IRC log for #asterisk on 20100619

`00:00.07`	`Kyosh`	`so 1.4 seems as if its pretty good compared to then`
`00:00.26`	`p3nguin`	`I'll use 1.4 until it is dead and buried.`
`00:00.35`	`pabelanger-lap`	`1.6.0 and 1.6.1 are in security mode`
`00:00.37`	`Kyosh`	`haha`
`00:00.38`	`Kyosh`	`so will i`
`00:01.28`	`*** join/#asterisk mindCrime (~chatzilla@64.134.184.64)`
`00:01.48`	`p3nguin`	`Actually, 1.6.1 is also scheduled for EOL eight months before 1.4.`
`00:01.58`	`p3nguin`	`1.6.2 will be a few days after 1.4.`
`00:02.39`	`p3nguin`	`Weird how they intend to kill off all these branches and only have one remaining.`
`00:03.07`	`Kyosh`	`which will remain? 1.8?`
`00:03.32`	`p3nguin`	`That's how it seems.`
`00:04.05`	`p3nguin`	`They might be planning another unstable branch that we haven't heard about.`
`00:04.38`	`Kyosh`	`well if they want me to relinquish my 1.4.2.1, theyll have to pry it from my cold dead hands!`
`00:04.41`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`00:06.05`	`*** join/#asterisk joobie (~joobie@CPE-124-181-130-239.vic.bigpond.net.au)`
`00:11.17`	`*** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net)`
`00:16.46`	`*** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com)`
`00:22.25`	`*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)`
`00:22.33`	`p3nguin`	`Since they still never added a changelog link near the downloads, I don't know if I need to upgrade my 1.4 systems or not.`
`00:23.01`	`joobie`	`if you are relying on a changelog to determine if you upgrade, then don't.`
`00:25.17`	`*** join/#asterisk aidinb (~Aidin@32.172.90.241)`
`00:25.50`	`p3nguin`	`Changelogs reveal the changes between versions, so I can find out if I need an upgrade by reading them. What's the problem with that?`
`00:27.35`	`*** join/#asterisk cp5 (~samy@cpe-76-173-222-231.socal.res.rr.com)`
`00:27.35`	`[TK]D-Fender`	`I see the changelogs just fine`
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`00:30.01`	`joobie`	`p3nguin, it's obviously not the only source on the internet to find out the differences`
`00:30.28`	`joobie`	`p3nguin, if you run out of fuel in your car, do you sit there pressing the accelerator waiting for it to start going again?`
`00:34.39`	`tzafrir`	`joobie, and that example is relevant because?`
`00:35.13`	`joobie`	`tzafrir, it just paints a picture in my mind of where p3nguin was at when the changelog link was broken`
`00:35.19`	`joobie`	`refresh refresh refresh`
`00:35.53`	`joobie`	`neway.. just being a smartass`
`00:36.18`	*** part/#asterisk MT`AwAy (~MagicalTu@2001:41d0:2:973::aeb)
`00:40.41`	`*** join/#asterisk aidinb (~Aidin@32.172.90.241)`
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`01:08.52`	`luke-jr`	`if two towns are in the same rate center, does that imply they have the same local-calling-from areas?`
`01:13.30`	`p3nguin`	`[tk]d-fender: I never said changelogs were invisible. What I said was there is no way to navigate to changelogs from asterisk.org, and it would make sense to put links to changelogs near the downloads; it still has not been done.`
`01:16.51`	`p3nguin`	`If you think I'm wrong, go to http://www.asterisk.org/ with me and show me where to navigate to changelogs. I'd love to be proven wrong on this matter.`
`01:17.29`	`*** join/#asterisk rocksfrow (~kyle@pool-96-244-85-128.bltmmd.fios.verizon.net)`
`01:18.10`	`rocksfrow`	`so, i have a primary/secondary asterisk failover system setup and working, but i have one problem... my phones don't show up registered fast enough`
`01:18.27`	`rocksfrow`	`is there some way from the asterisk box, to ping all extensions? to get the phones to register on the new system`
`01:19.08`	`rocksfrow`	`the virtual IP is being switched`
`01:19.14`	`rocksfrow`	`and the phones will slowly pop up after awhile`
`01:19.35`	`*** join/#asterisk cesar_CR (~cesar@201.201.41.242)`
`01:19.51`	`rocksfrow`	`any thoughts are appreciated.`
`01:20.57`	`[TK]D-Fender`	`[21:13]<p3nguin>[tk]d-fender: I never said changelogs were invisible. What I said was there is no way to navigate to changelogs from asterisk.org, and it would make sense to put links to changelogs near the downloads; it still has not been done. <- yup. Wrong`
`01:21.11`	`[TK]D-Fender`	`http://www.asterisk.org/downloads`
`01:21.16`	`[TK]D-Fender`	`Older Versions / Direct Access`
`01:21.23`	`[TK]D-Fender`	`http://downloads.asterisk.org/pub/telephony/asterisk/`
`01:21.53`	`[TK]D-Fender`	`Changelogs all nice & neat on top`
`01:22.07`	`[TK]D-Fender`	`Plus, IIRC they ar included IN the source tarball anyway`
`01:22.32`	`[TK]D-Fender`	`rocksfrow: You can't make a phone re-register`
`01:22.50`	`rocksfrow`	`[TK]D-Fender, what is making them pop up registered after a minute or so?`
`01:22.55`	`rocksfrow`	`is that a configuration on the phone?`
`01:23.07`	`[TK]D-Fender`	`rocksfrow: probably a regiter timeout`
`01:23.24`	`[TK]D-Fender`	`rocksfrow: You DO know they determine a re-reg frepquency... right?`
`01:23.38`	`rocksfrow`	`right, is there any problem with making the rereg frequency like 30s?`
`01:24.19`	`[TK]D-Fender`	`rocksfrow: think of the excess traffic you're adding`
`01:24.52`	`[TK]D-Fender`	`rocksfrow: you seem to be treating fail-over as a daily fact rather than a rare occurance`
`01:24.57`	`rocksfrow`	`[TK]D-Fender, well, my issue is that i need my customer service rep phones to come up ASAP once the failover occurs`
`01:25.19`	`TJNII`	`Think of the children!`
`01:25.21`	`TJNII`	`Think of how all the excess traffic could delay their pornos!`
`01:25.21`	`rocksfrow`	`[TK]D-Fender, of course i hope it to be rare, but i still feel good knowing the switch would happen quickly`
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`01:25.39`	`ChannelZ`	`Please, won't you give? Please.`
`01:25.43`	`rocksfrow`	`maybe just set the CS rep phones to rereg every 30s?`
`01:25.49`	`rocksfrow`	`its a lot less phones`
`01:26.06`	`rocksfrow`	`the other phones really aren't as important`
`01:26.21`	`rocksfrow`	`but... i mean i would prefer the phones to rereg quickly`
`01:26.32`	`rocksfrow`	`so, there a re-reg query you can send to particular phones?`
`01:26.39`	`rocksfrow`	`like, a broadcast`
`01:26.44`	`*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)`
`01:27.13`	`paul	`what asterisk function can i use so that i will be able to call a number then that number will give me a tone to input another number to call`
`01:27.14`	`paul	`?`
`01:27.28`	`ChannelZ`	`DISA`
`01:27.32`	`ChannelZ`	`or just write it yourself`
`01:28.10`	`TJNII`	`rocksfrow: If you mangle a packet bad enough you could crash them, causing them to reset....`
`01:29.06`	`[TK]D-Fender`	`paul: Or a recording of dialtone used in an IVR. Or in a Read() or in....
`01:29.07`	`paul	`thanks`
`01:29.37`	`paul	`oh yes thats an option to fender`
`01:30.00`	`paul	`so which is better DISA or Read() method?`
`01:32.09`	`ChannelZ`	`Disa is better in the sense you don't have to build out most of the logic yourself... but beyond those kinds of considerations it doesn't much matter`
`01:33.22`	`*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)`
`01:42.03`	`*** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net)`
`01:43.18`	`[TK]D-Fender`	`DISA offers no failover if you misdial, create extra wierd CDR's, etc.`
`01:43.43`	`jsidhu`	`how to I get the voicemail to automatically delete after an email is sent out? setting delete=yes in the extension doesn't seem to do it.. (500 => 500,Mailbox 500,users_email@domain.com,delete=yes)`
`01:45.14`	`WIMPy`	`jsidhu: You're missing a comma.`
`01:45.15`	`[TK]D-Fender`	`jsidhu: Because it isn't in the right parameter position.`
`01:45.38`	`jsidhu`	`ahh isthat the pager email`
`01:45.43`	`jsidhu`	`thanks guys`
`01:47.20`	`jsidhu`	`thank you, that was it`
`01:48.10`	`rocksfrow`	`does anybody have a perl/shell script to make a polycom phone restart?`
`01:48.17`	`rocksfrow`	`reregister***`
`01:48.58`	`rocksfrow`	`soundpoint 550s`
`01:50.07`	`puzzled`	`rocksfrow: iirc there is one on http://www.voip-info.org/`
`01:50.17`	`[TK]D-Fender`	`rocksfrow: Horrible option.`
`01:52.18`	`puzzled`	`turns on the horrible option in some xml files and waits what happens`
`01:54.39`	`rocksfrow`	`[TK]D-Fender, whats a better option?`
`01:54.49`	`rocksfrow`	`i need these customer service rep's phones to ring ASAP in case of a fialover`
`01:54.52`	`rocksfrow`	`failover***`
`01:55.29`	`rocksfrow`	`we only have 3 customer service phones atm, so i don't think making them reregister every 30s is that bad`
`01:55.38`	`rocksfrow`	`i actually find other people doing it with polycom phones to resolve issues`
`01:55.44`	`rocksfrow`	`on forums, atleast`
`02:03.10`	`[TK]D-Fender`	`rocksfrow: Considering the 2 minute boot cycles and the customers freaking out at not being able to place a call while their phone looks like it just crashed....`
`02:03.23`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`02:03.36`	`[TK]D-Fender`	`rocksfrow: crap idea. You need to think on this more. You are hacking together some really ugly ideas right now.`
`02:03.45`	`[TK]D-Fender`	`is of... show time...`
`02:03.48`	`[TK]D-Fender`	`off`
`02:04.07`	`rocksfrow`	`[TK]D-Fender, whaaat?`
`02:04.46`	`rocksfrow`	`you're wrong. the failover is working great. the ONLY issue here is that I need my customer service reps to re register after the failover occurs (the secondary server takes over the virtual IP, so the phones need to reregister)`
`02:05.08`	`rocksfrow`	`i don't see how this is hacking together ugly ideas`
`02:05.29`	`rocksfrow`	`and what are you talking about 2 minute boot cycles?`
`02:05.46`	`rocksfrow`	`reREGISTER, not reboot`
`02:06.24`	`rocksfrow`	`this time of failover setup is used all over`
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`02:34.10`	`jsidhu`	`so im evaluating asterisk for our company. we've got a bunch of branch offices.. 8-10.. lets say i have a asterisk box at each location, whats the best way to interconnect all of these boxes? I want any branch tobe able to call any other branch.. (with unique extensions across all). some of these asterisk systems will have a dynamic ip.. whats the best way to approach this? makinga bug of iax or sip trunks?`
`02:34.38`	`jsidhu`	`bug/bunch`
`02:35.25`	`TJNII`	`dundi, possibly`
`02:35.39`	`TJNII`	`Though I've never played with it, so I don't know how well it works.`
`02:36.07`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`02:36.17`	`jsidhu`	`yeah, was just reading about that on the side`
`02:36.26`	`jsidhu`	`anyone haveany links to get me started with DUnDI?`
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`02:56.03`	`pabelanger-lap`	`jsidhu: IAX2 trunks`
`02:57.00`	`pabelanger-lap`	`DUNDi is just a routing protocol for finding extensions`
`02:57.03`	`*** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com)`
`02:58.00`	`jsidhu`	`pabelanger: i think thatsgonna be a lot of extensions that I have to create on every single box`
`02:58.13`	`jsidhu`	`thats the scenario im trying to avoid..but lets see`
`03:09.02`	`pabelanger-lap`	`jsidhu: no way around it, each extension needs to be configure on each asterisk box. Dialplans can be simpler exten => _3XX,1,Macro(...)`
`03:09.06`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
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`04:30.17`	`ChannelZ`	`if all the branches have to have unique extensions, then just pick ranges for each branch.. 1xxx is location 1, 2xxx is location 2, etc.`
`04:31.22`	`ChannelZ`	`Each local dialplan handles their own extensions, and then single patterns direct calls to remote locations`
`04:48.38`	`jsidhu`	`ChannelZ: yeah but dont i still need to define trunks?`
`04:48.56`	`jsidhu`	`im confused`
`04:49.57`	`ChannelZ`	`yeah, I'd assume you'd want to`
`04:50.10`	`jsidhu`	`um`
`04:50.34`	`jsidhu`	`so ok, i define a IAX trunk on each box`
`04:50.49`	`jsidhu`	`then setup dialplans on each box over to that trunk`
`04:52.20`	`ChannelZ`	`well you'd define an IAX from box A to Box B,C,D, etc. On each one.. so each location could talk directly to any other.`
