IRC log for #asterisk on 20100317

`00:00.13`	`idespinner`	`and config on asterisk for DTMF is the same for all peers i'm guessing...?`
`00:01.56`	`fifer`	`I've tried all that. just trying to find anyone who has run into an issue like this.`
`00:02.04`	`fifer`	`Also trying to get a response from Aastra....`
`00:02.22`	`fifer`	`Just weird....`
`00:03.19`	`*** join/#asterisk jksM (jks@193.189.93.254)`
`00:03.21`	`*** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com)`
`00:03.28`	`fifer`	`It is only these new Aastra phones, my 480i works fine`
`00:03.46`	`fifer`	`I even tried using the same extension just to rule that out.`
`00:04.33`	`fifer`	`Well, have to pick it up tomorrow.`
`00:04.36`	`fifer`	`night all`
`00:05.28`	`*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)`
`00:09.49`	`manxpower`	`You could also just read the DAHDI README`
`00:10.00`	`*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)`
`00:16.35`	`*** join/#asterisk hipitihop (~denis@CPE-58-161-241-124.mzmy1.cha.bigpond.net.au)`
`00:17.39`	`*** part/#asterisk slinksh0t (~slinksh0t@wsip-68-227-95-60.pn.at.cox.net)`
`00:30.12`	`*** join/#asterisk Jhirley (~Jhirley@adsl-3-128-9.mia.bellsouth.net)`
`00:32.25`	`*** join/#asterisk DrDamnit (~DrDamnit@12.130.117.12)`
`00:37.40`	`*** join/#asterisk Jhirley_ (~Jhirley@adsl-161-132-136.mia.bellsouth.net)`
`00:41.36`	`QbY`	`if a channel is supposed to be FXO.. with DAHDI and * 1.6-- would it say 97 TE4/0/4/1 Clear (SWEC: MG2)`
`00:41.47`	`QbY`	`where in zaptel it showed fxo`
`00:43.38`	`p3nguin`	`Congratulations Your E-mail Address Have Been Pick`
`00:43.43`	`p3nguin`	`lame spammers`
`00:53.43`	`*** join/#asterisk nightrid3r (kvirc@41.92.18.208)`
`00:56.54`	`*** join/#asterisk cesar_CR (~cesar@201.201.41.242)`
`00:57.02`	`*** join/#asterisk niekie (~niek@CAcert/Assurer/niekie)`
`01:01.32`	`*** join/#asterisk JT (~j@unaffiliated/jt)`
`01:06.15`	`*** join/#asterisk niekie (~niek@CAcert/Assurer/niekie)`
`01:15.24`	`*** join/#asterisk michaely (~michaely@CPE-124-190-32-229.sdcz1.lon.bigpond.net.au)`
`01:15.47`	`michaely`	`Hello I have been trying to perform an automatic answer of the initial callback of a call to OriginateAction by setting the SIPADDHEADER variables when sending the Action. My variables in the Origination action are set like so:`
`01:15.47`	`michaely`	`Variable:__SIPADDHEADER01=Alert-Info: Ring Answer\|__SIPADDHEADER02=Call-Info: \;answer-after=0\|__SIPADDHEADER03=Alert-Info: \;info=alert-autoanswer`
`01:15.47`	`michaely`	`This works fine except that not only will the originator's extension auto answer but the destination extension will auto answer as well. I discovered that this was due to the use of the "__" perpended to the SIPADDHEADER key which caused the resulting channel to inherit the initiating channels variables during the call to Dial(). I removed the __s so that it looked like:`
`01:15.47`	`michaely`	`Variable:SIPADDHEADER01=Alert-Info: Ring Answer\|SIPADDHEADER02=Call-Info: \;answer-after=0\|SIPADDHEADER03=Alert-Info: \;info=alert-autoanswer`
`01:15.47`	`michaely`	`but now the auto answer fails completely on both ends and no SIP headers are sent to the device. Can someone tell me why the sip headers are not noticed when I do not have the "__" perpended to the SIPADDHEADER key. Note i have also tried with a single "_" and this caused bot extensions to auto answer.`
`01:22.45`	`*** join/#asterisk slinksh0t (~slinksh0t@wsip-68-227-95-60.pn.at.cox.net)`
`01:26.03`	`*** join/#asterisk chendy (~chatzilla@58.250.9.245)`
`01:31.18`	`*** join/#asterisk hipitihop (~denis@CPE-58-161-241-124.mzmy1.cha.bigpond.net.au)`
`01:33.02`	`sawgood`	`anyone good with digitmaps on Polycom phones (I have some which need a digitmap syntax for) 7 + 1 + area code + 7 digit number`
`01:45.02`	`*** join/#asterisk nightrid3r (~kvirc@41.214.142.172)`
`01:54.12`	`Naikrovek`	`oh was gonna answer him but he vamoosed`
`02:06.47`	`manxpower`	`71[2-9]XX[2-9]XXXXXX`
`02:08.18`	`Katty`	`hi`
`02:11.07`	`Katty`	`HI`
`02:12.31`	`Naikrovek`	`hola`
`02:12.41`	`Katty`	`:>`
`02:12.54`	`Katty`	`i was afraid i was going to have start saying sip trunk to get people to wake up`
`02:13.04`	`Naikrovek`	`heh`
`02:13.57`	`Naikrovek`	`i'm vpn'd into work, RDP'd into my laptop from a home VM and am monitoring a very high bandwidth application eat 100% of our bandwidth`
`02:14.08`	`Naikrovek`	`i'm honestly surprised i got in`
`02:14.40`	`Naikrovek`	`the protocol this thing uses ignores TCP's Slow Start mechanism`
`02:14.42`	`Naikrovek`	`so it rather unfairly hogs bandwidth`
`02:14.49`	`manxpower`	`chances are you have to escape the , or they will turn to \|`
`02:16.20`	`Katty`	`http://i.imgur.com/e0EcF.jpg`
`02:16.40`	`Katty`	`Naikrovek: what program is hogging bandwidth?`
`02:16.46`	`*** join/#asterisk xuser (~xuser@unaffiliated/xuser)`
`02:17.00`	`Naikrovek`	`Katty: Rational Clearcase`
`02:17.14`	`Naikrovek`	`i'm accepting a TON of data from Caterpillar`
`02:17.15`	`*** part/#asterisk michaely (~michaely@CPE-124-190-32-229.sdcz1.lon.bigpond.net.au)`
`02:17.21`	`Katty`	`ahhhh`
`02:17.27`	`Naikrovek`	`they have a rather large source code repository and I mirror it`
`02:17.30`	`Katty`	`and by ahhhh i mean i've never heard of it`
`02:17.30`	`Naikrovek`	`the parts we need, anyway`
`02:17.43`	`Naikrovek`	`several hundred gigabytes of source code`
`02:18.03`	`Katty`	`ah, i see`
`02:18.33`	`Naikrovek`	`rational clearcase is like CVS or Subversion or Perforce, or Git`
`02:18.46`	`Naikrovek`	`stores source code, keeps track of changes, etc.`
`02:19.53`	`Naikrovek`	`and the transfer has broken my phones and email is taking a hell of a long time to come and go`
`02:20.08`	`Katty`	`:<`
`02:21.08`	`Tech_Travis`	`Katty: Was that your party?`
`02:22.09`	`Katty`	`you kidding me?`
`02:22.12`	`Katty`	`i wouldn't drink that crap`
`02:22.22`	`Katty`	`i just thoguht it was cute`
`02:23.06`	`Tech_Travis`	`Katty: I'm not sure about cute, but depending on the number of people it could be impressive.`
`02:23.17`	`Naikrovek`	`it's likely that our phones are going to be out the rest of the week`
`02:23.31`	`Naikrovek`	`that looks like an Australian wedding.`
`02:23.41`	`*** join/#asterisk Ta^3 (~tacvbo@189.146.181.92)`
`02:23.58`	`Naikrovek`	`wow there was like a 5 minute lag from the time i typed that to when it showed up`
`02:23.59`	`Katty`	`could be`
`02:24.31`	`Naikrovek`	`go go gadget T1`
`02:24.38`	`Naikrovek`	`we desperately need more bandwidth`
`02:24.46`	`Naikrovek`	`maybe this will convince mgmt. that it's time`
`02:25.42`	`Katty`	`Naikrovek: you ever heard of a water bear?`
`02:26.00`	`Naikrovek`	`have not. /me heads to google`
`02:26.25`	`Naikrovek`	`that's that bug that can survive in a vacuum huh`
`02:26.28`	`Katty`	`http://www.youtube.com/watch?v=zlGZalFIVrE`
`02:26.32`	`Naikrovek`	`goes all dormanty`
`02:27.30`	`Katty`	`suspended animation`
`02:27.36`	`Katty`	`imagine what we could learn from this critter?`
`02:27.37`	`Naikrovek`	`yeah`
`02:27.42`	`Naikrovek`	`heh`
`02:27.55`	`Naikrovek`	`can it talk? i don't think it would make a good schoolteacher because it's tiny`
`02:28.08`	`Naikrovek`	`probably poops all over the desk`
`02:28.59`	`Katty`	`well the human eye can't see it`
`02:29.13`	`Katty`	`so i don't think pooping will have much of an effect`
`02:30.01`	`*** join/#asterisk Raszh (~Spoon@mail.nymble.net)`
`02:30.26`	`Raszh`	`Anyone here tried integrating * with DiaStar?`
`02:30.35`	`Katty`	`never heard of it`
`02:30.42`	`QbY`	`is it possible to specify the username and secret for a sip call in the dial command? or just before.. i have multple users coming from the same box going to the same proxy, but i need to authenticate them separately`
`02:31.14`	`[TK]D-Fender`	`QbY: I gave you a complete sample hours ago`
`02:31.23`	`QbY`	`doesn't work`
`02:31.29`	`[TK]D-Fender`	`QbY: Yes, it does`
`02:31.59`	`QbY`	`Dial(SIP/username:password@fqdn.domain.com/18885551212) fails`
`02:32.39`	`QbY`	`(06:06:20 PM) [TK]D-Fender: QbY: Dial(SIP/user:pass@host/extentodial)`
`02:32.42`	`[TK]D-Fender`	`QbY: And where do we get to see the attempt to find our why it failed?`
`02:32.44`	`p3nguin`	`Create a peer for it.`
`02:33.21`	`QbY`	`i seen it try to authetnicate with "asterisk"`
`02:33.28`	`Raszh`	`Katty: projectdiastar.org`
`02:34.44`	`manxpower`	`QbY: sounds like you have a [fqdn.domain.com] peer to me.`
`02:34.56`	`manxpower`	`QbY: the fact that it does not work for you does not mean it's wrong.`
`02:35.34`	`manxpower`	`You will be providing pastebin.ca link to the CLI output of a failed call?`
`02:35.41`	`ChannelZ`	`And a cookie?`
`02:35.46`	`manxpower`	`[TK]D-Fender: is QbY worth the trouble?`
`02:35.46`	`Naikrovek`	`show the error`
`02:35.48`	`QbY`	`sure`
`02:36.59`	`*** join/#asterisk x86 (~x86@p3m/member/x86)`
`02:38.53`	`QbY`	`http://www.pastebin.ca/1843169`
`02:39.10`	`*** join/#asterisk joako (~st0n3r@opensuse/member/joak0)`
`02:39.14`	`QbY`	`i've tried it in both ways`
`02:39.17`	`*** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com)`
`02:39.28`	`joako`	`How do I debug asterisk HTTP server?`
`02:39.44`	`Katty`	`intereseting`
`02:40.05`	`Katty`	`Pepsi says it will be pulling all 'sugary drinks' from schools worldwide by 2012`
`02:40.22`	`Katty`	`which is a real shame.`
`02:40.39`	`*** join/#asterisk NormB (~NormB@mercury.goes.com)`
`02:40.53`	`Katty`	`those artifical sweetners are horrible on the body, not to mention they cause hunger`
`02:41.44`	`manxpower`	`QbY: Too bad you can't do this the right way.`
`02:41.48`	`[TK]D-Fender`	`qbtWTF is the number doing BEFORE the host?`
`02:41.49`	`Naikrovek`	`aren't they dropping those in favor of things like juice`
`02:41.53`	`[TK]D-Fender`	`QbY: WTF is the number doing BEFORE the host?`
`02:42.01`	`[TK]D-Fender`	`[22:31]<QbY>Dial(SIP/username:password@fqdn.domain.com/18885551212) fails`
`02:42.13`	`QbY`	`[TK]D-Fender: that way it didn't even look up the host`
`02:42.16`	`[TK]D-Fender`	`QbY: you put the number BEFORE the "@"`
`02:42.17`	`manxpower`	`QbY: Calling YOURSELF is NOT a valid test!`
`02:42.48`	`[TK]D-Fender`	`QbY: You also didn't include SIP DEBUG`
`02:42.53`	`Katty`	`Naikrovek: not according to this article. they're putting in diet soda and sports drinks`
`02:43.09`	`manxpower`	`[TK]D-Fender: Too bad he didn't use the correct format.`
`02:43.28`	`Katty`	`Naikrovek: http://finance.yahoo.com/news/PepsiCo-cuts-sugary-drinks-apf-2624935934.html?x=0 <- for your reading pleasure.`
`02:43.43`	`ChannelZ`	`Hmm. My voicemail says it's playing "beep.ulaw" but it actually isn't.`
`02:43.46`	`manxpower`	`QbY: show [TK]D-Fender what he wants to see or he'll give up on you and then you'`
`02:43.55`	`ChannelZ`	`Either that or somehow my beep got replaced with silence`
`02:44.21`	`Katty`	`Naikrovek: ah, 'primary school' so 1st grade through 5th? we never had a vending machine`
`02:45.01`	`Katty`	`Naikrovek: that's just what we need, 5-10 year olds getting into the habit of raiding a vending machine every day`
`02:45.03`	`QbY`	`let me get calls flowing--then i will get all the debug--we've had a 9 hour outage because of dahdi and crapola`
`02:45.09`	`Katty`	`anywho.`
`02:45.12`	`Katty`	`goes to bed.`
`02:45.18`	`ChannelZ`	`nighty`
`02:47.26`	`manxpower`	`QbY: a better thing would be to pastebin the output if the format [TK]D-Fender originally recommended. Anyway, come back when you have time to concentrate on this issue.`
`02:47.56`	`QbY`	`i'm going to create a extension just in that format.. in just a second..`
`02:55.34`	`*** join/#asterisk hobbes615 (~hobbes@c-98-233-5-121.hsd1.dc.comcast.net)`
`03:02.46`	`*** join/#asterisk nighty^ (~nighty@210.188.173.245)`
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`03:10.35`	`*** join/#asterisk hobbes615 (~hobbes@unaffiliated/hobbes615)`
`03:16.15`	`QbY`	`[TK]D-Fender: http://www.pastebin.ca/1843201`
`03:19.13`	`p3nguin`	`If you didn't insist on using .ca instead of .com, I would look at your paste.`
`03:19.29`	`[TK]D-Fender`	`QbY: Try : SIP/16788521234@8645551212:o3hDxn6g5ix1en2r@sip.domain.net`
`03:19.32`	`p3nguin`	`With the way it is now, it'll either take 10 minutes to load the page or it might never load at all.`
`03:19.39`	`[TK]D-Fender`	`p3nguin: thats a real sad excuse...`
`03:19.47`	`QbY`	`http://pastebin.com/WTfZP6pR`
`03:19.54`	`QbY`	`i'm tellin ya.. everyone hating.`
`03:19.54`	`[TK]D-Fender`	`p3nguin: and .ca has been faster than .com for a long time`
`03:20.07`	`p3nguin`	`Except when it hasn't been.`
`03:21.19`	`[TK]D-Fender`	`p3nguin: Hardly ever... Your bile duct is misfiring again ;)`
`03:21.38`	`p3nguin`	`Is asterisk trying to resolve sip.domain.net/16788521234 ?`
`03:22.13`	`manxpower`	`[TK]D-Fender: I think he has an MTU issue.`
`03:22.26`	`QbY`	`[TK]D-Fender: http://pastebin.com/fYPdj7FS`
`03:23.01`	`QbY`	`i think i'm going to have to create peers--but i was hoping to build something a little faster and more flexible..`
`03:23.39`	`manxpower`	`QbY: you still have to create peers on the remote server`
`03:23.57`	`QbY`	`the remote is a proxy`
`03:24.15`	`QbY`	`it wants username/passwrod`
`03:28.27`	`manxpower`	`QbY: eventually you'll learn to stop fighting Asterisk`
`03:29.08`	`[TK]D-Fender`	`QbY: I have seen that host issue with the exten tacked on the end... I believe that qualifies as a bug...`
`03:29.41`	`[TK]D-Fender`	`QbY: What ver are you on?`
`03:30.03`	`QbY`	`Asterisk 1.6.2.6 built by root @ REMPBX01 on a i686 running Linux on 2010-03-16 21:23:01 UTC`
`03:30.10`	`[TK]D-Fender`	`manxpower: http://www.pastebin.ca/1843201 <--- * parsing flaw`
`03:30.26`	`[TK]D-Fender`	`QbY: Ok... they still haven't fixed this part yet.... gah`
`03:30.36`	`QbY`	`ok, no worries`
`03:30.48`	`[TK]D-Fender`	`QbY: Submit a bug on mantis for it`
`03:31.16`	`QbY`	`will do--i guess i could use realtime and just create a view of the same table`
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`03:36.18`	`dlynes`	`As I understand it, one should always use DeadAGI, rather than AGI?`
`03:36.27`	`dlynes`	`i.e. in case the caller hangs up on the call?`
`03:38.38`	`p3nguin`	`I thought DeadAGI was for use in hangup extensions ('h') after the channel has exited.`
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`06:05.40`	`pcrack`	`whats the best way to fine tune rxgain and txgain?`
`06:06.43`	`pcrack`	`i have the TDM800P all are FXO`
`06:06.52`	`pcrack`	`FXO`
`06:28.59`	`*** join/#asterisk mchou (~quassel@unaffiliated/mchou)`
`06:32.00`	`*** join/#asterisk the_weard (~mitch@196.212.100.148)`
`06:32.54`	`mchou`	`I just had a buddy buy a pap2-nt. Problem is when I talk to him (& he is using his ATA) I hear echo; if he uses PSTN I dont hear any echo. Any particular parameters on his ATA that should be tweaked?`
`06:33.00`	`the_weard`	`hey i have a problem with my call recordings can someone plz help me`
`06:33.21`	`the_weard`	`all my calls are recorded and not showing up in ari`
`06:33.44`	`the_weard`	`using asteriskNOW 86_64`
`06:34.28`	`the_weard`	`everything is working from sip trunk to making koffe just the damm call recordings`
`06:34.42`	`the_weard`	`plz aney help will be apreciated`
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`06:56.34`	`ChannelZ`	`ask #asterisknow`
`06:56.44`	`ChannelZ`	`or #freepbx`
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`07:19.08`	`pisg`	`hi, i set up CDR for asterisk, all ok, but asterisk send to databases strange entry, only time, channel and destynation : "s", what is this ?`
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`07:49.37`	`ChannelZ`	`pisg: 's' is the sort of 'default' extension when a call enters the dialplan with no specific extension`
`07:50.10`	`ChannelZ`	`Mostly used for DAHDI channels for incoming calls, but can happen on SIP and such too when no extension is defined`
