IRC log for #asterisk on 20121109

`00:00.25`	`kstange`	`yep, I know how to build it`
`00:03.11`	`ChrisInSydney`	`cool.`
`00:04.12`	`ChrisInSydney`	`I've been patchng things to get some slightly different behavior in call pckups and dial privacy opts, so I need to build from scratch`
`00:18.14`	`*** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d)`
`00:25.54`	`ChrisInSydney`	`pached app_dial.c so that, when a prvacy call comes in, and the announcement is useless, you can press 9 to delete the announcement file and connect the call :D`
`00:26.25`	`ChrisInSydney`	`was also so excited, he couldn't type properly`
`00:37.47`	`*** join/#asterisk Scottyob (~scott@27-33-131-215.static.tpgi.com.au)`
`00:37.59`	`Scottyob`	`[TK]D-Fender: Yo, are you around?`
`00:38.51`	`[TK]D-Fender`	`yup`
`00:38.54`	`[TK]D-Fender`	`but not for long`
`00:39.10`	`[TK]D-Fender`	`Scottyob: Managed to swap out that AirPort?`
`00:39.31`	`Scottyob`	`[TK]D-Fender: nope, turned it off and on at the wall, seems to have fixed it`
`00:39.37`	`Scottyob`	`[TK]D-Fender: http://www.scottyob.com/2012/11/08/address-book-for-a-the-blind/`
`00:39.40`	`Scottyob`	`works perfectly :)`
`00:42.46`	`[TK]D-Fender`	`Scottyob: Nice little write-up. You've been very sharp through this process and I'm glad it didn't take you too long to beat your way through this last bump`
`00:43.13`	`Scottyob`	`[TK]D-Fender: yeah, still disheartned I couldn't use a stun server or something of the like to get it working behind the NAT though`
`00:44.12`	`[TK]D-Fender`	`Scottyob: Well you didn't even need STUN.. isn't that better? :)`
`00:44.32`	`Scottyob`	`yeah, but I need somewhere to host it`
`00:44.49`	`[TK]D-Fender`	`Scottyob: ?`
`00:45.10`	`Scottyob`	`a VM on my laptop isn't really the best place for it :P`
`00:48.01`	`[TK]D-Fender`	`Oh well.... that's another matter...`
`00:49.13`	`Scottyob`	`yeah, anyway I've got to dash but thanks again for your help [TK]D-Fender :)`
`00:53.49`	`*** join/#asterisk joren (~joren@70.98.240.130)`
`00:55.40`	`joren`	`So now my ISP is starting to try to say my packet loss is beyond his control, because my asterisk server is accepting reinvites I guess. I set canreinvite=no on his peer, but I still seem to just be talking to a server besides his directly. anyone have any thoughts?`
`00:58.47`	`joren`	`namely thoughts on how to force my asterisk server to only talk to his`
`00:59.04`	`joren`	`I guess I could try dropping traffic to everything but his server`
`01:02.05`	`*** join/#asterisk serafie (~erin@76.73.167.231)`
`01:03.19`	`*** join/#asterisk tyrion (~tyrion@skynet.skynet.ie)`
`01:37.08`	`SeRi`	`p3nguin: you around?`
`01:38.01`	`*** part/#asterisk kstange (~kevin@2607:f128:0:1:222:4dff:fe51:2ed5)`
`01:38.02`	`SeRi`	`Is there a way I can jump in to a conversation when is in progress?`
`01:38.06`	`*** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net)`
`01:42.10`	`*** join/#asterisk ponyofdeath (~vladi@cpe-75-80-173-129.san.res.rr.com)`
`01:42.24`	`ponyofdeath`	`anyone else having issues with calls to google voice not making it to asterisk server?`
`01:57.46`	`p3nguin`	`seri: ChanSpy or ExtenSpy`
`02:04.20`	`SeRi`	`p3nguin: MhhMhhhh`
`02:04.42`	`SeRi`	`looking in to it now`
`02:06.47`	`p3nguin`	`For what it's worth, I like options b, B, E and q.`
`02:07.07`	`SeRi`	`p3nguin: I see.`
`02:07.35`	`SeRi`	`Interesting I am using chanspy to enable the speaker phones on my polycoms`
`02:07.51`	`p3nguin`	`Hmm?`
`02:07.56`	`SeRi`	`I didnt know I can jump in to a conversation using it as well.`
`02:07.58`	`p3nguin`	`How/why?`
`02:08.04`	`SeRi`	`I might be wrong`
`02:08.05`	`SeRi`	`one sec`
`02:08.11`	`SeRi`	`let me look at my dial plan`
`02:08.13`	`p3nguin`	`You realize that ChanSpy is to Spy on a Channel, right?`
`02:08.19`	`SeRi`	`yes`
`02:08.20`	`SeRi`	`lol`
`02:08.32`	`p3nguin`	`How does that have anything to do with speaker phones?`
`02:09.02`	`SeRi`	`never mind`
`02:09.11`	`SeRi`	`for what ever reason I had it tittle chanspy`
`02:09.19`	`SeRi`	`my bad`
`02:09.47`	`SeRi`	`ok so back at usinf chanspy`
`02:11.15`	`SeRi`	`s/usinf/using/`
`02:14.51`	`SeRi`	`p3nguin: Is there a way I can specify an ext? here is what I have so far http://pastebin.com/YDdSwwnn`
`02:15.06`	`SeRi`	`I ask the question wrong`
`02:15.11`	`p3nguin`	`ExtenSpy is for specifying extensions to spy.`
`02:15.17`	`SeRi`	`Can I have the system what ext I want to listen?`
`02:15.21`	`SeRi`	`ah!`
`02:15.23`	`SeRi`	`Nice`
`02:15.24`	`SeRi`	`got it`
`02:15.44`	`SeRi`	`ok so is it possible to have me ask for input?\`
`02:16.10`	`p3nguin`	`You've confused DEVICE with EXTENSION again.`
`02:16.21`	`p3nguin`	`ChanSpy(SIP/ASKFOREXTHERE,b,q) <--- no`
`02:16.38`	`p3nguin`	`ChanSpy(SIP/Device,bq)`
`02:17.34`	`SeRi`	`sorry I was looking for ExtenSpy instead.`
`02:17.49`	`p3nguin`	`ExtenSpy(extension@context,bq)`
`02:17.56`	`SeRi`	`so the idea is to dial a extention and be prompted what ext I want to spy on`
`02:18.06`	`SeRi`	`ah ok`
`02:18.23`	`p3nguin`	`You can do that, or you can just not specify any channel with ChanSpy and cycle through channels in use.`
`02:18.32`	`p3nguin`	`ChanSpy(,bq)`
`02:18.53`	`p3nguin`	`I think you press the # key to change channels.`
`02:19.06`	`SeRi`	`awesome. I am going to give it a try now.`
`02:22.27`	`*** join/#asterisk cyford (~allen@c-76-97-235-69.hsd1.ga.comcast.net)`
`02:25.23`	`SeRi`	`MhhMhhh`
`02:27.38`	`SeRi`	`It works very nice....`
`02:28.53`	`SeRi`	`NOTICE[31796]: app_chanspy.c:484 start_spying: Attaching SIP/device-00000140 to SIP/voipms`
`02:28.59`	`SeRi`	`nice`
`02:29.05`	`SeRi`	`so thats one.`
`02:29.16`	`SeRi`	`one more to go.`
`02:30.41`	`SeRi`	`channel redirect is my other small project`
`02:30.51`	`SeRi`	`I think I got that one nail down though`
`02:31.09`	`SeRi`	`Is time to start working on the new server`
`02:31.40`	`SeRi`	`esxi server with pfsense guest and ast guest.`
`02:32.17`	`SeRi`	`not sure if I really want to do that but I just might.`
`02:32.39`	`SeRi`	`I might just get rid of my alix... though I just love that damn board`
`02:32.43`	`*** join/#asterisk timahvo1 (~rogue@41.212.120.182)`
`02:32.48`	`WIMPy`	`Good morning`
`02:33.20`	`SeRi`	`good morning WIMPy`
`02:33.30`	`SeRi`	`adn good evning`
`02:33.37`	`SeRi`	`s/adn/and/`
`02:34.51`	`p3nguin`	`Channel redirect isn't really something that requires work.`
`02:35.03`	`p3nguin`	`You just... redirect a channel to a specific place in the dial plan.`
`02:35.07`	`SeRi`	`p3nguin: nope. just learning how to use it`
`02:35.29`	`p3nguin`	`In your CLI, type in channel redirect and press Enter.`
`02:36.01`	`SeRi`	`I have not work on live cli calls so I am just doing that now`
`02:36.08`	`SeRi`	`I been using it to send calls to vmail`
`02:36.48`	`WIMPy`	`Doing it manually leaves a good chance to send them to /dev/null.`
`02:36.59`	`SeRi`	`even better`
`02:37.00`	`SeRi`	`LOL`
`02:37.11`	`p3nguin`	`channel request hangup`
`02:37.12`	`SeRi`	`I been flood it with telemarketers calls this past few months`
`02:37.22`	`p3nguin`	`Configure call screening.`
`02:37.40`	`SeRi`	`p3nguin: Thats another thought.`
`02:37.54`	`p3nguin`	`Force callers to press a key to continue.`
`02:38.19`	`WIMPy`	`Ja, I did a CAPTCHA for anonymous calls.`
`02:38.20`	`p3nguin`	`Both methods are very effective.`
`02:38.51`	`SeRi`	`nice.`
`02:38.54`	`SeRi`	`good ideas`
`02:40.02`	`p3nguin`	`I have used both. The screening still makes your phone ring and you have to accept the call; the other never rings your phone if they don't press whatever you tell them to press.`
`02:40.17`	`p3nguin`	`Robo-dialers don't press keys on the phone. :)`
`02:40.31`	`WIMPy`	`Not yet.`
`02:41.13`	`SeRi`	`lol`
`02:41.27`	`SeRi`	`this damn one dollar a month razor is everywhere`
`02:42.06`	`SeRi`	`I might just do it.`
`02:50.38`	`p3nguin`	`What is it?`
`02:52.11`	`SeRi`	`checkit out`
`02:52.31`	`SeRi`	`p3nguin: http://www.dollarshaveclub.com`
`02:53.02`	`WIMPy`	`Is that filth?`
`02:53.58`	`SeRi`	`reviews seem to say is worth it`
`02:54.39`	`SeRi`	`http://www.dollarshaveclub.com/ref/index/r/4ynk/ubu3t9/`
`02:55.34`	`SeRi`	`click on that link if you are going to order ^^ :)`
`02:56.33`	`WIMPy`	`I don;t know how that happens, but reviews seem to always be positive to enthusiastic.`
`02:56.43`	`SeRi`	`I use cheap 2 blade razors and spend about 7 dollars a pack and usually last me about one month and a half`