`04:53.11`	`ChannelZ`	`Then if you do as I suggest, make each location have a specific range of extension numbers, you can send a call to any location just by looking at the first digit or whatever.`
`04:53.33`	`jsidhu`	`yes, each location already has their own seperate extensions`
`04:53.37`	`ChannelZ`	`Thus avoiding having to program each location with every individual extension for every other location.`
`04:53.57`	`jsidhu`	`ok`
`04:54.19`	`jsidhu`	`can you link to to a wiki/forum/something that will walkme thru on setting up a trunk between two boxes`
`04:55.02`	`ChannelZ`	`So from location 1 if someone dial 1xxx, the dialplan handles that and calls whatever local device. If they dial 2xxx, you just send to IAX2/location2/exten - a single entry in your dialplan. Then location 2's own dialplan routes to the proper device on their end`
`04:55.10`	`ChannelZ`	`~book`
`04:55.11`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`04:56.29`	`jsidhu`	`thank you`
`05:06.23`	`*** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb)`
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`05:33.02`	`jsidhu`	`hey ChannelZ, following thatguide, on page http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html, they have a dial out to the IAX in the [remote] context (at the bottom ofthe page). exten => _2XXX,n,Dial(IAX2/toronto/${EXTEN}) what is the "toronto" in this?? hostname? is this a bug? toronto should bedialing osaka..`
`05:35.34`	`kaldemar`	`jsidhu: "toronto" is [toronto] in iax.conf. looks like a bug.`
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`05:40.10`	`jsidhu`	`yeah`
`05:40.18`	`jsidhu`	`i thinkthey gotem switched`
`05:41.13`	`jsidhu`	`anyways, after fixing that and placing a call, i get a error saying Call rejected by 10.1.1.1: No such context/extension (The IP has been changed,but its correct)`
`05:44.10`	`kaldemar`	`show the dial line for the originating end and iax.conf and extensions.conf for the receiving end.`
`05:45.44`	`jsidhu`	`exten => _1XX,n,Dial(IAX2/sanjose/${EXTEN})`
`05:46.12`	`jsidhu`	`bah`
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`05:46.21`	`jsidhu`	`i think pastebin would be a good option`
`05:46.27`	`kaldemar`	`definitely`
`05:53.41`	`jsidhu`	`probably not what ur looking for`
`05:53.42`	`jsidhu`	`http://pastebin.com/WbyPYTiT`
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`05:54.02`	`jsidhu`	`ivegot freepbx installed, so thhese are justthe _custom.confs`
`05:56.47`	`jsidhu`	`the iax2 trunks are up tho, iax2 show peers confirms on both ends`
`05:59.28`	`x-demon`	`hi all`
`05:59.35`	`x-demon`	`i can't get extension working`
`05:59.41`	`kaldemar`	`jsidhu: not at all what i'm looking for.`
`05:59.48`	`x-demon`	`exten => _9749[59].,1,Dial(SIP/${EXTEN:1}@voipx)`
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`06:00.05`	`x-demon`	`in logs i see that there is no such extension`
`06:00.37`	`ChannelZ`	`ok well what are you dialing`
`06:00.49`	`x-demon`	`i want dial numbers such as 974951234567 and route them via voipx trunk`
`06:01.10`	`x-demon`	trunk configured and registered, i can see it via `sip show peers`
`06:01.19`	`kaldemar`	`jsidhu: assuming your boxes are sanjose and longbeach. sanjose should have Dial(IAX2/longbeach/...) in extensions.conf and [longbeach] in iax.conf. longbeach should have Dial(IAX2/sanjose/...) in extensions.conf and [sanjose] in iax.conf.`
`06:02.33`	`ChannelZ`	`pastebin the whole console output for the call attempt`
`06:03.25`	`x-demon`	`well there is just one line`
`06:03.29`	`x-demon`	`chan_sip.c: Call from 'me' to extension '974951234567' rejected because extension not found.`
`06:04.00`	`ChannelZ`	`core set verbose 4`
`06:04.19`	`ChannelZ`	`My guess is the context of the device you're dialing from isn't right`
`06:04.38`	`x-demon`	`well i have context lexcontext`
`06:04.55`	`x-demon`	`rule to route via voipx called voipx-outbound`
`06:05.14`	`ChannelZ`	`pastebin configs because obviously something isn't setup right`
`06:05.20`	`ChannelZ`	`extensions.conf and sip.conf`
`06:05.54`	`*** join/#asterisk k-man (~jason@unaffiliated/k-man)`
`06:06.04`	`jsidhu`	`kaldemar: thats what i have. but the problem is with the context. If I call from SAN JOSE to LONGBEACH, i get the following error (as seen on the longbeach side): Rejected connect attempt from 10.9.0.1, request '500@default' does not exist where is it getting @default from ?`
`06:06.05`	`x-demon`	`which of sip configs? i separated files for users and trunks`
`06:07.28`	`ChannelZ`	`mostly users`
`06:07.42`	`ChannelZ`	`whatever device you're dialing out from`
`06:08.41`	`x-demon`	`ChannelZ, http://pastebin.com/y9uShm0B`
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`06:10.51`	`ChannelZ`	`does 'dialplan show' show your extensions?`
`06:11.29`	`x-demon`	`huh, no`
`06:11.37`	`x-demon`	`only empty 'parkedcalls'`
`06:11.49`	`ChannelZ`	`did you reload your dialplan after constructing this?`
`06:12.01`	`x-demon`	`yes i reloaded asterisk`
`06:12.03`	`ChannelZ`	`Either that or it doesn't like all the spaces at the beginnings of the lines. Never tried.`
`06:12.09`	`x-demon`	`btw i have only chan_sip.so modules`
`06:12.19`	`x-demon`	`maybe that's a problem?`
`06:12.20`	`ChannelZ`	`or your extensions.conf isn't readable by the user asterisk is running at and it's not even parsing it`
`06:12.45`	`ChannelZ`	`uhh, as in you don't have pbx_config loaded?`
`06:12.46`	`p3nguin`	`You'll want the pbx_config module, too.`
`06:13.50`	`x-demon`	`yeah`
`06:13.56`	`x-demon`	`now i see dialplans`
`06:14.15`	`x-demon`	`sorry me, i'm a complete newbie in voip stuff`
`06:14.24`	`kaldemar`	`jsidhu: show the configs i asked for and i'll tell.`
`06:14.46`	`kaldemar`	`jsidhu: feel free to mask any passwords.`
`06:15.11`	`x-demon`	`WARNING[31044]: pbx.c:3680 pbx_extension_helper: No application 'Dial' for extension (lexcontext, 74951234567, 1)`
`06:15.16`	`x-demon`	`another module?`
`06:15.32`	`ChannelZ`	`you should leave autoload ON in modules.conf until you know what you don't need`
`06:15.45`	`x-demon`	`ChannelZ, it loads all 200+ modules`
`06:15.46`	`ChannelZ`	`you're likely missing dozens of crucial functions`
`06:16.11`	`ChannelZ`	`how is it that you didn't have pbx_config then?`
`06:16.20`	`x-demon`	`ChannelZ, no i loaded it`
`06:16.26`	`x-demon`	`and can see dialplans`
`06:16.39`	`[TK]D-Fender`	`x-demon: pastebin your modules.conf and the entire modules folder`
`06:16.43`	`ChannelZ`	`My question was why wasn't it loaded to begin with`
`06:17.32`	`x-demon`	`i found good documentation about modules, nevermind`
`06:19.54`	`ChannelZ`	`jsidhu: I think you have a context problem as well`
`06:20.10`	`x-demon`	`okay, i got it to work with one provider, but not with voipx`
`06:20.22`	`x-demon`	`[Jun 19 07:19:20] -- Called 74951234567@voipx [Jun 19 07:19:20] -- SIP/voipx-00000003 is circuit-busy`
`06:20.52`	`[TK]D-Fender`	`ok, this is going to be a piece-meal debugging nightmare. I'm out.`
`06:21.06`	`x-demon`	`and that error is really annoying - chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on transmission 122684039336@95.211.0.236 for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt. - i googled for it but found nothing`
`06:21.08`	`ChannelZ`	`That's a them-problem`
`06:21.13`	`p3nguin`	`For some reason we have yet to uncover, plenty of people have shown up here without dialplans being loaded, only to eventually find out that pbx_config isn't loaded.`
`06:21.20`	`[TK]D-Fender`	`ChannelZ: Not necessarily.`
`06:21.39`	`[TK]D-Fender`	`ChannelZ: God luck squeezing out enough usable backup to validate either way`
`06:24.23`	`jsidhu`	`yeah, i do`
`06:24.24`	`*** join/#asterisk jetlag (jetlag@pool-173-61-213-35.cmdnnj.east.verizon.net)`
`06:24.31`	`jsidhu`	`but I got it working one way`
`06:24.51`	`jsidhu`	`cant make it work from Long Beach to San Jose`
`06:25.13`	`jsidhu`	`San Jose's a bigsetup, andhas freepbx.. so its got a boatload of contexts`
`06:25.34`	`*** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com)`
`06:27.06`	`*** join/#asterisk GameGamer43\|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)`
`06:27.58`	`p3nguin`	`That's probably part of the problem.`
`06:28.06`	`devdvd`	`hey everyone, running into a bit of an odd issue. I have a feeling its somehow related to nat but ive never experienced this behavior. I have a polycom 321 running on asterisk 1.4.32. When I dial the phone from an extension, it works fine, i can hear the other side and they can hear me. When they call an external phone through the sip trunk, both parties can hear just fine but if an outside party calls and connects to the phone, the pe`
`06:28.24`	`devdvd`	`thoughts?`
`06:29.42`	`p3nguin`	`Your detailed description was truncated.`
`06:29.58`	`devdvd`	`how far did you see to`
`06:30.01`	`p3nguin`	`outside party calls and connects to the phone, the pe...`
`06:30.17`	`devdvd`	`, the person on the polycom can be heard but not the person on the other side`
`06:30.55`	`p3nguin`	`So you have one-way audio... but only when calling inbound?`
`06:31.07`	`devdvd`	`yes`
`06:31.35`	`p3nguin`	`One-way audio is often a NAT issue, but I would expect it to show its face on calls going either direction.`
`06:31.52`	`devdvd`	`extension to polycom works, polycom to extension works, polycom to external works but external to polycom has one way audi`
`06:31.52`	`devdvd`	`o`
`06:31.53`	`devdvd`	`yea`
`06:31.56`	`devdvd`	`that was my thought as well`
`06:32.17`	`devdvd`	`which is why im sitting here with blood on my hands from scratching my head going wtf`
`06:32.31`	`devdvd`	`http://pastebin.com/UQ0ZxiYs`
`06:32.35`	`p3nguin`	`Are you using SIP for all connections?`
`06:32.42`	`devdvd`	`theres a pastebin of my sip.conf entry for that phone`
`06:32.46`	`devdvd`	`yea`
`06:33.10`	`p3nguin`	`I'd run through the sipnat stuff just to make sure I didn't miss something.`
`06:33.13`	`p3nguin`	`~sipnat`
`06:33.15`	`infobot`	`[~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`
`06:36.54`	`*** join/#asterisk uqlev (~yuriy@91.184.221.31)`
`06:44.39`	`*** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net)`
`06:45.51`	`jsidhu`	`when A calls B over a IAX2 trunk, B says "No such context/extension". How can i find out what context B is complaining about? since teh extesion does infact exist, how do i look at the context it says doesnt exist`
`06:47.23`	`kaldemar`	`jsidhu: find out what peer the call matches to on B, see its context parameter and check your dialplan.`
`06:47.24`	`ChannelZ`	`it's the entry for A on B set to the wrong context probably`
`06:48.20`	`jsidhu`	`its not`
`06:48.35`	`jsidhu`	`as far as i can see, everythins correct`
`06:48.41`	`jsidhu`	`B can call A's extensions justfine`
`06:48.45`	`jsidhu`	`A cant call B's`
`06:49.18`	`p3nguin`	`B has a problem, then. Show us the configs.`
`06:49.25`	`ChannelZ`	`Turn up verbosity a little (3 or 4), make sure verbose is turned on in logger.conf for your console, and it should be showing you where it thinks it's going`
`06:49.40`	`kaldemar`	`jsidhu: you can also show what has been asked for.`
`06:50.35`	`p3nguin`	`Maybe you can tell us what context and extension is being looked for.`
`06:50.47`	`*** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com)`
`06:51.21`	`jsidhu`	`ok, let put the fules up`
`06:51.55`	`ChannelZ`	`that sounds festive`
`06:59.35`	`jsidhu`	`heres the longbeach side`
`06:59.37`	`jsidhu`	`http://pastebin.com/eJypgMT2`
`06:59.51`	`jsidhu`	`gimme a few minutes, the freepbs muck is a big challenging to weed thru`
`06:59.58`	`jsidhu`	`which files shall i paste from San Jose?`
`07:02.20`	`ChannelZ`	`well already I see "context=incoming_sanjose" but you have no context named that in your extensions`
`07:02.36`	`jsidhu`	`?`
`07:02.46`	`jsidhu`	`ur looking at the longbeachside`
`07:02.55`	`jsidhu`	`incoming_sanjose would exist on the san jose side`
`07:02.57`	`jsidhu`	`http://pastebin.com/bFnSBx4t`
`07:03.01`	`*** join/#asterisk Micc (~Micc@c-98-225-57-96.hsd1.wa.comcast.net)`
`07:03.42`	`ChannelZ`	`If you get an incoming connection that matches the peer 'sanjose', it's going to try to send it to the "incoming_sanjose" context on that box`