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`07:56.25`	`pisg`	`ChannelZ: http://pastebin.com/ykJmH3tH this is my extensions.conf so what i must change ?`
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`08:03.38`	`MAbbas`	`Hi all .. Asterisk 1.6.0.X and 1.6.1.X have which verions of "Call Manager" 1.1 or 1.0 ?`
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`08:12.56`	`rvleij`	`Goodmorning all!`
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`08:15.05`	`*** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com)`
`08:16.06`	`EmleyMoor`	`has a SIP Trunk between his own asterisk and pbxes.com, and is wondering how to send a call to pbxes.com for a particular extension using thes arrangement`
`08:17.36`	`*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)`
`08:22.42`	`EmleyMoor`	`(I use pbxes.com for two reasons - less problems with NAT and STUN, and transcoding to ilbc)`
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`09:03.05`	`ThoMe`	`hiho`
`09:04.15`	`ThoMe`	`i have asterisk 1.6.2.6 on my (debian) mashine`
`09:04.26`	`ThoMe`	`and the 23-phonius:~# ps aux \|grep asterisk`
`09:04.26`	`ThoMe`	`root 2760 99.9 1.3 28900 14304 ? Rl Mar16 684:42 /usr/sbin/asterisk -c`
`09:04.35`	`ThoMe`	`but he need ~100% from my cpu`
`09:04.37`	`ThoMe`	`any ideas?`
`09:05.03`	`*** join/#asterisk c0rnoTa (~c0rnoTa@178.176.217.179)`
`09:05.26`	`EmleyMoor`	`I think I understand the principle of what to do at the pbxes side but not how it works at the asterisk side`
`09:06.26`	`ThoMe`	`what is at the moment the stable version from asterisk?`
`09:09.18`	`*** join/#asterisk tamiel (~tamiel@213.30.183.226)`
`09:10.13`	`EmleyMoor`	`(surely I don't need a trunk per user)`
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`09:43.25`	`bragon`	`Hi`
`09:43.44`	`bragon`	`i want to build app_meetme.so for my asterisk-1.2.37`
`09:43.47`	`bragon`	`how can i do that ?`
`09:44.12`	`bragon`	`it's possible to build only the module ?`
`09:47.12`	`TommyBotten`	`The module not depending on zaptel/dahdi, right?`
`09:48.15`	`bragon`	`i can't build zaptel because i can't reboot and it' need a kernel module to work.`
`09:48.26`	`*** join/#asterisk Z_God (~julius@wlan225137.mobiel.utwente.nl)`
`09:48.41`	`bragon`	`i just want to have a app_meetme.so working :)`
`09:49.30`	`TommyBotten`	`Well, the timing part that has replaced dahdi/zaptel for app_meetme won't be usable in 1.2.37 ... so no, is unfortunately the answer`
`09:50.24`	`bragon`	`TommyBotten: so i can't make meetme confroom with my 1.2.37 ?`
`09:52.25`	`bragon`	`http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe`
`09:53.58`	`TommyBotten`	`Sure, but you need the zaptel timing module compiled and loading`
`09:54.36`	`TommyBotten`	`as in kernel module. If you don't have any cards with a timing source, you should use the dummy kernel module`
`09:57.32`	`bragon`	`TommyBotten: my kernel is build without module support ...`
`10:00.09`	`*** join/#asterisk krion (~seb@unaffiliated/krion)`
`10:01.17`	`TommyBotten`	`ouch.`
`10:01.23`	`TommyBotten`	`is this a production system?`
`10:01.58`	`TommyBotten`	`Because asterisk 1.6.2.x supports conferences with only an internal timing source. which means no dahdi, and no zaptel`
`10:02.09`	`TommyBotten`	`but I wouldn't recommend it in production without proper testing`
`10:03.47`	`bragon`	`TommyBotten: it's a production box.`
`10:03.59`	`bragon`	`I can't reboot it for Module kernel support.`
`10:04.19`	`bragon`	`so i have it in my ass :x`
`10:07.19`	`TommyBotten`	`is asterisk in production?`
`10:07.58`	`TommyBotten`	`or are there other services?`
`10:08.24`	`TommyBotten`	`and if you really need this feature without rebooting, you could always setup another system behind asterisk, doing conferences only`
`10:08.31`	`bragon`	`other services`
`10:08.34`	`TommyBotten`	`aha`
`10:08.43`	`TommyBotten`	`in that case, you can install asterisk 1.6.2.6`
`10:08.53`	`TommyBotten`	`then you won't need the module support`
`10:08.57`	`bragon`	`the asterisk isn't in production, but other service in this box there are another service.`
`10:09.08`	`TommyBotten`	`but be aware, much has changed`
`10:10.09`	`*** join/#asterisk pif (~ldm@zenon.apartia.fr)`
`10:11.29`	`bragon`	`TommyBotten: if i test it on a asterisk 1.2.37 it's because i must deploy meetme after on a production asterisk :)`
`10:11.47`	`bragon`	`i don't have the time to migrate my dialplan :x`
`10:13.35`	`*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-gjiwfwvkbaqlyzyp)`
`10:13.57`	`TommyBotten`	`I don't understand what you are trying to achieve`
`10:16.23`	`dlynes`	`bragon, unless your box is built with module support, you're trying to achieve the unachievable without a reboot of the box`
`10:17.57`	`dlynes`	`TommyBotten, also the internal timing source...is it actually working for dahdi trunking and meetme now? In 1.6.1 it was working for everything but`
`10:18.05`	`*** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr)`
`10:19.03`	`dlynes`	`bragon, that being said, what you can do is set up a new box with app_meetme.so on it, and redirect any conference calls to that box`
`10:19.34`	`bragon`	`alucard asterisk-1.2.37 # lsmod`
`10:19.34`	`bragon`	`Opening /proc/modules: No such file or directory`
`10:19.46`	`bragon`	`srv74 asterisk # lsmod`
`10:19.46`	`bragon`	`Opening /proc/modules: No such file or directory`
`10:19.56`	`bragon`	`All my production box are build without module kernel`
`10:20.00`	`bragon`	`All off it ...`
`10:21.58`	`dlynes`	`bragon, get a new box that's either build with a kernel that can use modules, or get a new box that's got the rtc compiled in, and try using 1.6.2.x, and hope the internal timing is working for app_meetme now`
`10:22.59`	`TommyBotten`	`dlynes: app_conference works fine without any kernel modules`
`10:23.01`	`dlynes`	`bragon, none of the digium drivers are available as being built into the kernel`
`10:23.27`	`dlynes`	`TommyBotten, It's no longer called app_meetme in 1.6.2?`
`10:23.46`	`TommyBotten`	`dlynes: App_meetme is the good old one. app_conference is the new one, using internal timing`
`10:23.59`	`dlynes`	`TommyBotten, ah`
`10:25.22`	`dlynes`	`TommyBotten, is dahdi trunking working properly using internal timing in 1.6.2 now, too?`
`10:26.51`	`dlynes`	`Anyone heard of a possible timeline for 1.8 yet?`
`10:29.23`	`TommyBotten`	`dlynes: Don't really know about dahdi trunking. Haven't tested that in 1.6.x`
`10:29.35`	`TommyBotten`	`and 1.8 is expected in june or so`
`10:29.45`	`TommyBotten`	`see the roadmap on digiums bugtracker`
`10:30.31`	`TommyBotten`	`>`
`10:30.32`	`TommyBotten`	`Release date: TBD (estimated Q3 2010)`
`10:30.40`	`TommyBotten`	`https://issues.asterisk.org/roadmap_page.php`
`10:32.25`	`*** join/#asterisk pentanol (~pentanol@77-35-6-037.pppoe.primorye.net.ru)`
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`10:34.48`	`dlynes`	`TommyBotten, ah...thanks`
`10:35.38`	`dlynes`	`TommyBotten, third quarter would be more like September, though (quarter expectations usually fall at the end of the quarter, not the beginning)`
`10:35.38`	`TommyBotten`	`:)`
`10:36.15`	`TommyBotten`	`Thats right... the "june" answer was by memory... and my memory does not have ECC ;)`
`10:36.34`	*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
`10:37.16`	`dlynes`	`TommyBotten, nah...I mean in corporate speak, it usually means it'll be released in the fall`
`10:37.48`	`dlynes`	`TommyBotten, i wasn't suggesting that it won't be 'third quarter' :)`
`10:37.49`	`petern_`	`autumn`
`10:37.51`	`petern_`	`:s`
`10:38.04`	`dlynes`	`fall...autumn..same difference :)`
`10:38.11`	`TommyBotten`	`Being a non native speaker... aren't those the same?`
`10:38.24`	`*** join/#asterisk gego (~quassel@b238085.customer.hansenet.de)`
`10:38.29`	`*** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br)`
`10:38.43`	`dlynes`	`TommyBotten, yes...he's just being facetious :)`
`10:39.02`	`petern_`	`i like that word, all the vowels in order...`
`10:39.10`	`TommyBotten`	`stop using those ten cent words ;)`
`10:39.36`	`dlynes`	`TommyBotten, In Europe (and Canada), it's known as Autumn`
`10:39.46`	`dlynes`	`TommyBotten, In the US (and Canada), it's known as fall`
`10:41.01`	`dlynes`	`btw`
`10:41.16`	`TommyBotten`	`ah... so canada is kind of promiscous?`
`10:41.41`	`dlynes`	`I've heard that if you're expecting the call to get hung up, you should use DeadAGI(), rather than AGI(), because apparently AGI() doesn't play nice with channels that have hung up?`
`10:41.48`	`petern_`	`hmm, iax2 encryption... something to test`
`10:41.48`	`dlynes`	`TommyBotten, Yes, very`
`10:42.08`	`dlynes`	`TommyBotten, We love both our sex and our beer`
`10:42.08`	`*** join/#asterisk fofware (~chatzilla@186.125.41.133)`
`10:42.18`	`dlynes`	`TommyBotten, not necessarily in that order`
`10:42.36`	`dlynes`	`TommyBotten, sorta like a what came first...chicken or the egg syndrome`
`10:43.12`	`TommyBotten`	`haha :D`
`10:43.13`	`*** join/#asterisk bmg505 (~leon@196-209-74-160-rndf-esr-3.dynamic.isadsl.co.za)`
`10:47.11`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`10:50.39`	`petern_`	`dlynes, depends on your AGI? DeadAGI will continue running even if the call is hung up`
`10:51.20`	`dlynes`	`petern_, ok, so I want DeadAGI, then...thanks...I don't want to leave asterisk in an undefined state`
`10:51.55`	`TommyBotten`	`I have an issue with my digium FXS card.. using dahdi_monitor, i always hit max audio on my TX channel.`
`10:51.55`	`dlynes`	`Does anyone know if the application HasVoicemail(), or whatever it was still exists in 1.6.1?`
`10:52.14`	`dlynes`	`TommyBotten, you can adjust your input/output gains`
`10:52.49`	`TommyBotten`	`dlynes: Yep... but even with the gain set yo -63.0 I hit max audio`
`10:52.53`	`TommyBotten`	`which is insane`
`10:53.14`	`TommyBotten`	`and regarindg the hasvoicemail... is it the func_vmcount.so you are thinking of`
`10:53.17`	`TommyBotten`	`?`
`10:55.09`	`dlynes`	`TommyBotten, exactly what i wanted...thanks`
`10:55.58`	`dlynes`	`TommyBotten, I'm guessing people are complaining the audio is too loud?`
`10:56.37`	`dlynes`	`TommyBotten, or does it just appear to be inaccuracy in dahdi_monitor?`
`10:57.51`	`TommyBotten`	`well, I'm trying to send faxes over the analog line`
`10:58.11`	`TommyBotten`	`But that goes very wrong, when the audio peaks, thus losing data`
`10:58.16`	`dlynes`	`TommyBotten, ah...ok...so the VOLUME() function won't help you, then`
`10:58.20`	`TommyBotten`	`so even the handshake won't work properly`
`10:58.23`	`TommyBotten`	`nope :(`
`10:59.15`	`dlynes`	`TommyBotten, so you're trying to bridge an fxo and an fxs port, then?`
`11:00.11`	`TommyBotten`	`well, my fax is a regular analog fax (fxo), and my digium card is fxs - the rest of the transport is done via a SIP trunk`
`11:02.30`	`dlynes`	`So, again...are you bridging the fxo and fxs ports to send faxes? Or is SIP somehow involved?`
`11:03.22`	`dlynes`	`TommyBotten, Also, did you buy this card from Digium, or from some third party supplier?`
`11:03.58`	`*** join/#asterisk Malky (~oliv@lon92-1-82-67-212-172.fbx.proxad.net)`
`11:04.05`	`dlynes`	`Btw...does anyone know if you can specify multiple keys to stop playback using ControlPlayback()?`
`11:04.54`	`Faustov`	`Hi, when running asterisk from an internet gateway, is there a good way to prioritize VoIP traffic? What's recommended?`
`11:05.55`	`dlynes`	`Faustov, best would be if you control the internet gateway _and_ your customer's internet...then you can provide end-to-end QoS`
`11:06.31`	`*** join/#asterisk Rajmohan (~raj@122.164.14.59)`
`11:06.57`	`dlynes`	`Faustov, you can prioritize using vlans, TOS flags, ...`
`11:08.18`	`pisg`	`anyone have good how-to create callback ? i call to my asterisk, asterisk busy and asterisk call to me and give dialtone`
`11:08.52`	`dlynes`	`pisg, call files`
`11:09.13`	`*** join/#asterisk adnc (~numer@unaffiliated/adnc)`
`11:09.27`	`pisg`	`i search FAQ / how to`
`11:09.29`	`dlynes`	`pisg, you can have an agi answer your initial call, which in turn creates the call file and hangs up on you`
`11:10.11`	`Faustov`	`dlynes: I have control, just asking for the best method (smallest overhead)`
`11:10.37`	`*** join/#asterisk razu (~razu@razu.data.ee)`
`11:10.56`	`dlynes`	`pisg, you wouldn't be able to do that using fastagi, unless the fastagi socket server is running on the same machine as asterisk`
`11:11.16`	`dlynes`	`Faustov, More than likely checking the tos flags`
`11:11.33`	`dlynes`	`Faustov, assuming your CPE is capable of reading TOS flags`
`11:13.07`	`dlynes`	`Faustov, actually...on the CPE side (if they're not using asterisk, or even if they are), you can control the qos based on what switch/router port their SIP device is plugged into`
`11:13.28`	`dlynes`	`Faustov, just make sure you're not relying on linksys garbage to handle this`
`11:13.45`	`dlynes`	`Faustov, the rv082/rv042 chokes on this kinda thing`
`11:13.49`	`Faustov`	`dlynes: I got everything based on iptables`
`11:14.19`	`Faustov`	`lan --- switch --- gateway with asterisk on it and iptables --- wan`
`11:14.21`	`*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-hyyvsmlzejsooesu)`
`11:14.36`	`Faustov`	`looks like I can disregard lan and switch`
`11:15.18`	`Faustov`	`I'd just like to prioritize WAN traffic, giving VoIP top priority, and lower priority to the rest of the world`
`11:16.16`	`manxpower`	`Faustov: You're not going to get real QoS on a consumer WAN service.`
`11:17.45`	`manxpower`	`Faustov: Also you can only QoS transmitted data, not received data.`
`11:18.43`	`Faustov`	`manxpower: I'm aware of that`
`11:19.40`	`Faustov`	`However by just having the VoIP traffic processed and responded to without delay would be good`
`11:19.54`	`TommyBotten`	`dlynes: From digium directly`
`11:20.22`	`Faustov`	`manxpower: how would you approach the outbound data QoS? by ToS flags?`
`11:20.39`	`manxpower`	`Faustov: you read doc/ip-qos.txt ?`
`11:20.57`	`manxpower`	`I think it's ip-tos.txt, actually`
`11:21.02`	`*** join/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang)`
`11:21.59`	`Faustov`	`once again, the question is for recommended methods, priority is for minimum overhead (and not how to do it, I can google that part)`
`11:22.53`	`*** join/#asterisk Akiraaa (~Akira@92.81.196.238)`
`11:23.56`	`dlynes`	`manxpower, Faustov said he had control of both ends of communication`
`11:24.12`	`manxpower`	`Faustov: there is no answer to your question.`
`11:24.30`	`manxpower`	`What you do with QoS depend on your network hardware.`
`11:24.50`	`*** join/#asterisk dennisG (~root@84.30.136.208)`
`11:24.52`	`manxpower`	`Usually you tag the data in Asterisk, then have your network do QoS based on the tagged packets from Asterisk`
`11:24.55`	`*** join/#asterisk ccesario_ (~ccesario@189.88.1.100)`
`11:25.18`	`manxpower`	`exactly what the tag needs to be set to depends on how you have your network set up.`
`11:25.35`	`dlynes`	`pisg, http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out`
`11:25.56`	`dlynes`	`TommyBotten, I would send it back in for warranty replacement, then`
`11:26.13`	`*** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il)`
`11:26.19`	`dlynes`	`TommyBotten, but digium will have to log into your box first to make sure they're satisfied that's what the problem is`
`11:26.33`	`manxpower`	`We have our network to prioritize packets tagged with DSCP EF.`
`11:27.46`	`*** join/#asterisk pentanol (~pentanol@77-35-6-037.pppoe.primorye.net.ru)`
`11:27.55`	`manxpower`	`Faustov: which routers are being used on your WAN?`
`11:27.59`	`TommyBotten`	`dlynes: Hmm... that can't really happen.`