`02:57.17`	`SeRi`	`so for 1 dollar plus shipping is not a bad deal`
`02:57.46`	`SeRi`	`one less thing I have to worry about it`
`02:57.59`	`SeRi`	`this site has gone viral...`
`02:59.30`	`SeRi`	`WIMPy: lol.... I am not an enth... I did some research seems to be ok. I am not in to spending a shit load of money on razors. damn gillette is like 20 dollars for 5 razors`
`02:59.33`	`*** join/#asterisk last1 (~last1@modemcable153.206-57-74.mc.videotron.ca)`
`03:08.43`	`SeRi`	`has anybody seen this?`
`03:08.47`	`SeRi`	`http://www.kickstarter.com/projects/limemouse/lifx-the-light-bulb-reinvented`
`03:08.55`	`p3nguin`	`It's cheap, but I doubt I spend that much.`
`03:09.33`	`WIMPy`	`SeRi: Where have you been in the last years?`
`03:10.44`	`SeRi`	`WIMPy: LOL.... I been working`
`03:10.59`	`SeRi`	`sense January of 2012 I have not had a life`
`03:11.02`	`SeRi`	`only work`
`03:11.04`	`SeRi`	`:PO`
`03:11.07`	`WIMPy`	`In a cave?`
`03:11.09`	`SeRi`	`:P`
`03:11.21`	`WIMPy`	`That's less than one year.`
`03:11.29`	`SeRi`	`LOL`
`03:12.08`	`SeRi`	`working almost 13/15hr days :(`
`03:13.07`	`WIMPy`	`Next time get a cave with internet at least.`
`03:13.23`	`SeRi`	`WIMPy: I'll call you.`
`03:13.28`	`SeRi`	`:)`
`03:13.36`	`p3nguin`	`I haven't seen those bulbs further back than the past year.`
`03:13.52`	`SeRi`	`well hold on a sec`
`03:14.05`	`SeRi`	`I hope you know this is not just an led bulb...`
`03:14.14`	`p3nguin`	`WiFi-enabled LED bulbs`
`03:14.32`	`SeRi`	`wtf. wow than shit I have been in a cave`
`03:14.32`	`p3nguin`	`Not just standard LED bulbs.`
`03:14.41`	`SeRi`	`first time I ever heard of them was the past few months`
`03:14.50`	`SeRi`	`I had no clue they been out`
`03:14.52`	`p3nguin`	`Yep, they aren't too old.`
`03:14.53`	`WIMPy`	`Yes, usually they only have IR remotes, but not a big deal to connect them to something else.`
`03:14.59`	`*** join/#asterisk timahvo1 (~rogue@41.212.120.182)`
`03:15.02`	`slav3_kitten`	`http://pastie.org/private/0tx6uiougrxajhb15johvg < that's what i ended up doing`
`03:15.34`	`SeRi`	`WIMPy: I seen those and been able to use them in home automation....`
`03:15.40`	`slav3_kitten`	`70 bucks for a bulb...`
`03:15.53`	`SeRi`	`but didnt know that wifi enable bulbs where out`
`03:16.03`	`*** join/#asterisk serafie (~erin@76.73.167.231)`
`03:16.12`	`SeRi`	`'well now I know.`
`03:16.46`	`SeRi`	`shit I do feal like I was in a cave`
`03:16.52`	`*** join/#asterisk linocisco (~linocisco@193.134.242.12)`
`03:17.19`	`WIMPy`	`Might save you lots of money if you DIY.`
`03:17.47`	`SeRi`	`WIMPy: Yea. would be a nice DYI project.`
`03:18.09`	`linocisco`	`hi all, I want to buy very cheap asterisk based box like switchvox, for 1 FXO and 3 FXS(analog port)`
`03:18.15`	`linocisco`	`I dont know what to buy`
`03:18.38`	`WIMPy`	`A friend suggested to (ab)use some spare PWM controlled fan connectors of the mobo.`
`03:18.59`	`SeRi`	`cheap? Hit ebay. even than you would be paying 300+`
`03:19.01`	`p3nguin`	`Those WiFi bulbs are not out yet.`
`03:19.15`	`SeRi`	`p3nguin: Not the ones I link.`
`03:19.16`	`p3nguin`	`The project is estimated to be fully funded next week.`
`03:20.39`	`SeRi`	`p3nguin: If I would of found it earlier I would of probably pledge in`
`03:20.52`	`p3nguin`	`would HAVE`
`03:21.18`	`p3nguin`	`There is no "would of." It's WOULD HAVE.`
`03:21.22`	`SeRi`	`yes sr. :)`
`03:21.48`	`p3nguin`	`You can still invest your dollars in it right now.`
`03:21.55`	`slav3_kitten`	`linocisco, why does it need to be cheap?`
`03:22.13`	`SeRi`	`p3nguin: I cant all pledges are locked.`
`03:22.36`	`p3nguin`	`If you click on the dollar pledge button, it rejects you?`
`03:22.38`	`WIMPy`	`linocisco: Get some plastic router with a fitting spec from ebay?`
`03:22.47`	`linocisco`	`slav3_kitten, for home use`
`03:23.06`	`SeRi`	`p3nguin: with one dollar you wont get the bulbs. :)`
`03:23.18`	`p3nguin`	`That's for sure.`
`03:23.22`	`slav3_kitten`	`linocisco, d-link made a cheap fxo, and a number o heap fxs's exist`
`03:23.33`	`slav3_kitten`	`number of cheap*`
`03:24.31`	`SeRi`	`I'll watch it and see if somebody drops off`
`03:25.02`	`last1`	`I'm trying to use this function: SpeechBackground(Sound File\|Timeout)`
`03:25.07`	`slav3_kitten`	`linocisco, got an old laptop? i'm running my home setup on an old 800mhz laptop`
`03:25.08`	`last1`	`but part of an agi script`
`03:25.15`	`last1`	`how can I specify the timeout value ?`
`03:25.33`	`slav3_kitten`	`old laptop + sip fxo + sip fxs = cheap`
`03:25.44`	`last1`	`I tried something like this: $AGI->exec('SpeechBackground', 'ivr/press', 3);`
`03:25.48`	`SeRi`	`linocisco: buy an atom board or an alix board for cheap and build your own. You can also buy a thin client of ebay and build it there`
`03:26.01`	`last1`	`but it still waits indefinitely`
`03:26.52`	`SeRi`	`linocisco: http://www.ebay.com/itm/IP-PBX-for-SOHO-2FXO-port-Asterisk-ZYCOO-IVR-30-extension-call-center-/281008886199?pt=Phone_Switching_Systems&hash=item416d6f51b7`
`03:27.41`	`SeRi`	`linocisco: you can also buy a raspberypi and slap asterisk`
`03:27.50`	`SeRi`	`cheap as it canm get`
`03:28.04`	`SeRi`	`s/canm/can/`
`03:28.14`	`slav3_kitten`	`SeRi, the pi will handle it?`
`03:28.22`	`SeRi`	`ooo yes it will`
`03:28.28`	`SeRi`	`and now it comes with 512MB`
`03:28.33`	`SeRi`	`for the same price`
`03:28.35`	`slav3_kitten`	`well assuming you put a distro with hardware fpu enabled`
`03:28.44`	`slav3_kitten`	`i've got a first gen pi`
`03:28.45`	`SeRi`	`arch is all you need`
`03:28.55`	`WIMPy`	`None of these suggestions seem cheap to me, but certainly a lot easier.`
`03:29.03`	`SeRi`	`WIMPy:`
`03:29.10`	`slav3_kitten`	`pi is 30 bucks`
`03:29.18`	`SeRi`	`WIMPy: how a raspberrypi not cheap?`
`03:29.55`	`WIMPy`	`Compared to an old plastic router that already contains the telephony ports?`
`03:29.55`	`slav3_kitten`	`pi + dlink fxo (50 bucks) + some fxs's =150 bucks maybe max`
`03:30.11`	`slav3_kitten`	`WIMPy, not the number of ports he wanted`
`03:30.24`	`WIMPy`	`Why not?`
`03:30.36`	`slav3_kitten`	`because he wanted 1 fxo + 3 fxs`
`03:30.43`	`WIMPy`	`Ok, mos have only 2 FXS, but there are ones with 3.`
`03:31.16`	`WIMPy`	`Or you could use one with 2FXS and a BRI plus a PRI o FXS adapter.`
`03:31.21`	`SeRi`	`http://www.ebay.com/itm/AA50-Asterisk-Appliance-IP-PBX-S800i-With-SIP-IAX-/281001692847?pt=Phone_Switching_Systems&hash=item416d018eaf`
`03:31.37`	`SeRi`	`not cheap :)`
`03:32.03`	`slav3_kitten`	`maybe i should run this on my pi instead of that old toughbook`
`03:32.09`	`WIMPy`	`You should be able to find somethign fo max 50 bucks.`
`03:32.55`	`WIMPy`	`You can sometimes get them new for 12.99, but you want something that has enough space left to install Asterisk, off course.`
`03:36.12`	`*** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr)`
`03:36.32`	`linocisco`	`SeRi, raspberypi has no 1 FXO and 2 FXS ports`
`03:36.43`	`WIMPy`	`And mind you that DLink even had a dedicated Asterisk box.`
`03:37.48`	`WIMPy`	`Unfortunately it's from the 1.2 days. I managed to upgrade mine to 1.8 earlier this year, but I don't know if the FXO/FXS ports work with new (kernel, Asterisk) versions.`
`03:38.18`	`SeRi`	`WIMPy: you have a link to that model?`
`03:38.38`	`WIMPy`	`The two S0 ports work with a bog standard kernel, however. Which happens to be all I was interested in.`
`03:38.47`	`linocisco`	`SeRi, raspberypi has no 1 FXO and 2 FXS ports`
`03:38.48`	`WIMPy`	`HorstBox`
`03:39.05`	`SeRi`	`linocisco: I heard you the first time. :)`
`03:39.21`	`linocisco`	`SeRi, where?`
`03:39.36`	`slav3_kitten`	`you said 1 and 3 previously linocisco`
`03:39.40`	`WIMPy`	`DVA-G3342SB`
`03:40.01`	`SeRi`	`slav3_kitten: Thanks :)`
`03:40.02`	`WIMPy`	`1+2 is a lot easier to find than 1+3.`
`03:40.05`	`SeRi`	`WIMPy: Nice!`
`03:40.21`	`linocisco`	`SeRi, yes. one and 3 is most I wanted`
`03:40.26`	`slav3_kitten`	`WIMPy, obviously`
`03:40.42`	`slav3_kitten`	`SeRi, != slav3_kitten`
`03:40.48`	`SeRi`	`linocisco: are you trying to address me or the channel`
`03:40.54`	`SeRi`	`you keep saying my name`
`03:41.14`	`linocisco`	`SeRi, because only u r answering me`
`03:41.21`	`slav3_kitten`	`it's a nice name SeRi`
`03:41.34`	`SeRi`	`slav3_kitten: LOL`
`03:41.40`	`WIMPy`	`has a feeling of not being heard.`
`03:41.58`	`SeRi`	`linocisco: everybody here is been talking to you`
`03:42.09`	`SeRi`	`slav3_kitten and WIMPy`
`03:42.32`	`linocisco`	`hi all, what is the asterisk based cheap set top box of 50 USD with 1 FXO and 3 FXS analog ports?`