`07:04.03`	`Micc`	`is there an internal asterisk database file somewhere? Somehow when I start asterisk on the backup server it thinks it has calls to make from the spool outgoing directory, but there are no call files there.`
`07:04.08`	`*** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com)`
`07:04.49`	`Micc`	`And it thinks a bunch of peers are registered when they are not. Where does asterisk keep that kind of information?`
`07:05.10`	`ChannelZ`	`there is the ASTDB`
`07:05.16`	`ChannelZ`	`database show from the console`
`07:05.58`	`Micc`	`I need to wipe it out without asterisk running.`
`07:06.05`	`Micc`	`can I just rm astdb?`
`07:07.02`	`ChannelZ`	`if you like`
`07:09.44`	`jsidhu`	`ChannelZ:so wheres the problem? ifit sends itto incoming_sanjose, then that should get picked up by the [incoming_sanjose] contextas defined in extensions_custom.conf on the san jose side`
`07:12.06`	`ChannelZ`	`yeah. But right now you're sending to "incoming_longbeach" which doesn't exist in the dialplan you showed`
`07:13.43`	`ChannelZ`	`on San Jose, you have a peer defined, 'longbeach'. Presumably when you send a call from longbeach to sanjose, sanjose will match the peer 'longbeach' for the incoming call. sanjose in turn sends the call to the 'incoming_longbeach' context of it's local dialplan.... which doesn't exist`
`07:17.17`	`*** join/#asterisk Micc (~Micc@c-98-225-57-96.hsd1.wa.comcast.net)`
`07:17.32`	`jsidhu`	`HAH!`
`07:17.35`	`ChannelZ`	`--> incoming call from XYZ -> asterisk matches peer XYZ -> sends call to context defined for peer XYZ`
`07:17.37`	`jsidhu`	`ur a genious`
`07:17.43`	`Micc`	`How can I prevent peers from registering on the backup server when the primary server is still up?`
`07:17.46`	`jsidhu`	`i had it backwards`
`07:18.02`	`jsidhu`	`i thought it was "sending" that context`
`07:18.07`	`jsidhu`	`guess i need sleep`
`07:18.21`	`Micc`	`some devices register on the backup when the network may be having issues, then it sticks with it for a while.`
`07:18.29`	`ChannelZ`	`No, you don't get to choose what context the other side uses`
`07:18.40`	`jsidhu`	`i didnt realize that`
`07:18.48`	`jsidhu`	`thanks man`
`07:18.50`	`jsidhu`	`appreciate it`
`07:19.02`	`jsidhu`	`now i can finally sleep`
`07:19.08`	`jsidhu`	`and let this info sync`
`07:19.14`	`ChannelZ`	`contexts are basically a main function of access control in asterisk`
`07:19.44`	`Micc`	`Is there any dialplan context or script that can run when registration happens that I could have asterisk reject based on some condition?`
`07:19.46`	`jsidhu`	`yeah, now i have a better understanding`
`07:19.57`	`jsidhu`	`thanks man, i'll see you later`
`07:20.02`	`ChannelZ`	`good luck`
`07:20.46`	`jsidhu`	`haha yeah`
`07:21.05`	`Micc`	`I suppose I could modify the asterisk source to only allow registrations if the main server peer is offline.`
`07:21.11`	`jsidhu`	`openwrt + asterisk appearsto be a pretty good solution for branch offices`
`07:21.42`	`Micc`	`openwrt have good qos?`
`07:21.43`	`jsidhu`	`aight im out....beofrei passout`
`07:21.53`	`ChannelZ`	`heh`
`07:23.04`	`ChannelZ`	`Micc: Are you on BeOS?`
`07:23.10`	`Micc`	`does asterisk have any way to distribute peers accross multiple servers?`
`07:23.16`	`Micc`	`ChannelZ, Haiku :)`
`07:23.35`	`ChannelZ`	`I really miss programming for BeOS...`
`07:24.01`	`coppice`	`I really miss programming for huge piles of cash`
`07:24.11`	`ChannelZ`	`I loved the API, it was actually fun to write in`
`07:24.13`	`*** join/#asterisk nezZario (~nezZario@67.237.23.232)`
`07:24.15`	`Micc`	`ChannelZ, I know, me too. I have been running Haiku for a couple weeks on my desktop. Its not bad but its missing a few key things still.`
`07:24.16`	`nezZario`	`Two questions..`
`07:24.31`	`nezZario`	`1) what's the cheapest way I can plug my landline into my freebsd server running asterisk?`
`07:24.45`	`nezZario`	`(it's sitting next to a telephone jack)`
`07:24.53`	`ChannelZ`	`An SPA3102 probably`
`07:25.39`	`Micc`	`ChannelZ, I can't wait for Haiku to get polished up and have vpn, wireless security and flash. Then I can use it every day.`
`07:25.52`	`nezZario`	`2) what's the big difference between asterisk 1.6.0`
`07:25.58`	`Micc`	`ChannelZ, and I'll write some Be apps for managing asterisk.`
`07:26.07`	`nezZario`	`and asterisk 1.6.2`
`07:26.32`	`nezZario`	`because freebsd ports is trying to install 1.6.0 ...`
`07:26.46`	`nezZario`	`it would be a lot easier to let it .. but if there is some major reason I should be using 1.6.2, let it be known`
`07:26.47`	`Micc`	`ChannelZ, have you tried Haiku yet?`
`07:27.04`	`ChannelZ`	`Micc: I did quite some time ago`
`07:27.28`	`Micc`	`ChannelZ, alpha 2 is pretty stable and functional.`
`07:27.36`	`nezZario`	`ahh SPA3102 is like a reverse ATA`
`07:27.43`	`nezZario`	`i was thinking more of a PCI card, but that works too`
`07:27.52`	`ChannelZ`	`What's a reverse ATA?`
`07:28.02`	`ChannelZ`	`The SPA3102 has an FXO and an FXS port.`
`07:28.13`	`ChannelZ`	`You can get a Digium card (or clone) but you said "cheapest"`
`07:28.32`	`nezZario`	`define: ata -- An adapter converts a traditional or analog phone into a VoIP phone without the need to buy a separate headset or terminal.`
`07:28.34`	`ChannelZ`	`Although I guess an Openvox card can be had for a little under $100 maybe`
`07:29.06`	`nezZario`	`e.g. i plug my 2 year old PAP2NA into the network and I can plug a $6 regular phone from wal-mart into it`
`07:29.10`	`ChannelZ`	`You said "plug my landline into my freebsd..."`
`07:29.27`	`nezZario`	`yes i know .. so I said the SPA3102 is kinda like a reverse ATA =)`
`07:29.28`	`ChannelZ`	`to me that means FXO`
`07:29.51`	`ChannelZ`	`The 3102 is both`
`07:29.58`	`nezZario`	`OH!`
`07:29.59`	`nezZario`	`gotcha`
`07:30.09`	`nezZario`	`can it multitask though?`
`07:30.29`	`nezZario`	`e.g. can I plug a landline IN and a PSTN phone OUT and have the asterisk server in the "middke"`
`07:30.34`	`ChannelZ`	`Yes`
`07:30.41`	`nezZario`	`middle*`
`07:30.51`	`nezZario`	`sweet, i might just try for one of those`
`07:31.10`	`ChannelZ`	`I was doing that for awhile, so calls come in, Asterisk does something, and rings my analog handset if it decided to`
`07:31.22`	`nezZario`	`there is no immediate need for one now .. but i was just curious if it can be done for $100 or less so when I can afford an office`
`07:31.29`	`ChannelZ`	`or I'd send the call SIP over the net to my phone at work`
`07:31.37`	`nezZario`	`but i really don't think my wife wants me setting up IVR prompts on our home telephone`
`07:31.46`	`nezZario`	`=)`
`07:31.56`	`ChannelZ`	`What is it you want to do with it then?`
`07:32.13`	`nezZario`	`when i get an office, they'll force me to buy a landline`
`07:32.16`	`nezZario`	`to get internet service`
`07:32.23`	`nezZario`	`so might as well use that DID for something..`
`07:32.43`	`nezZario`	`plus the major point is I could do outbound calls on the landline`
`07:32.47`	`nezZario`	`right?`
`07:32.54`	`ChannelZ`	`I dunno where you are but you can get what is essentially a DSL-only line with no telephone service on it`
`07:33.01`	`ChannelZ`	`but yeah`
`07:33.05`	`nezZario`	`ChannelZ: can't do it man`
`07:33.10`	`nezZario`	`I use embarq/centurylink`
`07:33.14`	`nezZario`	`trust me, i tried`
`07:33.36`	`ChannelZ`	`Hmm. I did on Qwest. Glad I finally moved off them though, considering Centurylink is buying them.`
`07:33.47`	`nezZario`	`yep they just snapped them up`
`07:34.00`	`nezZario`	`They're not BAD ... CenturyLink is better than Embarq was on it's own`
`07:34.17`	`nezZario`	`e.g. embarq would shut me off if i was a day late on the bill, no notice`
`07:34.18`	`ChannelZ`	`I moved my office lines and DSL away from them a few months ago, and just moved my home phone a couple weeks ago.`
`07:34.40`	`nezZario`	`centurylink calls me 3 times and sends a letter giving me payment arragements if I need them`
`07:35.27`	`Micc`	`I've been all voip at home and work for about 6 years. I'm glad I don't have to deal with qwest anymore.`
`07:35.32`	`nezZario`	`=D`
`07:35.36`	`nezZario`	`anyway ..`
`07:35.42`	`nezZario`	`about the 1.6.0 and 1.6.2 ?`
`07:35.45`	`*** join/#asterisk khussein78 (~khussein7@188.225.192.238)`
`07:35.51`	`nezZario`	`i'm thinking i'm just going to compile this thing manually`
`07:35.51`	`ChannelZ`	`anyways as for what changed between 1.6.0 and 1.6.2, read http://svn.asterisk.org/svn/asterisk/branches/1.6.2/CHANGES`
`07:35.56`	`nezZario`	`THANK YOU!`
`07:35.59`	`ChannelZ`	`I would`
`07:36.04`	`khussein78`	`hi`
`07:36.05`	`nezZario`	`i was looking for that ChannelZ, but couldn't find it`
`07:36.23`	`ChannelZ`	`it's broken up into two chunks, 1.6.1 to 1.6.2 and 1.6.0 to 1.6.1`
`07:36.26`	`Micc`	`nezZario, I doubt you'll need any of the changes in 1.6.1 or 1.6.2`
`07:36.29`	`khussein78`	`i am trying to connect cisco 7911 phone to asterisk but it cannot register`
`07:36.34`	`*** join/#asterisk Wildy (~simba@83.149.9.92)`
`07:36.45`	`khussein78`	`it always say unprovisioned`
`07:36.51`	`ChannelZ`	`but I'd just build 1.6.2 manually anyway, it's dead easy (well not sure of any extra freebsd-isms)`
`07:37.01`	`Micc`	`nezZario, 1.6.2 has siren7 and siren14 and some multi tenant parking fixes.`
`07:37.06`	`khussein78`	`i created SEP<MAC>.cnf.xml file`
`07:37.14`	`nezZario`	`i don't know what that means Micc, but i want it`
`07:37.23`	`nezZario`	`am I stupid, or does no one use mySQL anymore?`
`07:37.23`	`khussein78`	`here is my file syntax`
`07:37.41`	`nezZario`	`when i went to compile it under freebsd, the menuconf listed postgre and sqlite, but not mysql`
`07:37.43`	`ChannelZ`	`I use MySQL`
`07:37.58`	`ChannelZ`	`although not with asterisk`
`07:38.09`	`nezZario`	`any reason?`
`07:38.17`	`ChannelZ`	`no real need`
`07:38.28`	`nezZario`	`i fiddled with it on asterisk 1.4 eons ago, but never really got serious with it`
`07:38.29`	`khussein78`	`http://paste.ubuntu.com/451951/`
`07:38.31`	`Micc`	`nezZario, siren7 and siren14 are only on some high end conference phones and Polycom phones, so you won't need it. And multi-tenant parking you would only use if you had multiple offices you wanted to seperate the parking lots.`
`07:38.38`	`khussein78`	`can any one help me in this`
`07:38.43`	`ChannelZ`	`mysql can be used for CDR, but I don't care about that much, and realtime config which again I don't care about (not a big enough installation to need it)`
`07:39.00`	`nezZario`	`ChannelZ, i just like to be prepared for the future..`
`07:39.01`	`nezZario`	`=)`
`07:39.04`	`khussein78`	`i upgraded the phones to SIP by cisco callmanager ? is this may cause problems`
`07:39.14`	`nezZario`	`plus i like things managable from mySQL`
`07:39.33`	`nezZario`	`e.g. i run a postfix/dovecot mail server install that's totally backed by mySQL`
`07:39.35`	`nezZario`	`LOVE it`
`07:39.58`	`ChannelZ`	`I just switched over to dovecot a little while ago`
`07:40.05`	`nezZario`	`from courier-imap ?`
`07:40.11`	`ChannelZ`	`yeah. though I'm using qmail and vpopmail`
`07:40.16`	`nezZario`	`ahh gotcha`
`07:40.21`	`nezZario`	`i use dovecot LDA too`
`07:40.22`	`ChannelZ`	`so vpopmail is using mysql`
`07:40.31`	`nezZario`	`i guess everybody has their preference`
`07:40.35`	`ChannelZ`	`(well I suppose dovecot is too actually for the auth module, derr.)`
`07:40.36`	`Micc`	`sounds pretty cool.`
`07:40.53`	`nezZario`	`anyway, off to manually compile asterisk`
`07:41.27`	`Micc`	`I don't care much what flavor of sql, but I like my stuff in sql too.`
`07:41.48`	`Micc`	`I use a little postgresql, mysql, and mssql.`
`07:41.49`	`ChannelZ`	`khussein78: I don't know anything about that phone. Are you seeing failed registration attempts or any indication of anything happening on the console?`
`07:41.52`	`khussein78`	`on the phone i see error verifying config info`
`07:42.35`	`ChannelZ`	`hmm sounds like something happening on the phone then, I have no idea there. (You could turn on SIP debug on Asterisk and see if you're even getting any traffic from it..)`
`07:42.45`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`07:43.34`	`khussein78`	`ChannelZ, let me check sip debug`
`07:44.16`	`nezZario`	`the fun part is setting up this server to also act as a VPN router over GRE tunneling and throwing in some QoS for asterisk`