`11:28.21`	`TommyBotten`	`dlynes: Not that the money in this case really is the issue... i just need it to work`
`11:29.03`	`manxpower`	`"money is not an issue, but we are too cheap to but a spare card"?`
`11:29.13`	`*** join/#asterisk gardo (~gardo@125.212.88.255)`
`11:31.28`	`manxpower`	`TommyBotten: You just need FaxOverVoiceOverIPOverInternet to "just work" What are you? Crazy?`
`11:31.36`	`dlynes`	`TommyBotten, You can always get digium to say the card's bad, that they'll replace it, and ask them to send you an advance replacement (you get a replacement card before you ship your current one back)`
`11:32.17`	`dlynes`	`manxpower, I don't think he's doing foip....he's got fxo and fxs, so I believe he's doing bridged analog channel faxing`
`11:32.24`	`*** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)`
`11:32.31`	`dlynes`	`manxpower, i've asked him a few times though, and he's never answered my question`
`11:32.48`	`dlynes`	`manxpower, faustov's using iptables for his routers`
`11:32.53`	`manxpower`	`(06:00:11 AM) TommyBotten: well, my fax is a regular analog fax (fxo), and my digium card is fxs - the rest of the transport is done via a SIP trunk`
`11:33.16`	`manxpower`	`That looks to me he is sending Fax over SIP`
`11:33.23`	`TommyBotten`	`yeah`
`11:33.23`	`TommyBotten`	`I am`
`11:33.29`	`TommyBotten`	`it worked earlier`
`11:33.34`	`manxpower`	`dlynes: ah. No point in helping someone that doesn't want help.`
`11:33.41`	`manxpower`	`~foip`
`11:33.42`	`infobot`	`methinks foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject`
`11:33.48`	`TommyBotten`	`hehe..calm down, manxpower`
`11:34.14`	`*** join/#asterisk aandrade (~aandrade@187.58.239.123)`
`11:34.17`	`TommyBotten`	`I'm having trouble with my connection here. And the IRC session is a screen`
`11:34.36`	`manxpower`	`TommyBotten: People have been trying to get reliable Fax over Voice Over IP Over Internet for 10 years, very few are successful.`
`11:34.53`	`TommyBotten`	`Is it likely to be the card in this case?... I mean, usually it's me doing stupid things, not the hardware failing`
`11:35.14`	`manxpower`	`No, it's likely a failure on your part to research the issue first.`
`11:35.30`	`TommyBotten`	`Well, the thing is, it worked earlier on.`
`11:35.41`	`Akiraaa`	`How much power does an IP phone use? 5w?`
`11:35.52`	`TommyBotten`	`And now someone has been messing with my setup, over the last six months... and now it doesn't work anymore.`
`11:35.56`	`manxpower`	`TommyBotten: Yes, and chances are it will work int he future. That the thing about Faxoverip, it's not reliable.`
`11:36.08`	`manxpower`	`Akiraaa: ask your phone?`
`11:36.31`	`TommyBotten`	`manxpower: I see... is using T.38 considered a viable option?`
`11:36.34`	`manxpower`	`TommyBotten: In any case, I wish you the BEST of luck.`
`11:37.13`	`manxpower`	`your most viable option is to stop trying to route faxes thru VoIP.`
`11:37.43`	`dlynes`	`Akiraa, it depends on the phone's power consumption...every phone is different...the manual should tell you what its consumption is`
`11:38.21`	`TommyBotten`	`Ok... thanks for the advice.. but I'll probably have to spend some time on this`
`11:38.31`	`Akiraaa`	`TommyBotten: I've seen several FoIP implementations in live environments, with terrible performance usually`
`11:38.36`	`dlynes`	`TommyBotten, if t.38 was actually the panacea the industry says it is, it would be a viable option...the problem is major incompatibility between the t.38 implementations`
`11:38.46`	`Akiraaa`	`clients eventually gave up and designed a separate PSTN line for fax only`
`11:39.04`	`manxpower`	`Spend all the time you want. It's kind of funny watching people bang their head against a wall trying to solve a problem that's already been solved.`
`11:39.34`	`manxpower`	`Akiraaa: That is very typical.`
`11:40.06`	`Akiraaa`	`manxpower: can't figure out what size UPS to use for around 20 IP phones (with LCD diplay)`
`11:40.25`	`Akiraaa`	`power use is not mentioned anywhere in the electrical features leaflet`
`11:40.42`	`manxpower`	`Akiraaa: which phones?`
`11:41.03`	`Akiraaa`	`manxpower: http://www.voip-info.org/wiki/view/PH802 -- these`
`11:42.09`	`mazpe`	`is there a recommended SOHO router that will behave good with VoIP traffic? QoS,VLAN, DSL`
`11:42.21`	`manxpower`	`Akiraaa: You can't complete your task without the power requirements of your phones`
`11:42.58`	`manxpower`	`mazpe: yes and know. yes there are routers that do that, no it doesn't really matter if your carrier does not do QoS`
`11:43.05`	`manxpower`	`..yer... yes and no.`
`11:43.07`	`adnc`	`i'm trying to get some status informations from my peers. and i connect via php but when i ask for status i get "Unmonitored" back. is there a switch for a peer that needs to be enabled?`
`11:43.21`	`manxpower`	`adnc: yes! qualify=`
`11:43.28`	`adnc`	`ahh`
`11:43.32`	`Akiraaa`	`manxpower: hm... the power adaptor says "5V, 1A" -- so 5W max power use?`
`11:43.33`	`adnc`	`manxpower, thankyou`
`11:44.00`	`*** part/#asterisk manxpower (~ewieling@216.186.151.147)`
`11:44.11`	`mazpe`	`manxpower: so i guess its better to focus on the switch support vlan and QoS`
`11:51.37`	`*** join/#asterisk ManxPower-work (~manxpower@216.186.151.147)`
`11:52.02`	`TommyBotten`	`When a user tries to register, passing 'Registration from '<sip:test@x.x.x.x;user=phone>' .. what argument is the latter user= ? .. and how can I match that in my config?`
`11:52.32`	`TommyBotten`	`as of now, it fails when I have my both the user 'test' and its secret set correctly.`
`11:52.39`	`ManxPower-work`	`the user=phone means nothing in this case.`
`11:52.45`	`ManxPower-work`	`it's not USERNAME it's USERTYPE`
`11:52.55`	`ManxPower-work`	`in this case the type of user is a phone.`
`11:53.40`	`ManxPower-work`	`TommyBotten, exactly how do you "have other user test and the setret set correctly". username=test or [test] or what?`
`11:54.03`	`ManxPower-work`	`[thisinhereistheincomingusername]`
`11:54.59`	`TommyBotten`	`well, tried both actually... and at least the latter works with my other setups`
`11:55.12`	`ManxPower-work`	`pastebin your sip.conf removing only the passwords`
`11:55.35`	`TommyBotten`	`Sure.. let me try the usertype first`
`11:55.47`	`ManxPower-work`	`huh? usertype?????`
`11:55.52`	`ManxPower-work`	`you don't change the usertype?`
`11:56.05`	`adnc`	`ManxPower-work, qualify=yes didn't help.`
`11:56.15`	`ManxPower-work`	`adnc, then you did it wrong`
`11:56.55`	`ManxPower-work`	`if it shows "unmonitored" in "sip show peers" then you don't have the qualify option set correctly for that peer.`
`11:57.04`	`TommyBotten`	`ManxPower-work: Not quite following you. I have just left the default values there`
`11:57.20`	`ManxPower-work`	`TommyBotten, less talk, more pastebin action`
`11:57.59`	`adnc`	`do you know which permissions in the manager user for read and write needs to be enabled for this=`
`11:58.15`	`ManxPower-work`	`adnc, does or does not sip show peers show that peer as monitored?`
`11:58.45`	`adnc`	`ManxPower-work, indeed it does show it for that peer`
`11:59.01`	`ManxPower-work`	`adnc, good. Then your qualify=yes worked.`
`11:59.43`	`ManxPower-work`	`adnc, why do you need to write the status?`
`11:59.56`	`TommyBotten`	`ManxPower-work: http://pastebin.ca/1843470`
`12:00.23`	`adnc`	`ManxPower-work, i do not need to write the status, only read it. http://www.venturevoip.com/peer-status.phps this is what i wanted to use`
`12:00.39`	`ManxPower-work`	`TommyBotten, I can't help you with users.conf stuff.`
`12:01.06`	`TommyBotten`	`ManxPower-work: That's fine.. they all work fine`
`12:01.11`	`TommyBotten`	`ouch... low on battery :(`
`12:01.16`	`ManxPower-work`	`TommyBotten, then you don't need our help`
`12:01.42`	`ManxPower-work`	`TommyBotten, but you might want to actually add a [test] user to your setup.`
`12:02.12`	`ManxPower-work`	`You said you had one, but you don't.`
`12:03.09`	`TommyBotten`	`faxtest is one`
`12:03.17`	`TommyBotten`	`just removed the fax part`
`12:03.21`	`ManxPower-work`	`but that is not the message you pasted.`
`12:03.42`	`ManxPower-work`	`I give up. TommyBotten in the future make sure your configs match what you say they are.`
`12:04.34`	`ManxPower-work`	`first you say 'Registration from '<sip:test@x.x.x.x;user=phone>' then you provide configs with no [test] user.`
`12:04.50`	`TommyBotten`	`I changed the test to faxtest.`
`12:04.52`	`TommyBotten`	`Sorry about that`
`12:05.14`	`ManxPower-work`	`TommyBotten, now repastebin with the CORRECT stuff.`
`12:05.26`	`ManxPower-work`	`as well as the error messges you get with THAT setup.`
`12:08.41`	`*** join/#asterisk fish-bulb (~cstewart@nat/digium/x-ctccwldlbjplbfrt)`
`12:08.49`	`ManxPower-work`	`TommyBotten, oh, and stop indenting your configs.`
`12:09.22`	`adnc`	`ahh, i get a permission denied when executing "Action: Command" and "Command: sip show peers"`
`12:09.47`	`ManxPower-work`	`TommyBotten, Also remove the usertype=phone it is useless and not a valid option.`
`12:13.28`	`Tim_Toady`	`the B410 digium cards have hardware echo canceler, do i have to add something like echocanceller=foo in system.conf for it to be enabled?`
`12:16.23`	`*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)`
`12:17.12`	`ManxPower-work`	`[TK]D-Fender, RUN! RUN!`
`12:17.22`	`*** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23)`
`12:17.32`	`[TK]D-Fender`	`runs in circles`
`12:18.37`	`Dovid`	`lol`
`12:18.50`	`*** join/#asterisk rttrey (~trey@209.208.18.121)`
`12:19.58`	`*** join/#asterisk ccesario_ (~ccesario@189-19-6-236.dsl.telesp.net.br)`
`12:21.26`	`fish-bulb`	`Tim_Toady: shouldn't have to add anything, it will be automatically enabled`
`12:22.38`	`Tim_Toady`	`ok i ask because i get this when run dahci_cfg: Channel 05: Clear channel (Default) (Echo Canceler: none) (Slaves: 05)`
`12:22.39`	`*** join/#asterisk kotp (~vgoff@96.2.187.66)`
`12:23.29`	`adnc`	`i was following a tutorial where a php lib called starastapi is described, unfortunately the domain is not anymore reachable, are there alternatives to it you could point me to?`
`12:23.49`	`ManxPower-work`	`Tim_Toady, sounds like you messed up your config file. Remember set the options, THEN specify the channel`
`12:24.41`	`*** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk)`
`12:25.07`	`fish-bulb`	`yeah that means software echo canceler, which you don't need since you have hardware echo cancellation`
`12:25.30`	`ManxPower-work`	`fish-bulb, cite your source`
`12:25.53`	`ManxPower-work`	`Tim_Toady, add echocancel=yes`
`12:26.58`	`Tim_Toady`	`ManxPower-work i just did... i get the same: Channel 05: Clear channel (Default) (Echo Canceler: none) (Slaves: 05)`
`12:27.00`	`fish-bulb`	`ManxPower-work: he is asking about the DAHDI configuration, not chan_dahdi. But yes, echocancel=yes must be enabled for any echo canceler to work`
`12:27.16`	`fish-bulb`	`ManxPower-work: it is in the system.conf sample file`
`12:27.16`	`ManxPower-work`	`Tim_Toady, pastebin.ca your config`
`12:27.54`	`ManxPower-work`	`fish-bulb, nobody seems to read the sample configs`
`12:28.12`	`petern_`	`hurr`
`12:28.28`	`ManxPower-work`	`Or the readme for that matter.`
`12:28.33`	`Tim_Toady`	`simple config generated with dahdi_genconf: http://pastebin.com/a6BHJWwb`
`12:28.51`	`dlynes`	`ManxPower-work, or the core show, or the ..., or the .... :)`
`12:28.56`	`ManxPower-work`	`Tim_Toady, no, dahdi_genconf does NOT generate working configs.`
`12:29.01`	`Tim_Toady`	`oh`
`12:29.05`	`Tim_Toady`	`nice :D`
`12:29.20`	`ManxPower-work`	`it generates SAMPLE configs that you ADAPT to your system.`
`12:29.20`	`Tim_Toady`	`i thought it was ok for system.conf`
`12:29.51`	`Tim_Toady`	`ok, anyway this conf seems ok, echocanceller=yes is in its place`
`12:29.53`	`petern_`	`i set up an IAX2 trunk, which works from a local phone, but if an external call is routed over it, asterisk on the machine with the ISDN interface crashes`
`12:29.55`	`ManxPower-work`	`now, pastebin the DAHDI configs. Make sure to remove any #include lines and replace them with the included data in your paste.`
`12:30.02`	`*** join/#asterisk AlHafoudh (~AlHafoudh@213.151.217.151)`
`12:30.28`	`[TK]D-Fender`	`Tim_Toady: echocanceller=yes <- this should have an EC module, not "yes"`
`12:30.41`	`[TK]D-Fender`	`Tim_Toady: echocanceller=mg2,1 <- example`
`12:30.57`	`Tim_Toady`	`[TK]D-Fender i have a B410 with hardware echo canceller`
`12:31.17`	`Tim_Toady`	`the guys here told me to add 'yes'`
`12:31.17`	`ManxPower-work`	`Oh, I'm SURE Tim_Toady read the DAHDI readme files so he knows that stuff.`
`12:31.42`	`[TK]D-Fender`	`Tim_Toady: What does * say at CLI?`
`12:31.45`	`ManxPower-work`	`Tim_Toady, it's up to you to map old commands to the new commands.`
`12:31.47`	`Kobaz`	`[Mar 16 19:52:45] WARNING[18380]: tcptls.c:141 ast_tcptls_server_root: Accept failed: Too many open files`
`12:31.54`	`Kobaz`	`i got like 32894239874983247 of those on a server this morning`
`12:32.07`	`Kobaz`	`and then it died`
`12:32.22`	`ManxPower-work`	`Kobaz, nod There's various patches on issues.digium.com having to do with Asterisk not closing sockets under some situations.`
`12:32.24`	`dlynes`	`Tim_Toady, not guys...just fish-bulb`
`12:32.33`	`Kobaz`	`well it didn't die on it's own, my asterisk watcher script killed asterisk since it wasn't responding anymore`
`12:32.37`	`Kobaz`	`ManxPower-work: ah good`
`12:32.42`	`Kobaz`	`ManxPower-work: this is an older 1.6.0`
`12:32.58`	`ManxPower-work`	`Kobaz, I found them when NOBODY WOULD EVEN RESPOND to a question I had yesterday.`
`12:33.31`	`Tim_Toady`	`[TK]D-Fender wait a bit to write chan_dahdi.conf`
`12:33.31`	`Kobaz`	`heh`
`12:33.37`	`Kobaz`	`it happens`
`12:33.38`	`fish-bulb`	`Tim_Toady: do not put echocanceller=yes in system.conf, make sure you have echocancel=yes in chan_dahdi.conf`
`12:33.45`	`Kobaz`	`i'm working on a libpri crash that noone cares about`
`12:34.11`	`ManxPower-work`	`Kobaz, we rolled back to a much older version of Asterisk.`
`12:34.23`	`*** join/#asterisk hluesea (~hulusikah@88.247.127.66)`
`12:34.25`	`dlynes`	`Kobaz, i'm at the point where from all the grief voip has caused me over the last several years, I don't care about voip anymore :)`
`12:34.28`	`Kobaz`	`you really need to be a c coder, and know all the asterisk internals to be a successful asterisk based vendor`
`12:34.33`	`ManxPower-work`	`Tim_Toady, ignore everyone and read the DAHDI readmes.`
`12:34.40`	`fish-bulb`	`Tim_Toady: the "echocanceller" definition is just for software echo cancelers, which you do not need since you have hardware echo cancellation`
`12:34.44`	`Tim_Toady`	`:)`
`12:34.53`	`Kobaz`	`dlynes: but this isn't even voip really... well... voip on the internal side... tdm/t1 on the outside`
`12:34.54`	`Tim_Toady`	`ok ok let me see`
`12:35.21`	`ManxPower-work`	`fish-bulb, again, cite your source.`
`12:35.28`	`Kobaz`	`so it's a good thing we have some c coders (myself included), that work on asterisk`
`12:35.39`	`Kobaz`	`otherwise only half my problems would get fixed`
`12:35.45`	`fish-bulb`	`ManxPower-work: The sample DAHDI configuration files and experience with the product`
`12:35.53`	`*** join/#asterisk lesouvage (~lesouvage@82.73.69.76)`
`12:36.14`	`Kobaz`	`this year alone i've probably contributed 100 hours worth of asterisk bug reports and debugging`
`12:36.38`	`Kobaz`	`i find a new bug roughly once a week`
`12:37.04`	`*** join/#asterisk javb (~Anonimo@190.166.53.135)`
`12:37.21`	`ManxPower-work`	`The DAHDI changelog is AWESOME!`
`12:37.52`	`Kobaz`	`ManxPower-work: the libpri code is a wreck`
`12:38.01`	`Kobaz`	`i want to gouge my eyes out`
`12:38.21`	`fish-bulb`	`ManxPower-work: specifically "it is very important that you specify one here if you do not have hardware echo cancellers and need echo cancellation"`