`03:42.48`	`slav3_kitten`	`linocisco : does not exist`
`03:42.55`	`WIMPy`	`linocisco: Scroll up and read the answers.`
`03:42.59`	`slav3_kitten`	`3 fxs is the hard part, why do you need 3 fxs?`
`03:43.29`	`p3nguin`	`Why even one? SIP phones are so much better.`
`03:43.30`	`slav3_kitten`	`i can get the fxo bit, but sip IP phones are ass cheap and offer more then an analog phone`
`03:43.33`	`linocisco`	`slav3_kitten, I have 3 rooms`
`03:43.46`	`*** join/#asterisk timahvo1 (~rogue@41.212.120.182)`
`03:43.57`	`slav3_kitten`	`linocisco, run some cat5 to them an put in 3 cheap sip phones`
`03:43.58`	`linocisco`	`slav3_kitten, I have only one CO line, meaning one FXO line`
`03:44.19`	`WIMPy`	`slav3_kitten: Like drawing lots of electricity and that way cash out of your pocket?`
`03:44.39`	`slav3_kitten`	`like i said, i can understand the need for an FXO, i have one in my router.`
`03:45.16`	`slav3_kitten`	`WIMPy, lets look at this 3x sip phones draw maybe 15 watts each. so 45 watts for all 3. electricity is 10 cents/kwh...`
`03:45.59`	`WIMPy`	`For us it's at least three times as expensive.`
`03:46.12`	`slav3_kitten`	`39.42 dollars per year`
`03:46.29`	`slav3_kitten`	`or 10.8 cents a day to run 3 sip phones`
`03:47.21`	`slav3_kitten`	`hell most people lose more coinage a day then that`
`03:47.40`	`p3nguin`	`And they lose more THAN that, too.`
`03:47.56`	`linocisco`	`slav3_kitten, that is why I am looking into normal analog phone. also cables are inexpensive`
`03:48.53`	`slav3_kitten`	`maybe i'm odd then because i ran a shit load of cat5 in the house and absolutely no cat3`
`03:49.02`	`*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)`
`03:49.29`	`linocisco`	`slav3_kitten, how about running used astribank with analog desktop phones?`
`03:49.50`	`slav3_kitten`	`linocisco, why, i have 6 cisco phones with sip images`
`03:49.58`	`slav3_kitten`	`that are full of features`
`03:50.44`	`SeRi`	`my whole house is Cat6e`
`03:51.04`	`slav3_kitten`	`if you want to run fxs's all over the place then more power to you, but i think it's going to be hard to find a 50 dollar asterisk box that has 1 fxo and 3 fxs`
`03:51.09`	`WIMPy`	`I use only one IP phone.`
`03:51.15`	`linocisco`	`slav3_kitten, the reason why I would like to deploy asterisk system urgently is that I hate using very expensive proprietary Nortel Business COmmunication Manager which costs 30 K USD at least. I want to beat that`
`03:51.16`	`SeRi`	`lies!`
`03:51.17`	`slav3_kitten`	`SeRi, i couldn't afford cat6`
`03:51.20`	`SeRi`	`LOL`
`03:51.55`	`WIMPy`	`linocisco: Do you want Asterisk or just any PBX?`
`03:51.58`	`SeRi`	`slav3_kitten: I found a open spindle at the local microcenter and got it for cheap`
`03:52.06`	`slav3_kitten`	`lucky`
`03:52.19`	`SeRi`	`:)`
`03:52.23`	`slav3_kitten`	`i ran 1500 feet of cat5e`
`03:52.23`	`linocisco`	`slav3_kitten, but stupid management at work or Head Quarter knows only to use expensive Avaya Nortel or Cisco Call manager`
`03:52.29`	`slav3_kitten`	`house is long an server closet is at the far end`
`03:52.37`	`SeRi`	`ah`
`03:52.48`	`p3nguin`	`1500 ft in one run?`
`03:52.56`	`slav3_kitten`	`1500 feet for 24 runs`
`03:52.57`	`linocisco`	`WIMPy, any other PBX?`
`03:53.01`	`SeRi`	`not possible`
`03:53.06`	`SeRi`	`slav3_kitten: ah`
`03:53.20`	`slav3_kitten`	`SeRi, it is possible if you put some repeaters in the run`
`03:53.24`	`p3nguin`	`I was going to ask if it even worked at 1500 feet since the max is 328 ft.`
`03:53.27`	`SeRi`	`of course`
`03:53.36`	`SeRi`	`slav3_kitten: ^^`
`03:53.46`	`SeRi`	`is not common practice any more`
`03:53.50`	`slav3_kitten`	`but then you have a run with a lot of room for problems, better to run fiber at that point`
`03:53.55`	`WIMPy`	`linocisco: As I said earlier: Just some old plastic router with 1+3 ports. If you don't have to worry about having to install Asterisk, but just use it as is, that should be easy.`
`03:54.17`	`SeRi`	`ok guys g/n`
`03:54.26`	`slav3_kitten`	`night SeRi`
`03:54.28`	`linocisco`	`WIMPy, I have Link sys WRT110 N`
`03:54.45`	`slav3_kitten`	`linocisco, does it have 1fxo+3fxs ports?`
`03:55.25`	`linocisco`	`slav3_kitten, it has all RJ45 ports. one Internet WAN port and 4 LAN ports`
`03:55.50`	`WIMPy`	`So how's that going to help?`
`03:56.02`	`linocisco`	`slav3_kitten, i have one phone line with RJ11 head. I need FXO port with RJ11 fits, and extension FXO that RJ11 can fit`
`03:56.17`	`linocisco`	`WIMPy, that is the problem`
`03:56.34`	`slav3_kitten`	`well then that's going to be a problem no? look at some sip fxo an fxs adapters and run asterisk on something low power like a old router or raspberry pi`
`03:56.51`	`slav3_kitten`	`or like i said an replace them with 3 sip phones that hook into your existing home network`
`03:59.22`	`linocisco`	`slav3_kitten, raspberry is settopbox ready for sale on the market?`
`04:02.56`	`*** join/#asterisk radic (~radic@dslb-178-010-031-126.pools.arcor-ip.net)`
`04:05.07`	`slav3_kitten`	`linocisco, i'm absolutely confused now`
`04:05.30`	`slav3_kitten`	`are you home use or trying to sell it?`
`04:06.25`	`drmessano`	`He's bouncing back and forth between here and #freepbx with the same half-questions`
`04:06.46`	`slav3_kitten`	`offers drmessano some chili`
`04:06.59`	`drmessano`	`ohhhh`
`04:07.10`	`drmessano`	`Perfect night for it`
`04:07.24`	`slav3_kitten`	`it really is, nice home made chili :)`
`04:07.33`	`jpsharp`	`yum`
`04:08.02`	`slav3_kitten`	`offers jpsharp and everyone but linocisco a bowl`
`04:08.03`	`*** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806)`
`04:08.05`	`*** join/#asterisk xnt14 (~xnt14@2001:470:1f07:14cf:212:3fff:fe4f:3149)`
`04:08.17`	`slav3_kitten`	`he/she an have some when they decide what they want`
`04:08.18`	`*** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell)`
`04:08.19`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`04:08.47`	`slav3_kitten`	`i can't stand indecisive people`
`04:09.03`	`*** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1003:215:5dff:fe07:4806)`
`04:09.07`	`*** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez)`
`04:09.53`	`*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)`
`04:10.54`	`jpsharp`	`I'm originally from Texas, so I always enjoy a good shot of chili.`
`04:11.43`	`slav3_kitten`	`i really want to try real chili from texas`
`04:23.27`	`drmessano`	`They buy it from New York City`
`04:24.46`	`slav3_kitten`	`lol`
`04:25.40`	`linocisco`	`hi all, I have another question. Is there any diagram for typical asterisk setup in LAN avoiding voice and data traffic`
`04:26.16`	`drmessano`	`Yeah`
`04:26.25`	`drmessano`	`VLAN it or put it on a different physical network`
`04:28.59`	`slav3_kitten`	`linocisco, what do you men avoiding voice and data traffic?`
`04:29.37`	`slav3_kitten`	`typically you QoS the vlan for voice traffic at a higher priority so you have uninterrupted audio from your voip`
`04:30.13`	`linocisco`	`slav3_kitten, I will not run extra cables for VOICE using asterisk. I will use only same LAN cables that are used for data traffice like internet.`
`04:30.55`	`linocisco`	`slav3_kitten, yes . that is what I meant. QoS. is there any typical settings for a smooth running of both in the same network like peaceful coexistance.`
`04:31.04`	`linocisco`	`?`
`04:31.36`	`drmessano`	`How many users do you have?`
`04:31.40`	`drmessano`	`How many endpoints?`
`04:31.41`	`slav3_kitten`	`me i have mine setup so i not only have vlans different i also have some inter vnet routing of ssh only for administration. that way if someone gets in via a security hole they are denied access to my data network. my phones have a 3 port switch built into them so ethernet hooks to phone, computer hooks to pc port, everything is put on the right vlan via cdp telling the phone what to send where.`
`04:31.55`	`slav3_kitten`	`drmessano, he said home use, 3 extensions`
`04:32.00`	`drmessano`	`Uhhh`
`04:32.09`	`drmessano`	`I dont think you need a VLAN or QoS for 3 phones`
`04:32.11`	`slav3_kitten`	`on a 10/100 network it could all be in the same pipe wihtout problem`
`04:33.27`	`linocisco`	`drmessano, 100 users`
`04:33.37`	`slav3_kitten`	`linocisco, hold the freaking phone here`
`04:34.01`	`slav3_kitten`	`how did you jump from home user, 50 dollar max budget, 3 fxs endpoints.... to a 100 user phone network`
`04:34.17`	`linocisco`	`slav3_kitten, I told u that I have another question`
