IRC log for #asterisk on 20090129

`00:00.14`	`rue_mohr`	`well you havn't heard of kb1 by name either :)`
`00:00.51`	`rue_mohr`	`which is kb1kanobe`
`00:01.17`	`rue_mohr`	`I dont hold it against you, ya see quite a lot of people in here`
`00:01.41`	`rue_mohr`	`I think kb1kanobe primarily spoke with Qwell`
`00:02.02`	`[TK]D-Fender`	`rue_mohr: thats a name I do recognize at least..`
`00:02.21`	`[TK]D-Fender`	`rue_mohr: if you really wanted help you'd have taken invites for others to look at your system...`
`00:02.38`	`rue_mohr`	`you mean irl or in channel?`
`00:02.54`	`rue_mohr`	`I can set up if you want to walk all over it`
`00:03.56`	`rue_mohr`	`I really dont think anyone would find anything I havn't`
`00:04.24`	`rue_mohr`	`I think they might erase it all and try rebuilding it from scratch`
`00:05.14`	`rue_mohr`	`I chose asterisk 1.4 cause I understood it would give me less trouble`
`00:05.28`	`rue_mohr`	`I suspect that 1.6 might have better odds of working with oslec`
`00:05.50`	`rue_mohr`	`but I also suspect it would have more odds of unexpectidly going belly up`
`00:05.59`	`[TK]D-Fender`	`rue_mohr>I really dont think anyone would find anything I havn't <- You know... given how long you beat yourself over the head on the simplest little speed-dial issues... you should have more faith in us :)`
`00:06.30`	`*** join/#asterisk xfighter (n=xfighter@a100-1.adsl.paltel.net)`
`00:06.39`	`rue_mohr`	`my issues are how to make a system that is emulating 4 seperate systems in one`
`00:06.39`	`xfighter`	`hello there`
`00:06.59`	`xfighter`	`hiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiii`
`00:07.19`	`[TK]D-Fender`	`rue_mohr: ...."Would you like fries with that, sir?"`
`00:07.45`	`xfighter`	`am newbie dude and am lookin for answers the internet won't help me answering it`
`00:07.52`	`rue_mohr`	`I didn't quite get just how simple my system at home is`
`00:07.53`	`[TK]D-Fender`	`rue_mohr: the "* will replace 4 completely different key systems in a way that'll blow your mid" sales-pitch will be the death of your business.`
`00:08.14`	`rue_mohr`	`no, 4 analog sets on each desk`
`00:08.18`	`[TK]D-Fender`	`xfighter: .... we ARE "the internet" :p`
`00:08.20`	`rue_mohr`	`thats what they have now`
`00:08.25`	`[TK]D-Fender`	`xfighter: so...`
`00:08.26`	`rue_mohr`	`4 analog sets and one voip set`
`00:08.27`	`[TK]D-Fender`	`~ask`
`00:08.28`	`jbot`	`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.`
`00:08.29`	`[TK]D-Fender`	`^^^`
`00:08.39`	`xfighter`	`:) thanx`
`00:09.13`	`xfighter`	`about asterisk , it needs the linux platform , what exactly must be installed also beside that OS??`
`00:09.39`	`rue_mohr`	`ftp`
`00:09.58`	`xfighter`	`? why?`
`00:10.07`	`rue_mohr`	`config download for voip sets`
`00:10.25`	`rue_mohr`	`html server for serving xml menus`
`00:10.46`	`*** join/#asterisk siera08 (n=sosoriri@218.207.141.90)`
`00:10.46`	`*** join/#asterisk RoyK (n=roy@ip-132-21-149-91.dialup.ice.no)`
`00:10.51`	`rue_mohr`	`ssh for maintenance`
`00:12.10`	`xfighter`	`hmmm , the story guyz is that am a 3rd year IT student and am planning for the graduation startin today I know about the ftp and the IIS and the databasing and I tend to be expert in the microsoft OS systems more than the linux sope I have no idea about voip , well not exactly I used to use an html softphone the thing is that ------>`
`00:12.17`	`xfighter`	`I need to get started`
`00:12.29`	`rue_mohr`	`got linux installed?`
`00:12.35`	`xfighter`	`what is the best ever refrence for that`
`00:12.43`	`xfighter`	`I got ubuntu live CD`
`00:12.44`	`rue_mohr`	`I suggest debian WITHOUT A GUI DESKTOP INSTALLED`
`00:13.00`	`xfighter`	`ubuntu sucks?`
`00:13.11`	`nix8n82`	`I really like OpenSuse`
`00:13.25`	`rue_mohr`	`the gui will eat up important resources`
`00:13.38`	`rue_mohr`	`xfighter, are you interfacing with phone lines?`
`00:13.38`	`*** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk)`
`00:14.01`	`xfighter`	`yea am planning to do all interfaces computer and phones`
`00:14.17`	`xfighter`	`like the (OCX idea)`