`12:38.32`	`Kobaz`	`f = pri->txqueue, s = r.l.l.v; q = r.l.l.d;`
`12:38.43`	`Kobaz`	`who writes code like that anymore?`
`12:39.35`	`ManxPower-work`	`fish-bulb, now convince Tim_Toady to read that.`
`12:39.45`	`fish-bulb`	`Tim_Toady: Go read that`
`12:39.49`	`Kobaz`	`if i had time, i would rewrite libpri from scratch and write unit tests to go along with it`
`12:40.53`	`ManxPower-work`	`Kobaz, the only bugs we seem to find are on productions servers.`
`12:40.59`	`Kobaz`	`ManxPower-work: me too`
`12:41.04`	`ManxPower-work`	`So we can't really troubleshoot the problem.`
`12:41.07`	`Kobaz`	`ManxPower-work: this libpri crash is on a production box`
`12:41.34`	`Kobaz`	`and only on a production box... AND, it's the same linux kernel. same libpri, same asterisk, same wanpipe drivers as 50 other installations`
`12:41.40`	`Kobaz`	`and none of them have this problem`
`12:41.43`	`ManxPower-work`	`All attended transfers broke on our Asterisk yesterday.`
`12:42.25`	`ManxPower-work`	`Turns out Asterisk may or may not be generating a packet which may or may not be valid, depending on who you ask.`
`12:42.30`	`Kobaz`	`heh`
`12:42.50`	`Kobaz`	`do you have an in-house asterisk developer?`
`12:42.55`	`ManxPower-work`	`no.`
`12:42.59`	`Kobaz`	`hire one :P`
`12:43.05`	`ManxPower-work`	`not my call`
`12:43.23`	`Kobaz`	`i think it's _required_, honestly`
`12:43.45`	`ManxPower-work`	`Kobaz, oh, I tend not to put up with Asterisk bugs. We'll roll back to a known working version if we have to.`
`12:44.03`	`Kobaz`	`ManxPower-work: the problem is... i need to roll up to fix bugs i've encountered`
`12:44.14`	`Kobaz`	`and if i roll back, i'll still hit the same bugs i'm trying to get away from`
`12:44.17`	`benngard`	`http://store.digium.com/productview.php?product_code=1TE220BF <- any using that card?`
`12:44.23`	`ManxPower-work`	`Kobaz, we have different needs.`
`12:44.29`	`Kobaz`	`if the new version is much worse than the old one, i generally apply the patches that I need to my version`
`12:44.44`	`petern_`	`hmm, damn, it's chan_lcr that's crashing :(`
`12:44.49`	`ManxPower-work`	`If we REALLY need some coding done, we'll post a bounty.`
`12:44.49`	`Kobaz`	`i have about 1000 lines of my own patches now`
`12:45.09`	`ManxPower-work`	`I have 2 lines of my own patches now.`
`12:45.35`	`Katty`	`testosterone spill in aisle 2`
`12:45.36`	`ManxPower-work`	`plus Dawg knows how many lines the CONNECTEDLINE and SIPRemoveHeader patches are.`
`12:45.53`	`Kobaz`	`heh`
`12:46.04`	`Kobaz`	`i haven't needed sipremoveheader yet`
`12:46.33`	`ManxPower-work`	`Kobaz, Our NexTone MSX SIP proxy freaks out if it gets a call with Alert-Info: header.`
`12:46.40`	`Kobaz`	`haha`
`12:46.47`	`Kobaz`	`i had a gateway like that`
`12:46.56`	`ManxPower-work`	`freak out like drop the call and stomp it into the floor`
`12:47.43`	`ManxPower-work`	`Kobaz, You know how much I hate Microsoft, right? Well I think NexTone is worse.`
`12:47.48`	`Kobaz`	`hehe`
`12:48.05`	`[TK]D-Fender`	`benngard: probably lots of people. Got a more specific question?`
`12:48.10`	`ManxPower-work`	`They do seem the have the biggest and slowest java application in the workd and use it to manage the box.`
`12:49.33`	`Kobaz`	`reminds me of the software to manage a livingston portmaster`
`12:51.50`	`ManxPower-work`	`I didn't deal with those much, but when I did, telnet was the tool I used.`
`12:51.51`	`Nugget`	`telnet is eeeeeeevil!`
`12:52.18`	`ManxPower-work`	`we used it as a "KVM switch, but for serial consoles" kind of server.`
`12:52.20`	`Kobaz`	`i think someone broke the telnet interface on the one i had to use`
`12:52.35`	`Kobaz`	`kvm switch?`
`12:52.38`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`12:52.49`	`ManxPower-work`	`I was always an Ascend guy rather than Livingston`
`12:52.50`	`Kobaz`	`hmm, the portmaster i used was for a dialup modem pool`
`12:53.17`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`12:54.12`	`*** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk)`
`12:55.36`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`12:56.04`	`ManxPower-work`	`"Beware of all enterprises that require new clothes." --Henry David Thoreau`
`12:57.23`	`petern_`	`hmm`
`12:57.28`	`coppice`	`ManxPower-work: what about fashion resellers?`
`12:57.30`	`petern_`	`is it valid for ast_frame.data to be null?`
`12:57.52`	`benngard`	`[TK]D-Fender: yupp, realized my question was bad, i was looking for some1 who had used the card in sweden`
`12:58.15`	`ManxPower-work`	`coppice, Ascend? No, we were an ISP.`
`12:58.35`	`ManxPower-work`	`petern_, that is a question for #asterisk-dev`
`12:58.47`	`petern_`	`true`
`12:58.50`	`[TK]D-Fender`	`benngard: Got an even MORE specific question like "Can you explain how to fix this exact problem I'm having?"`
`12:58.59`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`12:59.18`	`coppice`	`ManxPower-work: your quote says beware of all ....`
`13:00.17`	`ManxPower-work`	`coppice, Ah. Yes, I would be especially wary of fashion reseller job.`
`13:00.24`	`benngard`	`[TK]D-Fender: i am planning to buy that card, but wanna be 100% sure it works in sweden`
`13:00.48`	`ManxPower-work`	`benngard, as long as Sweden is still in Europe, it should work just fine.`
`13:01.11`	`ManxPower-work`	`I personally recommend Sangoma.`
`13:01.46`	`[TK]D-Fender`	`benngard: Why wouldn't it?`
`13:03.04`	`benngard`	`guess it would work, but it would be nice to "talk" to some1 in sweden who actually use that card, but what the heck, i tell my boss to order a card`
`13:04.11`	`Kobaz`	`benngard: call up sangoma, they are a very friendly canadian company`
`13:04.25`	`Kobaz`	`benngard: they will know if their card works with sweden telco`
`13:05.23`	`TommyBotten`	`ManxPower-work: Sorry for the delay... battery ran out. The usertype=phone which you mentioned fixed it.`
`13:05.33`	`TommyBotten`	`ManxPower-work: And you mentioned not identing configs. Why not?`
`13:05.47`	`TommyBotten`	`ManxPower-work: And also, thanks for the fix :)`
`13:05.54`	`ManxPower-work`	`sigh`
`13:06.02`	`TommyBotten`	`?`
`13:06.37`	`[TK]D-Fender`	`benngard: I would also state that I would go for a Sangoma instead as ManxPower-work advised`
`13:07.31`	`*** join/#asterisk kotp (~vgoff@96.2.187.66)`
`13:08.41`	`coppice`	`ManxPower-work: I generally avoid work in the clothing and weapons industries. I've have enough of this cloak and dagger stuff`
`13:09.18`	`ManxPower-work`	`coppice, SSSHHHHH!!!!!! The Theatans are listening!`
`13:10.17`	`*** join/#asterisk the1_ (~x@122.52.175.49)`
`13:10.17`	`*** join/#asterisk brad_mssw (~brad@shop.monetra.com)`
`13:10.29`	`*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)`
`13:10.49`	`the1_`	`hello can anyone help with dahdi channel groupings?`
`13:11.00`	`*** join/#asterisk xpot-mobile (~xpot@66.60.101.91)`
`13:11.13`	`Kobaz`	`the1_: you put channels into groups, and you dial the group, and all channels are used`
`13:11.24`	`Kobaz`	`all channels in that group`
`13:11.57`	`the1_`	`ok here is my config http://pastie.org/873702`
`13:12.53`	`the1_`	`am i doing it correctly? becuase when i dial g1, it goes out to channel 15 which is group 2`
`13:13.51`	`*** join/#asterisk dennisG (~root@84.30.136.208)`
`13:13.59`	`Kobaz`	`no`
`13:14.02`	`Kobaz`	`you have it backwards`
`13:14.37`	`the1_`	`how so?`
`13:14.42`	`Kobaz`	`er, wait hmm, i haven't manually configured a dahdi in bit`
`13:15.07`	`the1_`	`channels 1 to 14 are actually fx0 (using fxs signalling)`
`13:15.22`	`the1_`	`channel 15 to 16 are fxs (using fx0 signalling)`
`13:15.23`	`Kobaz`	`okay that's correct... group goes before channel =`
`13:15.29`	`*** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98)`
`13:15.43`	`the1_`	`hmm this is odd`
`13:15.47`	`elliot98`	`honks his friendly horn`
`13:15.54`	`elliot98`	`hello everyone`
`13:15.57`	`Kobaz`	`g1 only goes out 15?`
`13:15.59`	`Kobaz`	`or 15 and 16`
`13:17.01`	`the1_`	`sometimes it goes out correctly through port 14 (only line with actually telco line), but sometimes it goes out to port 15`
`13:17.24`	`the1_`	`btw, what happens when a channel is configured but theres no line/dial tone`
`13:17.35`	`the1_`	`will asterisk ignore the line?`
`13:17.44`	`Kobaz`	`it'll try and dial it, but it will fail`
`13:17.44`	`ManxPower-work`	`the1_, then you send your call to nothing`
`13:18.01`	`ManxPower-work`	`Depending on the card it may or may not consider the port down.`
`13:18.14`	`Kobaz`	`yeah... digium cards have a habit of just answering and giving you dead air`
`13:18.18`	`the1_`	`oh ic`
`13:18.36`	`*** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru)`
`13:18.45`	`the1_`	`im using the atcom ax 1600p`
`13:18.52`	`Kobaz`	`never used it`
`13:19.01`	`ManxPower-work`	`Kobaz, I was going to say that we have a newer digium card that detects the line, but then I remembered, no, we replaced the Digium card with a Sangoma. 8-)`
`13:19.04`	`the1_`	`drivers are digium`
`13:19.06`	`benngard`	`sangoma was a little bit more expensive than digium, but no big difference`
`13:19.12`	`Kobaz`	`but it's recomended to take out the channels that aren't actually in working order`
`13:19.24`	`Kobaz`	`benngard: sangoma cards have much better debugging toold`
`13:19.25`	`Kobaz`	`tools`
`13:19.45`	`benngard`	`seems like most recommend me to buy sangoma`
`13:19.58`	`the1_`	`i tried taking that out with config channel=>14`
`13:20.14`	`the1_`	`so that with dahdi, it only sees one fx0`
`13:20.17`	`the1_`	`will that work?`
`13:20.22`	`Kobaz`	`that's fine`
`13:20.35`	`Kobaz`	`fxo not fx0`
`13:20.59`	`the1_`	`yep :) my ultimate goal is to be able to dialout (7digit) without using 9`
`13:21.05`	`the1_`	`so i have this dialplan`
`13:21.07`	`Kobaz`	`that's the easy part`
`13:21.20`	`the1_`	`exten => _NXXXXXX,1,Dial(dahdi/g1/${EXTEN:0})`
`13:21.23`	`elliot98`	`who is in charge of the asterisk.org website?`
`13:21.40`	`Kobaz`	`${EXTEN:0} is the same as ${EXTEN}`
`13:22.08`	`the1_`	`yeah i thought so, i got varying results so i tried :0 also`
`13:22.33`	`Kobaz`	`${EXTEN:skip:limit}`
`13:22.46`	`Kobaz`	`skip = skip this many digits... and give me the rest`
`13:22.52`	`*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`13:22.52`	`*** mode/#asterisk [+o leifmadsen] by ChanServ`
`13:23.02`	`Kobaz`	`skip:limit = skip this many digits, and give me <limit> digits after`
`13:23.17`	`Kobaz`	`so 0:4 will give you the first four digits`
`13:23.36`	`Kobaz`	`using just 0, will skip 0 digits.. and give you the rest`
`13:23.45`	`the1_`	`ahhhhh 0:7 will give me 7 digits? startting from the left?`
`13:23.46`	`*** join/#asterisk slinksh0t (~slinksh0t@66.90.110.157)`
`13:23.56`	`Kobaz`	`but, you are already matching on 7`
`13:23.59`	`Kobaz`	`so you will have 7`
`13:24.18`	`Kobaz`	`only 7 digit dialing will run that bit of code`
`13:24.27`	`the1_`	`ahhh i got it`
`13:24.28`	`Kobaz`	`because of the NXXXXXX`
`13:24.35`	`the1_`	`heres my error msg http://pastie.org/873720`
`13:24.59`	`Kobaz`	`your dialplan is correct, your dahdi is not configured correctly`
`13:26.30`	`Kobaz`	`either that or the number you dialed is actually busy or not available`
`13:26.41`	`Katty`	`hi`
`13:26.50`	`Kobaz`	`first make sure you can pick up a regular phone and make a call on that line`
`13:27.02`	`Kobaz`	`it's teh katty`
`13:27.15`	`the1_`	`i know its not busy`
`13:27.25`	`the1_`	`and i can make a call`
`13:27.35`	`Kobaz`	`so it's your configuration, or the card itself`
`13:27.37`	`the1_`	`http://pastie.org/873731 thats the instruction from atcom pdf`
`13:27.54`	`the1_`	`notice it only has one group= config`
`13:28.04`	`the1_`	`i have a feeling that is incorrect`
`13:28.24`	`Kobaz`	`generally you'll want to split them`
`13:28.52`	`Kobaz`	`paste your output of dahdi_cfg -vvvv also`
`13:29.22`	`the1_`	`http://pastie.org/873733`
`13:29.25`	`Katty`	`i had a really weird dream last night.`
`13:29.31`	`*** join/#asterisk Arsenick (~y@69.70.231.230)`
`13:29.57`	`Kobaz`	`me too... i dreamed that..`
`13:29.59`	`*** join/#asterisk babbio (~matteo@host-78-13-24-238.cust-adsl.tiscali.it)`
`13:30.02`	`Kobaz`	`HOLY CRAP IT'S BACK`
`13:30.07`	`Kobaz`	`it's gonna eat me!`
`13:30.12`	`Katty`	`:<`
`13:30.21`	`Kobaz`	`anyways`
`13:30.23`	`Kobaz`	`you were saying?`
`13:30.33`	`Katty`	`that sounds like a skeery dream.`
`13:30.34`	`*** join/#asterisk Jhirley (~Jhirley@mail.mmdlaw.com)`
`13:31.04`	`babbio`	`i guya i'm new on asterisk...i have installed asterisknow, then i'm trying to write my little test extensions.conf but when i run "dialplan show" i have no extensions.....why? please help`
`13:32.03`	`Kobaz`	`~asterisknow`
`13:32.04`	`infobot`	`rumour has it, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead.`
`13:32.04`	`Katty`	`i dreamt that (i think) vampires were invading the house. and i had to use my magical spells (from fable 2 for xbox) to defeat them. somehow my dog got a back leg chopped in half and that i got chopped in half around the torso area. but some person assisting in the battle knew how to put people back together. It involved using a crayon to trace a line where they should go back together, and then singing a little song`
`13:32.22`	`asteriskATmarmuD`	`:)`
`13:32.26`	`Kobaz`	`Katty: sounds fun`
`13:32.37`	`asteriskATmarmuD`	`ok, so I wont ask about vicidial...`
`13:32.49`	`Katty`	`Kobaz: i suppose.`
`13:33.05`	`Katty`	`Kobaz: i think i was more heart broken over my dog's legt han i was about gettin chopped in half`
`13:33.09`	`Kobaz`	`i wouldn't think many people know about vicidial either`
`13:33.09`	`the1_`	`Kobaz thanks for the help, will troubleshoot tomorrow, have to go home now`
`13:33.15`	`Kobaz`	`the1_: k`
`13:33.21`	`the1_`	`thanks again`
`13:33.23`	`[TK]D-Fender`	`the1_: did you restart * completely after setting the group?`
`13:33.30`	`the1_`	`yep`
`13:33.40`	`the1_`	`tried to stop asterisk`
`13:33.41`	`Katty`	`i was out walking riddick yesterday evening and the folks at the end of the road half this white dog always chained up outside their house.`
`13:33.43`	`the1_`	`restart dahdi`
`13:33.49`	`the1_`	`start asterisk`
`13:34.12`	`the1_`	`also tried to restart dahdi from the cli`
`13:34.19`	`Katty`	`well their kids had the dog off the chain and were walking him around the neighborhood (off leash). the dog decided he was going to attack either me or riddick. not sure to be honest, because the dog never actually got to me if he was after me.`
`13:35.07`	`babbio`	`nobody answer me on #asterisknow.....my question is very trivial for you...please help me....`
`13:35.08`	`Katty`	`thankfully riddick was not hurt and the owner of the other dog pulled him off riddick before much damage could be done.`
`13:35.58`	`Katty`	`been debating whether or not to contact the city about it. these people at the end of the road are....people who would probably slash tires and put sugar in the gas tank.`
`13:37.15`	`Kobaz`	`mmm sugar`
`13:37.15`	`[TK]D-Fender`	`babbio: do a "reload" and see if it takes. If not, pastebin "ls -la /etc/asterisk", and your extensions.conf`
`13:37.22`	`leifmadsen`	`video cameras can help solve that problem`
`13:38.14`	`Katty`	`i'm sure they would certainly help`
`13:38.19`	`Katty`	`but they have 3 children who love that dog`
`13:39.56`	`Katty`	`i don't want the city to take the kid's dog away....`
`13:40.06`	`Katty`	`i just dont' want it attacking riddick anymore.`
`13:40.20`	`Katty`	`as it is riddick is probably going to be very cranky everytime we walk by now :<`
`13:41.56`	`babbio`	`[TK]D-Fender: ok give me a minute because asterisk is on my server in the other room`
`13:42.08`	`[TK]D-Fender`	`~pb`
`13:42.09`	`infobot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit`
`13:42.10`	`[TK]D-Fender`	`^^^^`
`13:42.15`	`Jhirley`	`o/`
`13:42.28`	`Katty`	`hi Jhirley`