`04:34.17`	`drmessano`	`He's mormon`
`04:34.28`	`drmessano`	`Happens fast`
`04:34.39`	`drmessano`	`1 kid, 100 kids..`
`04:34.44`	`slav3_kitten`	`fair point drmessano`
`04:35.21`	`slav3_kitten`	`linocisco, normally if you come into a channel noobing it up, and you say you have another question. it's related to the original noob reason you came in`
`04:35.33`	`linocisco`	`drhonk\|afk, slav3_kitten , I know all of u are funny about my questions. It is really very diffcult to meet very interactive asterisk geeks due to different GMT`
`04:36.06`	`slav3_kitten`	`linocisco, 100 users. so it's a business, what's your current network load?`
`04:36.42`	`linocisco`	`drmessano, slav3_kitten , please be patience and sorry for my stupid questions in no time. how can I check network load?`
`04:37.01`	`slav3_kitten`	`sighs`
`04:37.24`	`ChrisInSydney`	`afternoon all`
`04:38.28`	`slav3_kitten`	`how much do you know about switches, vlans, connecting multiple switches and preserving throughput?`
`04:38.35`	`slav3_kitten`	`yo ChrisInSydney`
`04:39.50`	`ChrisInSydney`	`slav3_kitten: You could tell him (linocisco) about cactus`
`04:40.20`	`ChrisInSydney`	`ive just come in on this`
`04:41.22`	`slav3_kitten`	`ChrisInSydney, he/she/it came in here asking about a 50 dollar or less asterisk server with 1 fxo and 3 fxs for home use. suddenly it's now a 100 user phone network, asking about qos, etc`
`04:41.31`	`ChrisInSydney`	`hmm`
`04:41.57`	`ChrisInSydney`	`100 users @ 100k per active user`
`04:42.01`	`drmessano`	`It should really be the same`
`04:42.09`	`drmessano`	`I mean, 3 users x 33...`
`04:42.19`	`drmessano`	`You just need 33x more oomph`
`04:42.56`	`ChrisInSydney`	`go in with 10/100 setup and split / upgrade when people complain. Simple :-)`
`04:43.00`	`slav3_kitten`	`100 users that's a minimum of 3 48 port switches, hopefully with gigabit interconnect interfaces or even 10 gig if they are modern switches. then you have to consider what the current network load is, most business i've ever helped network had their gear at 80 or 90% capacity to keep costs down.`
`04:43.34`	`ChrisInSydney`	`you could stack a bunch of C3750-48ps-s`
`04:43.56`	`ChrisInSydney`	`gig gbic to the * box and you're away`
`04:44.07`	`*** join/#asterisk unicron (~unicron@64.135.37.5)`
`04:44.08`	`slav3_kitten`	`mmm cisco :D`
`04:44.31`	`ChrisInSydney`	`3750s are cheap 2nd hand`
`04:45.06`	`linocisco`	`ChrisInSydney, not cheap, not available in my country. but need to find how to acquire anyway`
`04:45.06`	`ChrisInSydney`	`$350-500 for a 24 port PoE depending on age / condtion her in au`
`04:45.10`	`drmessano`	`I have a 3750-48ps someone accidentally ordered with PoE`
`04:45.17`	`drmessano`	`Only $3000 difference`
`04:45.19`	`drmessano`	`No biggue`
`04:45.21`	`drmessano`	`biggie`
`04:45.29`	`slav3_kitten`	`but if they don't understand the basics of networking this is all moot`
`04:45.42`	`slav3_kitten`	`drmessano, i have a 3500xl inline power`
`04:45.46`	`drmessano`	`You mean like Facebook?`
`04:45.50`	`slav3_kitten`	`i picked it up for 30 bucks on ebay`
`04:45.54`	`ChrisInSydney`	`drmessano: not if ts not your wallet`
`04:45.54`	`drmessano`	`Nice`
`04:46.07`	`drmessano`	`Well, it was the companys wallet`
`04:46.09`	`slav3_kitten`	`yea i was pretty proud of my auction sniping`
`04:46.30`	`ChrisInSydney`	`I traded a couple of 3750 48 PoEs`
`04:46.34`	`linocisco`	`slav3_kitten, 30 Bucks for 3500 cisco catalyst switch?`
`04:46.40`	`ChrisInSydney`	`10/100 but so are the phones`
`04:46.42`	`epaphus`	`Hello all. So I configured an asterisk server with a DID number. The problem is that when i dial the DID fromt he outside all i get is a fast busy signal. In the asterisk server i have verbosity to 30 and I dont see not even the try... however I placed tcpdump to listen and I do see that the remote server is sending the call. How else can i troubleshoot the issue`
`04:46.43`	`linocisco`	`slav3_kitten, holdy goat.`
`04:46.44`	`slav3_kitten`	`linocisco, free shipping`
`04:46.51`	`ChrisInSydney`	`they wanted something with a web gu`
`04:47.33`	`linocisco`	`ChrisInSydney, all friends here are welcome if you all need cisco router/switch config as I am CCNA certified with 1000/1000`
`04:47.48`	`linocisco`	`all questions, Free of charge`
`04:47.53`	`ChrisInSydney`	`<epaphus> sip rego or direct IP connect ? BTW check your context / destination extension`
`04:48.06`	`ChrisInSydney`	`calls dont always show up f you have these bits wrong`
`04:48.16`	`slav3_kitten`	`linocisco, i actually had an issue with my 3500 and phone ports, had to set them up as trunks`
`04:48.38`	`ChrisInSydney`	`linocisco, so where in the world are you ?`
`04:48.49`	`slav3_kitten`	`i was under the impression i just needed to set access an voice vlan; and that cdp would take care of things`
`04:48.54`	`linocisco`	`ChrisInSydney, I am from south east asia`
`04:49.07`	`linocisco`	`ChrisInSydney, GMT+6:30`
`04:49.31`	`linocisco`	`slav3_kitten, did you configure sip traffice allowed?`
`04:49.33`	`ChrisInSydney`	`so am I, just more south and more east than most, but not as much as them damned kiwis ;-)`
`04:50.02`	`linocisco`	`ChrisInSydney, but I know there is blacktown in sydney`
`04:50.13`	`linocisco`	`ChrisInSydney, my friends are there`
`04:50.25`	`ChrisInSydney`	`we dont talk about there. its too far west ;-)`
`04:50.27`	`epaphus`	`ChrisInSydney, my DID is configured based on a SIP trunk.. and the context and destination extension... I made sure that the REGISTRY string does contain the correct extension to ring.. what do you mean by context sorry iam a newbie`
`04:50.40`	`slav3_kitten`	`linocisco, nope, didn't know that was a command`
`04:51.23`	`slav3_kitten`	`i mean i set it as a trunk port with default untagged traffic to vlan 2 and voice traffic to vlan 3 and it worked fine`
`04:51.50`	`ChrisInSydney`	`epaphus: in your sip.conf stuff, you'll have a context thingy which will be where the sip trunks calls land`
`04:52.00`	`ChrisInSydney`	`context=default`
`04:52.16`	`ChrisInSydney`	`context=sip-in etc`
`04:53.44`	`ChrisInSydney`	`now, there is a small gotcha, f you have two registrations to the same SIP server all calls will go to the first (or last I cant remember which) context you define for that SIP service`
`04:54.03`	`*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)`
`04:54.27`	`ChrisInSydney`	`so you can only effectively have one context per SIP service for inbound`
`04:54.50`	`WIMPy`	`hopefully`
`04:54.52`	`epaphus`	`ChrisInSydney, what i have is context=from-external in sip.conf`
`04:55.35`	`ChrisInSydney`	`cool so if you have register => 12345:password@somesipbloke.com/12345`
`04:56.21`	`ChrisInSydney`	`then the call will go to exten => 12345,1,NoOp(Yay found it) in the context [from-external] in your extensions.conf`
`04:56.37`	`ChrisInSydney`	`for example`
`04:56.46`	`*** join/#asterisk mchou (~quassel@unaffiliated/mchou)`
`04:57.01`	`ChrisInSydney`	`Hey WIMP :-)`
`04:57.17`	`ChrisInSydney`	`s/Hey WIMP :-)/Hey WIMPy :-)`
`04:57.40`	`ChrisInSydney`	`hopes that WIMPy realises that that was NOT deliberate`
`04:58.14`	`ChrisInSydney`	`also doesnt care if he does as he doesnt know what I look like ;-)`
`04:58.16`	`WIMPy`	`Yes, I know that you sometimes hit return one too early :-)`
`04:58.37`	`ChrisInSydney`	`and I still havent fixed the <I> key`
`05:00.02`	`epaphus`	`ChrisInSydney, yeah i think the context as well as the desitnation extension all matches. I foudn out that i can enable sip debug.. i think that has more info.. sec`
`05:00.43`	`ChrisInSydney`	`thats what you need`
`05:01.11`	`ChrisInSydney`	`see which context the call is hitting`
`05:03.16`	`slav3_kitten`	`ah one cig before bed`
`05:03.27`	`slav3_kitten`	`i really need to stop smoking`
`05:04.18`	`ChrisInSydney`	`Playing with queues and my modifiied (kludged) app_dial.c. have found that it helps if you Answer() the call before sendnig t to the queue. A cup of coffee and a few hours of somethng else helped me find that one :-/`
`05:05.04`	`WIMPy`	`Yes, it isn;t really obvious, which applications will auto-answer.`
`05:05.47`	`ChrisInSydney`	`should be embarrassed by that but he has no shame`
`05:06.42`	`epaphus`	`ChrisInSydney, this is the sip debug text when the calls comes in... I dont know what iam looking for though www.pastebin.ca/2249879`
`05:06.46`	`ChrisInSydney`	`WIMPy: I have hacked the dial app so that I can now specify different announcement files`