`07:44.29`	`nezZario`	`ohh so much to do for absolutely no pay =D`
`07:57.24`	`*** join/#asterisk uqlev (~yuriy@91.184.221.31)`
`07:57.25`	`khussein78`	`ChannelZ, nothing appears in SIP debug from this phones`
`08:14.05`	`nezZario`	`wth`
`08:14.07`	`nezZario`	`this thing says`
`08:14.17`	`nezZario`	`===> dahdi-2.3.0rc2 requires firmware(9) support for kernel modules with hyphens.`
`08:15.40`	`ChannelZ`	`what thing says that`
`08:16.05`	`nezZario`	`looks like a freebsd issue`
`08:16.08`	`nezZario`	`gotta update my gd kernel`
`08:16.41`	`ChannelZ`	`you don't really need dahdi anyway unless you've got telephony hardware`
`08:17.28`	`ChannelZ`	`or want to use MeetMe, but 1.6.2 has the new ConfBridge which doesn't use DAHDI timing`
`08:18.07`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`08:18.42`	`nezZario`	`might as well`
`08:20.07`	`ChannelZ`	`needs a shower`
`08:23.10`	`*** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za)`
`08:23.57`	`x-demon`	`is it possible to configure sip trunk failover?`
`08:24.03`	`doolittlework`	`hi there does anyone here have experience in dumping cdr records to a com port?`
`08:25.03`	`doolittlework`	`x-demon yes`
`08:25.33`	`x-demon`	`doolittlework, any how to? i see only guides for trixbox`
`08:26.02`	`doolittlework`	`doo you hav more than one sip trunk?`
`08:26.38`	`x-demon`	`doolittlework, yes`
`08:26.47`	`x-demon`	`oh looks like i found info`
`08:26.51`	*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
`08:27.05`	`x-demon`	`just add second trunk to context?`
`08:27.20`	`doolittlework`	`thats it`
`08:27.43`	`doolittlework`	`are u using freebpx?`
`08:27.53`	`x-demon`	`nope, plain asterisk`
`08:28.03`	`*** join/#asterisk Dark_Overload (~Dark_Over@36.140.100.97.cfl.res.rr.com)`
`08:28.16`	`Dark_Overload`	`hello`
`08:28.18`	`Dark_Overload`	`anyone awake?`
`08:28.28`	`x-demon`	`sort of`
`08:28.31`	`Dark_Overload`	`hi`
`08:28.34`	`doolittlework`	`yes Dark_Overload`
`08:28.39`	`Dark_Overload`	`you ever used ipcomms?`
`08:29.04`	`doolittlework`	`nope is it a sip trunk or hardware`
`08:29.39`	`Dark_Overload`	`http://www.ipcomms.net/product-freedid.html`
`08:29.48`	`Dark_Overload`	`its a trunk`
`08:30.09`	`doolittlework`	`whats the problem`
`08:30.37`	`Dark_Overload`	`ive got it routing calls to my asterisk, but it keeps saying the call is rejected because the extension is not found, and the extension is my phone number`
`08:30.44`	`x-demon`	`huh? signup page gives me 404...`
`08:30.50`	`Dark_Overload`	`yeah i know`
`08:31.01`	`Dark_Overload`	`you have to actually call them to sign up`
`08:32.00`	`Dark_Overload`	`lemme get a copy of the relevant contexts and messages into a pastebin hang on`
`08:32.12`	`doolittlework`	`Dark_Overload paste you cli call info`
`08:33.22`	`*** join/#asterisk BANSAL (~bansal@117.199.123.121)`
`08:38.30`	`Dark_Overload`	`http://pastebin.com/q5LPwYbu`
`08:38.32`	*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
`08:38.39`	`Dark_Overload`	`thats the error, i masked my last 4`
`08:38.54`	`Dark_Overload`	`also tagged on all changes that i have made beyond default config`
`08:39.47`	`joobie`	`Dark_Overload,`
`08:40.03`	`Dark_Overload`	`yeS?`
`08:40.12`	`joobie`	`401830xxxx do you have an extension like that defined in your extensions.conf/ael?`
`08:40.49`	`doolittlework`	`paste your extensions.conf?`
`08:41.01`	`Dark_Overload`	`it is still default`
`08:41.16`	`ChannelZ`	`the error says what what it means. There apparently is no matching extension in the 'from-pstn' context of your dialplan.`
`08:41.17`	`Dark_Overload`	`i haven't made any changes beyond the default asterisk config`
`08:41.23`	`joobie`	`that's ur problem`
`08:41.25`	`ChannelZ`	`well then how do you expect it to do anything?`
`08:41.31`	`joobie`	`something is coming in on your from-pstn context`
`08:41.34`	`joobie`	`to extesion 401830xxxx`
`08:41.35`	`*** join/#asterisk wubbla (~patrick@93.185.128.231)`
`08:41.36`	`wubbla`	`hi!`
`08:41.38`	`doolittlework`	`then it would no work you need 401830xxxx,1,Dial(SIP/myphone)`
`08:41.41`	`joobie`	`and because u have the default extensions.conf, you cant handle i`
`08:41.42`	`joobie`	`tt`
`08:41.57`	`Dark_Overload`	`i thought it came preconfigured with a menu system?`
`08:42.00`	`doolittlework`	`in your default contect`
`08:42.08`	`joobie`	`dood this aint windows`
`08:42.12`	`Dark_Overload`	`it includes demo`
`08:42.17`	`doolittlework`	`lol`
`08:42.19`	`ChannelZ`	`are you using FreepBX?`
`08:42.27`	`Dark_Overload`	`lol, no asterisk`
`08:42.57`	`joobie`	`format ur box`
`08:42.57`	`ChannelZ`	`well that's at least smart`
`08:42.59`	`joobie`	`and put windows on it`
`08:43.11`	`Dark_Overload`	`not on a server`
`08:43.20`	`Dark_Overload`	`none of my servers run windows`
`08:43.25`	`joobie`	`asterisk doesnt give u a gui itnerface`
`08:43.25`	`ChannelZ`	`anyways you have your answer`
`08:43.28`	`joobie`	`there are other projects`
`08:43.31`	`joobie`	`like freepbx`
`08:43.31`	`wubbla`	`can I somehow run external applications from the dialplan (for example with System()...) in the background i.e. so that asterisk does not need to wait until that application terminates again...`
`08:43.34`	`joobie`	`that give u a gui`
`08:43.39`	`joobie`	`out of th ebox`
`08:43.40`	`joobie`	`but`
`08:43.44`	`joobie`	`fuck all that.. just learn the cli`
`08:43.46`	`joobie`	`and use asterisk`
`08:43.48`	`joobie`	`it's more flexible`
`08:43.55`	`Dark_Overload`	`i love CLI`
`08:43.58`	`doolittlework`	`hi there does anyone here have experience in dumping cdr records to a com port?`
`08:44.00`	`joobie`	`good then fuk the gui`
`08:44.00`	`Dark_Overload`	`i just need to get the hang of it`
`08:44.06`	`Dark_Overload`	`everything else i use is clu`
`08:44.08`	`Dark_Overload`	`cli*`
`08:44.11`	`joobie`	`good`
`08:44.13`	`joobie`	`welcome`
`08:44.13`	`wubbla`	`something like the "&" operator for System() would be needed.... any ideas?`
`08:44.14`	`joobie`	`:P`
`08:44.15`	`Dark_Overload`	`meta... w3af`
`08:44.38`	`joobie`	`Dark_Overload, u give that iax a context`
`08:44.43`	`joobie`	`then when u receive a call on it`
`08:44.46`	`joobie`	`it comes in on that context`
`08:44.53`	`joobie`	`then u need to setup an extension to handle it`
`08:44.55`	`Dark_Overload`	`i thougt it came in on from-pstn`
`08:45.01`	`joobie`	`all this is in extensions.conf`
`08:45.02`	`joobie`	`brb gamen`
`08:45.31`	`Dark_Overload`	`ok so how do i define an extension and have it go into the default menu`
`08:45.41`	`ChannelZ`	`What default menu?`
`08:45.43`	`x-demon`	`by the way... do i really need to configure separate sip accounts for receiving and sending calls?`
`08:45.55`	`ChannelZ`	`and dolittlework told you above`
`08:46.00`	`Dark_Overload`	`i dunno mine is incoming only`
`08:46.12`	`x-demon`	`one provider, two accounts... or that's not needed?`
`08:46.20`	`ChannelZ`	`x-demon: well, yes in the sense of an ITSP. Not for a single device`
`08:46.28`	`x-demon`	`ITSP?`
`08:46.31`	`Dark_Overload`	`what specifically do i do to the extensions.conf to do this though?`
`08:46.32`	`doolittlework`	`joobie, do you know how to edit the master.csv file to change the duration into hhmmss format?`
`08:47.02`	`ChannelZ`	`exten => 401830xxxx,1,Dial(SIP/yourphone)`
`08:47.11`	`Dark_Overload`	`that will just sent it to my softphone`
`08:47.21`	`ChannelZ`	`or 'phone1' in your case I guess`
`08:47.30`	`Dark_Overload`	`i want it to goto the demo menu so i can learn the structure of it`
`08:47.50`	`doolittlework`	`then it would no work you need 401830xxxx,1,Dial(SIP/myphone&SIP/grandma)`
`08:47.56`	`Dark_Overload`	`thats how i learn things is by playing with them and looking at the code`
`08:48.08`	`Dark_Overload`	`wouldnt that dial two phones?`
`08:50.08`	`ChannelZ`	`There is nothing of real interest in the sample extensions.conf pre-built; it's mostly all fragments that explain specific functions`
`08:50.25`	`ChannelZ`	`you really need to read the book first`
`08:50.27`	`ChannelZ`	`~book`
`08:50.37`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`08:50.38`	`ChannelZ`	`kicks infobot`
`08:50.48`	`ChannelZ`	`~itsp`
`08:50.49`	`infobot`	`[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.`
`08:50.52`	`ChannelZ`	`x-demon: ^^^`
`08:51.07`	`Dark_Overload`	`i never do that, i jump in head first and learn as i go, referencing a manual and/or channel when neede`
`08:51.10`	`doolittlework`	`ChannelZ , do you know how to edit the master.csv file to change the duration into hhmmss format?`
`08:51.12`	`x-demon`	`Oh, so i need to configure separate in- and out- accounts`
`08:51.26`	`x-demon`	`thanks`
`08:51.26`	`ChannelZ`	`Generally for talking to an ITSP you have 2 SIP peers, one for incoming, one for outgoing.`
`08:51.42`	`ChannelZ`	`Only because they usually have different hostnames for each`
`08:52.40`	`Dark_Overload`	`thanks ChannelZ and joobie im gonna go see what i cant mess up in this conf ;]`
`08:53.41`	`ChannelZ`	`doolittlework: well it would be in cdr_custom.conf or such, but I dunno if there is a variable that outputs in the format you want`
`09:05.35`	`*** join/#asterisk ChannelZ (~bobm@burner.com)`
`09:05.36`	`x-demon`	`so for every incoming SIP i need register account with type=user?`
`09:11.53`	`ChannelZ`	`well yes but it depends`
`09:12.12`	`joobie`	`doolittlework, there is a config file custom_cdr.conf or sumthen`
`09:12.25`	`*** part/#asterisk ChannelZ (~bobm@burner.com)`
`09:12.26`	`joobie`	`u can define the way master.csv logs in there`
`09:12.31`	`*** join/#asterisk ChannelZ (~bobm@burner.com)`
`09:12.32`	`ChannelZ`	`oops`
`09:14.06`	`doolittlework`	`joobie & ChannelZ: i know how to get the data using master,csv and customxx but i need the billsec in hhmmss format currently only in ssssss seconds`
`09:15.46`	`ChannelZ`	`I told you, there probably isn't a variable that has it in the format that you want.`
`09:17.24`	`ChannelZ`	`are you parsing this in some other language to build a report? convert it yourself..`
`09:28.46`	`*** join/#asterisk QaDeS (~mklaus@p4FC71CD2.dip0.t-ipconnect.de)`
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`10:24.59`	`puzzled`	`hi`
`10:43.48`	`x-demon`	`what means "circuit-busy"?`
`10:44.08`	`x-demon`	`it happens on voip.ms provider on some destinations`
`10:44.18`	`x-demon`	`while working on some others`
`10:58.00`	`*** join/#asterisk sgimeno (~chatzilla@95.122.8.54)`
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`11:35.10`	`WIMPy`	`x-demon: It means there were insufficient resources on the network.`
`11:35.25`	`x-demon`	`WIMPy, on my server or?`
`11:36.59`	`WIMPy`	`Whereever that message came from.`
`11:37.56`	`WIMPy`	`AFAIK on SIP there is no such thing as the location field known from PSTN.`
`11:40.22`	`WIMPy`	`circuit busy usually translates from 'no circuit or channel available'.`
`11:47.10`	`x-demon`	`WIMPy, so it can be problems on sip trunk?`
`11:49.01`	`WIMPy`	`I'd expect another message if it happens on voip, but it could be anywhere.`
`11:49.25`	`WIMPy`	`Anywhere from the calling partys line up to the called partys line.`
`11:52.32`	`WIMPy`	`I usually end up using a real phone line, to find out, what went wrong. One of the drawbacks of using voip. (so far)`
`12:08.24`	`*** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69)`
`12:11.10`	`iscario`	`hi, i would like to know if it is possible to launch an AGI script from the dialplan in background (my script writes in the asterisk native db and has a wait function , so i can't wait to go to the next extend until it finished)`
`12:26.57`	`*** join/#asterisk BANSAL (~bansal@117.207.81.112)`
`12:32.07`	`*** join/#asterisk Godfather_ (~aaaa@135.Red-88-11-95.dynamicIP.rima-tde.net)`
`12:37.30`	`*** join/#asterisk danilom (~danilom@186.25.4.247)`
`12:38.43`	`danilom`	`can i add a trunk with ata+pstn line?? But the ata have dynamic ip`
`12:59.37`	`*** join/#asterisk viq (~viq@unaffiliated/viq)`
`13:04.44`	`*** join/#asterisk lost_soul (shackett@devio.us)`
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`13:24.30`	`*** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt)`
`13:24.34`	`[sr]`	`hi :P`
`13:29.14`	`WIMPy`	`lo`