`13:42.28`	`[TK]D-Fender`	`babbio: SSH doesn't care where you are`
`13:42.33`	`smooth_p[work]`	`hey!`
`13:42.38`	`Katty`	`hugs smooth_p[work]`
`13:42.47`	`Kobaz`	`ssh cares! ssh misses me when i'm not around`
`13:43.04`	`babbio`	`[TK]D-Fender: i know but here i'm on windows`
`13:43.19`	`[TK]D-Fender`	`babbio: http://www.chiark.greenend.org.uk/~sgtatham/putty/`
`13:43.26`	`smooth_p[work]`	`:>`
`13:43.27`	`[TK]D-Fender`	`babbio: SSH also doesn't care what platform`
`13:43.31`	`Brooklyn`	`pokes dennisG with some Asterisk 1.4 server`
`13:43.35`	`[TK]D-Fender`	`babbio: My PHONE could do this`
`13:43.52`	`babbio`	`[TK]D-Fender: great...i'll check it out`
`13:43.57`	`Jhirley`	`ssh has feeling too, just because its a cold heartless protcol !`
`13:44.24`	`Jhirley`	`it does care, now telnet is a slut.`
`13:44.44`	`Katty`	`psh`
`13:44.47`	`Jhirley`	`k, I will shut up now. :)`
`13:44.55`	`Katty`	`telnet has booking fees`
`13:44.58`	`Katty`	`do you?`
`13:45.07`	`babbio`	`[TK]D-Fender: ok i'm in`
`13:45.08`	`babbio`	`now`
`13:45.42`	`Katty`	`checks on the varmits`
`13:45.57`	`Katty`	`oh dear :< almost out of food.`
`13:47.08`	`babbio`	`this is the reload output http://pastebin.ubuntu.com/396689/`
`13:47.31`	`*** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp)`
`13:48.20`	`Katty`	`Kobaz: humming bird migration starts soon (=`
`13:49.10`	`Kobaz`	`mmm`
`13:50.06`	`Katty`	`will have to break out the feeder here soon. i got a pretty red rainbow one`
`13:50.22`	`babbio`	`[TK]D-Fender: do u see something wrong?`
`13:50.25`	`[TK]D-Fender`	`babbio: pastebin the LS dump I asked you for, alogn with the extensions.conf I asked you far, and while you're at it... modules.conf`
`13:51.23`	`*** join/#asterisk adnc (~numer@unaffiliated/adnc)`
`13:51.39`	`*** join/#asterisk moy (~chatzilla@74.12.130.209)`
`13:53.39`	`babbio`	`[TK]D-Fender: here u are http://pastebin.ubuntu.com/396694/`
`13:53.39`	`*** join/#asterisk shader (~user@cpe-174-097-217-107.nc.res.rr.com)`
`13:53.55`	`[TK]D-Fender`	`babbio: -rw-r--r-- 1 root root 107 Mar 17 11:05 extensions.conf <--- owned by ROOT instead of ASTERISK like the otehrs. * has no rights to it`
`13:54.01`	`[TK]D-Fender`	`babbio: chown it`
`13:54.35`	`Kobaz`	`asterisk can read it`
`13:54.49`	`*** join/#asterisk monotek (~abauer@line1.magix.net)`
`13:54.50`	`[TK]D-Fender`	`don't care`
`13:55.10`	`Kobaz`	`unless you're creating dialplan during runtime it's not an issue`
`13:55.15`	`[TK]D-Fender`	`Kobaz: IIRC * also typically tries to reserve the right to write to it as well`
`13:55.26`	`*** part/#asterisk monotek (~abauer@line1.magix.net)`
`13:55.27`	`Kobaz`	`all my config files are owned by root`
`13:55.30`	`[TK]D-Fender`	`Kobaz: If we're loking for a culprit, fix the damn rights.`
`13:55.41`	`*** join/#asterisk monotek (~abauer@line1.magix.net)`
`13:55.41`	`[TK]D-Fender`	`Kobaz: yes, but is your ASTERISK running as root?`
`13:55.50`	`[TK]D-Fender`	`Kobaz: His isn't.`
`13:55.59`	`Kobaz`	`my asterisk runs as the pbx user`
`13:56.44`	`[TK]D-Fender`	`[09:55]<Kobaz>my asterisk runs as the pbx user <- which is?`
`13:56.53`	`Kobaz`	`the user called 'pbx'`
`13:57.13`	`monotek`	`Hi.`
`13:57.13`	`monotek`	`I want to configure wrapuptime in a ringgroup or in general but dont want to us a queue.`
`13:57.14`	`monotek`	`Is this possible or does this only work for queues?`
`13:57.14`	`monotek`	`Thanks i advance`
`13:57.14`	`monotek`	`AndrÃ©`
`13:57.23`	`Kobaz`	`babbio: that's your entire console output? looks like you're not loading the pbx_* modules`
`13:57.24`	`[TK]D-Fender`	`Kobaz: interesting choice. But "whatever"`
`13:57.43`	`Kobaz`	`[TK]D-Fender: what's it matter what user asterisk runs as?`
`13:57.46`	`[TK]D-Fender`	`monotek: Wrapuptime is a queue feature. The End`
`13:57.50`	`Kobaz`	`you could call it foobar`
`13:57.57`	`Kobaz`	`it'll still run :P`
`13:58.05`	`babbio`	`i changed the owner now?`
`13:58.24`	`Kobaz`	`babbio: that's not going to do anything necessarily... ls -al /etc/asterisk`
`13:58.25`	`[TK]D-Fender`	`babbio: Are you asking us what you just did? We're only psychic on TUESDAYS`
`13:58.32`	`monotek`	`i know. any kind of workaround possible?`
`13:58.52`	`*** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek)`
`13:59.05`	`Katty`	`i like how they show these photos of hummingbird feeders with multiple hummingbirds on them.`
`13:59.12`	`Katty`	`that just doesn't happen.`
`13:59.14`	`[TK]D-Fender`	`monotek: make aq lot of dialplan to log when they hung up, and to check if enough time has passed before including them in a call.`
`13:59.48`	`*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)`
`13:59.53`	`*** join/#asterisk Rahr (~Rahr@212.178.158.35)`
`14:00.03`	`babbio`	`sorry [TK]D-Fender i'm only trying to understand hwo to make it run....i have bought "asterisk: the future of telephony" and i've just followed the instruction for the first extension test but it doesn't work`
`14:00.21`	`Kobaz`	`babbio: if that's your entire console output, not all the asterisk modules are loading`
`14:00.22`	`[TK]D-Fender`	`babbio: Did you fix the permissions like I said?`
`14:00.40`	`babbio`	`yes i do`
`14:00.40`	`Kobaz`	`[TK]D-Fender: permissions aren't the issue`
`14:01.02`	`Kobaz`	`babbio: paste your /etc/asterisk/asterisk.conf and /etc/asterisk/modules.conf`
`14:02.03`	`babbio`	`Kobaz: here you are http://pastebin.ubuntu.com/396697/`
`14:03.47`	`*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)`
`14:03.47`	`*** mode/#asterisk [+o putnopvut] by ChanServ`
`14:04.07`	`*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)`
`14:04.14`	`*** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465)`
`14:04.44`	`Katty`	`hi angryuser`
`14:04.46`	`Katty`	`oh`
`14:04.47`	`*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)`
`14:04.48`	`Katty`	`hi anthm`
`14:05.07`	`anthm`	`hello`
`14:05.08`	`monotek`	`@ d-fender: sorry. i dont understand. how to do this?`
`14:05.08`	`monotek`	`I hoped i could implement some kind of "Wait(60)" into extension...`
`14:06.38`	`[TK]D-Fender`	`monotek: No. You have to track when the device ends its last call (probably monitor via AMI) and check with every call placed if enough time has passed`
`14:10.28`	`pisg`	`some have scripts load extensions.conf and insert to some mysql databases ?`
`14:11.27`	`*** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23)`
`14:11.37`	`*** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23)`
`14:12.08`	`Katty`	`it's gonna be a long day, isn't it`
`14:15.31`	`*** join/#asterisk OlafsenM (~omorika@193.198.31.85)`
`14:15.34`	`OlafsenM`	`hi guys`
`14:15.40`	`Katty`	`ahem.`
`14:15.51`	`OlafsenM`	`how can I get two separate CDRs for bridged channels?`
`14:16.05`	`OlafsenM`	`it seems that only one is generated`
`14:16.25`	`OlafsenM`	`when i bridge IN and OUT call with manager Bridge action`
`14:19.39`	`*** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh)`
`14:20.16`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
`14:20.48`	`*** join/#asterisk Brooklyn (miracle@the.city.that.never-sleeps.us)`
`14:20.59`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`14:21.52`	`*** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net)`
`14:22.16`	`*** join/#asterisk andres833 (~andres833@190.144.75.22)`
`14:22.42`	`*** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net)`
`14:26.25`	`*** join/#asterisk tamiel (~tamiel@213.30.183.226)`
`14:31.34`	`*** join/#asterisk sourcode (~code@ppp-58-11-76-109.revip2.asianet.co.th)`
`14:31.34`	`OlafsenM`	`anyone here?`
`14:31.39`	`OlafsenM`	`helloooo?`
`14:32.00`	`*** join/#asterisk sourcode (~code@ppp-58-11-76-109.revip2.asianet.co.th)`
`14:32.02`	`Katty`	`impatient much.`
`14:33.00`	`OlafsenM`	`i just need an info, nothing else`
`14:33.43`	`Katty`	`mhmm`
`14:34.01`	`tzafrir_laptop`	`ForkCDR?`
`14:35.16`	`OlafsenM`	`ForkCDR?`
`14:37.01`	`Kobaz`	`forking cdrs`
`14:38.03`	`OlafsenM`	`well, i've checked the description`
`14:38.21`	`OlafsenM`	`and it doesn't look like something that i need`
`14:38.44`	`OlafsenM`	`the problem is`
`14:38.45`	`OlafsenM`	`that`
`14:38.50`	`OlafsenM`	`i got A and B calls`
`14:38.57`	`OlafsenM`	`at some point I merge them`
`14:39.12`	`OlafsenM`	`and then at some point i UNBRIDGE them`
`14:40.55`	`*** join/#asterisk anonymouz666 (~anonymouz@189.24.87.110)`
`14:44.19`	`Katty`	`hi anonymouz666`
`14:44.26`	`anonymouz666`	`hello Katty!`
`14:55.27`	`*** join/#asterisk UQlev (~yuriy@212.50.99.8)`
`14:57.12`	`*** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler)`
`14:59.10`	`*** join/#asterisk E-bola (~bola@smtp.techbiz.dk)`
`14:59.25`	`E-bola`	`I'm looking for something cool/fun to offer in my queue`
`14:59.36`	`E-bola`	`Anybody know of any neat scripts or other interesting stuff`
`14:59.52`	`E-bola`	`i was thinking like small games or trivia or something that users could opt-in for instaid of just moh`
`15:02.11`	`*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)`
`15:04.04`	`*** part/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net)`
`15:08.00`	`*** join/#asterisk andrebarbosa (~andrebarb@62.28.143.14)`
`15:08.56`	`andrebarbosa`	`Hi`
`15:09.08`	`andrebarbosa`	`I have an asterisk that has random segfaults`
`15:09.22`	`andrebarbosa`	`I built a version with DONT_OPTMIZE`
`15:09.28`	`andrebarbosa`	`but with that option`
`15:09.37`	`andrebarbosa`	`it crashes like every 5 minutes`
`15:09.42`	`andrebarbosa`	`any reason for that?`
`15:09.43`	`andrebarbosa`	`:s`
`15:09.55`	`[TK]D-Fender`	`E-bola: You can't take input while sitting in a queue, only to exit`
`15:11.20`	`E-bola`	`[TK]D-Fender: Couldnt u make some hacks to get around that?`
`15:11.22`	`anonymouz666`	`andrebarbosa: look at the core file...`
`15:11.28`	`E-bola`	`i mean simple ones`
`15:12.08`	`andrebarbosa`	`kern.log:Mar 17 14:39:22 edgebox kernel: asterisk[10192]: segfault at 1e890118 ip b6885fcf sp b6862b90 error 4 in libisdnnet.so[b6875000+22000]`
`15:12.17`	`andrebarbosa`	`anonymouz666, yes I will`
`15:12.37`	`*** join/#asterisk ebroad (~EB@72.11.213.195)`
`15:12.37`	`andrebarbosa`	`but my question is, why with DONT_OPTIMIZE crash so many times then a regular build`
`15:12.51`	`andrebarbosa`	`I'm trying to figure out, what dont_optimize does.`
`15:12.57`	`*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)`
`15:14.24`	`*** join/#asterisk fofware (~chatzilla@186.125.41.133)`
`15:14.37`	`*** join/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk)`
`15:15.30`	`t_j`	`hi, i am trying to set up a IAX2 trunk between a server with a static IP and one with a dynamic one, but the static IP server never sees the trunk come up even after the client has registered`
`15:16.59`	`[TK]D-Fender`	`E-bola: vi app_queue.c`
`15:18.23`	`[TK]D-Fender`	`t_j: show us it registering, and failing to look like it right after`
`15:19.14`	`t_j`	`i am getting this right now: [Mar 17 10:18:57] NOTICE[2072]: chan_iax2.c:5800 register_verify: Host 192.168.176.42 failed MD5 authentication for 'tomj' (a67f11be5c57e3dc3d364191938fb1a3 != 5bca7916190edef478cecde97bc76b8e)`
`15:19.39`	`E-bola`	`[TK]D-Fender: i can just simulate a queue by using Dial(Local/queue1,999,m) and then dialing whoever underneath`
`15:20.01`	`[TK]D-Fender`	`t_that looks like a failed registration to me...`
`15:20.46`	`[TK]D-Fender`	`e-you can't interactively do anything while in dial like that`
`15:20.51`	`t_j`	`ok i was missing the host= on in the inbound settins of the client trunk`
`15:20.51`	`[TK]D-Fender`	`E-bola: everything escapes`
`15:21.42`	`E-bola`	`...`
`15:26.30`	`E-bola`	`Think ile make something so ppl can either have the weather forecast read to them or select what moph they wuld like`
`15:27.33`	`*** join/#asterisk norrec (~norrec@76-201-85-140.lightspeed.frokca.sbcglobal.net)`
`15:28.55`	`t_j`	`humm now it comes up but as a new outbound user :(`
`15:29.52`	`[TK]D-Fender`	`t_j: what makes a peer "outbound"?`
`15:30.04`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`15:30.27`	`t_j`	`[TK]D-Fender: am using freepbx web interface`
`15:30.36`	`[TK]D-Fender`	`t_j: then you're in the wrong channel`
`15:30.38`	`[TK]D-Fender`	`~freepbx`
`15:30.39`	`infobot`	`[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx`
`15:30.48`	`t_j`	`[TK]D-Fender: i am in there and its dead`
`15:32.00`	`*** join/#asterisk AlHafoudh (~AlHafoudh@2002:4e62:8b76:1234:223:dfff:fe8b:b4e6)`
`15:34.32`	`*** join/#asterisk eppigy (~eppigy@c-69-180-16-188.hsd1.ga.comcast.net)`
`15:34.34`	`eppigy`	`hello`
`15:34.37`	`eppigy`	`i am dave`
`15:39.58`	`Katty`	`hey stranger`
`15:40.03`	`Katty`	`i was wondering when you were gonna show up`
`15:41.05`	`E-bola`	`Can anybody explain precisely what the option T does to the queue command? Its described as: T allow the calling user to transfer the call.`
`15:41.16`	`E-bola`	`But how does the calling user transfer the call?`
`15:43.09`	`p3nguin`	`DTMF transfer as defined by features.conf`
`15:43.11`	`*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)`
`15:43.35`	`*** join/#asterisk darkdrgn2k (~darkdrgn2@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com)`
`15:43.48`	`darkdrgn2k`	`hi, can 2 sip clients share the same extension?`
`15:43.55`	`darkdrgn2k`	`IE: IPphone and Softclient`
`15:43.56`	`[TK]D-Fender`	`darkdrgn2k: Sure`
`15:43.57`	`E-bola`	`hmm ok, i wonder if i coudl somehow use that to let users waiting in queue to change moh`
`15:44.04`	`p3nguin`	`darkdrgn2k: yes, but they cannot share the same peer device name.`
`15:44.31`	`[TK]D-Fender`	`[11:43]<p3nguin>darkdrgn2k: yes, but they cannot share the same peer device name. <- they can .... and will fight ove who gets calls to that device`
`15:44.43`	`[TK]D-Fender`	`E-bola: Nope`
`15:44.45`	`p3nguin`	`darkdrgn2k: exten => 123,1,Dial(SIP/phone1&SIP/phone2)`
`15:45.51`	`p3nguin`	`darkdrgn2k: That makes both phones use extension 123.`
`15:45.54`	`*** part/#asterisk shader (~user@cpe-174-097-217-107.nc.res.rr.com)`
`15:46.10`	`Nugget`	`0`
`15:46.14`	`Nugget`	`erp`
`15:47.00`	`*** join/#asterisk QbY (~QbY@c-24-126-145-123.hsd1.ga.comcast.net)`
`15:47.27`	`darkdrgn2k`	`p3nguin: so basicaly im better of making a seperate extension then :-P`
`15:47.28`	`QbY`	`is there an upper limit to the number of registrations (extensions) an Asterisk box should make to an upstream provider?`
`15:47.39`	`p3nguin`	`darkdrgn2k: not really`
`15:48.04`	`p3nguin`	`qby: Extensions are limited by what you write into extensions.conf.`
`15:48.19`	`p3nguin`	`qby: Registrations are something completely different.`
`15:48.20`	`QbY`	`ok, in sip.conf i want to register nearly 100 lines`
`15:48.34`	`QbY`	`and i don't see a single one registering, nor do i see them in sip show registry`
`15:48.53`	`p3nguin`	`You have 100 different register statements?`
`15:49.02`	`Katty`	`HI NUGGET`
`15:49.03`	`[TK]D-Fender`	`QbY: Nor do we see your configs. Nor do we see your SIP debug.`
`15:49.14`	`*** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net)`
`15:49.28`	`QbY`	`hehehee`
`15:49.37`	`[TK]D-Fender`	`reaches for his ClueBat (tm)`
`15:49.43`	`QbY`	`[TK]D-Fender.. Bustin my balls.. I got it--come correct or don't come at all..`
`15:49.46`	`QbY`	`stand by`
`15:50.01`	`p3nguin`	`wonders if the Cluepon has expired yet`
`15:50.22`	`Katty`	`heh cluepon`
`15:51.24`	`Katty`	`that's cute.`
`15:51.34`	`[TK]D-Fender`	`~cluepon`
`15:51.34`	`infobot`	`[~cluepon] Get yours now! http://www.bored.com/photos/cluepon_4.html`
`15:51.47`	`*** join/#asterisk disposable (disposable@blackhole.sk)`
`15:52.59`	`p3nguin`	`100 register statements... I'm having a hard time believing that one.`
`15:53.07`	`*** join/#asterisk cusco (~trilili@2001:0:53aa:64c:24bd:1445:2ac0:762d)`
`15:53.08`	`cusco`	`hello!`
`15:53.29`	`Katty`	`hi`
`15:53.32`	`cusco`	`I am wondering.... we have several peers in realtime mysql configuration...`