`05:07.33`	`WIMPy`	`ChrisInSydney: I could have used that for ConfBridge, but it was way easier to just do it in the dialplan.`
`05:07.57`	`WIMPy`	`It actually has further benefits. You can play user count and the join announcement at the same time that way.`
`05:08.00`	`ChrisInSydney`	`<PROTECTED>`
`05:08.30`	`ChrisInSydney`	`I also configured 9 to connect the call but delete the callers announcement file`
`05:08.56`	`ChrisInSydney`	`epaphus looking now`
`05:09.50`	`ChrisInSydney`	`SIP/2.0 401 Unauthorized`
`05:09.55`	`ChrisInSydney`	`line 41`
`05:12.07`	`epaphus`	`ChrisInSydney, yes i see that... how do i fix this though... :S`
`05:12.12`	`ChrisInSydney`	`hang on, so you have a register => something:somepass@sp.callmyway.net/4000 ??`
`05:12.20`	`epaphus`	`exactly`
`05:12.50`	`ChrisInSydney`	`may not be an issue as it will send an invte, get rejected then resend wth an auth hash`
`05:13.59`	`ChrisInSydney`	`and you have your syntax correct. Can you call that extension at that context internally ??`
`05:14.50`	`ChrisInSydney`	`paste bin the sip.conf and extensions.conf if you're cool with that. Change passwords please :-)`
`05:15.00`	`epaphus`	`sure one min`
`05:15.38`	`epaphus`	`i dont think i can call that extension, it was meant to be in a context called from the outside`
`05:16.27`	`ChrisInSydney`	`WIMPy, so now I can have a simple call screen for mobiles/cell phones that are part of a queue, without having their voicemail eat the calls`
`05:17.30`	`ChrisInSydney`	`it also gives me more fexibility with the call screen as I can skip the recording phase all together and just have it work simlar to followme()`
`05:19.28`	`WIMPy`	`To me, followme used to mean reverse forwarding.`
`05:19.50`	`*** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru)`
`05:19.51`	`ChrisInSydney`	`:-)`
`05:20.52`	`WIMPy`	`All the well known words having a different meaning when using Asterisk ist not very user friendly.`
`05:21.02`	`ChrisInSydney`	`its means that when I get an incomming call on my mobiile I can now send to voicemal, send to recepton, send to the service queue etc rather than be stuck with the basic options`
`05:21.34`	`ChrisInSydney`	`epaphus: Hows the paste bn gong, i have to run. I havea gig to go and set up for :D`
`05:22.45`	`epaphus`	`SEC`
`05:22.50`	`ChrisInSydney`	`cool`
`05:23.03`	`slav3_kitten`	`IPSEC?`
`05:23.15`	`epaphus`	`www.pastebin.ca/2249888 and www.pastebin.ca/2249887`
`05:23.24`	`epaphus`	`ChrisInSydney,`
`05:23.34`	`slav3_kitten`	`you're a canadian?`
`05:23.36`	`ChrisInSydney`	`:-)`
`05:23.53`	`epaphus`	`iam not. I changed 400 to 40001293 as the extension btw`
`05:24.12`	`epaphus`	`all should be accordingly`
`05:26.15`	`ChrisInSydney`	`codecs in sip.conf`
`05:26.28`	`ChrisInSydney`	`disallow=all`
`05:26.32`	`ChrisInSydney`	`allow=ulaw`
`05:26.36`	`ChrisInSydney`	`allow=alaw`
`05:26.45`	`slav3_kitten`	`epaphus, i'm a noob but it looks like you have no default codecs and didn't specify one for your sip trunk`
`05:27.06`	`epaphus`	`hmm ok...!...`
`05:27.14`	`ChrisInSydney`	`just humour me and add those to the sip entries for the trunk and two extensions`
`05:27.39`	`ChrisInSydney`	`leave out 729`
`05:27.44`	`ChrisInSydney`	`for now`
`05:28.15`	`epaphus`	`done, retesting`
`05:28.45`	`epaphus`	`same, call doesnt go through`
`05:29.10`	`ChrisInSydney`	`it wll make it easier if you get rid of the default "stuff" in those files`
`05:29.22`	`ChrisInSydney`	`just keep a copy somewhere you can refer to`
`05:29.34`	`epaphus`	`your right. :S`
`05:29.40`	`ChrisInSydney`	`sip show registry`
`05:29.52`	`ChrisInSydney`	`from the CLI`
`05:30.33`	`epaphus`	`www.pastebin.ca/2249890`
`05:30.35`	`*** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)`
`05:30.39`	`epaphus`	`sorry i had to paste it there hehe`
`05:32.32`	`epaphus`	`so all we are left with is the sip debug.. and that doesnt say much :S`
`05:33.15`	`ChrisInSydney`	`check your passwords. I have seen where things register but dont work when the password it wrong. Makes hacking harder, but also debugging too`
`05:33.44`	`ChrisInSydney`	`BTW, are you calling yourself from one of your extensions ??`
`05:33.56`	`ChrisInSydney`	`whats the public number ??`
`05:34.09`	`epaphus`	`ChrisInSydney, no. iam calling from th eoutside.. so it reaches the correct context. DID number is 40001293`
`05:34.50`	`epaphus`	`i checked the password.`
`05:34.55`	`epaphus`	`passswords all ok`
`05:35.27`	`ChrisInSydney`	`ok so what is the full e.164 number, I'll give you a call`
`05:35.45`	`epaphus`	`ChrisInSydney, its in Costa Rica.. you cant ;)`
`05:35.56`	`epaphus`	`but believe me.. i tried using the PSTN`
`05:36.12`	`ChrisInSydney`	`I'll check my tarrif`
`05:36.42`	`epaphus`	`i dont think these numbers are advertised for international carriers, they are only valid in CR PSTN`
`05:38.14`	`ChrisInSydney`	`thats ok`
`05:42.45`	`ChrisInSydney`	`Costa Rica Rate $10 buys you`
`05:42.46`	`ChrisInSydney`	`Costa Rica 0.0629 159 min`
`05:42.46`	`ChrisInSydney`	`Costa Rica Mobile 0.0629 159 min`
`05:45.26`	`epaphus`	`:)`
`05:45.31`	`epaphus`	`i removed all the comments from my config files`
`05:45.39`	`ChrisInSydney`	`cool`
`05:45.53`	`epaphus`	`so i guess no ideas for now? ChrisInSydney`
`05:46.24`	`ChrisInSydney`	`I have a few but 'm out of time. have to go and load in for another crappy covers band gig`
`05:47.05`	`ChrisInSydney`	`thats what we have on our poster Xp`
`05:47.22`	`ChrisInSydney`	`thinks that thats what we should have on our poster`
`05:47.49`	`ChrisInSydney`	`sip debug on`
`05:48.09`	`epaphus`	`ok..`
`05:48.24`	`epaphus`	`thanks though.`
`05:48.48`	`*** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au)`
`05:48.54`	`ChrisInSydney`	`crash :-/`
`05:49.23`	`ChrisInSydney`	`mate, I have to bolt. There are plenty of capable people here to help`
`05:51.36`	`epaphus`	`ChrisInSydney, its ok, thanks man`
`05:52.57`	`ChrisInSydney`	`c y`
`05:53.04`	`ChrisInSydney`	`ya`
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`07:07.04`	`devdvd`	`hey all, is there a way to retrieve the filename that a voicemail was recorded to in the dialplan?`
`07:07.39`	`devdvd`	`so say someone leaves a voicemail, i want to be able to pull into a variable what filename the vm got recorded to`
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`07:25.36`	`kaldemar`	`devdvd: VM_MESSAGEFILE`
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`08:23.41`	`aurs`	`oh, so "watch the video" actually means "fill out this registration form" ? Ok. (www.asterisk.org)`
`08:23.48`	`aurs`	`:P`
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`08:47.40`	`ChannelZ`	`English is such a tricky language.`
`08:49.23`	`pppingme`	`ChannelZ no it aint..`
`08:52.21`	`ChannelZ`	`did you mean: no i taint`
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`08:55.57`	`paul	`help. dahdi is properly installed but meetme is disabled in make menuselect :(`
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`09:02.16`	`*** join/#asterisk StaRetji (~LittleAll@91.142.129.1)`
`09:02.22`	`StaRetji`	`Howdy folks`
`09:02.41`	`StaRetji`	`I need little help. I have a client which doesn't want to work with me because I use Asterisk`
`09:02.43`	`StaRetji`	`lol`
`09:02.44`	`aurs`	`paul: do you have any dahdi compatible hardware? I vaguely remember that I needed the dahdi_dummy module to get meetme
`09:02.56`	`StaRetji`	`he said he had some bad experience with asterisk partner`
`09:03.02`	`aurs`	`paul: unless you have any dahdi compatible hw, that is
`09:03.24`	`StaRetji`	`so, is there a way he would know if I tell him I use NASA supergigamega switch? :D`
`09:03.27`	`paul	`i dont have any hardware`
`09:03.36`	`paul	`@aurs`
`09:03.47`	`StaRetji`	`if yes, can I change it to other name?`
`09:03.48`	`StaRetji`	`thx`
`09:03.51`	`paul	`im using asterisk 1.4 btw`
`09:04.20`	`bulkorok`	`StaRetji: you should change the user-agent header`
`09:04.39`	`StaRetji`	`bulkorok: thx man`
`09:04.45`	`StaRetji`	`is that all? nothing else?`
`09:05.07`	`bulkorok`	`this is the point where Asterisk 1. bla is`
`09:05.20`	`bulkorok`	`change it to whatever you want...`
`09:05.21`	`aurs`	`paul: try to compile dahdi with 'make MODULES_EXTRA="dahdi_dummy"' and 'make MODULES_EXTRA="dahdi_dummy" install'
`09:05.36`	`paul	`ok thanks... will try`
`09:06.31`	`aurs`	`paul: let me know if it works!