`00:14.23`	`beek`	`xfighter: Ubuntu is an excellent "windows-user" converting to "Linux-user" OS. Really sweet on the desktop.`
`00:14.51`	`xfighter`	`thanx beek :)`
`00:15.02`	`beek`	`xfighter: If you're an IT student and are expecting to learn Enterprise-level Linux then I'd suggest CentOS (a spin-off of RHEL) or OpenSuSE.`
`00:15.04`	`[TK]D-Fender`	`rue_mohr: not necessary`
`00:15.58`	`beek`	`xfighter: I use CentOS with my Asterisk box, as well as any Linux machine around the place not running RHEL. I use Ubuntu on my friends' and parents' desktops`
`00:16.33`	`xfighter`	`hmmm so you all sugguest that I dumb ubuntu,, I See , so what's the BEST : Centos? OpenSuSe ?`
`00:16.36`	`[TK]D-Fender`	`xfighter: * can run on any sane distro as long as you satisfy its dependencies`
`00:16.54`	`*** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au)`
`00:17.01`	`xfighter`	`-->>dependencies<<--`
`00:17.08`	`[TK]D-Fender`	`xfighter: Ubuntu can work fine... it also causes a little trouble and doesn't offer anything of value in compensation for it`
`00:17.17`	`beek`	`xfighter: [TK]D-Fender is correct, however since you're going into the IT biz then it's worth it to learn distros commonly used in business.`
`00:17.35`	`rue_mohr`	`xfighter, I'm also always in #garfield if you run out of help`
`00:18.03`	`xfighter`	`:) thanx rue_mohr it seems that I'll disturb you a lot :D`
`00:18.32`	`rue_mohr`	`ask these guys first`
`00:18.45`	`[TK]D-Fender`	`xfighter: Trust me... he was disturbed LONG before you came around ;)`
`00:18.56`	`xfighter`	`ok depending on your opinions folks am going for the centos option`
`00:19.08`	`xfighter`	`after that ?`
`00:19.17`	`[TK]D-Fender`	`xfighter: But seriously... Ubuntu is a GREAT desktop distro... There's just no way I'd run a server on it personally..`
`00:19.43`	`[TK]D-Fender`	`xfighter: where it wants to upgrade your kernel every other week, whic... BTW.. would BREAK Zaptel/DAHDI`
`00:20.06`	`xfighter`	`I've downloaded it because I felt it's the most updated linux OS I dunno maybe am mistaken`
`00:20.14`	`beek`	`xfighter: and if you're into some heavy masochism, try Gentoo.`
`00:20.38`	`xfighter`	`:S mercy beek am already confused enough :D`
`00:21.14`	`beek`	`xfighter: My recommendation is CentOS5. When you install it, turn off SELINUX. That's what I use and it's enterprise-level.`
`00:21.21`	`xfighter`	`it's ok am downloading centos now :)`
`00:21.39`	`xfighter`	`can I install it beside the XP?`
`00:22.01`	`beek`	`xfighter: With some prep work (e.g. making sure that there is freespace on your hard drive) then yes, you can.`
`00:22.08`	`xfighter`	`I mean 2 OS's at a time`
`00:22.17`	`[TK]D-Fender`	`xfighter: business lesson : the problem with bleeding-edge is the risk of exsanguination`
`00:22.19`	`xfighter`	`PERFECT`
`00:22.21`	`beek`	`xfighter: And by freespace I mean UNALLOCATED, UNPARTITIONED space.`
`00:22.44`	`xfighter`	`hmm I can use Acronis , right?`
`00:22.54`	`xfighter`	`Acronis Disk Director`
`00:23.49`	`beek`	`Never used it, but if your WinXP installation has hogged the entire drive the usual route is to do defrag, then reduce the size of the disk. There is plenty of info on the net on how to prepare a WinXP box for dual-booting with Linux`
`00:24.15`	`beek`	`xfighter: And please note: TURN OFF SELINUX on installation.`
`00:25.08`	`xfighter`	`thats great to know beek , and I'll take that note in notice :) I have 8 computers at home 5 notebooks and 3 destops no need to worry I'll back my files before doin anything ;)`
`00:26.05`	`xfighter`	`now lets go after installing the OS`
`00:26.22`	`xfighter`	`whats next`
`00:26.31`	`[TK]D-Fender`	`xfighter: Go read the BOOK.`
`00:26.33`	`[TK]D-Fender`	`~book`
`00:26.34`	`jbot`	`it has been said that book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook`
`00:26.57`	`[TK]D-Fender`	`xfighter: and go download the latest 1.4 release and its associated bits and go follow the instructions in the source tarball`