`15:53.47`	`disposable`	`i'm trying to set up a provisioning server for sipura/linksys phones (https). i've generated the key and csr but the cisco guide doesn't say where i can get the key signed by cisco. anyone willing to share the knowledge?`
`15:53.49`	`Katty`	`yeah i'm wondering about lunch myself`
`15:54.06`	`*** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il)`
`15:54.07`	`*** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23)`
`15:55.19`	`benngard`	`lunch, how about dinner? ;)`
`15:55.30`	`Katty`	`mmmmmmno`
`15:55.33`	`Katty`	`i want lunch first.`
`15:55.39`	`p3nguin`	`Dinner could be lunch.`
`15:55.44`	`benngard`	`had lunch dor some hours ago`
`15:55.50`	`Katty`	`it could, but i wouldn't want to skip dinner`
`15:55.54`	`Katty`	`perhaps we could have dinner twice today`
`15:55.59`	`benngard`	`/s/dor/for`
`15:56.07`	`cusco`	`when we register them in the queue_member_table they have a certain penalty`
`15:56.08`	`p3nguin`	`Dinner is defined at the chief meal of the day -- it could be lunch or supper.`
`15:56.12`	`QbY`	`here is the syntax of my register => line and the output from the cli`
`15:56.12`	`QbY`	`http://pastebin.com/tEkUAege`
`15:56.24`	`p3nguin`	`benngard: Oh, so close!`
`15:56.38`	`benngard`	`:)`
`15:56.38`	`cusco`	`then when a call comes in, several peers with the same penalty ready to answer the same queue...`
`15:56.48`	`cusco`	`the call always goes to the same softphone`
`15:56.53`	`cusco`	`strategy is ringall`
`15:56.54`	`Katty`	`what about second breakfast?`
`15:56.59`	`*** join/#asterisk ickmund (~magnus@cli-5b7ee15c.bcn.adamo.es)`
`15:57.00`	`p3nguin`	`brunch?`
`15:57.04`	`Katty`	`no.`
`15:57.06`	`Katty`	`second breakfast`
`15:57.07`	`cusco`	`any hints on why this happens?`
`15:57.10`	`Naikrovek`	`i don't think he knows about second breakfast, pip`
`15:57.11`	`Katty`	`brunch is after second breakfast.`
`15:57.14`	`Katty`	`and then is elevenses`
`15:57.15`	`p3nguin`	`oh`
`15:57.20`	`*** join/#asterisk hluesea (~hulusikah@88.247.127.66)`
`15:57.21`	`[TK]D-Fender`	`QbY: Not enough to prove its valid, no CLI output from an attempt to load sip, no SIP debug for attempts, etc`
`15:57.35`	`QbY`	`ok`
`15:57.39`	`Katty`	`actually i think brunch might be elevenses`
`15:57.42`	`QbY`	`will gladly producet aht fro you`
`15:58.03`	`cusco`	`QbY: sip show registry`
`15:58.15`	`p3nguin`	`REMPBX01*CLI> sip show registry`
`15:58.15`	`QbY`	`that's in the paste`
`15:58.15`	`p3nguin`	`Host dnsmgr Username Refresh State Reg.Time`
`15:58.18`	`p3nguin`	`0 SIP registrations.`
`15:58.22`	`p3nguin`	`Should have read the pastebin!`
`15:58.44`	`cusco`	`:(`
`15:58.50`	`cusco`	`I was better off quiet`
`15:58.53`	`QbY`	`p3nguin.. and i even put it on the .com one :)`
`15:59.05`	`p3nguin`	`I know, it loaded right up for me!`
`15:59.20`	`cusco`	`QbY: sip reload`
`15:59.40`	`cusco`	`should show information about trying to register`
`15:59.48`	`p3nguin`	`I just do not understand the problem with the .ca site. It may take 10 minutes to load the page or it may never load at all, resulting in the browser displaying a timeout message.`
`16:00.15`	`[TK]D-Fender`	`p3nguin: Just for you`
`16:00.19`	`p3nguin`	`maybe`
`16:00.24`	`fifer`	`Anyone know the deal with xo and Asterisk? They have only certified Trixbox for use with their services, no other * flavor. That game is digium playing here?`
`16:00.26`	`QbY`	`[TK]D-Fender.. From this next paste, I'm sure you don't care to see all of the regular sip invites/acks do you`
`16:00.41`	`[TK]D-Fender`	`QbY: egt the rest`
`16:01.10`	`fifer`	`I mean switchvox`
`16:01.12`	`fifer`	`:-)`
`16:02.01`	`E-bola`	`cant believe he found something asterisk doesnt apear to be able to do`
`16:02.26`	`p3nguin`	`e-bola: What's that?`
`16:02.27`	`QbY`	`output from sip reload.. http://pastebin.com/WeMuDxBN -- sip debugging is on, but there are NO Register messages`
`16:02.49`	`E-bola`	`p3nguin: To let callers waiting in a queue make choices about things, for example to let them pick some new music on hold while waiting`
`16:02.59`	`[TK]D-Fender`	`fifer: It means XO is willing to actually help you configure Switchvox to work with them. They don't to train their techs on 50 GUI's jsut to satisfy every loser out there.`
`16:03.06`	`[TK]D-Fender`	`fifer: So they aimed "corporate"`
`16:04.49`	`p3nguin`	`e-bola: If you do, it will cause the caller to lose his place in line.`
`16:05.08`	`p3nguin`	`e-bola: Use extensions to kick the person out of the queue, Set(CHANNEL(musicclass)=whatever), then go back into the queue.`
`16:05.11`	`leifmadsen`	`E-bola: well, you could potentially do it if you are able to control the penalties and weights correctly to put the caller back into the correct place in line :)`
`16:05.38`	`*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)`
`16:05.46`	`p3nguin`	`If you could solve that issue, everything else should work just fine.`
`16:05.48`	`cusco`	`any hints on how to find out why asterisk always dials to the same peer with ringall strategy? Actually it rings on all, at the same time... all of them have auto answer, and alwyas the same one gets the call`
`16:05.52`	`E-bola`	`p3nguin: use extensions to kick him out of the queue how, he cant do anything when he's in the queue?`
`16:06.10`	`[TK]D-Fender`	`E-bola: What part of "no" are you failing to understand?`
`16:06.29`	`E-bola`	`leifmadsen: asuming i can find a way to pull him out of the queue, i had considered to just give him top priority/weigth when puttting him back in`
`16:06.39`	`p3nguin`	`e-bola: queues.conf: context`
`16:06.39`	`leifmadsen`	`cusco: means that phone is answering first every time`
`16:06.41`	`QbY`	`a sip peer does not do dns srv lookup does it...`
`16:06.48`	`cusco`	`:/`
`16:06.52`	`E-bola`	`[TK]D-Fender: Why do you insist on being negative, clearly its possible since 2 nice people just offered solutions`
`16:07.00`	`[TK]D-Fender`	`QbY: a SIP peer has nothing to do with your registrations`
`16:07.10`	`leifmadsen`	`E-bola: because he's a crabby old man`
`16:07.10`	`cusco`	`I could use leastrecent then...`
`16:07.22`	`beek`	`leifmadsen: he's not old.`
`16:07.23`	`cusco`	`would "leastrecent" strategy respect penaltys?`
`16:07.27`	`*** join/#asterisk voipmonk (~shido6@67.204.1.83)`
`16:07.28`	`leifmadsen`	`beek: I know, I've met him`
`16:07.35`	`leifmadsen`	`beek: he's much nicer in person`
`16:07.44`	`[TK]D-Fender`	`E-bola: Aren't real solutions... you aren't sitting in queue... your caller loses their place and other callers get answered, and they have to choose to go back to to the queue so forget "interactive"`
`16:07.57`	`[TK]D-Fender`	`E-bola: I'm negative because thats the outcome of this line of thought.`
`16:08.09`	`[TK]D-Fender`	`E-bola: You are trying to do things that * can't really do right now.`
`16:08.52`	`E-bola`	`[TK]D-Fender: Well try to consider a situation where there are rarely more than 1 person in a queue never any more than 3, and the waiting times are very low, in that case position isnt extremely important for us, and in that case the above options are valid solutions`
`16:08.56`	`[TK]D-Fender`	`E-bola: And trying to weasel around after we've already answered your questions. Makes us feel like we're wasting our time.`
`16:09.03`	`E-bola`	`So no need to just yell "its not possible" right away`
`16:09.16`	`[TK]D-Fender`	`E-bola: If wait time is so low... just let them sit with the MoH they've got`
`16:09.22`	`E-bola`	`[TK]D-Fender: Referring to yourself as we, is sort of childish :)`
`16:09.30`	`*** join/#asterisk Godfather_ (~Godfather@157.Red-88-11-88.dynamicIP.rima-tde.net)`
`16:09.41`	`*** join/#asterisk nickaugust (~anonymous@rrcs-71-42-53-182.se.biz.rr.com)`
`16:09.43`	`[TK]D-Fender`	`E-bola: The voices don't like your sarcastic tone! :p`
`16:09.53`	`Godfather_`	`o/`
`16:10.03`	`E-bola`	`hehe`
`16:10.27`	`cusco`	`16:06 < E-bola> leifmadsen: asuming i can find a way to pull him out of the queue, i had considered to just give him top priority/weigth when puttting him back in - we do that...`
`16:10.35`	`[TK]D-Fender`	`[12:04]<p3nguin>e-bola: If you do, it will cause the caller to lose his place in line.`
`16:10.40`	`[TK]D-Fender`	`E-bola: And it wasn't just us`
`16:11.00`	`E-bola`	`cusco: rofl thats sort of hilarious given [TK]D-Fender previous statements :)`
`16:11.00`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`16:11.06`	`*** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`16:11.24`	`E-bola`	`cusco: Just out of curiosity what do you use it for? I mean why do you pull the callers out?`
`16:11.31`	`cusco`	`the caller loses his place.. but we do that because we want him to pi picked up first`
`16:11.39`	`p3nguin`	`If the queued callers are few and the wait time is low, starting over might not be so bad.`
`16:11.40`	`cusco`	`so we don't care about his place in line`
`16:12.30`	`cusco`	`E-bola: by law, callers can only wait 1 minute here... we have to offer them an option to leave their contat and be called back in 48h, or ask him if he wants to keep wating`
`16:12.33`	`E-bola`	`Isnt there a way to log/store his current position when kicking him out of the queue?`
`16:12.38`	`[TK]D-Fender`	`cusco: If he's there so short... whats this fuss about changing MoH for?`
`16:12.43`	`[TK]D-Fender`	`E-bola: rather`
`16:12.44`	`*** join/#asterisk eppigy (~eppigy@c-69-180-16-188.hsd1.ga.comcast.net)`
`16:13.06`	`cusco`	`[TK]D-Fender: he's just being picky, I guess`
`16:13.12`	`[TK]D-Fender`	`[12:12]<cusco>E-bola: by law, callers can only wait 1 minute here... we have to offer them an option to leave their contat and be called back in 48h, or ask him if he wants to keep wating <- that's quite easy as-is`
`16:13.20`	`E-bola`	`[TK]D-Fender: Well as i started out by stating i was looking for fun/cool stuff to do, there isnt much point other than showing off asterisk functions to impress callers i guess`
`16:14.40`	`[TK]D-Fender`	`E-bola: this is a really sad area to target in an attempt to loko "impressive"`
`16:14.55`	`p3nguin`	`Create an extension in some arbitrary context which will set the moh class and send the caller back into the queue where he came from. Problem solved. Next!`
`16:15.00`	`E-bola`	`Thats just your oppinion`
`16:15.21`	`E-bola`	`p3nguin: that does solve it in the most basic form yep, but would be neat if u could register the callers position somehow`
`16:15.37`	`*** join/#asterisk slinksh0t_ (~slinksh0t@66.90.110.157)`
`16:15.45`	`*** join/#asterisk eppigy (~eppigy@c-69-180-16-188.hsd1.ga.comcast.net)`
`16:15.50`	`p3nguin`	`Based on leif's statement, it sounds like that can also be accomplished.`
`16:15.51`	`[TK]D-Fender`	`E-bola: Correct, and I announced it as a personal opinion. We are gramatically valid!`
`16:17.44`	`E-bola`	`leifmadsen or others interested: Where can i find information about what events/vars are set when a user dials a 1 digit extension to go to an extesion in the same context? Im hoping there's some way of registering his place in he queue upon leaving it`
`16:18.08`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`16:20.06`	`[TK]D-Fender`	`E-bola: core show application queue`
`16:20.11`	`[TK]D-Fender`	`E-bola: and the sample config`
`16:20.13`	`*** join/#asterisk paulc (~paulc@unaffiliated/paulc)`
`16:20.16`	`[TK]D-Fender`	`E-bola: anything set should be there`
`16:20.27`	`E-bola`	`Hmm wonder what ${QUEUESTATUS} is`
`16:20.40`	`p3nguin`	`Not what you need, I'm quite sure.`
`16:20.56`	`p3nguin`	`QUEUESTATUS The status of the call as a text string, one of TIMEOUT \| FULL \| JOINEMPTY \| LEAVEEMPTY \| JOINUNAVAIL \| LEAVEUNAVAIL`
`16:21.31`	`E-bola`	`ya but the onyl queue var that popped up`
`16:22.33`	`QbY`	`[TK]D-Fender.. I know a SIP Peer has nothing to do with Registraitons`
`16:24.06`	`p3nguin`	`e-bola: I would use my simple method and allow the caller to start over in the queue for now. Work on re-entry positioning later.`
`16:25.06`	`E-bola`	`p3nguin: ile probably end up doing that`
`16:25.25`	`E-bola`	`Just anoys me something as seemingly simple as interpreting user input isnt possible with queues and asterisk :(`
`16:25.26`	`*** join/#asterisk Brack10 (~tbrackett@66.35.2.50)`
`16:26.04`	`Brack10`	`Hey. Has asterisk GUI been abandoned? I can only find stuff that links to dead pages on asterisk.org`
`16:26.16`	`p3nguin`	`Yes, it has been.`
`16:26.57`	`Brack10`	`Gotcha. So the other user friendly way to to moves/adds/changes is freepbx?`
`16:27.02`	`Brack10`	`to do*`
`16:27.09`	`p3nguin`	`ssh + vim`
`16:28.00`	`p3nguin`	`Users shouldn't really be administering Asterisk.`
`16:29.27`	`*** join/#asterisk jameswf (~james@unaffiliated/jameswf-home)`
`16:29.34`	`[TK]D-Fender`	`Brack10: Or any of the other GUI's that are being maintained`
`16:30.03`	`Brack10`	`p3nguin: I'm thinking like help desk staff adding extensions, changing names etc in the absense of our telco guy`
`16:30.52`	`*** join/#asterisk slinksh0t (~slinksh0t@66.90.110.157)`
`16:32.26`	`p3nguin`	`I wouldn't want help desk staff messing with extensions.`
`16:35.03`	`E-bola`	`Hmm QEORIGINALPOS original position of the caller in the queue`
`16:35.21`	`E-bola`	`sounds about right, although its listed as being set when the caller is bridged the a queue member`
`16:35.42`	`*** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com)`
`16:36.11`	`*** join/#asterisk ariel_ (~chatzilla@63.214.236.169)`
`16:36.17`	`ariel_`	`hello everyone`
`16:36.18`	`[TK]D-Fender`	`E-bola: Because after they get answered dialplan does not progress`
`16:37.33`	`E-bola`	`[TK]D-Fender: Sure ofcourse is doesnt progress when they get answered, question is if its set if the caller dials out of the queue`
`16:38.14`	`[TK]D-Fender`	`E-bola: Guess there is no place it keeps that info for you then`
`16:39.11`	`E-bola`	`Its possible its not exposed as a userfriendly variable i can read, that would be a bit odd though, should be easy to implement since its used in the QEORIGINALPOS var as well as in the queue position announcements`
`16:43.01`	`p3nguin`	`Now I have to try it.`
`16:43.57`	`p3nguin`	`e-bola: Where did you discover that variable?`
`16:44.20`	`*** join/#asterisk joako (~joako@opensuse/member/joak0)`
`16:44.42`	`*** join/#asterisk dynamicpulse (~dynamicpu@adsl-99-172-50-102.dsl.emhril.sbcglobal.net)`
`16:44.45`	`E-bola`	`http://www.asterisk.org/docs`
`16:45.07`	`E-bola`	`i started reading there sometimes after lmadsen or whoever it was was bitching at me for only using voip-info.org hehe`
`16:45.15`	`E-bola`	`its under "Various application variables"`
`16:45.45`	`E-bola`	`ohh wait thats not it, i think i just found it on voip-info.org under the queues.conf`
`16:45.49`	`E-bola`	`its described in the sample configs`
`16:47.26`	`p3nguin`	`Yeah, but it is described as being set "just prior to the caller being bridged" with an agent.`
`16:47.41`	`p3nguin`	`So we must test.`
`16:47.56`	`E-bola`	`It wont work, im quite sure`
`16:50.29`	`*** part/#asterisk c0rnoTa (~c0rnoTa@178.176.217.179)`
`16:50.59`	`hardwire`	`http://www.phonelosers.org/issue/025/`
`16:51.02`	`E-bola`	`Unfortunately i dont have a new enough asterisk to test with here, and im heading home, but ile continue the hunt for a solution later`
`16:51.04`	`p3nguin`	`Does that exist in 1.4? My sample doesn't have it.`
`16:51.12`	`E-bola`	`p3nguin: pretty sure its 1.6 only`
`16:51.17`	`p3nguin`	`:(`
`16:51.41`	`E-bola`	`this bug which was related to it explains it pretty well https://issues.asterisk.org/view.php?id=12490`
`16:51.53`	`E-bola`	`i think it was created to be used from the macro u can have called when a caller is bridged`
`16:52.19`	`E-bola`	`I guess nobody expressed a need to have a general queue position var to a developer so far....`
