g729
23:45.28vk2dgyallow=ulaw
23:45.33ManxPower-workyou would not want to allow g729
23:45.44vk2dgyso when I'm trying to use g729, I uncomment it and comment ulaw, and vice versa.
23:45.59ManxPower-worktry g729 AFTER you get it working
23:46.12vk2dgyManx- why not? They specifically state that G729 is their preferred codec.
23:46.15skriteok, built the dahdi from source, and can modprobe dahdi and it is ok, but when i modprobe wctdm24xxp, i am getting this... http://pastie.org/836017
23:46.21vk2dgybut I can't get it going with 711u either :(
23:46.29*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
23:48.27vk2dgy"sip show peer (them)" confirms the codec too btw:
23:48.34vk2dgy<PROTECTED>
23:48.35vk2dgy<PROTECTED>
23:52.23ManxPower-workvk2dgy: because I doubt you know enough to make g729 passthru work at this point.
23:52.44ManxPower-workvk2dgy: do a sip debug, pastebin the results
23:52.46ManxPower-work~pb
23:52.46infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
23:53.16ManxPower-workskrite: "service dahdi start". don't modprobe them manually
23:53.29skriteok, thanks
23:54.18adnchello, for matching local calls like 123456 would an entry like _N. be enough or do i need to consider anything else?
23:54.32ManxPower-workwell N won't match 1
23:54.36vk2dgyManxpower - it doesn't work with g711u either, so it's not specifically a g729 issue. I had it all working with broadvoice, it's only when I changed to future-nine that the wheels have fallen off - hence my question about them specifically.
23:54.42adncmaxagaz, ohh
23:54.50adncthat is bad, so _X. is necessary?
23:54.53ManxPower-workvk2dgy: I'm wiating for the pastebin.
23:55.16ManxPower-workadnc: What are you trying to match?
23:55.50adncmanxPower-work i would like to match to local numbers and route them through a particular provider
23:56.10adnclocal numbers in germany are from 4 to ...
23:56.13vk2dgysip debug will flood you with stuff - there are a bunch of other trunks doing stuff at the moment. Will a debug of that trunk only help?
23:56.29ManxPower-workthen just do a debug on that peer.
23:56.31ManxPower-work~trunk
23:56.32infobotextra, extra, read all about it, trunk is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term.
23:58.31adncManxPower-work so is _X. enough?
23:59.23*** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net)

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