`00:27.22`	`beek`	`xfighter: And for CentOS specific info google Asterisk CentOS to find out what packages you'll need to include on installation.`
`00:27.52`	`xfighter`	`Am downloading 1.6 is that ok?`
`00:28.07`	`beek`	`That's bleeding-edge, but it's what I'm using with no problems.`
`00:28.39`	`xfighter`	`so 1.6 is still in beta?`
`00:28.51`	`[TK]D-Fender`	`xfighter: I'd suggest 1.4 for now so you can follow the book and WIKI more`
`00:28.56`	`beek`	`No, it's not beta. But it's young.`
`00:29.17`	`[TK]D-Fender`	`xfighter: Many things changed with 1.6 which aren't as well documented in one place for a beginner`
`00:29.19`	`xfighter`	`canceled... goin for the 1.4`
`00:30.25`	`xfighter`	`th funny thing is that am the first student in my university to do a project about VOIP , even my doctors know shit about it`
`00:30.30`	`xfighter`	`the*`
`00:30.43`	`*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)`
`00:31.29`	`xfighter`	`ok , now after downloading asterisk 1.4`
`00:32.10`	`xfighter`	`is there any documentation a full complete documentation about establishing the server and make it ready to go`
`00:33.19`	`rue_mohr`	`giz, standard editors DO NOT like vi commands`
`00:33.40`	`xfighter`	`???`
`00:33.46`	`jaytee`	`~book`
`00:33.46`	`jbot`	`i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook`
`00:33.55`	`rue_mohr`	`xfighter, so you have regular phone lines your going to hook into your asterisk system?`
`00:35.13`	`xfighter`	`yea am planning to use voip gateway of dirct ethernet telephones`
`00:35.38`	`xfighter`	`am even thinking about making a java small prog for N series (Nokia)`
`00:35.47`	`xfighter`	`it'll be a new idea :D`
`00:36.00`	`rue_mohr`	`so you are NOT planning on interfacing with telephone lines`
`00:37.30`	`xfighter`	`well for the university ,, nope but for me why not :D I think am gonna make the whole house Voiped , but am thinking about the internet link more than the phonelink`
`00:37.34`	`xfighter`	`(cheaper)`
`00:37.47`	`xfighter`	`in other words (sort of free)`
`00:38.15`	`xfighter`	`imagine when I link all my friends on a server at my place`
`00:38.55`	`rue_mohr`	`ok, as long as your not trying to interface with real phone lines your ok`
`00:39.10`	`[TK]D-Fender`	`rue_mohr: And if he did... what goes wrong then?`
`00:39.18`	`rue_mohr`	`everything goes to hell`
`00:39.27`	`*** join/#asterisk legis (n=wad@unaffiliated/legis)`
`00:39.37`	`*** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net)`
`00:39.49`	`rue_mohr`	`interface cards and drivers and echo and dahdi vs zaptel`
`00:39.49`	`xfighter`	`?? why rue_mohr`
`00:40.13`	`xfighter`	`thats a thing surely I know damn nthn about`
`00:40.19`	`xfighter`	`who are these people :D?`
`00:40.41`	`*** join/#asterisk lilkid (n=chatzill@87-194-38-230.bethere.co.uk)`
`00:40.47`	`xfighter`	`dahdi zaptel??`
`00:40.50`	`rue_mohr`	`around you are 10% experts, 30% lost souls and 70% idlers`
`00:41.01`	`rue_mohr`	`yea I know about how that ads up`
`00:42.06`	`xfighter`	`I guess I'll need to study the book on the asterisk website before I establish anything`
`00:42.21`	`[TK]D-Fender`	`xfighter: rue_mohr is just jaded here :p`
`00:42.22`	`rue_mohr`	`setting up a bunch of voip sets is easy`
`00:42.59`	`lilkid`	`having one way audio problems, i know its probably a NAT issue but i want to see if anyone can narrow it down. when I get incoming calls from another sip phone registered from WAN, incoming audio works fine but outgoing audio takes about 20seconds to kick in, and sometimes doesnt at all. any ideas?`
`00:43.09`	`[TK]D-Fender`	`rue_mohr: Oh... should I remind you about the weeks it took to get your phone running? or then presence afterwards?`
`00:43.10`	`rue_mohr`	`yea, nobody wants to help me anymore cause I got spun around a few times with everyone pointing different directions and everyone said I didn't do what they told me to, which conflicted`
`00:43.29`	`[TK]D-Fender`	`rue_mohr: No, I haven't given up.. but you appear to have sigh`
`00:43.44`	`rue_mohr`	`I was trying to not end up with the speeddial / prescence thing I have now`