`09:06.35`	`StaRetji`	`bulkorok: hm, it was changed already to SIP server, so...`
`09:06.48`	`StaRetji`	`useragent=SIP Server`
`09:07.05`	`bulkorok`	`StaRetji: it's ok... reload/restart asterisk ?!`
`09:07.19`	`StaRetji`	`well, no need, this conf was here for months :)`
`09:07.31`	`bulkorok`	`you can check with 'sip set debug on' on cli and issue a call...`
`09:07.34`	`StaRetji`	`so, either he is psychic or there is other way he knows :)`
`09:07.49`	`bulkorok`	`then look for the INVITE message and look what User-Agent: tells`
`09:07.57`	`StaRetji`	`got it`
`09:08.18`	`bulkorok`	`it could be in the SDP-header...`
`09:08.19`	`StaRetji`	`thx dude`
`09:08.28`	`bulkorok`	`check the INVITE`
`09:08.56`	`kaldemar`	`paul: did you re-run the configure script for asterisk?
`09:09.23`	`kaldemar`	`paul: also, which 1.4 version are you using?
`09:09.47`	`paul	`make menuselect for asterisk : XXX 32. app_meetme`
`09:09.49`	`paul	`sad`
`09:10.02`	`paul	`need to run ./configure first?`
`09:10.08`	`kaldemar`	`of course.`
`09:10.18`	`aurs`	`./configure AFTER installing dahdi, indeed`
`09:10.22`	`paul	`asterisk asterisk-1.4.21.2`
`09:10.22`	`kaldemar`	`do you have a specific reason to run a 1.4.X version?`
`09:10.27`	`paul	`ok will try brb`
`09:10.52`	`paul	`using vicidial :( ithink its only compatible with 1.4 :(`
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`09:12.07`	`paul	`:( still XXX 32. app_meetme on menuselect`
`09:12.19`	`*** part/#asterisk StaRetji (~LittleAll@91.142.129.1)`
`09:12.32`	`kaldemar`	`paul: your version of asterisk does not support DAHDI.
`09:12.44`	`kaldemar`	`paul: first version to do so was 1.4.22.
`09:13.43`	`*** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme)`
`09:13.46`	`paul	`:(`
`09:13.54`	`kaldemar`	`upgrade to the newest 1.4 unless you can use a recent branch that's actually alive, unlike 1.4.`
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`09:14.29`	`paul	`is it ok to reinstall it on top of 1.4.21?`
`09:15.19`	`bulkorok`	`paul: check configure --help for directory-options
`09:16.39`	`paul	`hmmm i mean if i download latest 1.4 then do a ./configure make make install should be ok without removing old version?`
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`09:16.57`	`kaldemar`	`paul: no. uninstall the current version first.
`09:17.25`	`bulkorok`	`but you will lose the "compatible" version to vicidial...`
`09:17.32`	`kaldemar`	`paul: you can do that with "make uninstall"
`09:17.54`	`paul	`according to guide in vicidial wget http://download.vicidial.com/required-apps/asterisk-1.4.27.1-vici.tar.gz so its compatible :)`
`09:18.09`	`kaldemar`	`vicidial says "Asterisk server(any version before Asterisk 1.6)"`
`09:18.10`	`bulkorok`	`:-)`
`09:18.23`	`paul	`will make uninstall remove config files?`
`09:18.24`	`bulkorok`	`wasn't sure about that`
`09:18.28`	`bulkorok`	`no`
`09:18.42`	`bulkorok`	`but backup them is everytime a good one...`
`09:19.03`	`kaldemar`	`no, it does not remove config files. uninstall-all would remove configs too.`
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`09:22.00`	`paul	`finally!`
`09:22.09`	`paul	`meetme is enabled in menuselect lolz`
`09:22.23`	`paul	`thanks guys`
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`09:42.19`	`ChannelZ`	`just in time to learn about ConfBridge!`
`09:42.54`	`ChannelZ`	`does the upgrade dance`
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`09:54.47`	`jacekowski`	`hi, i've got question about DPMA - basically, initially when phone is started for the first time i can see list of extensions and phone can download it, but after resetting phone to factory defaults i can't see that extension any more`
`09:59.04`	`ectospasm`	`jacekowski: do you have any of the PIN or MAC settings defined in res_digium_phone.conf?`
`09:59.16`	`jacekowski`	`no`
`09:59.41`	`jacekowski`	`basically, once phone is provisioned it disappears from the list, deleting astdb makes it appear again`
`09:59.57`	`ectospasm`	`what version of DPMA?`
`10:01.14`	`ectospasm`	`jacekowski: "digium_phones show version"`
`10:01.41`	`ectospasm`	`...and what version of Asterisk?`
`10:02.04`	`jacekowski`	`Digium Phone Module for Asterisk Version 1.8.11_1.3.0`
`10:02.11`	`jacekowski`	`Asterisk 1.8.11-cert8`
`10:02.23`	`*** join/#asterisk sekil (~sekil@78.24.104.73)`
`10:03.24`	`ectospasm`	`jacekowski: sounds like you may have come across a bug. I recommend you contact Digium support`
`10:04.04`	`ectospasm`	`either that or you have a configuration issue I've not come across`
`10:06.56`	`*** join/#asterisk ruied (~AndChat66@241.208.37.188.rev.vodafone.pt)`
`10:15.21`	`jacekowski`	`how does your res_digium_phones.conf look?`
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`10:21.14`	`jacekowski`	`deleting DigiumPhoneUsers entry from asterisk database makes that extension reappear`
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`10:33.01`	`ectospasm`	`jacekowski: pastebin your res_digium_phone.conf... I don't have one I can paste right now...`
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`10:57.09`	`jacekowski`	`ectospasm: http://pastebin.com/wN3uk7tM`
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`10:59.33`	`jmls1`	`good morning all.`
`10:59.35`	`jmls1`	`has anyone used camel netty to connect to the asterisk ami ?`
`11:01.32`	`ectospasm`	`jacekowski: nothing there looks immediately wrong with your configuration`
`11:01.49`	`jacekowski`	`it's not that much of a problem to be honest`
`11:02.12`	`*** join/#asterisk timahvo1 (~rogue@41.212.120.182)`
`11:02.13`	`jacekowski`	`in case i have to reprovision a phone i can always delete it's entry from database`
`11:04.22`	`ectospasm`	`jacekowski: I'm not seeing the same behavior on my lab machine, but it's running DPMA 1.2.0 and Asterisk 1.8.11-cert7`
`11:04.32`	`ectospasm`	`...upgrading now to see if I run into the same prolem.`
`11:04.43`	`ectospasm`	`s/prolem/problem/`
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`11:09.12`	`ectospasm`	`jacekowski: I appear to be getting the same thing with 1.3.0 and cert8`
`11:09.52`	`ectospasm`	`jacekowski: wait, my phone is updating firmware... gonna see if that makes a difference (I doubt it)`
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`11:22.29`	`ectospasm`	`jacekowski: I see the same as you, but apparently deleting the astdb family/keys doesn't seem to be enough to make the extensions show back up`
`11:24.03`	`ectospasm`	`jacekowski: looks like I have to restart Asterisk to get the full list of extensions back`
`11:25.11`	`ectospasm`	`jacekowski: if you want to get in on the bug I'm going to file, call Digium technical support (+1.256.428.6161) so I can associate it with a support case.`
`11:37.27`	`ectospasm`	`jacekowski: the more I think about it, I shouldn't file the bug until you open a support case or call...`
`11:38.58`	`*** join/#asterisk funky1 (~jarvis@p5481A4B6.dip.t-dialin.net)`
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`11:40.47`	`funky1`	`I have a question I run asterisk 1.8 I have an extension that registers via the internet, works great, i forward my incoming calls to that extension, works as well, extension rings when being called but sound only kicks in after exactly 1 minutes passes, during the first minute no sound at all, then regular 2 way sound, always exactly one minute, what could be the problem?`
`11:42.03`	`jacekowski`	`ectospasm: i'm not in the US`
`11:42.21`	`jacekowski`	`ectospasm: it may be better idea if you call them and file the bug`
`11:42.22`	`ectospasm`	`sounds like a NAT or firewall issue, as if the firewall doesn't allow two-way RTP/audio traffic at first`
`11:42.32`	`ectospasm`	`jacekowski: no, I work for Digium support, and I'm at work now`
`11:42.42`	`ectospasm`	`I really can't file a bug without an associated support case`
`11:42.47`	`jacekowski`	`ahm`
`11:42.48`	`jacekowski`	`ok`
`11:42.59`	`ectospasm`	`jacekowski: and that's an international number... I can get you the EMEA number if you'd like`
`11:43.06`	`jacekowski`	`UK ideally`
`11:43.14`	`funky1`	`ectospasm: thought so as well that it could be NAT/firewall problem but i found it weird that it is exactly one minute (to the second)`
`11:43.16`	`ectospasm`	`yeah, it's a UK number`
`11:43.42`	`ectospasm`	`jacekowski: +44 (207) 183-7577`
`11:44.08`	`ectospasm`	`funky1: yeah, that's strange`
`11:44.58`	`funky1`	`will play around some more...`
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`11:51.33`	`ghost75`	`is jitterbuffer only usable for incoming traffic ?`
`11:56.00`	`ectospasm`	`ghost75: the jitter buffers are designed to work on the receiving end, which means inbound traffic`
`11:56.20`	`ectospasm`	`...if you need a jitter buffer on outbound traffic, the far end would have to implement the jitter buffer`
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`12:04.58`	`MatthewJava89`	`hello`
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`12:07.58`	`jmls`	`trying to use starpy, think that I've installed all I need (man, I miss the gem functionality - is there an equivalent in python ?)`
`12:08.36`	`jmls`	`however, have a problem trying to run a sample program`
`12:08.43`	`jmls`	`ImportError: No module named utilapplication`
`12:08.53`	`*** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain)`
`12:13.49`	`mute`	`i just setup asterisk for gvoice<->sip, is cdr defaults suppose to create call logs??`
`12:14.32`	`jacekowski`	`ectospasm: i've just checked version 1.2 of DPMA and it works fine there`
`12:14.41`	`jacekowski`	`ectospasm: so it looks more like DPMA problem than asterisk`