`13:29.38`	`[sr]`	`hi WIMPy`
`13:38.28`	`*** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au)`
`13:38.51`	`x-demon`	`asterisk is really amazing`
`13:39.14`	`x-demon`	`anyone can recommend good books/mans for better understanding it?`
`13:39.26`	`WIMPy`	`~book`
`13:39.34`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`13:39.34`	`x-demon`	`preferably free, but paid is also okay`
`13:40.15`	`x-demon`	`whoa, book from oreilly for free?`
`13:40.41`	`*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)`
`13:40.51`	`*** join/#asterisk DennisG (~DennisG@84.30.136.208)`
`13:41.32`	`lost_soul`	`g'day everyone`
`13:47.32`	`[sr]`	`hi lost_soul, did you sleep well?`
`13:47.33`	`[sr]`	`:p`
`13:48.58`	`*** join/#asterisk danilom (~danilom@186-24-167-250.genericrev.telcel.net.ve)`
`13:49.27`	`lost_soul`	`[sr]: not to bad, and yourself?`
`13:49.34`	`danilom`	`please im a newbie, can someone answer me a question!`
`13:49.37`	`*** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d)`
`13:49.53`	`[sr]`	`im ok`
`13:49.53`	`[sr]`	`:p`
`13:50.04`	`WIMPy`	`danilom: Not unless you ask it.`
`13:50.07`	`lost_soul`	`danilom: they can't until you ask t`
`13:51.51`	`danilom`	`is possible to add a sip/trunk in asterisk server.. Using hostname and no a static ip? I have ata+fxo with pstn line, but have dynamic ip, so i need to use ddns, or hostname of the ata`
`13:52.29`	`[sr]`	`WIMPy: im going to setup my final box with asterisknow, a new compiled kernel, and LCR, this way i can have freepbx working and LCR`
`13:52.52`	`[sr]`	`already did a test on a instalation for that porpose and worked great with the compiled kernel`
`13:53.21`	`WIMPy`	`[sr]: Does Freepbx allow you to add things easily?`
`13:54.05`	`[sr]`	`WIMPy: yap, the problem is that it won't be just me adding things.. and the other persorns are just normal users, so i need freepbx`
`13:56.22`	`[sr]`	`WIMPy: the only think i don't like is the EXT3 filesystem :(, the fedora that's not a problem, i have several years of redhat also`
`13:56.22`	`*** join/#asterisk guilhermebr (~Guilherme@200.103.96.98)`
`13:57.37`	`[sr]`	`WIMPy: sorry not fedora, centOS, but that's the same`
`13:58.26`	`WIMPy`	`So you're kind of doing your own distro?`
`13:59.35`	`[sr]`	`sort of`
`13:59.43`	`[sr]`	`i have a instalation with asterisk+LCR+freepbx`
`13:59.59`	`*** join/#asterisk sgimeno (~chatzilla@95.122.8.54)`
`14:00.12`	`[sr]`	`with debian, but something on freepbx not working OK ... i have that's different context's`
`14:00.20`	`[sr]`	`havent got time to check it`
`14:03.33`	`[sr]`	`WIMPy: i believe my fault with freepbx, is that i install the asterisk samples, and then freepbx one's`
`14:03.52`	`[sr]`	`and that turns into a mismatch on the files/context's`
`14:05.09`	`WIMPy`	`Oh, almost certainly, yes.`
`14:06.17`	`[sr]`	`:)`
`14:06.35`	`[sr]`	`i have to find some time to confirm that, but that's not gonna be this weekend`
`14:06.52`	`[TK]D-Fender`	`[09:51]<danilom>is possible to add a sip/trunk in asterisk server.. Using hostname and no a static ip? I have ata+fxo with pstn line, but have dynamic ip, so i need to use ddns, or hostname of the ata <- Of course`
`14:07.03`	`[sr]`	`i want to make an install with LCR for ISDN, and DAHDI for FXS/FXO`
`14:07.58`	`jblack`	`Yeah, I do something like that. I hope your ip doesn't change too much, though, as there's a dns lookup hole.`
`14:08.22`	`jblack`	`They do a lookup, your ip changes before the lookup cache expires, you're not getting your call`
`14:09.27`	`beardy`	`Quite bad to run any server in such conditions. But it's better than not getting any call of course.`
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`15:35.26`	`mloe`	`Hi, guys, I buy VoIP number and the people from the company told me that I need to make SIP server to forward the calls there`
`15:35.46`	`mloe`	`I started SIP server, give them my IP and they told me that is ready`
`15:36.00`	`mloe`	`but what I need to do now?`
`15:36.32`	`mloe`	`they give me this information:IP Termination: 255.255.255.255, Manufacturer: OpenSer (Kamailio), SW Version: 1.4.3, IP Origination: 255.255.255.255, Manufacturer: Asterisk, SW Version: 1.4, H323 ID: n/a, Codec: G711a, g711, g729, gsm, Number Format: E.164 intl CC, Protocol: SIP`
`15:36.39`	`mloe`	`I changed the IPs :)`
`15:39.08`	`*** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)`
`15:49.00`	`puzzled`	`mloe: read the Asterisk book`
`15:50.57`	`mloe`	`ok`
`15:51.05`	`*** join/#asterisk Benwa (~Benwa@78.129.51.74)`
`15:53.31`	`*** join/#asterisk sputnick (~sputnick@unaffiliated/sputnick)`
`15:53.37`	`sputnick`	`hi there`
`16:06.57`	`*** join/#asterisk slackell (~zues@190.213.184.244)`
`16:08.07`	`slackell`	`hi all, i'm new to voip and asterisk as a whole and i want to setup a small home/office phone system`
`16:08.52`	`p3nguin`	`~answers`
`16:08.55`	`infobot`	`[~answers] 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`
`16:08.55`	`WIMPy`	`slackell: Sounds like you want to read`
`16:08.57`	`WIMPy`	`~`
`16:09.00`	`WIMPy`	`~book`
`16:09.01`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`16:09.07`	`slackell`	`is there a basic tutorial i can follow`
`16:09.08`	`slackell`	`?`
`16:09.54`	`p3nguin`	`I just provided five items.`
`16:10.08`	`*** join/#asterisk QaDeS (~mklaus@p4FC71CD2.dip0.t-ipconnect.de)`
`16:10.42`	`slackell`	`is one of them a step by step tutorial because i downloaded the oreily and it was a huge book with everything`
`16:10.46`	`p3nguin`	`That'll keep you busy the rest of the day.`
`16:10.48`	`*** join/#asterisk QaDeS_ (~mklaus@p4FC71CD2.dip0.t-ipconnect.de)`
`16:11.24`	`slackell`	`i know it make sound odd but i want to just setup a system without getting bogged down by many details`
`16:11.39`	`*** join/#asterisk QaDeS (~mklaus@p4FC71CD2.dip0.t-ipconnect.de)`
`16:11.51`	`p3nguin`	`Maybe you want someone else to do the work for you, then.`
`16:12.29`	`slackell`	`lol no no just wondering if there's proper tutorial with explicit instructions in existence`
`16:12.54`	`p3nguin`	`Yep, it's called The Book.`
`16:13.09`	`WIMPy`	`Or just use a phone without Asterisk.`
`16:13.26`	`lost_soul`	`lots of tutorials, but everyone has different needs/hardware`
`16:13.42`	`slackell`	`anyway I want to have 2 PC's be able to call each using my own PBX, the asteriskNOW, without need for a voip provider`
`16:14.04`	`p3nguin`	`Heck, with AsteriskNOW, you really don't even need The Book.`
`16:14.13`	`*** join/#asterisk ideaman (~ihaveapla@74-81-241-158.static.sdyl005.digis.net)`
`16:14.14`	`slackell`	`lol, thats part of the problem, many different ways`
`16:14.19`	`p3nguin`	`It certainly helps, though.`
`16:14.38`	`slackell`	`i can skim through portions of the book but i'm not looking to be a guru in this`
`16:16.00`	`lost_soul`	`slackell: it's always nice knowing how your equipment works though, just in case it needs to be fixed. So you should not dismiss researching and reading so quickly`
`16:16.19`	`slackell`	`so in summary i want to call pc1 from pc2 without using a pay-for voip provider. I'll have my own pbx the asteriskNOW on pc3.`
`16:16.42`	`lost_soul`	`or you could hire someone to set it all up if you want to remain hands off`
`16:16.43`	`p3nguin`	`And what was the problem?`
`16:17.00`	`WIMPy`	`Why don't you just call pc2 from pc1 without using a third party?`
`16:17.11`	`p3nguin`	`If you only want to call between two computers (using soft phones), you really don't need Asterisk.`
`16:17.30`	`slackell`	`i'm not downplaying its importance i just want directions/guidance on how to set it up, like a guide. With so many out there i figure someone here might have it done it b4 and can point me in the right direction`
`16:18.01`	`slackell`	`well pc1 to pc2 is simplified version`
`16:19.17`	`WIMPy`	`We certainly cannot help if you give false information.`
`16:19.54`	`ideaman`	`I have flowroute, outbound works fine, but on inbound, the one Sip phone I have registered will always ring, but 90% of the time, it will ring once, and I can't even answer it if I do get it in time. No Nat, no firewall. I'm not too great at reading the sip debugs, but when it goes bad, i get the Call leg/transaction does not exist message in the debug. Any thoughts?`
`16:20.00`	`slackell`	`i understand that sorry, let me start over, confusin myself here lol`
`16:22.04`	`slackell`	`Want: to be able to use the net, ie voip, to make/receive calls for free, from my home to other phones`
`16:22.19`	`p3nguin`	`ideaman: You aren't using a Google Voice phone number, forwarding it to your flowroute account, are you?`
`16:22.27`	`ideaman`	`no`
`16:22.44`	`ideaman`	`just using a did i purchased from flowroute and stuck it in the dialplan`
`16:23.16`	`ideaman`	`i only have about 4 lines in my extensions.conf for this feature`
`16:23.57`	`p3nguin`	`While the problem likely does not exist in the dialplan, go ahead and paste it in pastebin.com.`
`16:24.23`	`ideaman`	`and i dont even know that it's something related to this phone, because i tried replacing the dial line with just a say digits command and still same problem`
`16:24.36`	`ideaman`	`only said digits about 10 percent of the time`
`16:24.47`	`p3nguin`	`I experience that exact problem when I use google voice forwarded to my ipkall phone number (which is delivered to me via sip).`
`16:25.02`	`ideaman`	`ahh`
`16:25.28`	`p3nguin`	`But calling the ipkall number directly works properly.`
`16:25.35`	`ideaman`	`interesting`
`16:25.52`	`p3nguin`	`annoying, too!`
`16:26.08`	`ideaman`	`so I've never used pastebin.com before since I'm new at this whole asterisk thing. Is it just where I can past my code, and get help on it?`
`16:27.18`	`p3nguin`	`You copy any text that you want other people to see, paste it in the box on the pastebin.com site, select the options at the bottom, submit the paste, then provide the URL to the people whom you want to view it.`
`16:27.52`	`ideaman`	`got it. gimme a sec to remove some secure info and ill create an account and paste it`
`16:28.19`	`p3nguin`	`There shouldn't be a whole lot of private information in a dialplan.`
`16:28.48`	`ideaman`	`so i just need to paste in the extensions.conf?`
`16:29.06`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`16:29.26`	`p3nguin`	`I'm interested in seeing the extension that is reached when a call comes in (where the problem exists).`
`16:29.33`	`ideaman`	`k`
`16:30.26`	`p3nguin`	`Also, information that you think is private probably isn't, and hiding it often makes it hard for people to help. The standard rule when providing pastes of configuration and debug info is to hide ONLY your passwords.`
`16:30.48`	`ideaman`	`k`
`16:33.51`	`slackell`	`so is it possible to be able to use the net, ie voip, to make/receive calls for free, from my home to other phones? How to go about setting up this system with minimal/no cost/additional hardware and is there a guide/tutorial that i can follow?`
`16:34.27`	`WIMPy`	`It's still the book.`
`16:34.34`	`slackell`	`oki`
`16:34.36`	`lost_soul`	`slackell: pc to pc free, pc to normal phone you will need to pay`
`16:34.45`	`lost_soul`	`generally anyways`
`16:35.02`	`*** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com)`
`16:35.13`	`slackell`	`ok i'm currently goin through the links provided earlier`
`16:35.28`	`lost_soul`	`I used a magicjack.. cheapest deal I could find`
`16:35.31`	`p3nguin`	`~itsp`
`16:35.32`	`infobot`	`[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.`
`16:36.11`	`ideaman`	`p3nguin > http://pastebin.com/hp7pBabZ`
`16:36.46`	`slackell`	`cool lost_soul. thats where the pc to pc thing and ip phones came in.`
`16:37.11`	`ideaman`	`p3nguin > did that work?`
`16:37.24`	`*** join/#asterisk MooseMorals (~chatzilla@cpc2-newc2-0-0-cust234.gate.cable.virginmedia.com)`
`16:37.32`	`p3nguin`	`ideaman: You want me to be able to call your phone number and dial out to other phone numbers, making you pay for the phone bill?`
`16:38.07`	`ideaman`	`ummm no`
`16:38.12`	`p3nguin`	`When you include [outgoing] in your [incoming] context, that's what you're doing.`
`16:38.23`	`ideaman`	`ohh really?`
`16:38.27`	`p3nguin`	`yes`