`00:43.54`	`rue_mohr`	`I'm hurt, I'm a slow healer`
`00:43.56`	`xfighter`	`wow whos the real expert here :D`
`00:44.44`	`[TK]D-Fender`	`xfighter: Don't worry about anything. Go install * and here, go read this for a SAMPLE of what a simple setup could look like :`
`00:44.47`	`[TK]D-Fender`	`~jerjerguide`
`00:44.48`	`jbot`	`[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/`
`00:45.14`	`rue_mohr`	`[TK]D-Fender, is the expert, and Qwell, bmoraca is pretty smart`
`00:45.36`	`*** part/#asterisk satish2437 (n=root@122.167.115.41)`
`00:45.47`	`[TK]D-Fender`	`rue_mohr: and you were warned from the start that * does not support SLA and that the closest you'll get is a bastardized fake-out that pretends to be SLA`
`00:45.58`	`xfighter`	`one more last question guyz`
`00:46.23`	`xfighter`	`how many numbers can I register (create)`
`00:46.37`	`[TK]D-Fender`	`xfighter: in terms of?`
`00:46.40`	`xfighter`	`oh and there'll be another question after that :`
`00:46.41`	`xfighter`	`:D`
`00:46.52`	`xfighter`	`I mean how many telephone numbers`
`00:47.02`	`xfighter`	`(accounts)`
`00:47.03`	`eppigy`	`hello`
`00:47.05`	`eppigy`	`i am dave`
`00:47.06`	`[TK]D-Fender`	`xfighter: clarify that a bit will you...`
`00:47.17`	`rue_mohr`	`[TK]D-Fender, my later understanding was that the only thing I cant do is have mixed hold/transfer functionality, which i tried to point out I wasn't worried about. the only reason I had to give up is because the phones cannot support line keys that go 'busy' using their lamp`
`00:47.19`	`*** join/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-fdccc56c2bf76523)`
`00:47.41`	`xfighter`	`I mean my number is 001 my friends is 002 and so how many numbers can I create`
`00:47.51`	`rue_mohr`	`xfighter, in your closed system, you can make as many number as you want`
`00:48.13`	`[TK]D-Fender`	`xfighter: YOU could have sent back a "busy" if it was in use and they tried hitting it...`
`00:48.23`	`xfighter`	`cool , how many calls can my server process in the same time`
`00:48.26`	`xfighter`	`at*`
`00:48.37`	`rue_mohr`	`...`
`00:48.40`	`[TK]D-Fender`	`xfighter: If you're talking "extensions" for like "how many SIP devices can I set up", then the answer is "as many as you want"`
`00:49.31`	`[TK]D-Fender`	`rue_mohr: on hitting a speed-dial that was lit you have your dialplan check the lit status and jsut do "Busy()"`
`00:49.33`	`rue_mohr`	`[TK]D-Fender, having them manually dial hunt for a line is stupid :)`
`00:50.00`	`[TK]D-Fender`	`rue_mohr: Pressing s stupid "line" button to pick a line to go out is "stupid" :p`
`00:50.13`	`rue_mohr`	`its done, each line has an extension with a hint to the zaptel channel that gives back prescence info`
`00:50.15`	`xfighter`	`got that , I meant in my second question that if I call one of my friends and one of my friends calls another and so , how many connection will my server accept at the same time??`
`00:50.41`	`xfighter`	`connections*`
`00:50.47`	`rue_mohr`	`[TK]D-Fender, not when the person making the call has to make sure it goes out on the analog line that belongs to the company the person is making the call for`
`00:50.48`	`[TK]D-Fender`	`xfighter: as many as you have bandwidth to support`
`00:51.06`	`[TK]D-Fender`	`xfighter: www.ekiga.net <-`
`00:51.30`	`*** join/#asterisk nightrid3r (n=kvirc@78-20-228-200.access.telenet.be)`
`00:51.41`	`[TK]D-Fender`	`xfighter: have them sign up here and they can call each other more directly all they want. You can then set Asterisk up with an account instead of a soft-phone, and YOU are using *.`
`00:52.05`	`[TK]D-Fender`	`xfighter: then they can call you and if you want to let them do other stuff specifically then you can configure that in your dialplan`
`00:52.33`	`xfighter`	`genius :D`
`00:53.09`	`xfighter`	`but I'd like to learn it first then I'll go for these tricks :D`
`00:53.17`	`[TK]D-Fender`	`xfighter: Have them register to you if you want to control everything they dial... if you just want to chat with them... then have them go 'generic" using the Ekiga.net free service and you're just one more destinatin they can reach`