`12:15.01`	`ectospasm`	`jacekowski: so DPMA 1.3.0 is the key piece here.`
`12:15.27`	`jacekowski`	`yes`
`12:19.17`	`ectospasm`	`Putting that into my report to engineering`
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`12:23.53`	`leifmadsen`	`jmls: seems like you might just be missing a dependency`
`12:24.26`	`leifmadsen`	`jmls: some python library (and yes, I like gem)`
`12:26.17`	`jmls`	`leifmadsen: i think that utilapplication is defined in starpy`
`12:26.41`	`leifmadsen`	`jmls: which is trying to import something...`
`12:26.46`	`jmls`	`just trying to run the examples / priexhaustionbare.py sample`
`12:27.24`	`jmls`	`which has "import utilapplication" on the 5th line`
`12:27.35`	`leifmadsen`	`right...`
`12:27.46`	`leifmadsen`	`which is a way to import libraries`
`12:27.58`	`*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)`
`12:28.02`	`leifmadsen`	`at least as far as I can tell (I'm not a python guy)`
`12:28.30`	`jmls`	`is not a python guy either ;)`
`12:29.14`	`leifmadsen`	`I'm just saying, it's trying to import something that doesn't exist, so you're missing a library or dependency`
`12:31.12`	`jmls`	`leifmadsen: sure, appreciate that.`
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`12:33.03`	`MatthewJava89`	`how get information about phone i.e how many phone have a line`
`12:33.25`	`leifmadsen`	`from where?`
`12:34.21`	`MatthewJava89`	`i write app in java`
`12:34.31`	`MatthewJava89`	`and i look a manager`
`12:34.53`	`MatthewJava89`	`i don't now where is information abaout line`
`12:34.55`	`leifmadsen`	`you're going to have to elaborate on what you're expecting`
`12:35.43`	`leifmadsen`	`what are you expecting to poll about number of lines? I'm not sure you can ask for that information dynamically. You'll have to allow a device type to be specified, then just know the configuration of that device`
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`12:38.09`	`MatthewJava89`	`how the best solution on get information who call now`
`12:38.37`	`MatthewJava89`	`i listening manager and parse dial, bridge`
`12:39.14`	`MatthewJava89`	`whether is good road?`
`12:39.31`	`MatthewJava89`	`whether this is good road?`
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`12:39.58`	`leifmadsen`	`using the manager interface is the most likely method to listen for current call activity, yes.`
`12:40.55`	`MatthewJava89`	`ok thx`
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`12:52.22`	`SeRi`	`good morning all`
`12:53.58`	`unicron`	`hi`
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`12:54.10`	`wonderworld`	`hey`
`12:56.06`	`kjs`	`hi`
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`13:17.13`	`boch`	`hi all`
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`13:21.41`	`boch`	`i am having a problem with asterisk 1.8.17. it is not closing some socket/fifo and it finally crashes because of too many open files.. any clue on what is not being closed and why ?? thanks in advance`
`13:27.28`	`*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)`
`13:29.40`	`leifmadsen`	`boch: that is pretty much an impossible question to answer without backtraces and other various debugging information`
`13:34.32`	`boch`	`leifmadsen, i know, what info will be helpful ?`
`13:34.56`	`leifmadsen`	`my answer already describes at least 1 bit of information`
`13:35.16`	`leifmadsen`	`boch: https://wiki.asterisk.org/wiki/display/AST/Debugging`
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`13:40.56`	`jmls`	`anyone use lxc containers ?`
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`13:47.12`	`fprior`	`Hi again. Another day testing webrtc; I followed this tutorial http://code.google.com/p/sipml5/wiki/Asterisk#Configure_sipml5 but no voice in my call. Do you know another HowTo like that ?`
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`13:58.03`	`SeRi`	`jmls: http://slackware.ponce.cc/blog/2012/11/02/lxc-containers-on-slackware-14-0/`
`13:59.40`	`jmls`	`was wanting to ask a question on networking / ip address`
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`14:00.36`	`jmls`	`i want to run lxc on amazon ec2 , which has several elastic ip addresses, and was hoping to be able to allocate ip to each lxc`
`14:00.44`	`SeRi`	`jmls: I setup lxc on 13.37 some time a go. Is been a while now`
`14:02.21`	`pabelanger`	`jmls: Why even bother with lxc? Just launch another instance`
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`14:12.02`	`ollii`	`is it somehow possible to preserve or to pass channel variables while doing attended transfer via asterisk 1.8?`
`14:12.23`	`ollii`	`without user interaction`
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`14:58.13`	`FireAndIce`	`leifmadsen, Thanks a lot for your help. I got yesterday's issue solved.. :)`
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`15:04.43`	`Katty`	`HI LADS`
`15:06.38`	`*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)`
`15:10.50`	`n3hxs`	`Katty! how are you?`
`15:12.18`	`Katty`	`goodly :>`
`15:12.20`	`Katty`	`how'rechu?`
`15:12.31`	`*** join/#asterisk WolWid (WolWid@pD9ED5127.dip.t-dialin.net)`
`15:13.13`	`Katty`	`infobot: crittercam?`
`15:13.14`	`infobot`	`hmm... crittercam is Katty's live broadcast of The Nut House at http://www.ustream.tv/channel-popup/squirrel-critter-cam`
`15:13.18`	`Katty`	`infobot: forget crittercam`
`15:13.19`	`infobot`	`Katty: i forgot crittercam`
`15:13.23`	`Katty`	`infobot: critter cam?`
`15:14.18`	`Katty`	`infobot: crittercam is The Birdie Breakfast Buffet!!! http://www.ustream.tv/channel-popup/birdie-breakfast-buffet`
`15:14.18`	`infobot`	`Katty: okay`
`15:14.23`	`Katty`	`infobot: critter cam`
`15:14.26`	`Katty`	`infobot: crittercam`
`15:14.26`	`infobot`	`crittercam is probably The Birdie Breakfast Buffet!!! http://www.ustream.tv/channel-popup/birdie-breakfast-buffet`
`15:14.26`	`n3hxs`	`Pretty good considering it is Friday and the weekend is upon us.`
`15:15.00`	`cklimos`	`it looks ugly outside`
`15:15.15`	`Katty`	`n3hxs: woot friday :>`
`15:15.21`	`Katty`	`cklimos: it's a bit chilly out :<`
`15:15.41`	`Katty`	`the leaves are nearly all gone now`
`15:16.25`	`cklimos`	`is that an easter bunny preparing hot choc for the birds?`
`15:16.56`	`cklimos`	`oh a mouse`
`15:18.14`	`*** join/#asterisk ziz212 (~chatzilla@203.115.2.202)`
`15:18.18`	`ziz212`	`hi,`
`15:18.40`	`n3hxs`	`Not hi yet, but working on it...`
`15:19.01`	`ziz212`	`I need to use FXO port of a ATA to connect my PSTN line to Asterisk PBX? is this possible?`
`15:19.06`	`ziz212`	`pls let me know`
`15:20.00`	`WIMPy`	`ziz212: Technicalli not exately an ATA, but if you have one with an XO port, yes.`
`15:20.02`	`Katty`	`cklimos: eh?`
`15:20.19`	`Katty`	`cklimos: OH`
`15:20.26`	`Katty`	`cklimos: you're on the old page, viewing last winter`
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`15:20.45`	`Katty`	`cklimos: it's not there anymore...see the new birdie breakfast buffet link`
`15:21.09`	`*** join/#asterisk vlad_starkov (~vlad_star@194.67.37.86)`
`15:21.29`	`Katty`	`disregard the poor video quality, i've not really tweaked anything yet`
`15:21.31`	`n3hxs`	`You would think the page would have melted during the warm summer days.`
`15:21.32`	`cklimos`	`I see now`
`15:21.57`	`Katty`	`the feeder has been up...hmm, 3 days now? there are visitors but no 'frequenters' just yet`
`15:22.41`	`n3hxs`	`I took my feeder down so the hurricane wouldn't blow it 3 blocks down the road, and a mouse had a heyday.`
`15:22.51`	`n3hxs`	`or a birdfeed day.`
`15:23.01`	`Katty`	`i'm sure the mouse was very happy those nights`
`15:23.02`	`ziz212`	`Thanks .`
`15:23.37`	`Katty`	`n3hxs: i must show you my two latest Fresh Knits!`
`15:23.48`	`ziz212`	`This ATA came with FXS, FXO ports. if sip goes fail you can use FXO port . so how can I config`
`15:24.20`	`n3hxs`	`wonders which ATA ziz212 speaks of.`
`15:24.22`	`ziz212`	`that to take sip-->astersik-->ATA--->pstn call?`
`15:25.17`	`Katty`	`n3hxs: http://4.bp.blogspot.com/-0dYHCoFtfD4/UJp3KhpcwII/AAAAAAAADro/4dIxCVxv3DU/s1600/IMG_6718.JPG <- skeptical doc oct, and the derpy mister bun`
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`15:25.29`	`ziz212`	`my one is Welltech ATA-171M`
`15:25.56`	`cklimos`	`ziz your pstn line must be connected to the fxo module yes`
`15:26.05`	`Katty`	`n3hxs: they were celebrating election night.`
`15:26.32`	`ziz212`	`I have connected that to FXO port of the ATA... Welltech ATA-171M`
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`15:27.03`	`ziz212`	`need to config ata to take calls sip client-->astersik-->ATA--->pstn`
`15:27.19`	`orn`	`Hello there. Is there any way to force down a SIP channel that won't hang up if channel request hangup doesn't work?`
`15:27.20`	`ziz212`	`because I dont have option to use pci card`
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`15:27.45`	`*** mode/#asterisk [+o sruffell] by ChanServ`
`15:27.50`	`ziz212`	`i am using a pc with only one pci and that is connected to network card`
`15:33.42`	`n3hxs`	`ziz212, have you read http://www.manualslib.com/manual/408187/Welltech-Welltech-Asterisk-Applications.html#manual`
`15:33.46`	`n3hxs`	`yet?`
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`15:38.00`	`ziz212`	`thanks for the simple step that i didnt find because of my error....`
`15:38.03`	`ziz212`	`thanks`
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`15:39.10`	`slav3_kitten`	`well i now feel dumb, totally forgot about the "same" feature when writing my dialplan`
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`16:03.03`	`leifmadsen`	`same => ftw`