`16:38.55`	`ideaman`	`how do i make a phone do what it does with it's incoming context, but still be able to use flowroutes outgoing context?`
`16:39.53`	`p3nguin`	`In sip.conf, each phone has a peer definition. Each phone's peer definition needs to include a context for dialing. That context could be outgoing if that's the only context you want phones to have access to.`
`16:40.05`	`WIMPy`	`Hmm. Is SayDigits a backdoor like SayUnixTime?`
`16:40.16`	`p3nguin`	`context=outgoing`
`16:40.50`	`ideaman`	`so if i set phone 8000 up with the context for outgoing, that's the only place i need to worry about it getting it's outgoing context from?`
`16:40.59`	`sputnick`	`What is the best way to make a call from a script from a "server A" to asterisk "server B" ? I just need to get the idea to take the good way ;)`
`16:41.06`	`p3nguin`	`I personally use context=phones, then the phones context includes some other contexts that I want the phones to have access to.`
`16:41.34`	`MooseMorals`	`can I use astersisk with softphones that are on a different lan to the server? (ie, with the server in a well connected rack somwhere and the softphone on a laptop behind ADSL)?`
`16:41.51`	`p3nguin`	`but my incoming never includes outgoing.`
`16:42.00`	`ideaman`	`thanks for the heads up`
`16:42.10`	`ideaman`	`im just figuring things out as i go`
`16:42.15`	`WIMPy`	`sputnick: Call files or AMI.`
`16:42.26`	`p3nguin`	`Unfortunately that does not solve the issue that you were having. :/`
`16:42.27`	`WIMPy`	`MooseMorals: yes`
`16:42.30`	`ideaman`	`true`
`16:42.35`	`ideaman`	`but learned something very important`
`16:42.53`	`ideaman`	`one of the good reasons at least that i keep a very low balance in my flowroute as i am learning`
`16:42.54`	`MooseMorals`	`Ok, thanks, I'll keep looking for docs then :)`
`16:43.04`	`p3nguin`	`Good idea!`
`16:43.12`	`sputnick`	`WIMPy: I use Call files, si I will have to fill a file by example via ftp and then a script on asterisk server have to move it to the queue ? that's it ?`
`16:43.24`	`sputnick`	`s/si/so`
`16:43.33`	`*** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)`
`16:44.00`	`p3nguin`	`Oh, so close!`
`16:44.18`	`p3nguin`	`s/so close/almost/`
`16:44.29`	`WIMPy`	`sputnick: Ah, that way from another server. Well, it's still possible, yes, and you can move via ftp as well.`
`16:44.46`	`WIMPy`	`sputnick: But maybe ftp isn't the most secure way.`
`16:45.14`	`sputnick`	`scp or rsync need a blank passphrase, so...`
`16:45.26`	`p3nguin`	`ideaman: When you get a call, how many digits are being played before the call breaks?`
`16:45.31`	`WIMPy`	`Or a key.`
`16:45.35`	`sputnick`	`a rsync daemon is a better solution`
`16:45.53`	`sputnick`	`no need key exchange with this`
`16:45.56`	`ideaman`	`it plays them all when it plays them, it's just sometimes it will ring once then hangup and not play anything`
`16:46.03`	`ideaman`	`so it rings every single time`
`16:46.11`	`ideaman`	`but 90% of the time it's a ring once and hangup`
`16:46.23`	`ideaman`	`and i cant even answer it in that one ring anyway if i try`
`16:46.29`	`WIMPy`	`sputnick: IIRC rsync always generates a temp file and renames it after successfull transfer, so that might be very hand.`
`16:46.31`	`WIMPy`	`y`
`16:46.39`	`ideaman`	`but 10% of the time, it works great, rings plays all the digits`
`16:46.51`	`p3nguin`	`ideaman: Do you know how to get a sip debug of a call?`
`16:47.15`	`*** join/#asterisk ruben23 (~ITadmin@125.212.40.2)`
`16:47.18`	`ideaman`	`i think so, but i dont have remote access to the server today`
`16:47.25`	`slackell`	`hey i'm going to be a long while reading these links so i just want to say Thank You! to those that helped me`
`16:47.26`	`ideaman`	`i will on monday`
`16:47.52`	`p3nguin`	`I'd be interested in a sip debug to see if something says why the call terminates early.`
`16:47.53`	`sputnick`	`thanks WIMPy. I will search AMI API too, maybe there's client/server features and output error to process`
`16:48.01`	`ideaman`	`i do remember one line out of there`
`16:48.21`	`ideaman`	`2 of the last 4 responses were a 481 call leg not.......`
`16:48.29`	`ideaman`	`does not exist`
`16:49.22`	`*** part/#asterisk slackell (~zues@190.213.184.244)`
`16:59.12`	`*** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr)`
`17:01.02`	`*** join/#asterisk Wildy (~simba@178.17.198.201)`
`17:01.13`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`17:01.59`	`ideaman`	`<PROTECTED>`
`17:02.35`	`p3nguin`	`Just come back here, describe the problem, and provide the sip debug. If I'm not here, surely someone else will be.`
`17:02.44`	`ideaman`	`k`
`17:04.20`	`ideaman`	`do you have any ideas on what it might be, or too many possibilities?`
`17:06.42`	`*** join/#asterisk x303 (~x303@97.101.116.106)`
`17:06.47`	`p3nguin`	`Without seeing a sip debug, I couldn't even try to guess.`
`17:07.08`	`p3nguin`	`Even if I see the debug, I might not know -- someone else might be able to tell, though.`
`17:07.18`	`[sr]`	`WIMPy: doesn't have any option for FXO/FXS card's right?`
`17:07.20`	`ideaman`	`k. thanks for all your help today!`
`17:07.44`	`WIMPy`	`[sr]: Pardon?`
`17:07.53`	`[sr]`	`WIMPy: ops, LCR, forgot to say that`
`17:08.29`	`WIMPy`	`No. Ist pure ISDN.`
`17:08.37`	`WIMPy`	`Almost`
`17:08.55`	`[sr]`	`nice`
`17:09.01`	`WIMPy`	`It has support for OpenBSC.`
`17:09.02`	`[sr]`	`i have to really use dahdi`
`17:09.27`	`WIMPy`	`It used to have H.323 support but since chan_lcr that's gone.`
`17:10.24`	`[sr]`	`i see`
`17:10.46`	`[sr]`	`what i need is to route certain number to a fisical fax machine, i'll dahdi plus a FXS card for that`
`17:10.53`	`[sr]`	`i'll use dahdi i mean`
`17:11.46`	`WIMPy`	`You could also use an ISDN TA. Either behind or even in parallel to LCR/Asterisk.`
`17:12.26`	`[sr]`	`hum`
`17:12.31`	`WIMPy`	`That should be a lot cheaper and easier, especially in the later case.`
`17:12.36`	`[sr]`	`i have some TA modems`
`17:12.47`	`[sr]`	`i see what u mean`
`17:12.54`	`[sr]`	`the TA works in NT mode`
`17:12.56`	`[sr]`	`right?`
`17:13.01`	`WIMPy`	`TA a/b`
`17:13.20`	`WIMPy`	`Ne. TAs are always TE.`
`17:13.24`	`WIMPy`	`no`
`17:13.33`	`[sr]`	`hum`
`17:13.50`	`[sr]`	`i'll do the test`
`17:13.58`	`[sr]`	`i'll save arround 200â¬ in a FXS card`
`17:14.35`	`[sr]`	`WIMPy: thanks for the idea :)`
`17:15.24`	`WIMPy`	`If you already planned to spend the 200 quid, you can just send them to me :-))`
`17:15.37`	`[sr]`	`hum`
`17:15.48`	`[sr]`	`i'll send half then :P`
`17:16.30`	`[sr]`	`WIMPy: the TA stuff, in the past i used to use it to have analog lines, using it`
`17:16.36`	`[sr]`	`so i'm into it how it works`
`17:34.19`	`*** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net)`
`17:34.26`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`17:39.07`	`*** join/#asterisk naif (5d209ada@gateway/web/freenode/ip.93.32.154.218)`
`17:39.08`	`*** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt)`
`17:39.10`	`[sr]`	`im back`
`17:39.21`	`naif`	`I am looking for a base platform to do SIP/TLS+SRTP proxying along with AMR -> ULaw transcoding . Can asterisk be a based to do so?`
`17:39.30`	`*** join/#asterisk QaDeS (~mklaus@p4FC71CD2.dip0.t-ipconnect.de)`
`17:39.31`	`naif`	`FreeSWITCH does not act as a SIP proxy`
`17:39.51`	`naif`	`Kamailio support SIP/TLS but RTPProxy does not support SRTP nor transcoding`
`17:40.08`	`naif`	`Can asterisk do that? otherwise there is a good hole in opensource voip support stuff and need to do coding`
`17:44.13`	`Tim_Toady`	`wait till asterisk 1.8 :P`
`17:45.07`	`Tim_Toady`	`atm asterisk 1.6 supports sip/tls but no srtp`
`17:45.19`	`*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)`
`17:45.20`	`Tim_Toady`	`1.8 will have srtp support too`
`17:53.35`	`*** join/#asterisk sekil (~sekil@80.93.247.26)`
`17:57.00`	`*** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com)`
`17:58.39`	`[sr]`	`for what i've read, asterisk 1.8x will the best version ever!`
`18:00.00`	`*** join/#asterisk Alagar (~Administr@122.164.34.30)`
`18:05.24`	`*** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)`
`18:07.28`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`18:08.23`	`Wildy`	`[sr] uh huh, it'd support new exciting glitches and misfeatures :P`
`18:09.13`	`Wildy`	`okay okay, but i'm kinda stuck with (overly) sophisticated FreePBX setup on underprovisioned server with a lot of load on it`
`18:11.32`	`[sr]`	`:)`
`18:30.21`	`devdvd`	`hey everyone, running into a bit of an odd issue. I have a feeling its somehow related to nat but ive never experienced this behavior. I have a polycom 321 running on asterisk 1.4.32. When I dial the phone from an extension, it works fine, i can hear the other side and they can hear me.`
`18:30.24`	`devdvd`	`When they call an external phone through the sip trunk, both parties can hear just fine but if an outside party calls and connects to the phone, the person on the polycom can be heard but not the person on the other side`
`18:30.42`	`devdvd`	`p3nguin pasted me some links to check out last night`
`18:30.53`	`devdvd`	`and i already had what was there`
`18:30.58`	`devdvd`	`as far as settings`
`18:31.21`	`devdvd`	`now heres something thats really perplexing me, as if the whole sitation wasnt odd enough`
`18:31.41`	`*** join/#asterisk MooseMorals (~chatzilla@cpc2-newc2-0-0-cust234.gate.cable.virginmedia.com)`
`18:31.42`	`devdvd`	`eventhough i can hear the person on the polycom phone`
`18:31.49`	`devdvd`	`i see nothing in rtp debug`
`18:32.05`	`WIMPy`	`"reinvite"`
`18:34.40`	`devdvd`	`i dont even see anything leaving the box for rtp traffic when i do a tcpdump eventhough i can hear the person on the polycom phone`
`18:34.50`	`devdvd`	`WIMPy: i set reinvite=no`
`18:35.24`	`WIMPy`	`"canreinvite" might work better :-)`
`18:35.32`	`devdvd`	`LOL`
`18:37.06`	`*** join/#asterisk trumee (~nobody@cpc2-cmbg15-0-0-cust1000.5-4.cable.virginmedia.com)`
`18:37.16`	`trumee`	`<PROTECTED>`
`18:37.23`	`trumee`	`<PROTECTED>`
`18:37.30`	`trumee`	`<PROTECTED>`
`18:37.30`	`devdvd`	`WIMPy: i love you...i fancy myself as someone with at least a minor intelligence..but there are times when i even have to look in the mirror and go wtf dude`
`18:37.56`	`WIMPy`	`devdvd: :)`
`18:38.02`	`*** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com)`
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`18:38.42`	`[TK]D-Fender`	`trumee: tried "fromdomain="?`
`18:39.08`	`trumee`	`[TK]D-Fender: no didnot. but i will check this out`
`18:40.57`	`trumee`	`[TK]D-Fender: thanks that worked`
`18:41.28`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`18:42.02`	`[TK]D-Fender`	`trumee: You might try realing the sample files and looking for obvious looking parameters... at like... stuff`
`18:43.04`	`trumee`	`[TK]D-Fender:yup, will look into them.`
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`19:15.36`	`Dark_Overload`	`hey anyone know how to call stdexten, i keep getting invalid address errros`
`19:16.28`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`19:17.11`	`*** part/#asterisk sekil (~sekil@80.93.247.26)`
`19:24.29`	`[TK]D-Fender`	`Dark_Overload: How are we to know what "stdexten" is supposed to do? This is YOUR dialplan.`
`19:24.39`	`[TK]D-Fender`	`DarkPerhaps you should actually show us the failed call to look at`
`19:24.43`	`[TK]D-Fender`	`~pb`
`19:24.44`	`infobot`	`[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.`
`19:24.46`	`[TK]D-Fender`	`^^^^^^^`
`19:31.17`	`hardwire`	`heh`
`19:41.18`	`*** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001)`
`19:47.53`	`*** join/#asterisk x303 (~x303@97.101.116.106)`
`19:51.08`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`19:52.28`	`ChannelZ`	`weird. The first time I use Zapateller it only makes a small little beep. The second time I use it, it's correct.`
`19:52.35`	`*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)`
`19:54.26`	`[TK]D-Fender`	`ChannelZ: I'd bet the audio stream isn't up fast enough. Answer, Playback(silence/2), then call it`
`19:56.47`	`*** join/#asterisk guilhermebr (~Guilherme@189.63.66.47)`
`19:59.00`	`ChannelZ`	`well I was doing an Answer(1500) which is enough for normal calls (which usually do get cut off at the head if I don't do that).`