`00:53.21`	`rue_mohr`	`I'v explained it so many times, the 4 busineses and how there are 4 lines and answering and making calls and everything`
`00:54.00`	`[TK]D-Fender`	`rue_mohr: Yes, I know. I've known way back...`
`00:54.01`	`*** join/#asterisk Mavericks (n=chatzill@135.251.100.97.cfl.res.rr.com)`
`00:54.36`	`rue_mohr`	`and you keep teling me that people have to dial each line to find out if its busy or not so they can make their call`
`00:55.03`	`[TK]D-Fender`	`rue_mohr: its petter to have them press the speed dial/BLF that "represents" the line they want to call out. Configuring it for BLF = easy. Configuring it to report "busy" if they try using it while already in use = easy.`
`00:55.13`	`rue_mohr`	`the idea to make a predigit for each line is almost acceptable`
`00:55.26`	`[TK]D-Fender`	`rue_mohr: No need for any predigit.`
`00:55.41`	`[TK]D-Fender`	`rue_mohr: have them press the speed-dial in order to get out and you're already in control.`
`00:55.46`	`*** part/#asterisk Mavericks (n=chatzill@135.251.100.97.cfl.res.rr.com)`
`00:55.53`	`rue_mohr`	`thats what I do have`
`00:55.54`	`*** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis)`
`00:56.05`	`rue_mohr`	`you do know I have 3 speed dials with presence right?`
`00:56.13`	`[TK]D-Fender`	`rue_mohr: then you've done an incomplete job if you can't do the 2 things I've just described`
`00:56.17`	`[TK]D-Fender`	`rue_mohr: Yes.`
`00:56.39`	`rue_mohr`	`it cannot be done with the line keys on the aastra, their leds cannot do that`
`00:56.46`	`rue_mohr`	`I cant hear what your saying`
`00:56.50`	`xfighter`	`[TK]D-Fender : You're the man buddy , thanx for the gr8 info you provided me , as soon as I install everything I'll be back for you with more questions and I hope questions only not problems :) , thanx guyz and excuse me I g2g flee :)`
`00:57.29`	`rue_mohr`	`I'd swear I'v done what your saying and your still saying I'm doing it wrong`
`00:57.34`	`[TK]D-Fender`	`rue_mohr: their speed dials with BLF are just that... speed-dials. Press the "fake line key thats actualy a BLF speedi-dial", then you give them an ASTERISK DIALTONE. and let them dial.`
`00:57.43`	`[TK]D-Fender`	`xfighter: np`
`00:58.01`	`rue_mohr`	`the key with the lettering "LINE 1" stamped in it cannot be a speed dial with blf`
`00:58.03`	`[TK]D-Fender`	`rue_mohr: Where is * providing them tone in your setup?`
`00:58.11`	`beek`	`xfighter: Get ready to enjoy yourself... Asterisk is a boatload of fun.`
`00:58.13`	`[TK]D-Fender`	`rue_mohr: NOR SHOULD IT`
`00:58.17`	`rue_mohr`	`its not`
`00:58.23`	`[TK]D-Fender`	`rue_mohr: and it should not even be CALLED "line 1"`
`00:58.39`	`beek`	`That's so key-system`
`00:58.43`	`rue_mohr`	`the only time they get tone is when they press what I'm calling 'intercom' to dial someone else in the office`
`00:58.45`	`[TK]D-Fender`	`rue_mohr: it should be like "100" or whatever represents the popular "extension" they are known as.`
`00:58.51`	`[TK]D-Fender`	`beek: thats the point`
`00:59.05`	`rue_mohr`	`[TK]D-Fender, there is a key on the phone with LINE 1 stamped on it by the factory`
`00:59.06`	`[TK]D-Fender`	`rue_mohr: it IS like the "intercom" key on a Norstar`
`00:59.21`	`rue_mohr`	`there is an led beside it`
`00:59.31`	*** part/#asterisk `paul (n=kutimoy@119.93.45.181)
`00:59.41`	`rue_mohr`	`the key cannot be used as a speed dial, the lamp on it cannot be used as a blf`
`00:59.41`	`[TK]D-Fender`	`rue_mohr: FUCK the label. they know their presence BLF keys are what counts.`
`00:59.51`	`rue_mohr`	`I'v been back and forth over the manual`
`00:59.57`	`*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)`
`01:00.19`	`*** part/#asterisk xfighter (n=xfighter@a100-1.adsl.paltel.net)`
`01:00.20`	`rue_mohr`	`the only way to do prescence BLF with a speed dial is to use one of their 7 programmable keys`
`01:00.32`	`[TK]D-Fender`	`rue_mohr: Correct`
`01:00.35`	`eppigy`	`IM BOUT MY PAPER`
`01:00.36`	`[TK]D-Fender`	`(for Aastra)`