`16:03.20`	`drmessano`	`hell yesh`
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`16:20.07`	`[TK]D-Fender`	`orn, AMI Redirect it off a cliff`
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`17:26.07`	`navaismo`	`Is there a way to allow people to call me via chan_motif without inviting first with google mail?`
`17:31.58`	`cusco`	`hello folks`
`17:32.38`	`cusco`	`I have two network interfaces in asterisk box`
`17:32.57`	`cusco`	`how can I decive via wich asterisk will try to communicate its sip?`
`17:37.23`	`vince_`	`I am trying to setup a wireless sip phone but it is not registering. I can see through tcpdump the phone sends a sip registration request but asterisk never responds. there is nothing in logs... however, if I change the extension/userid to one that doesn't exist then it will complain in logs about no peers otherwise with the right extension/userid I see nothing in logs rejecting or accepting the registration`
`17:40.13`	`navaismo`	`vince_, turn on the sip debug and pb the results`
`17:40.21`	`[TK]D-Fender`	`vince_, "asterisk -vrrrrrrrrrrrrrr" , "sip set debug on" ---> PASTEBIN`
`17:40.23`	`[TK]D-Fender`	`~pb`
`17:40.24`	`infobot`	`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/; or install pastebinit with yum or aptitude.`
`17:40.27`	`[TK]D-Fender`	`^^`
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`17:46.43`	`vince_`	`http://pastebin.com/Nkf4PmZk`
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`17:50.14`	`navaismo`	`hmm i see 2 different networks and no nat settings but ill wait the fender response`
`17:52.09`	`vince_`	`the wireless is on a seperate network. firewall was opened for 5060 communction as obvious from the sip request coming in. however even though I think I see it trying to transmit out I never see that packet in tcpdump`
`17:52.12`	`vince_`	`hmm?`
`17:53.50`	`navaismo`	`then set your asterisk with natsettings and your extension too`
`17:55.15`	`vince_`	`maybe it is transmitting out on the wrong ethernet device?`
`17:55.59`	`navaismo`	`check your route table`
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`18:20.53`	`vince_`	`ok so I had tcpdump listen to all interfaces and I see now that it is defiantly coming in one interface and going out another`
`18:21.11`	`vince_`	`will have to see how the network administrator wants to handle this`
`18:21.15`	`vince_`	`thanks for the help`
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`18:39.08`	`devdvd`	`kaldemar, thank you for the assist last night on the vm_messagefile thing.`
`18:45.47`	`devdvd`	`running into another issue. Trying to call a python script from asterisk. http://pastebin.com/PuuCsMQt that paste contains the code for my macro. the problem i'm running into is that the first call to the external script is working just fine, does exactly what its supposed to, but the second call to a different script isn't working. However, the permissions on the script are just fine and I can run the script manually passing in t`
`18:45.47`	`devdvd`	`he correct parameters and it works just fine. the second script is returning the word RAN , however, when i noop ${blah} it doesn't even return a value which tells me maybe asterisk isn't even trying to run the script`
`18:47.52`	`devdvd`	`actually...i might know why, ill let you know`
`18:49.53`	`devdvd`	`or not`
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`19:00.52`	`devdvd`	`got it!`
`19:00.53`	`devdvd`	`:)`
`19:01.04`	`devdvd`	`one of the variables had some extranious space`
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`19:14.15`	`JasonL`	`I've getting reports of one way voice, and looking at the RTP trace I see that the SSRC from asterisk to the phone is changing 3 times in the first 7 packets... any idea what causes that?`
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`19:17.28`	`JasonL`	`Even though the SSRC changes, the sequence number is still in order..`
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`19:23.08`	`leifmadsen`	`zerohalo: ohai :)`
`19:23.16`	`zerohalo`	`hi!`
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`20:12.00`	`devdvd`	`JasonL, one way audio issues I have found are almost always caused by nat problems. What is your layout? Is your * box behind a nat, are your phones behind a nat?`
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`20:13.06`	`AqD`	`hey all!`
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`20:38.55`	`jmetro`	`hm.. whats your opinions on snom`
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`20:45.42`	`jacekowski`	`jmetro: it's a phone`
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`20:50.47`	`jpsharp`	`snoms are good.`
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`20:54.40`	`JasonL`	`devdvd: The phone and * box is behind the same router... looking at a tshark capture, i see the packets being sent to the phone, and can hear them fine, but the person on the phone didn't hear anything... and I notice that asterisk is changing the SSRC right away, i dont know if that's related or not...`
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`21:18.58`	`jmetro`	`turn off alg on your router`
`21:20.42`	`JasonL`	`the asterisk server and the phone are on the same network/subnet, so the router is not in play....`
`21:26.49`	`Qwell`	`jmetro: My (highly biased) recommendation would be for Digium phones.`
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`21:27.08`	`polysics`	`hello`
`21:27.19`	`polysics`	`what license are the Asterisk sound files under, if anyone know?`
`21:27.31`	`jmetro`	`i think hooking a digium phone up to asterisk would create a compatability supernova.`
`21:27.38`	`Qwell`	`polysics: A LICENSE file should be included in the tarball.`
`21:30.12`	`polysics`	`Qwell: sounds are under the same license as Asterisk?`
`21:30.22`	`Qwell`	`polysics: No. In the sounds tarball.`
`21:30.34`	`polysics`	`and maybe you know the direct answer: am I allowed to use those sound files not with Asterisk?`
`21:30.48`	`Qwell`	`You'll need to read the license and consult an lawyer.`
`21:30.54`	`polysics`	`whoa`
`21:31.02`	`Qwell`	`We cannot provide you with legal advice.`
`21:31.22`	`polysics`	`I think I will just ask someone at Digium, but I hoped someone would know :-)`
`21:31.27`	`Qwell`	`I am at Digium.`
`21:31.33`	`Qwell`	`Digium cannot provide you legal advice.`
`21:32.08`	`polysics`	`ofc, I thought this has popped up before. I was wrong. Thank you all the same :-)`
`21:32.11`	`Qwell`	`The same comment would be true if you asked a question about the GPL. It's up to you (and your counsel) to make those determinations.`
`21:33.01`	`Qwell`	`Of course it's come up. That's why there's a LICENSE file. Do the terms fit with your usage? How could we possibly know the answer to that?`
`21:33.10`	`Nivex`	`Our LUG is having an Open Source Showcase next month, and I'm doing the Asterisk demo. Any suggestions of fun stuff to do? and Digium folk, any swag you might want me to give away? ;)`
`21:33.35`	`Qwell`	`Nivex: maybe talk to newtonr next week`
`21:33.57`	`jeffspeff`	`JasonL, still having that issue?`
`21:34.37`	`Nivex`	`Qwell: can do. I still have his email from talking to him at Southeast Linuxfest.`
`21:34.54`	`Qwell`	`Nivex: I'm not sure what kind of swag (if any) he can provide, but he'd be your guy, if anyone.`
`21:35.01`	`jeffspeff`	`JasonL, make sure you have localnet defined correctly in sip and that your nat settings are correct as well as your bind addresses.`
`21:36.02`	`drmessano`	`If you enter the coupon code DRMESSANO123 in the Digium store, you can buy one card for double the price and get the second one free. You could pass that out at your LUG meeting`
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`21:36.18`	`Qwell`	`WOAH, dude, we gave you that code in private.`
`21:36.21`	`jmetro`	`i would <3 some * swag`
`21:36.35`	`drmessano`	`I felt it wrong not to share`
`21:36.42`	`drmessano`	`:(`
`21:36.42`	`Nivex`	`I'm still digging through moving boxes, but I think I still have some Asterisk stickers I picked up at SELF`
`21:36.46`	`Qwell`	`jmetro: http://store.digium.com/products.php?category_id=22`
`21:37.11`	`drmessano`	`I got a handful of those Asterisk stickers`
`21:37.26`	`drmessano`	`I missed this years SELF though. So my stickers are a year behind`
`21:37.45`	`drmessano`	`I may have unsupported stickers`
`21:37.52`	`drmessano`	`I should get the LTS stickers next time`
`21:38.57`	`jblack`	`I love you`
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`22:09.15`	`[sr]`	`howdy`
`22:09.45`	`[sr]`	`does the asterisk console has some options simular to the console one: HISTCONTROL=ignoredups`
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`22:25.06`	`[TK]D-Fender`	`[sr]: no`
`22:29.55`	`Nivex`	`does Asterisk support Opus yet?`
`22:32.26`	`file`	`Digium has not produced an Opus codec module.`
`22:33.31`	`Nivex`	`cool, thanks.`
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`23:25.19`	`K0HAX-1`	`Random question: If I have N phones and a phone system behind a pfSense firewall, the phone system's trunk is SIP, and the phones and phone system are on the same network, why would outbound calls ring once and then lose inbound audio? The call can complete, but I can only hear one side of the conversation.`
`23:26.17`	`K0HAX-1`	`I've tried static outbound nat, 1-to-1 nat, UPnP (which is making the pfSense open ports), opening every port from 1-65535 UDP/TCP and natting them to the phone system.`
`23:27.07`	`K0HAX-1`	`also, it works just fine using a d-link router`
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