`16:52.44`	`p3nguin`	`I can't imagine something doesn't exist.`
`16:53.17`	`p3nguin`	`app_queue is able to keep track of the caller's position while in the queue, and I suspect the information is stored in a variable.`
`16:53.35`	`E-bola`	`Not in one you can reference in the dialplan, im rpetty sure i would ahve found it by now`
`16:53.40`	`E-bola`	`or leifmadsen would have mentioned it`
`16:54.13`	`E-bola`	`But i'd love for you to proove me wrong :)`
`16:54.19`	`p3nguin`	`Well, he did leave.`
`16:54.42`	`E-bola`	`Ohh well have to be another day`
`16:54.48`	`E-bola`	`thanks for your input p3nguin`
`16:54.49`	`p3nguin`	`If he would have stuck around, he probably could have contributed to the subject.`
`17:01.46`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`17:02.24`	`ebroad`	`"res_fax_spandsp.c: WARNING T.30 ECM carrier not found`
`17:02.24`	`ebroad`	`" anything to worry about?`
`17:02.37`	`ebroad`	`faxes tx/rx fine`
`17:03.56`	`ebroad`	`nvm`
`17:03.57`	`Chainsaw`	`ebroad: If anything, that warning generally means things are going well.`
`17:04.15`	`Chainsaw`	`ebroad: (As in, if I see that, I normally have an e-mail with a PDF soon afterwards)`
`17:04.55`	`ebroad`	`Chainsaw, thanx, i just realized that ECM is turned off on the hard fax that I am testing with, which usually means that no ECM will get carried over to the t.38 stream`
`17:06.07`	`Chainsaw`	`ebroad: nod That makes sense :)`
`17:10.05`	`*** join/#asterisk boch (~boch@190.174.5.144)`
`17:17.11`	`*** join/#asterisk ManxPower-work (~manxpower@216.186.151.147)`
`17:19.24`	`*** join/#asterisk Acro (~omgponies@unaffiliated/acro)`
`17:20.39`	`asteriskATmarmuD`	`someone already copied/cloned the quancept CATI dialer?`
`17:22.00`	`jdoe`	`sawgood: look on the bright side, gentoo's asterisk was stuck at 1.2.something for forever...`
`17:22.28`	`jdoe`	`whoops, apparently that was like... a day ago.`
`17:23.36`	`citrus`	`anyone from california? looking for good sip gateway here`
`17:24.16`	`*** join/#asterisk Tech_Travis (~Travis@mail.techglia.com)`
`17:24.17`	`boch`	`7 #a2billing`
`17:25.03`	`*** join/#asterisk lanning (~lanning@208.87.235.224)`
`17:25.50`	`boch`	`do i have to use reinvite when using D() option of Dial() app to play dtmf ?`
`17:26.48`	`jdoe`	`citrus: uhhh... voip.ms has something in LA I think, can't speak to how good they are though.`
`17:27.30`	`p3nguin`	`I use VoIP.ms, but I don't use their POP that is in California.`
`17:27.55`	`ebroad`	`i believe viatalk has some servers out there`
`17:28.17`	`ebroad`	`i use them out of ny though`
`17:29.28`	`ebroad`	`http://www.vtwiki.org/wiki/index.php/Server_Choices`
`17:38.13`	`citrus`	`looks like that voip.ms one i will check out`
`17:38.17`	`citrus`	`thanks jdoe`
`17:41.32`	`*** join/#asterisk davevg-btwtech (~davevg__@75.97.64.33.res-cmts.senj.ptd.net)`
`17:43.35`	`jdoe`	`yep.`
`17:49.13`	`*** join/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net)`
`17:55.26`	`*** join/#asterisk Z_God (~julius@wlan237049.mobiel.utwente.nl)`
`17:58.12`	`*** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23)`
`17:58.42`	`*** join/#asterisk theHub (~theHub@69.177.93.21)`
`18:04.32`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`18:08.10`	`*** join/#asterisk Arsenick (~y@69.70.231.230)`
`18:09.44`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`18:11.27`	`boch`	`is it possible to inject dtmf during conversation ?`
`18:12.39`	`*** join/#asterisk s34n (~chatzilla@ip-208-76-93-125.mvdsl.com)`
`18:13.19`	`s34n`	`I just got handed a Polycom Soundpoint IP 430 with 3.2.3 Bootrom`
`18:13.47`	`s34n`	`But I can't find a manual or config files for it on the Polycom site`
`18:14.10`	`s34n`	`Polycom doesn't seem to acknowledge anything newer than 3.2.2`
`18:15.33`	`s34n`	`I do see references to 3.2.3 from people besides Polycom`
`18:16.03`	`[TK]D-Fender`	`s34n: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html`
`18:16.27`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`18:16.34`	`[TK]D-Fender`	`s34n: 3.2.3 is only via resellers. Feel free to wait for it an dprep based on 3.2.2.`
`18:16.52`	`*** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net)`
`18:18.10`	`s34n`	`hm. so I need to downgrade this phone to something that is documented.`
`18:18.51`	`p3nguin`	`Why not configure it based on 3.2.2 firmware?`
`18:19.24`	`Chainsaw`	`jdoe: Oh yes, totally outdated. That Asterisk 1.6.2.6 ebuild is just a myth.`
`18:20.00`	`[TK]D-Fender`	`s34n: Hold on.. thats BOOTROM. I just pointed you to the SIP APP`
`18:20.11`	`[TK]D-Fender`	`s34n: Why would you need to change the BootROM?`
`18:20.25`	`*** join/#asterisk Imo (~Imo@p4FC5CA3A.dip0.t-ipconnect.de)`
`18:20.36`	`Imo`	`hello, i have a problem with asterisk 1.6#`
`18:20.44`	`p3nguin`	`~ask`
`18:20.45`	`infobot`	`Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will.`
`18:20.48`	`Imo`	`when i get an incomming call i get this error`
`18:20.59`	`Imo`	`<PROTECTED>`
`18:21.04`	`Imo`	`what can i do ?`
`18:21.14`	`[TK]D-Fender`	`Imo: Fix your auth`
`18:21.43`	`s34n`	`[TK]D-Fender: I'm just trying to bring this up to current`
`18:21.50`	`Imo`	`hmm how ? sorry bevor i'm using asterisk 1.4 and i have the same config but didnt work work with 1.6`
`18:21.58`	`[TK]D-Fender`	`s34n: Never mess with the BootROM unless you have to`
`18:22.00`	`s34n`	`I see your distinction between SIP app and bootrom`
`18:22.18`	`[TK]D-Fender`	`s34n: Its an important distinction`
`18:22.22`	`s34n`	`yes`
`18:22.30`	`*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)`
`18:22.40`	`s34n`	`sip app is at 2.0`
`18:23.15`	`s34n`	`To upgrade, I just put it on my tftp server, and the phone knows to upgrade?`
`18:23.34`	`[TK]D-Fender`	`s34n: Should`
`18:24.31`	`*** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org)`
`18:25.01`	`keith4`	`I'm having some trouble with realtime SIP friends. the account in question shows up in 'sip show peers' and 'sip show friends', but can't make any calls`
`18:25.12`	`keith4`	`asterisk reports "chan_sip.c:14441 handle_request_invite: Call from '710' to extension '715' rejected because extension not found."`
`18:25.45`	`keith4`	`this is the SIP debug output: http://pastebin.com/cftLQK9p`
`18:26.14`	`s34n`	`[TK]D-Fender: How do I backup existing config files first?`
`18:26.23`	`[TK]D-Fender`	`keith4DIALPLAN error. the SIP is fine`
`18:26.33`	`[TK]D-Fender`	`s34n: You don't`
`18:26.33`	`keith4`	`seriously?`
`18:26.37`	`[TK]D-Fender`	`s34n: there is no export`
`18:26.41`	`keith4`	`facepalms`
`18:26.41`	`*** join/#asterisk gardo (~gardo@125.212.88.255)`
`18:26.48`	`[TK]D-Fender`	`keith4: EXTENSION <-----------`
`18:26.56`	`[TK]D-Fender`	`keith4: Seriously...`
`18:27.03`	`keith4`	`sonofabitch!`
`18:27.09`	`keith4`	`hugs [TK]D-Fender`
`18:27.29`	`p3nguin`	`Call from '710' to extension '715' rejected because extension not found.`
`18:28.16`	`*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)`
`18:30.35`	`*** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58)`
`18:30.53`	`Imo`	`[TK]D-Fender: please help me`
`18:31.01`	`Imo`	`what should i do ?`
`18:31.49`	`[TK]D-Fender`	`[14:21]<[TK]D-Fender>Imo: Fix your auth`
`18:31.58`	`Imo`	`how ?`
`18:32.09`	`[TK]D-Fender`	`Imo: vi sip.conf`
`18:32.19`	`Imo`	`??`
`18:32.25`	`Imo`	`there all right`
`18:32.43`	`p3nguin`	`Maybe you want [tk]d-fender to do it for you.`
`18:33.49`	`Imo`	`??? what should i fix in my sip.conf ?`
`18:34.20`	`Imo`	`when the passwords and usernames are all right, when i use asterisk 1.4 there are worked fine`
`18:35.50`	`*** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net)`
`18:36.37`	`*** join/#asterisk netpro25_ (~mmanning@64-238-176-105.ksg.apt.gru.net)`
`18:38.11`	`[TK]D-Fender`	`Imo: You aren't showing anything.`
`18:40.09`	`Imo`	`?? should i past my sip.conf ?`
`18:40.25`	`[TK]D-Fender`	`Imo: and a COMPLETE call attempt with SIP DEBUG`
`18:40.27`	`[TK]D-Fender`	`~~pb`
`18:40.36`	`Imo`	`ok mom`
`18:42.37`	`Imo`	`here is my sip.conf http://pastebin.com/nZWBVMSe`
`18:44.19`	`Imo`	`and here my debug http://pastebin.com/tCbw7VLb`
`18:44.56`	`Imo`	`and here my extensions.conf`
`18:44.56`	`Imo`	`[sip-in]`
`18:44.56`	`Imo`	`exten => _620,1,Dial(SIP/109,60,tT)`
`18:45.11`	`*** join/#asterisk waa (~waa@balrog.credipar.com.br)`
`18:45.32`	`*** part/#asterisk aceio (~c2cbd7fe@gateway/web/freenode/x-owgomiqhmbnoebws)`
`18:46.11`	`*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)`
`18:46.30`	`[TK]D-Fender`	`[Mar 17 14:43:28] WARNING[7892]: chan_sip.c:12230 check_auth: username mismatch, have <sipFritzBox>, digest has <620>`
`18:47.10`	`*** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net)`
`18:47.42`	`Imo`	`yes i know, but i dont know what should i do :-) ? with asterisk 1.4 are worked fine :D`
`18:48.07`	`*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)`
`18:48.23`	`Imo`	`and when i use xlite worked too`
`18:49.48`	`*** part/#asterisk Imo (~Imo@p4FC5CA3A.dip0.t-ipconnect.de)`
`18:49.59`	`*** join/#asterisk Imo (~Imo@p4FC5CA3A.dip0.t-ipconnect.de)`
`18:50.02`	`Imo`	`??`
`18:53.01`	`p3nguin`	`Why would you use _620 instead of just 620?`
`18:53.08`	`Imo`	`ahhh i have this`
`18:53.13`	`Imo`	`very good :-)`
`18:53.14`	`p3nguin`	`620 doesn't include anything that isn't exact.`
`18:53.19`	`Imo`	`i used insecure=very`
`18:53.24`	`Imo`	`i must use insecure=INVITE`
`18:53.27`	`Imo`	`now worked`
`18:55.12`	`Kobaz`	`insecure=very is depricated`
`18:55.18`	`Kobaz`	`you should use insecure=port,invite`
`18:55.49`	`Imo`	`ok thanks`
`18:57.23`	`p3nguin`	`But if you don't need port, just use invite.`
`18:57.33`	`p3nguin`	`Or if you don't need invite, just use port.`
`18:59.40`	`*** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr)`
`19:07.14`	`*** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au)`
`19:10.37`	`Katty`	`if i Dial(anumber,,D(dtmf)) and it works 100% of the time, but using a polycom dtmf fails about 15-20% of the time, what should be looking for first.`
`19:10.54`	`Katty`	`perhaps a sip debug?`
`19:14.57`	`*** join/#asterisk RypPn (~TuMbL@rosscom.co.uk)`
`19:20.29`	`*** join/#asterisk michaely (~michaely@CPE-124-190-32-229.sdcz1.lon.bigpond.net.au)`
`19:21.07`	`*** join/#asterisk Ad-Hoc (~nimbus@62.1.225.167.dsl.dyn.forthnet.gr)`
`19:21.13`	`Katty`	`hrmm doesn't look like output of debug info on console shows dtmf`
`19:22.07`	`michaely`	`Hello I have been trying to perform an automatic answer of the initial callback of a call to OriginateAction by setting the SIPADDHEADER variables when sending the Action. My variables in the Origination action are set like so:`
`19:22.07`	`michaely`	`Variable:__SIPADDHEADER01=Alert-Info: Ring Answer\|__SIPADDHEADER02=Call-Info: \;answer-after=0\|__SIPADDHEADER03=Alert-Info: \;info=alert-autoanswer`
`19:22.07`	`michaely`	`This works fine except that not only will the originator's extension auto answer but the destination extension will auto answer as well. I discovered that this was due to the use of the "__" perpended to the SIPADDHEADER key which caused the resulting channel to inherit the initiating channels variables during the call to Dial(). I removed the __s so that it looked like:`
`19:22.07`	`michaely`	`Variable:SIPADDHEADER01=Alert-Info: Ring Answer\|SIPADDHEADER02=Call-Info: \;answer-after=0\|SIPADDHEADER03=Alert-Info: \;info=alert-autoanswer`
`19:22.07`	`michaely`	`but now the auto answer fails completely on both ends and no SIP headers are sent to the device. Can someone tell me why the sip headers are not noticed when I do not have the "__" perpended to the SIPADDHEADER key. Note i have also tried with a single "_" and this caused both extensions to auto answer.`
`19:22.17`	`Katty`	`which reload applies changes to logger.conf`
`19:22.31`	`p3nguin`	`core verbose might, and logs do if you enable the dtmf level.`
`19:22.44`	`Katty`	`hmm`
`19:22.46`	`keith4`	`is there any sort of shortcut to send a sip-notify to multiple phones? or all of them? or do i have to run sip notify through a for loop?`
`19:22.46`	`Katty`	`k`
`19:22.51`	`p3nguin`	`logger reload`
`19:23.13`	`Katty`	`k`
`19:23.25`	`Ad-Hoc`	`hi`
`19:23.52`	`Katty`	`excellent`
`19:25.00`	`*** join/#asterisk zieruz (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net)`
`19:25.04`	`zieruz`	`hi there`
`19:25.10`	`zieruz`	`is there anyone here ?`
`19:25.13`	`Katty`	`no`
`19:25.27`	`benngard`	`no`
`19:25.30`	`michaely`	`no`
`19:25.50`	`p3nguin`	`Did you really want all 248 other people to respond?`
`19:25.51`	`zieruz`	`well i need how to configure video conferencing system`
`19:25.52`	`*** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net)`
`19:25.55`	`zieruz`	`on asterisk`
`19:26.14`	`*** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)`
`19:26.23`	`QbY`	`where is the best documentation for dahdi? i've got a span saying "Unconfigured"`
`19:26.40`	`Katty`	`okay.`
`19:26.47`	`zieruz`	`i got 3 device like http://www.latco.ma/Visioconference/alt_hdx7000_sm.jpg`
`19:26.57`	`Katty`	`i put in 7744 at my polycom.`
`19:27.06`	`Katty`	`the debug log shows 77`
`19:27.13`	`Katty`	`and of course the dtmf playback only says 77`
`19:27.19`	`zieruz`	`i want to able to connect this three device`
`19:27.25`	`zieruz`	`from 3 different location`
`19:27.29`	`Katty`	`so the polycom isn't transmitting this dtmf right`
`19:27.34`	`zieruz`	`using through asterisk`
`19:27.38`	`zieruz`	`is that posssible ?`
`19:27.39`	`p3nguin`	`katty: in-band dtmf or out-of-band?`
`19:27.40`	`[TK]D-Fender`	`[15:10]-->\|azetu (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) has joined #freepbx`
`19:27.43`	`Micc`	`Some of my customers with 8+ aastra phones have problems with the blf lights being on when no one is on the phone. Maybe this is an aastra firmware bug.`
`19:27.46`	`[TK]D-Fender`	`[15:24]-->\|zieruz (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) has joined #asterisk`
`19:27.55`	`[TK]D-Fender`	`zieruz: I already answered you on this`
`19:28.06`	`Katty`	`p3nguin: i'm not sure i actually understand your question, but this is outgoing dtmf`
`19:28.08`	`zieruz`	`i think you don't know about that`
`19:28.09`	`[TK]D-Fender`	`zieruz: * does NOT do "video conferencing`
`19:28.14`	`Katty`	`p3nguin: which probably isn't what you're asking me`
`19:28.21`	`zieruz`	`do you have this equipement at u ?`
`19:28.32`	`zieruz`	`do you test it b4 ?`
`19:28.47`	`[TK]D-Fender`	`zieruz: ASTERISK has no such functionality. It will no MIX`
`19:28.51`	`[TK]D-Fender`	`not*`
`19:28.58`	`p3nguin`	`katty: Both sip.conf and the phone's config have settings for what method dtmf is send from the phone. Can you check to see how they are configured?`
`19:28.59`	`[TK]D-Fender`	`zieruz: And I have a VSX 7000`
`19:29.16`	`[TK]D-Fender`	`zieruz: as well as having used videophones * softphones with video.`
`19:29.30`	`Katty`	`p3nguin: well i know sip.conf is setup to use default of rfc2833`
`19:29.42`	`Katty`	`p3nguin: do you have any thoughts on where i can check the method of dtmf from the polycom?`
`19:29.52`	`Katty`	`p3nguin: sip.cfg?`
`19:30.10`	`p3nguin`	`Not sure, I use Cisco phones. :/`
`19:30.10`	`zieruz`	`so do you any opensource engine do that functionality ?`
`19:30.16`	`Katty`	`k`
`19:30.23`	`Kobaz`	`zieruz: probably not anything any good`
`19:30.28`	`Kobaz`	`if at all`
`19:30.54`	`Kobaz`	`zieruz: sourceforge.net freshmeat.net.... search`
`19:31.14`	`Katty`	`digs through sip.cfg looking for any signs of dtmf`
`19:31.37`	`beek`	`Hello Katty`
`19:31.42`	`Katty`	`hi beek`
`19:32.13`	`benngard`	`its always "fun" when u type "who" and see " 2010-03-17 19:48 (hq.telecom-service.ru)" when u are swedish ;)`
`19:32.29`	`Katty`	`here's something interesting`
`19:32.35`	`benngard`	`the u know u have a guest!`