`01:00.37`	`rue_mohr`	`which do not include the 3 buttons stamped "LINE"`
`01:00.41`	`rue_mohr`	`yes`
`01:00.45`	`rue_mohr`	`do we agree?`
`01:00.58`	`[TK]D-Fender`	`rue_mohr: but you need to ween them off assuming those 3 are their "lines"`
`01:01.14`	`rue_mohr`	`I have no problemw ith that`
`01:01.19`	`[TK]D-Fender`	`rue_mohr: Just take some f-ing LIQUID PAPER and label over them (intercom1), etc`
`01:01.21`	`eppigy`	`i have to switch out the corporate office PBX this weekend`
`01:01.30`	`eppigy`	`abotu a month of hell is going to ensue`
`01:01.32`	`rue_mohr`	`I'm going to tell them their all intercoms`
`01:01.41`	`[TK]D-Fender`	`rue_mohr: there, 1/2 done`
`01:01.48`	`rue_mohr`	`its all done`
`01:02.00`	`rue_mohr`	`Ihave 3 of the 7 programmable keys set up as speed dial`
`01:02.12`	`rue_mohr`	`to the extensions for the zaptel channels`
`01:02.13`	`[TK]D-Fender`	`rue_mohr: next is to fix what your BLF/SD keys DO when you dial`
`01:02.23`	`rue_mohr`	`which have hits that the phones use to know if they are busy or not`
`01:02.52`	`eppigy`	`I have to somehow get the receptionists polycom to show the status of the executives' lines`
`01:03.12`	`rue_mohr`	`:) this is like you saying "press the brake pedel" and me saying "its already pinned to the floor"`
`01:03.24`	`rue_mohr`	`its leaving me confused`
`01:03.34`	`[TK]D-Fender`	`rue_mohr: Well, what do they actually DIAL?`
`01:04.13`	`rue_mohr`	`they hit the speed dial for the 'line' they want, which connects them to the zaptel channel, they get the co dialtone and carry on as normal`
`01:04.29`	`[TK]D-Fender`	`rue_mohr: show me :)`
`01:04.53`	`rue_mohr`	`tell ya what,I'm gonna post all the config files, ona website, so their easy to browse`
`01:05.06`	`rue_mohr`	`not this instant`
`01:05.07`	`[TK]D-Fender`	`whatever works`
`01:05.13`	`rue_mohr`	`becorfe I call on ya again`
`01:05.29`	`rue_mohr`	`cause I think were going around in circles on stuff I already done`
`01:06.15`	`*** join/#asterisk killown (n=Yamato@unaffiliated/killown)`
`01:07.02`	`bmoraca`	`rue_mohr: so you finally got your hints working?`
`01:07.09`	`rue_mohr`	`long ago`
`01:07.27`	`bmoraca`	`cool`
`01:07.35`	`bmoraca`	`now go buy a HWEC card and be done with it :D`
`01:07.46`	`rue_mohr`	`just sent the order today`
`01:07.52`	`bmoraca`	`right on`
`01:08.25`	`rue_mohr`	`I asked and they said that the time for me to play with oslec which I might never get working would be better spent on other stuff`
`01:08.49`	`bmoraca`	`hey, they grew brains...awesome`
`01:08.53`	`rue_mohr`	`I spent most of yesterday trying to get 1 more freaking wire into a 1" pipe to the tel room for the asterisk server`
`01:09.11`	`coppice`	`you might never get the card with HWEC working, either :-)`
`01:09.11`	`bmoraca`	`rue_mohr: lol`
`01:09.20`	`rue_mohr`	`after 3 experianced electricians gave up, I ran pipe on the outside of the building`
`01:09.53`	`rue_mohr`	`coppice, as I understand I just need to say the echo canceling is on, and its all automatic`
`01:10.35`	`coppice`	`there is a rumour that on board EC "just works". this rumour is far from accurate`
`01:10.52`	`[TK]D-Fender`	`rue_mohr: You could be cursed. Had your karma checked recently?`
`01:11.03`	`rue_mohr`	`[TK]D-Fender, ok, that was easy, here: http://eds.dyndns.org/~ircjunk/not_public_dont_open/phonesys/`
`01:11.08`	`bmoraca`	`i really hope Veritek doesn't sign with boston...i'd love to see their pitching staff fall on their faces.`
`01:11.30`	`rue_mohr`	`[TK]D-Fender, my bed is 5' off the ground, if my carma were off, I'd know it`
`01:12.01`	`rue_mohr`	`floor, my house isn't that bad`
`01:12.04`	`[TK]D-Fender`	`rue_mohr: Ok, pick one person for me to look at in this`
`01:12.26`	`[TK]D-Fender`	`rue_mohr: I took "Charles" under polycom`
`01:12.28`	`rue_mohr`	`charles`
`01:12.32`	`rue_mohr`	`hah`
`01:12.40`	`rue_mohr`	`he's 2nd owner`
`01:12.52`	`rue_mohr`	`I started with his phone and copied the others from it`
`01:13.00`	`rue_mohr`	`the aastra started with jan`