`19:59.51`	`ChannelZ`	`But it's interesting, if I watch the console you can see it skip right past the first call of Zapateller really fast. IE you see the line come up of it executing that application, and then it immediately goes on to the next.`
`20:02.15`	`ChannelZ`	`I think there's something wrong with Answer(xxx). If I just Answer and then Wait(2) it works. If I Answer(2000) it doesn't.`
`20:02.58`	`ChannelZ`	`I also see if I do Answer(3000), I will get this "Exceptionally long voice queue length queuing to DAHDI/1-1" error`
`20:04.32`	`*** join/#asterisk aziyuu (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net)`
`20:05.12`	`aziyuu`	`i try to build kind of billing management system for asterisk for calshop`
`20:05.37`	`*** join/#asterisk mascool (~mascool@75.80.79.97)`
`20:05.38`	`*** join/#asterisk tris (tristan@camel.ethereal.net)`
`20:06.00`	`aziyuu`	`what asterisk file do i need to build it ?`
`20:06.28`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
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`20:11.50`	`p3nguin`	`Answer(1500) would cause a Wait() before the Answer() rather than an Answer() and then waiting, doesn't it? Maybe I have that backward.`
`20:12.18`	`p3nguin`	`I never use Answer(delay), so I'm possible mistaken.`
`20:12.52`	`[TK]D-Fender`	`The delay is BEFOER answer, not after`
`20:12.55`	`jblack`	`I didn't know answer took an option.`
`20:12.56`	`[TK]D-Fender`	`BEFORE*`
`20:13.00`	`[TK]D-Fender`	`hence no good`
`20:13.08`	`p3nguin`	`That's what I thought.`
`20:13.36`	`aziyuu`	`r u talking to me ?`
`20:15.29`	`p3nguin`	`So the better way would be Answer() followed by Wait(1.5)`
`20:15.51`	`p3nguin`	`or to play silence, as was already suggested.`
`20:16.36`	`[TK]D-Fender`	`aziyuu: Go read up on CDR in THE BOOK`
`20:16.37`	`[TK]D-Fender`	`~book`
`20:16.38`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`20:16.39`	`[TK]D-Fender`	`^^^`
`20:20.28`	`tzafrir`	`aziyuu, there are a number of asterisk-based call-shop systems. Though I'm not really familiar with any of them`
`20:20.41`	`*** join/#asterisk paule22 (~auf@erft-5d80ccec.pool.mediaWays.net)`
`20:20.45`	`paule22`	`hello`
`20:20.58`	`Chainsaw`	`Hi paule22.`
`20:21.12`	`ChannelZ`	`TK: really? The help probably shouldn't say "If a delay is specified,`
`20:21.12`	`ChannelZ`	`Asterisk will wait this number of milliseconds before returning to`
`20:21.12`	`ChannelZ`	`the dialplan after answering the call" then.`
`20:21.33`	`*** join/#asterisk sahafeez (~sahafeez@99-40-7-14.lightspeed.sntcca.sbcglobal.net)`
`20:21.49`	`paule22`	`i have download and make make install make samples`
`20:21.50`	`ChannelZ`	`It also does indeed answer the channel and pause, else I'd hear the line keep ringing for several seconds`
`20:22.15`	`paule22`	`1.6.2.9`
`20:22.49`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`20:22.52`	`paule22`	`now, how can i connect to a fritz nox?`
`20:23.01`	`paule22`	`now, how can i connect to a fritz box?`
`20:23.12`	`ChannelZ`	`You need to plug it into the froozle widget`
`20:23.13`	`*** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com)`
`20:23.31`	`ChannelZ`	`And then press the dingle buzzer 3 or 4 times`
`20:25.46`	`aziyuu`	`thanks for your reply tzafrir`
`20:25.59`	`aziyuu`	`what do you mean by that ?`
`20:26.03`	`paule22`	`when i start asterisk -vvvg and do iax2 show peers, i get message 0 online and 0 offline (demo/asterisk ...)`
`20:26.10`	`aziyuu`	`tzafrir:`
`20:27.33`	`tzafrir`	`aziyuu, I'm not really familiar with that, but a quick serch gives: http://duckduckgo.com/?q=asterisk+call+shop`
`20:29.19`	`*** join/#asterisk nicoAMG (~nicoamg@201.237.49.131)`
`20:33.06`	`aziyuu`	`thanks tzafrir`
`20:33.59`	`[sr]`	`WIMPy: is there any interface available/used for realtime function?`
`20:34.29`	`WIMPy`	`Realtime as in what?`
`20:34.41`	`[sr]`	`asterisk realtime`
`20:36.32`	`[sr]`	`im just investigating for now`
`20:38.15`	`WIMPy`	`I'm pretty sure I've seen functions for that, but I never used realtime.`
`20:38.20`	`[TK]D-Fender`	`[sr]: what "interface"? Could you be a little more vague please...`
`20:38.45`	`[sr]`	`[TK]D-Fender: web interface, to insert the records into the database`
`20:41.19`	`[TK]D-Fender`	`[sr]: ... phpmyadmin <-`
`20:41.54`	`*** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-72-83.home.otenet.gr)`
`20:43.47`	`[sr]`	`[TK]D-Fender: it's an option :)`
`20:44.25`	`*** join/#asterisk kdas (~user@c-24-5-192-88.hsd1.ca.comcast.net)`
`20:44.31`	`kdas`	`hey all`
`20:44.36`	`[sr]`	`[TK]D-Fender: ok but doesn't exists right?`
`20:46.17`	`[TK]D-Fender`	`[sr]: no`
`20:46.34`	`[sr]`	`[TK]D-Fender: hum oki, was just curious`
`20:46.53`	`*** join/#asterisk Poincare (~jefffnode@v74.ampersant.be)`
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`20:58.17`	`kdas`	`<PROTECTED>`
`20:58.22`	`kdas`	`what the hell does that mean ?`
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`21:12.14`	`*** part/#asterisk MooseMorals (~chatzilla@cpc2-newc2-0-0-cust234.gate.cable.virginmedia.com)`
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`21:13.38`	`ChannelZ`	`kdas: it means a call on a channel started, or at least tried to`
`21:14.04`	`kdas`	`ChannelZ: yea i had a woking sip.conf now it all went to hell`
`21:14.33`	`kdas`	`ChannelZ: i am just trying to register to voipgain and have all incoming calls forwarded to my mobile number`
`21:17.57`	`[TK]D-Fender`	`kdas: And so far that line you pasted means nothing. Try posting actual debug from your calls, or registration if that's all you've got for lack of inbound traffic at all`
`21:18.11`	`[TK]D-Fender`	`kdas: SIP DEBUG <-----------------`
`21:19.56`	`kdas`	`[TK]D-Fender: dude, i connect to asterisk -vvvr then i make a phone call to the number for voipgain and all i see is == Using SIP RTP CoS mark 5`
`21:20.26`	`kdas`	`[TK]D-Fender: hmm now i am seeing this too [Jun 19 14:18:15] NOTICE[3466]: chan_sip.c:20059 handle_request_invite: Call from 'kdasuser' to extension 's' rejected because extension not found.`
`21:21.10`	`[TK]D-Fender`	`kdas: SIP DEBUG <-----------------`
`21:21.37`	`[TK]D-Fender`	`kdas: Look ath what peer (if any) it is actually matching, and in which CONTEXT it's looking for that extension in`
`21:21.45`	`kdas`	`No such command 'SIP DEBIG' (type 'core show help SIP DEBIG' for other possible commands)`
`21:21.51`	`[TK]D-Fender`	`kdas: And then go make sure that it exists`
`21:21.51`	`kdas`	`debug*`
`21:22.14`	`[TK]D-Fender`	`kdas: help sip" <--- use some imagination here`
`21:22.24`	`kdas`	`[TK]D-Fender: i dont get what you mean by SIP DEBUG <-----------------------`
`21:22.38`	`[TK]D-Fender`	`kdas: Syntax varies by version and you didn't bother to tell us that either`
`21:22.57`	`kdas`	`Asterisk 1.6.2.6-1,`
`21:23.01`	`[TK]D-Fender`	`kdas: ENABLE SUIP DEBUG. Is this not clear?`
`21:24.24`	`kdas`	`[TK]D-Fender: i put it on but just spamms my screen`
`21:24.44`	`[TK]D-Fender`	`kdas: Maybe what it is saying is IMPORTANT and yo should look when you PLACE A CALL.`
`21:25.17`	`*** join/#asterisk MooseMorals (~chatzilla@cpc2-newc2-0-0-cust234.gate.cable.virginmedia.com)`
`21:28.51`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`21:37.09`	`[TK]D-Fender`	`ok, I'm off... BBL`
`21:40.47`	`[sr]`	`bye [TK]D-Fender`
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`22:26.51`	`kdas`	`ok there should be a better way of doing this... so please help me out. I have my ipkall -> callwithus <->asterisk is set that when i recieve a call from callwithus to make a call to MYphone@voipgain`
`22:27.18`	`kdas`	`in essence i am trying to get any call from the ipkall number forwarded to my pstn phone`
`22:29.55`	`*** join/#asterisk ManxPower (~manxpower@253.sub-75-235-20.myvzw.com)`
`22:30.26`	`p3nguin`	`Why do you need the extra bit between ipkall and asterisk?`
`22:31.13`	`p3nguin`	`IPkall is able to send calls to your asterisk system via SIP or IAX2. Based on the extension that you set it to call, you can then make Asterisk call whatever you want.`
`22:32.31`	`kdas`	`what do i put for ipkall sip phone number? just an entry to sip.conf?`
`22:32.48`	`p3nguin`	`I have two peers in sip.conf for ipkall (because they use two IP addresses). The peer definitions allow me to send calls into a given context, which contains the extension that I want the calls to go to (I use my IPkall phone number).`
`22:33.45`	`kdas`	`hmm`
`22:34.38`	`kdas`	`do you mind explaining to me how to do it?`
`22:35.00`	`p3nguin`	`In your ipkall settings on their web site...`
`22:35.20`	`p3nguin`	`You have to supply two pieces of information on how to get calls to your phone.`
`22:35.20`	`kdas`	`yea..`
`22:35.31`	`kdas`	`yea i have my asterisk box ip already`
`22:35.32`	`kdas`	`then ?`
`22:36.08`	`p3nguin`	`I do not remember what they are labeled, but one I enter my phone number and in the other goes my hostname.`
`22:36.30`	`kdas`	`SIP Phone Number: you enter what?`
`22:36.37`	`p3nguin`	`Now calls from ipkall will be delivered via SIP URI to me at myphonenumber@myhostname.`
`22:36.39`	`kdas`	`SIP Proxy: i entered my hostname`
`22:36.56`	`kdas`	`i actually entered my ip`
`22:37.04`	`p3nguin`	`Put in your IPkall phone number in the first one and your asterisk's hostname or address in the other.`
`22:37.37`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`22:38.20`	`kdas`	`i see the problem now`
`22:38.46`	`kdas`	`i have sip.conf generl to set all calls that are not registered to goto INVALID which just hangsup`
`22:38.54`	`kdas`	`so i need to change that to goto INCOMING ?`
`22:39.16`	`kdas`	`or is there a way for ipkall to go through a peer in my sip.conf ?`
`22:39.43`	`p3nguin`	`http://pastebin.com/VrgjVTA8`
`22:40.11`	`p3nguin`	`This is how I get my IPkall calls into their own context rather than the default one.`
`22:40.29`	`kdas`	`cant we have host=dynamic? or will it not catch it?`
`22:40.42`	`p3nguin`	`You don't need it to be dynamic.`
`22:40.49`	`p3nguin`	`You know their two IP addresses.`
`22:40.59`	`p3nguin`	`and they do not register to you.`
`22:41.11`	`kdas`	`so i have to have their ips`
`22:41.13`	`kdas`	`?`
`22:41.26`	`p3nguin`	`I've provided you with the EXACT info to put in your sip.conf.`
`22:41.36`	`p3nguin`	`copy what I provided and paste it into your sip.conf.`
`22:42.16`	`p3nguin`	`My config uses the context of [from-ipkall] for calls. You might want to call yours something else if you don't like what I called it.`
`22:43.14`	`kdas`	`yea i did`
`22:43.17`	`p3nguin`	`Then, in my [from-ipkall] context in extensions.conf, I make extensions for my IPkall phone numbers.`
`22:43.47`	`kdas`	`but i am just wondering do i have to put thier IPs ? why does dynamic not work? is it because they dont register to the asterisk box?`
`22:44.08`	`p3nguin`	`If you used your real IPkall phone number in the "SIP Phone Number:" box on their site, you'll use your real IPkall phone number in the extension.`
`22:44.20`	`p3nguin`	`1740.49) <p3nguin> You know their two IP addresses.`
`22:44.20`	`p3nguin`	`(1740.59) <p3nguin> and they do not register to you.`
`22:44.42`	`p3nguin`	`That's why. Dynamic is for hosts which you do not know their IP addresses and they DO register to you.`
`22:45.23`	`kdas`	`ok cool`
`22:45.28`	`kdas`	`thanks soo much for your help`
`22:45.46`	`kdas`	`now i just need to tweak my extensions and see if this will work`
`22:46.00`	`p3nguin`	`exten => 4251234567,1,Dial(something here)`
`22:46.19`	`p3nguin`	`If you want it to call a phone of yours, Dial(SIP/your-phone).`
`22:46.23`	`kdas`	`do you happen to know a cheap voip provider for uk mobile 4479244 prefix ?`
`22:46.47`	`p3nguin`	`~itsplist-uk`
`22:46.48`	`infobot`	`methinks itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google.`
`22:47.40`	`kdas`	`will check them out. I am getting 5.9 per min right now`
`22:50.17`	`p3nguin`	`You want a DID with that prefix or just a termination provider with good rates to it?`
`22:50.44`	`p3nguin`	`Are you in UK, US, or some other place?`
`22:52.24`	`kdas`	`i am in us wantting good rates to that prefix`
`22:54.15`	`p3nguin`	`Let me check with voip.ms (that's who I use).`