`19:32.51`	`Katty`	`tone.dtmf.rfc2833Control="1" and tone.dtmf.rfc2833Payload="101"`
`19:33.03`	`Katty`	`i would be willing to wager that means they're using rfc2833`
`19:33.10`	`p3nguin`	`benngard: I can't even understand what you're trying to say. Way too many letters not forming words there.`
`19:33.36`	`Kobaz`	`u a i o u i a u`
`19:33.43`	`Katty`	`l i b`
`19:33.45`	`Katty`	`m r ducks`
`19:33.49`	`zieruz`	`ok thx`
`19:34.07`	`Katty`	`c dem e d e d i's`
`19:34.08`	`p3nguin`	`katty: If you are interested in seeing different results, switch to inband dtmf for a while.`
`19:34.14`	`Kobaz`	`are people too lazy now to type three extra letters`
`19:34.20`	`Katty`	`p3nguin: well.`
`19:34.27`	`Kobaz`	`thanks is three more letters than 'thx'`
`19:34.30`	`Katty`	`p3nguin: asterisk sending dtmf inside the dial command is working just peachy`
`19:34.38`	`Katty`	`p3nguin: so that part is having no issue.`
`19:34.38`	`Kobaz`	`and makes it look like you're not an airhead`
`19:34.50`	`Katty`	`p3nguin: which also leads me to believe that because of that, there's not actually something wrong with the server`
`19:35.00`	`Katty`	`p3nguin: something specific with these polycoms and their configs`
`19:35.21`	`p3nguin`	`katty: No, it's the phone's dtmf that has a problem, hence changing to inband for a while to see different results.`
`19:35.23`	`Kobaz`	`srv brkn pls fix tks`
`19:36.02`	`Katty`	`p3nguin: define 'inband'`
`19:36.07`	`p3nguin`	`katty: I still cannot access Verizon's voicemail from my 7940, but it's perfectly fine on every other IVR I have ever encountered.`
`19:36.12`	`Katty`	`:<`
`19:36.14`	`Katty`	`bummer.`
`19:36.20`	`Kobaz`	`inband is dtmf via audio tones`
`19:36.24`	`Kobaz`	`rather than control packets`
`19:36.33`	`p3nguin`	`katty: dtmfmode=inband`
`19:36.40`	`Katty`	`if dtmf is encoded as sound...`
`19:36.51`	`Katty`	`well`
`19:36.54`	`Katty`	`it's worth a shot regardless`
`19:37.41`	`p3nguin`	`And I can access Verizon's voicemail from a 7912 on the same LAN... so I think some models are simply not friendly with dtmf.`
`19:38.41`	`p3nguin`	`Which is ridiculous, considering dtmf should be standard everywhere.`
`19:39.00`	`Katty`	`no doubt`
`19:40.35`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
`19:40.48`	`*** join/#asterisk philipp64 (~chatzilla@mail.redfish-solutions.com)`
`19:41.33`	`QbY`	`Digium is fucking stupid`
`19:41.52`	`Katty`	`watch your language please.`
`19:42.00`	`philipp64`	`howdy. was wondering if anyone had a link to the steps necessary to register an NANP number in the e164.arpa domain?`
`19:42.02`	`QbY`	`offer them money and they won't give you support without a serial number-- can't get a serial number because its in a remote location.. just great`
`19:42.04`	`QbY`	`wonderful`
`19:42.26`	`Katty`	`anyone know if polycoms have some sort of outgoing log that shows the dtmf digits it sent?`
`19:43.12`	`*** join/#asterisk andres833 (~andres833@190.144.75.22)`
`19:43.34`	`Katty`	`maybe this sip log debug will be useful`
`19:44.02`	`anonymouz666`	`Katty: what dtmf mode do you use?`
`19:44.26`	`Katty`	`well the server is using rfc2833`
`19:44.34`	`Katty`	`and sending dtmf in the dial command works like a charm`
`19:44.51`	`Katty`	`bug my debug claims my polycom only sent 77`
`19:44.57`	`Katty`	`when i really sent 7744 :<`
`19:46.28`	`Chainsaw`	`Katty: And it's not your digit map being annoying?`
`19:46.41`	`*** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net)`
`19:46.41`	`Katty`	`i did think of that`
`19:46.56`	`Chainsaw`	`Katty: I like to just dial from the main screen without hitting a line key so the digit map isn't applied.`
`19:47.08`	`Katty`	`oh i never hit a line key`
`19:47.23`	`Katty`	`these aren't registered with lines`
`19:47.29`	`Katty`	`they're just an extension.`
`19:47.38`	`Katty`	`they grab whatever free channel is open on the pri`
`19:47.45`	`Chainsaw`	`Ah, right.`
`19:48.02`	`Katty`	`polycom won't talk to me`
`19:48.05`	`Katty`	`:<`
`19:48.24`	`Chainsaw`	`Yeah, you buy that for the quality of the hardware & firmware, not the support.`
`19:48.35`	`Chainsaw`	`They're awful to everyone, I doubt it's personal.`
`19:48.52`	`Katty`	`yeah i know`
`19:49.12`	`Chainsaw`	`Does your SIP app not log it in the provisioning directory?`
`19:49.21`	`Chainsaw`	`(It seems to only write it when you reboot)`
`19:50.08`	`philipp64`	`ok. guess no one is using E.164.`
`19:51.07`	`Chainsaw`	`philipp64: I don't use that, no.`
`19:51.07`	`Katty`	`Chainsaw: well the polycom produces some boot logs`
`19:51.24`	`Katty`	`Chainsaw: i did see where to enable sip debug for logging purposes`
`19:51.45`	`Micc`	`is E.164 dundi?`
`19:53.18`	`*** join/#asterisk Deeewayne (~dwayne@adsl-070-148-053-233.sip.bhm.bellsouth.net)`
`19:53.18`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`19:53.48`	`Nugget`	`brand new, clean installation... ael or legacy extensions.conf? Seems every time I end up going the safe route.`
`19:53.59`	`Nugget`	`anyone want to talk me into trying ael2 finally?`
`19:54.28`	`*** join/#asterisk Ta^3 (~tacvbo@189.146.181.92)`
`19:54.42`	`*** join/#asterisk joako_ (~joako@opensuse/member/joak0)`
`19:55.23`	`Katty`	`not particularly`
`19:56.02`	`Chainsaw`	`uses "legacy" extensions.conf`
`19:56.13`	`philipp64`	`Micc: Dundhi can use e.164.`
`19:57.09`	`*** part/#asterisk rttrey (~trey@209.208.18.121)`
`20:01.26`	`[TK]D-Fender`	`Nugget: AEL can never offer you more than extensions.conf and guarantees less.`
`20:01.27`	`Qwell`	`Nugget: All the cool kids are using AEL`
`20:01.39`	`Qwell`	`[TK]D-Fender: except far cleaner syntax..`
`20:01.55`	`Katty`	`if i just one particular phone to use dtmfmode=inband`
`20:01.59`	`Katty`	`will that work?`
`20:02.03`	`Katty`	`or is it a global setting`
`20:02.06`	`[TK]D-Fender`	`Qwell: A clean looking pile of less is still less :)`
`20:04.05`	`Nugget`	`heh`
`20:05.07`	`benngard`	`Katty: u can pu dtmfmode per extension in sip.conf`
`20:05.12`	`benngard`	`put*`
`20:05.16`	`Katty`	`well i set inband as global and did a test`
`20:05.21`	`Katty`	`didn't help any`
`20:05.45`	`*** join/#asterisk mnick86 (~mnick86@whhem00002.cip.uni-regensburg.de)`
`20:06.05`	`*** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com)`
`20:07.27`	`Katty`	`infobot: phones`
`20:07.28`	`infobot`	`While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever.`
`20:07.40`	`*** join/#asterisk patrb (~asdf@64-150-178-3.kansascity.abac.net)`
`20:08.23`	`*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)`
`20:08.35`	`patrb`	`Does anyone have any experience with MWI/externnotify= with realtime voicemail config? If so, can I just add a new column to voicemail_users table?`
`20:09.09`	`*** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com)`
`20:09.34`	`*** join/#asterisk socain (~socain00@74.255.249.66)`
`20:09.47`	`p3nguin`	`benngard: Sorry, but there are no extensions in sip.conf.`
`20:09.58`	`p3nguin`	`katty: You can set dtmfmode per peer in sip.conf.`
`20:10.02`	`socain`	`is anyone here farmiliar with using Polycom elk keys?`
`20:10.39`	`*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)`
`20:11.19`	`socain`	`efk rather...`
`20:16.34`	`*** join/#asterisk norrec (~norrec@76-201-85-140.lightspeed.frokca.sbcglobal.net)`
`20:19.22`	`*** join/#asterisk rgsteele (~rgsteele@207.106.239.81)`
`20:22.17`	`socain`	`Anyone know how to send a speed dial contact button to an efk prompt that is triggered with a custom Polycom softkey for transfer to vmail, etc?`
`20:23.27`	`*** join/#asterisk Ad-Hoc (~nimbus@62.1.107.107.dsl.dyn.forthnet.gr)`
`20:24.10`	`Katty`	`that's interesting.`
`20:24.29`	`Katty`	`this aastra phone has the following dtmf options: RTP, SIP Info, or Both`
`20:24.35`	`Katty`	`is rtp rfc2833?`
`20:25.25`	`Kobaz`	`yeap that's rtp dtmf`
`20:27.41`	`Katty`	`k`
`20:30.01`	`[TK]D-Fender`	`checkout time, later all`
`20:30.14`	`*** join/#asterisk DotHack (~dothack@213.51.110.35)`
`20:33.19`	`Micc`	`So, infobot says no grandstream phones, I agree with that. But what about grandstream handytone 502? Seems like a pretty good little adapter.`
`20:37.10`	`Katty`	`infobot: phones`
`20:37.11`	`infobot`	`While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever.`
`20:38.03`	`Katty`	`what is the 5i series?`
`20:39.08`	`Katty`	`does that mean 5xi?`
`20:41.58`	`DotHack`	`anyone know a dummy sip provider where calles are just anwered and a demo is played for instance?`
`20:42.35`	`Chainsaw`	`DotHack: You want a test SIP address?`
`20:42.40`	`DotHack`	`and where is can also test dialins`
`20:43.10`	`DotHack`	`Chainsaw: yes i like to test sip dialout and dialin, im experimenting with asterisk`
`20:43.55`	`Katty`	`what is y'all think about the linksys spa931`
`20:44.06`	`Katty`	`telephonydepot is recommending it to me to test out`
`20:44.35`	`Chainsaw`	`I don't know, it scores quite low in the infobot list.`
`20:44.45`	`*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)`
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`21:04.40`	`*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)`
`21:06.29`	`*** part/#asterisk philipp64 (~chatzilla@mail.redfish-solutions.com)`
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`21:24.30`	`thehar`	`Anyone familiar with an NEC Electra Elite IPK II KTS in here by chance? :)`
`21:27.21`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`21:27.23`	`*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)`
`21:27.45`	`Kobaz`	`http://en.wikipedia.org/wiki/Lockheed_Model_10_Electra`
`21:27.50`	`Kobaz`	`there's that electra`
`21:28.06`	`thehar`	`heh`
`21:28.23`	`thehar`	`not familiar with the piece of crap`
`21:28.26`	`thehar`	`and need to get sip into it`
`21:30.35`	`*** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net)`
`21:34.14`	`timholum`	`hello, I just installed a new firewall and at the same time changed my internal ip range. When doing so my asterisk server can no longer register my sip account ( I have a sip account with voip.ms ) I have ports 10000 - 20000 ( udp and tcp ), 5060 ( udp and tcp ) , 4569 ( udp and tcp ) and 5036 ( udp and tcp ) all forwarded to my asterisk server. does anyone know what else I am missing`
`21:34.31`	`timholum`	`I can ping my voip providers ip it just does not register`
`21:34.44`	`[TK]D-Fender`	`~sipnat`
`21:34.45`	`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`
`21:34.46`	`[TK]D-Fender`	`^^^`
`21:42.07`	`*** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)`
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`21:45.36`	`timholum`	`I have double checked my setup, it is like that and the server was working behind a nat before. i am using pfsense as my firewall now i am not sure if that helps?`
`21:45.56`	`timholum`	`I have allowed just about everything I can think of and still nothing :(`
`21:48.42`	`Katty`	`timholum: call them.`
`21:48.48`	`Katty`	`timholum: they will tell you why it's not registering.`
`21:55.32`	`Katty`	`infobot: polycom admin guide`
`21:55.54`	`Katty`	`infobot: you're not as useful as google :<`
`21:55.55`	`infobot`	`no you're not as useful as google :<`
`21:56.02`	`Katty`	`well we all know that`
`21:56.42`	`[TK]D-Fender`	`Katty: pwned`
`22:05.51`	`*** join/#asterisk kbr (~kbr@ASte-Genev-Bois-154-1-111-194.w83-199.abo.wanadoo.fr)`
`22:06.55`	`Kobaz`	`makes the rounds and checks for crashed asteriskes`
`22:07.51`	`*** join/#asterisk manxpower (~ewieling@216.186.151.147)`
`22:08.08`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`22:08.10`	`*** join/#asterisk cguerrero (~cuauhtemo@201.140.187.122)`
`22:08.50`	`manxpower`	`~answers`
`22:08.51`	`infobot`	`answers is, like, 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`
`22:09.24`	`Jhirley`	`~42`
`22:09.24`	`infobot`	`i heard 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything`
`22:10.11`	`*** join/#asterisk andres833 (~andres833@190.144.75.22)`
`22:11.20`	`manxpower`	`http://yro.slashdot.org/story/10/03/17/2020246/Mississippi-Makes-Caller-ID-Spoofing-Illegal`
`22:12.03`	`*** join/#asterisk nightrid3r (kvirc@41.214.248.175)`
`22:14.51`	`manxpower`	`Seems mostly related to Telemarketing.`
`22:17.48`	`*** join/#asterisk mpe (~mpe@gate.ipvision.dk)`
`22:18.02`	`Qwell`	`I thought that was already illegal in quite a few places?`
`22:18.12`	`*** join/#asterisk hluesea (~hulusikah@88.247.127.66)`
`22:19.23`	`manxpower`	`Qwell: Fraud is illegal most places.`
`22:20.56`	`*** join/#asterisk snowboarder04 (~un@obi.bemail.co.uk)`
`22:21.26`	`*** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net)`
`22:22.24`	`citywok`	`i jsut had a massive explosion of asterisk in my call center, server hard locked for 2 minutes then came back... http://pastebin.com/ndnRiTu5`
`22:22.40`	`citywok`	`that's the dmesg info of the crash. looks to me like the dahdi timer caused the entire system to lock up?`
`22:23.31`	`*** part/#asterisk Tech_Travis (~Travis@mail.techglia.com)`
`22:26.01`	`manxpower`	`citywok: what version of Asterisk?`
`22:26.51`	`manxpower`	`citywok: and you might also try #asterisk-dev. Just follow the backtrace.txt file included with asterisk to file a bug report on issues.digium.com`
`22:27.14`	`manxpower`	`But unless you are running the latest version of the branch you are using chances are they will just laugh at you.`
`22:28.05`	`manxpower`	`Qwell: Telcos that don't allow Caller*ID Spoofing (Verizon SIP Wholesale, I'm taking to you) really annoy me.`
`22:28.41`	`manxpower`	`citywok: also you might try not loading every single dahdi kernel module under the sin.`
`22:28.59`	`manxpower`	`(yes, I know it's the default, but I still think that is a bad thing)`
`22:29.01`	`citywok`	`sorry max, 1.6.12`
`22:29.05`	`citywok`	`manx*`
`22:29.12`	`citywok`	`1.6.1.12, jeez long day`
`22:29.13`	`manxpower`	`citywok: you are wrong.`
`22:29.18`	`manxpower`	`there you go!`
`22:29.29`	`citywok`	`i have the fancy new build that comes out after .8 and .10`
`22:29.33`	`manxpower`	`looks like 1.6.1.18 is out.`
`22:30.00`	`manxpower`	`citywok: are you running any patches?`
`22:30.05`	`citywok`	`none`
`22:30.07`	`citywok`	`clean compiled`
`22:30.51`	`manxpower`	`[749750.734593] ioctl32(asterisk:13093): Unknown cmd fd(25) cmd(c0605402){t:'T';sz:96} arg(ffb81b3c) on /dev/dahdi/transcode Looks to me like you may have a dahdi/asterisk version mismatch or something similar.`
`22:33.52`	`citywok`	`i have had issues getting dahdi to work properly on this box, and plan to wipe it pretty soon because of them.`
`22:34.00`	`citywok`	`but this was a new kind of crash, lol`
`22:34.21`	`citywok`	`my standby server has 1.6.1.17, which i plan on putting in to production tonight because of this`
`22:37.47`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`22:38.45`	`*** join/#asterisk ruben23 (~ITadmin@122.55.48.243)`
`22:38.57`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`23:03.57`	`*** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)`
`23:04.07`	`ruben23`	`hi how do i compute jitter on my voip trunk..`
`23:09.53`	`*** join/#asterisk fofware (~chatzilla@186.125.41.133)`
`23:14.25`	`Micc`	`what causes an blf notification to be (queued) ?`
`23:15.33`	`Micc`	`It seems like aastra phones get (queued) a lot.`
`23:16.02`	`joako_`	`Does anyone know how I can use ${MEMBERINTERFACE} as part of the filename for recordings from a queue?`
`23:22.48`	`*** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)`
`23:24.21`	`*** part/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)`
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`23:40.04`	`idespinner`	`is it possible to store a dialplan command in a variable, then execute it? I.E. set(${mycmd}=dial/sip/123); exten = xx,2,${mycmd} ?`
`23:40.13`	`*** join/#asterisk fofware (~chatzilla@186.125.41.133)`
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