`01:13.32`	`[TK]D-Fender`	`rue_mohr: Ok, so far you've done most of it... if they hit a line thats in use they get rejected...`
`01:13.38`	`[TK]D-Fender`	`rue_mohr: What should it do instead?`
`01:13.57`	`[TK]D-Fender`	`rue_mohr: Infact... you give them RAW dialtone.`
`01:14.04`	`[TK]D-Fender`	`right our DAHDI`
`01:14.06`	`[TK]D-Fender`	`out*`
`01:14.43`	`rue_mohr`	`there are only a few tweeks needed, I need to fix the echo problem (card should be in the process of being sent) and I need to tune what happens when people just try to start dialng (which I expect is all phone side dialplan repari)`
`01:15.31`	`rue_mohr`	`yes, it gives them the dahdi right away, they can use the lamp to know if its busy before they hit it`
`01:16.12`	`beek`	`rue_mohr: While your at it, do yourself a favor and learn the templating format for your sip.conf file...`
`01:16.12`	`[TK]D-Fender`	`rue_mohr: Ok, so whats missing?`
`01:16.30`	`rue_mohr`	`beek, I tried to avoid modifying sip.conf at all, cause its replaced on upgrade, so the idea was to override only`
`01:16.56`	`rue_mohr`	`[TK]D-Fender, I think just phone side dialplan`
`01:16.59`	`beek`	`rue_mohr: then put everything into a separate file and just have an #include in the sip.conf file.`
`01:17.16`	`rue_mohr`	`beek, already done`
`01:17.29`	`rue_mohr`	`as I understand`
`01:17.31`	`[TK]D-Fender`	`rue_mohr: tip : change all of your 2X BLF extens to TEXT names, not "numbers".`
`01:17.50`	`[TK]D-Fender`	`rue_mohr: you could call it DAHDHI1 for example...`
`01:18.13`	`rue_mohr`	`I have a chart that maps everything out`
`01:18.42`	`rue_mohr`	`I was trying to make it in such a way I can use it for other systems with a few mods`
`01:19.04`	`rue_mohr`	`[TK]D-Fender, I hate to say this but I have to go home`
`01:19.21`	`[TK]D-Fender`	`alrighty`
`01:19.24`	`rue_mohr`	`there is someone standing at the door offering to lock me in or out for the night`
`01:19.33`	`rue_mohr`	`and my thermous is empty`
`01:20.11`	`rue_work`	`[TK]D-Fender, I'll be back :)`
`01:27.10`	`beek`	`GN all`
`01:36.05`	`*** join/#asterisk [gnubie] (n=[gnubie]@bb219-74-65-168.singnet.com.sg)`
`01:36.06`	`*** join/#asterisk Hymie (i=hymie@69.70.111.174)`
`01:36.19`	`*** join/#asterisk adam000 (n=adam@h198.213.18.98.dynamic.ip.windstream.net)`
`01:36.20`	`Hymie`	`http://www.voip-info.org/wiki/view/MBU-400`
`01:36.21`	`Hymie`	`bah!`
`01:36.46`	`[gnubie]`	`waves`
`01:37.24`	`[gnubie]`	`what are the advantages or disadvantages if tcp will be used over udp on using iax2 channel?`
`01:37.48`	`Hymie`	`points [TK]D-Fender to the url`
`01:38.52`	`[TK]D-Fender`	`Hymie: MBU-400? Never heard of it`
`01:39.16`	`[TK]D-Fender`	`[gnubie]: IAX does not support TCP`
`01:39.21`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`01:39.35`	`[gnubie]`	`[TK]D-Fender: ah, ok.. i thought it supports, too like sip`
`01:39.40`	`[gnubie]`	`[TK]D-Fender: thanks.. ;)`
`01:39.59`	`Hymie`	`[TK]D-Fender: it's new... just came out`
`01:40.13`	`Hymie`	`[TK]D-Fender: I can't believe how crappy aastra tech support was over this issue`
`01:40.17`	`[TK]D-Fender`	`[gnubie]: You seem to mistake "thought" with "randomly guess" a lot these days`
`01:40.42`	`[TK]D-Fender`	`Hymie: their 480i CT/57i CT DECT handset is pretty decent.`
`01:40.52`	`[TK]D-Fender`	`Hymie: just that the phone is not a real "base" for it`
`01:41.14`	`Hymie`	`[TK]D-Fender: I just wanted a bunch of dect phones with speaker phone on them, and this was the new boy on the block`
`01:41.20`	`*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.200)`
`01:41.36`	`[gnubie]`	`[TK]D-Fender: it's been months that i didn't asked here`
`01:41.44`	`[TK]D-Fender`	`Hymie: Looks similar the the scal of the Snom M3, etc`
`01:41.48`	`[TK]D-Fender`	`scale*`
`01:42.07`	`Hymie`	`[TK]D-Fender: it is definitely smaller sized.. like a cell phone`
`01:42.30`	`Hymie`	`[TK]D-Fender: I basically had to whisper to use the phone, althugh I do have a loud voice`
`01:42.45`	`Hymie`	`[TK]D-Fender: I'd say a normal person could never raise their voice for any reason, without significant distortion`