`22:54.33`	`kdas`	`ok`
`22:55.27`	`p3nguin`	`0.00790000 per minute`
`22:55.40`	`kdas`	`yea i got for 0.059 right now`
`22:55.46`	`p3nguin`	`That's cheaper than it is for me to call across the street!`
`22:55.57`	`p3nguin`	`0.79 cents`
`22:55.58`	`kdas`	`where u at?`
`22:56.04`	`p3nguin`	`You're paying 5.9 cents`
`22:56.09`	`p3nguin`	`US`
`22:56.38`	`kdas`	`so it costs u more than 5.9 cents to call to us ?`
`22:56.41`	`p3nguin`	`no`
`22:56.55`	`p3nguin`	`It costs me 0.79 cents to call that UK prefix.`
`22:56.58`	`kdas`	`it costs me 5.9 cents to call that uk number`
`22:57.08`	`p3nguin`	`It costs me 1.05 cents to call US numbers.`
`22:57.24`	`kdas`	`you mean 0.079 dollars or 7.9 cents`
`22:57.29`	`p3nguin`	`nope`
`22:57.31`	`kdas`	`not 0.79 cents`
`22:57.39`	`p3nguin`	`I mean 0.00790000 dollars`
`22:57.40`	`kdas`	`oh shit`
`22:57.43`	`p3nguin`	`or 0.79 cents`
`22:57.48`	`kdas`	`woa`
`22:57.52`	`kdas`	`to that prefix ?`
`22:58.05`	`p3nguin`	`http://voip.ms/rates.php`
`22:58.10`	`p3nguin`	`Put in 44 and press search.`
`22:58.44`	`kdas`	`United Kingdom Mobile 4479 6 0.18060000`
`22:59.00`	`kdas`	`thats 18cents`
`22:59.26`	`p3nguin`	`I don't have that.`
`22:59.50`	`p3nguin`	`Mine only shows United Kingdom Other 44 6 0.00790000`
`23:00.00`	`p3nguin`	`I'll try again.`
`23:00.40`	`kdas`	`you looking at landline`
`23:00.52`	`kdas`	`4479244 is a mobile prefix`
`23:01.33`	`p3nguin`	`Maybe their search is a failure.`
`23:01.49`	`p3nguin`	`I checked flowroute, and they want 22 cents per minute.`
`23:01.55`	`kdas`	`sounds right`
`23:02.07`	`kdas`	`damn ringing cmd not working properly`
`23:02.56`	`*** join/#asterisk jks (jks@193.189.93.254)`
`23:03.03`	`kdas`	`exten => 03030303,1,Ringing`
`23:03.03`	`p3nguin`	`You probably don't need it anyway.`
`23:03.03`	`kdas`	`exten => 03030303,2,Dial(SIP/447924121212@voipgain)`
`23:03.15`	`kdas`	`well i dont hear ringing when i call`
`23:03.18`	`p3nguin`	`Yeah, that's not how you should be doing it.`
`23:03.29`	`kdas`	`then?`
`23:03.31`	`p3nguin`	`If it doesn't ring when calling, use the r option in the Dial.`
`23:03.38`	`kdas`	`ohhhh`
`23:03.42`	`p3nguin`	`exten => 03030303,1,Dial(SIP/447924121212@voipgain,r)`
`23:03.51`	`kdas`	`thanks let me try that`
`23:04.17`	`p3nguin`	`The Ringing() command stops as soon as the Dial() command begins.`
`23:04.26`	`p3nguin`	`THat's why it didn't make any sounds.`
`23:04.32`	`ManxPower`	`The "r" option to dial has been shown to cause cancer. It is not recommended.`
`23:04.33`	`kdas`	`make sense`
`23:04.56`	`kdas`	`well the "r" option didnt give me rings`
`23:05.21`	`ManxPower`	`If you don't get ringback automatically, "r" seldom helps.`
`23:05.31`	`ManxPower`	`check /etc/asterisk/indications.conf`
`23:05.56`	`kdas`	`i heard it other times when i try other kinds of setups`
`23:06.03`	`kdas`	`but not like how you showed me`
`23:06.31`	`p3nguin`	`Using r actually helped get around a problem with a couple providers never providing a 180 Ringing to me. I never had 180 Ringing, so the call never got "connected."`
`23:07.02`	`kdas`	`you think cuz i am calling a UK number somehting funky is happening ?`
`23:07.12`	`p3nguin`	`no idea.`
`23:07.15`	`ManxPower`	`yes, "r" can help if you are using a brain dead provider`
`23:07.20`	`p3nguin`	`heh`
`23:07.24`	`ManxPower`	`kdas, Do you do an answer() before the Dial`
`23:07.53`	`kdas`	`ManxPower: no only one line "exten => 03030303,1,Dial(SIP/447924121212@voipgain,r)"`
`23:08.16`	`kdas`	`[ipkall] then that line`
`23:08.26`	`p3nguin`	`If the r didn't help, rip it back out of there so you don't get gangrene or something.`
`23:08.38`	`kdas`	`but the "r" flag didnt help`
`23:08.47`	`kdas`	`i did already ;)`
`23:09.09`	`p3nguin`	`You put 03030303 in the SIP Phone Number field in IPkall configuration?`
`23:09.16`	`kdas`	`ywea`
`23:09.18`	`kdas`	`yea*`
`23:09.34`	`p3nguin`	`Try using an explicit Answer() before you dial.`
`23:10.17`	`*** join/#asterisk guilhermebr (~Guilherme@189.63.66.47)`
`23:10.17`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`23:10.26`	`p3nguin`	`I guess that would bring up the channel before starting a new one.`
`23:10.28`	`kdas`	`ok so ipkall does a 30030303@myhost which in turns get caught by [ipkall0 and 1] which hands off to extention [ipkall]`
`23:10.49`	`kdas`	`and under [ipkall] answer then the dial ?`
`23:10.59`	`p3nguin`	`yeah, try it just for fun.`
`23:11.09`	`p3nguin`	`Like you did with the ringing before.`
`23:11.14`	`kdas`	`ok`
`23:11.14`	`kdas`	`lol`
`23:12.10`	`kdas`	`Auto fallthrough, channel 'SIP/ipkall0-00000009' status is 'UNKNOWN'`
`23:12.14`	`kdas`	`then busy tone`
`23:12.18`	`kdas`	`didnt like that`
`23:12.20`	`p3nguin`	`hmm`
`23:12.37`	`p3nguin`	`exten => 03030303,1,Answer()`
`23:12.48`	`p3nguin`	`exten => 03030303,n,Dial(SIP/447924121212@voipgain)`
`23:12.53`	`kdas`	`yea`
`23:12.54`	`p3nguin`	`You did it this way?`
`23:13.12`	`p3nguin`	`and it fell through... interesting.`
`23:13.15`	`kdas`	`oh`
`23:13.16`	`kdas`	`wait`
`23:13.18`	`kdas`	`duh`
`23:13.33`	`kdas`	`3030303,1,Dia instead of 3030303,2,Dia`
`23:13.33`	`ManxPower`	`Doing an Answer will REQUIRE /etc/asterisk/indications.conf to exist or you won't get ringback\`
`23:14.02`	`kdas`	`trying again..`
`23:14.20`	`kdas`	`nope`
`23:14.22`	`kdas`	`did not work`
`23:14.25`	`kdas`	`that sucks`
`23:14.43`	`p3nguin`	`At this point, I would forget about dialing another number and make it play sounds to me.`
`23:15.29`	`kdas`	`how you mean ?`
`23:15.46`	`kdas`	`i am just worried cuz my phone rings but the person who calls me dosent hear it ringing hahah`
`23:16.06`	`p3nguin`	`03030303,1,Answer()`
`23:16.15`	`kdas`	`dide that`
`23:16.18`	`kdas`	`did*`
`23:16.32`	`p3nguin`	`then 03030303,n,Ringing() and 03030303,n,Wait(60)`
`23:16.39`	`kdas`	`maybe an answer than the "r" flag?`
`23:16.56`	`kdas`	`where does the dial come in place ?`
`23:16.56`	`p3nguin`	`You should get a full minute of ringing sounds (if you have the correct indications).`
`23:17.17`	`p3nguin`	`<p3nguin> At this point, I would forget about dialing another number and make it play sounds to me. <--- Dial has been forgotten!`
`23:18.01`	`p3nguin`	`I would be checking that sounds are working.`
`23:18.05`	`kdas`	`i am using ipkall number to use my voip service to call my uk mobile`
`23:18.08`	`kdas`	`so how does that work ?`
`23:18.21`	`kdas`	`i heard it ring when i used ringing for a split second`
`23:18.23`	`*** join/#asterisk uqlev (~yuriy@91.184.221.31)`
`23:18.58`	`p3nguin`	`If you've used Ringing and Wait(60), you should hear 10 rings before it exits.`
`23:19.10`	`kdas`	`yea i hear ringing`
`23:19.13`	`kdas`	`when i use ringing`
`23:19.26`	`kdas`	`but the problem is if i dont use dial then how do i call my uk mobile ?`
`23:19.31`	`p3nguin`	`Okay, so sounds are working, then.`
`23:19.53`	`kdas`	`what i am doing is providing a us number for people to call which forwards/calls my uk mobile`
`23:20.00`	`kdas`	`yea sounds are working`
`23:20.19`	`p3nguin`	`Now that you've determined that your sounds ARE working, NOW you can worry with dialing again.`
`23:20.24`	`*** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com)`
`23:20.30`	`*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)`
`23:21.23`	`kdas`	`lol`
`23:21.25`	`kdas`	`yay!!`
`23:21.31`	`kdas`	`yea i have no idea what to do`
`23:21.36`	`p3nguin`	`Maybe MoH would work rather than ringing sounds?`
`23:21.43`	`kdas`	`MoH?`
`23:21.51`	`p3nguin`	`MusicOnHold()`
`23:21.59`	`kdas`	`would an answer then a dial with "r" flag work ?`
`23:22.06`	`p3nguin`	`use m instead of r`
`23:22.14`	`p3nguin`	`But make sure moh has been configured.`
`23:22.28`	`kdas`	`how do i do that?`
`23:22.46`	`p3nguin`	`breathes...`
`23:22.49`	`p3nguin`	`~book`
`23:22.50`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`23:22.56`	`p3nguin`	`~wiki`
`23:23.01`	`kdas`	`should i do a 3030,1,MusicOnHold() then a 3030,2,Dial(blah)`
`23:23.07`	`p3nguin`	`hmm...`
`23:23.09`	`p3nguin`	`~wikis`
`23:23.10`	`infobot`	`[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners`
`23:24.24`	`ChannelZ`	`you don't want to actually call MusicOnHold or the person will just sit there listening to music indefinately. It doesn't continue on in the dialplan.`
`23:24.50`	`kdas`	`yea i am reading about it`
`23:24.50`	`p3nguin`	`Dial(SIP/peer,,m) rather than Dial(SIP/peer,,r)`
`23:25.04`	`ChannelZ`	`what is the point of all this again?`
`23:25.05`	`kdas`	`what reason would the ringing not be working though ?`
`23:25.11`	`p3nguin`	`no clue`
`23:25.21`	`kdas`	`ChannelZ: i have my ipkall dial my uk number`
`23:25.25`	`kdas`	`but no ring ring`
`23:26.55`	`p3nguin`	`I really think that an Answer() first, then a Dial using the r option will play the ringing sound.`
`23:26.59`	`kdas`	`so p3nguin is trying to get me to music on hold instead`
`23:27.18`	`p3nguin`	`For music instead of ringing, you use m rather than r.`
`23:27.27`	`p3nguin`	`but moh has to be working.`
`23:27.50`	`russellb`	`also try Progress() before Dial()`
`23:28.11`	`ChannelZ`	`wonders why none of this providers indications are working in the first place`
`23:28.40`	`kdas`	`ChannelZ: ipkall is working`
`23:28.52`	`p3nguin`	`the other one apparently isn't.`
`23:29.03`	`kdas`	`p3nguin: answer than dial with r didnt work`
`23:29.14`	`kdas`	`p3nguin: yea voipgain not working`
`23:29.14`	`p3nguin`	`I know. We went over it.`
`23:30.00`	`ChannelZ`	`If your provider is hosed, no amount of flags will do it - if asterisk thinks the channel has been answered, it's over. Any ringing would have to come from the upstream provider at that point`
`23:30.45`	`russellb`	`not quite. the 'r' option do dial should produce inband ringing if the channel is up`
`23:30.49`	`russellb`	`s/do/to/`
`23:31.00`	`ChannelZ`	`How does it know when to stop?`
`23:31.16`	`russellb`	`app_dial knows to stop it before it bridges the channels together`
`23:31.51`	`ChannelZ`	`but that's what I'm saying, it's probably already been bridged..`
`23:32.48`	`russellb`	`well i didn't read all of the scrollback, so sorry if I misunderstand the problem`
`23:33.01`	`russellb`	`dial() won't bridge the channels until the outbound channel answers`
`23:33.23`	`kdas`	`Answer("SIP/ipkall0-00000001", "") in new stack`
`23:33.36`	`ChannelZ`	`Yeah and I think his problem is that the outbound channel is answering as soon as it's done dialing`
`23:33.40`	`kdas`	`Dial("SIP/ipkall0-00000001", "SIP/447924333@voipgain,r") in new stack`
`23:33.58`	`kdas`	`WARNING[4118]: app_dial.c:1847 dial_exec_full: Invalid timeout specified: 'r'. Setting timeout to infinite`
`23:34.05`	`kdas`	`SIP/voipgain-00000002 is making progress passing it to SIP/ipkall0-00000001`
`23:34.05`	`russellb`	`syntax fail`
`23:34.17`	`russellb`	`s/,r/,,r/`
`23:34.23`	`ChannelZ`	`oh well`
`23:34.26`	`ChannelZ`	`needs to go`
`23:34.29`	`russellb`	`cya`
`23:34.34`	`ChannelZ`	`tootles :)`
`23:35.20`	`p3nguin`	`What'd I do, typo the options syntax the first time?`
`23:35.40`	`kdas`	`what it should be ,,r instead of ,r?`
`23:36.04`	`p3nguin`	`yes`
`23:36.29`	`p3nguin`	`If you would have read "core show application Dial", you could have told me I screwed up.`
`23:37.26`	`kdas`	`lol`
`23:37.29`	`kdas`	`,,r works`
`23:37.37`	`p3nguin`	`yayz!`
`23:37.45`	`kdas`	`stupid syntax errors`
`23:38.03`	`kdas`	`p3nguin: THANK YOU for your time patience and kindness and most of all your help`
`23:38.36`	`p3nguin`	`adds voipgain to the list of "brain dead providers."`
`23:38.51`	`kdas`	`if you lived in bay area i would take you out to eat ;)`
`23:39.01`	`kdas`	`now the fun begins`
`23:39.10`	`kdas`	`now i need to do callback with my uk mobile`
`23:39.11`	`kdas`	`lol`
`23:39.20`	`kdas`	`ok i am going to eat`
`23:39.23`	`kdas`	`i am hungry`
`23:39.29`	`kdas`	`thanks for your help again`
`23:39.34`	`kdas`	`see ya later`
`23:39.39`	`p3nguin`	`laytor!`
`23:47.45`	`*** join/#asterisk plutopathz (~plutopath@38.126.213.64)`
`23:47.57`	`plutopathz`	`hello all`
`23:49.42`	`plutopathz`	`can someone please help me? i cannot run the ./configure-gv command. when i type it it returns "-bash: ./configure: No such file or directory"`

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