`01:43.16`	`[TK]D-Fender`	`Hymie: "craptastic" sounds like it sums it up.`
`01:43.26`	`[TK]D-Fender`	`Hymie: So serious gain issues. What else?`
`01:43.44`	`*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)`
`01:44.00`	`Hymie`	`[TK]D-Fender: I tried to place a piece of tape over the mic hole, but then the phone did not detect any background noise, and ramped up the amplification or some such.. so, every move of the phone against your face came through like sandpaper against a cat's ass`
`01:44.47`	`Hymie`	`[TK]D-Fender: and thge speaker phone was definitely quiet... I can't see using it in a room where there would be any other noise, like an open cubicle office environment or outside with traffic`
`01:44.49`	`[TK]D-Fender`	`Hymie: And I won't ask the original of your basis of comparison...`
`01:44.54`	`[TK]D-Fender`	`origin*`
`01:45.42`	`Hymie`	`[TK]D-Fender: I think these are all just firmware issues though, but I couldn't risk being stuck with a brick for 2 years, until they fix it`
`01:45.55`	`Hymie`	`[TK]D-Fender: and, of course, the speaker phone bit could just be underpowered, and unfixable`
`01:46.24`	`[TK]D-Fender`	`Hymie: Firmware has been a thorn with Aastra.. the 57i CT I have was flakey... would reboot on mass presence update, at random, etc....`
`01:46.33`	`Hymie`	`[TK]D-Fender: funny thing.. I hear that this same platform is going to be used by polycom too`
`01:46.42`	`[TK]D-Fender`	`Oh I'm sorry.. it would LOCK... I had to go about rebooting it.`
`01:46.54`	`[TK]D-Fender`	`Hymie: what "platform"?`
`01:46.56`	`Hymie`	`[TK]D-Fender: anything that defeats laziness is a bad thing`
`01:47.09`	`*** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178)`
`01:47.21`	`Hymie`	`[TK]D-Fender: I read an article on the MBU-400.. apparently Aastra just bought it from some other manufacturer and branded it...`
`01:47.31`	`Hymie`	`[TK]D-Fender: apparently polycom is going to do the same`
`01:48.03`	`[TK]D-Fender`	`Hymie: It might not be the bas at issue...`
`01:48.19`	`[TK]D-Fender`	`Hymie: and Polycom bout out SpectraLink which was always higer end.`
`01:48.27`	`[TK]D-Fender`	`Hymie: I hate the look of them however`
`01:48.31`	`Hymie`	`[TK]D-Fender: hmm`
`01:48.40`	`[TK]D-Fender`	`Hymie: I swear some looking like dildos...`
`01:49.04`	`[TK]D-Fender`	`You won't be able to just whip one out with a straight face`
`01:49.18`	`[TK]D-Fender`	`scores 500 points for the multiple double-entendres`
`01:49.44`	`Hymie`	`http://www.mgraves.org/voip/2008/11/first-look-from-afar-aastra-mbu-400-sipdect-system/`
`01:49.54`	`Hymie`	`[TK]D-Fender: hehe`
`01:50.19`	`Hymie`	`[TK]D-Fender: anyhow, I might buy the polycom based on the same, just to see how a nice firmware works with that hardware ;)`
`01:50.26`	`Hymie`	`when / if it appears`
`01:51.43`	`*** join/#asterisk bmoraca (n=bmoraca@adsl-75-12-126-173.dsl.skt2ca.sbcglobal.net)`
`01:52.18`	`Hymie`	`http://www.mgraves.org/voip/tag/ip200w/`
`01:52.19`	`Hymie`	`<PROTECTED>`
`01:52.26`	`Hymie`	`so, who knows`
`01:52.29`	`Hymie`	`grain of salt and all`
`01:52.32`	`coppice`	`everyone seems to have a DECT IP phone now. most are very basic but expensive.`
`01:53.17`	`bmoraca`	`the Snom M3 is a steaming pile of crap`
`01:53.26`	`bmoraca`	`i'd take an analog phone with a PAP2T over it any day`
`01:53.47`	`Hymie`	`heh, wow... the config screens I see in this article, are almost identical to the Aastra MBU-400 config screens`
`01:53.55`	`Hymie`	`basically, they're identical`
`01:54.06`	`Hymie`	`(that is, for the M3)`
`01:54.07`	`coppice`	`and very expensive`
`01:54.18`	`bmoraca`	`although i'd trust a polycom DECT IP phone over the Snom anyway`
`01:54.28`	`Hymie`	`coppice: I don't want to hear any more of that, it is your job to keep the eoncomy moving.. so spend!`
`01:54.35`	`coppice`	`$150-$200 for a $40 handset is a bit over the top`
`01:54.55`	`[TK]D-Fender`	`bmoraca: I've heard "iffy" about the M3... never anything so strong. Maybe they were too kind`
`01:55.12`	`bmoraca`	`i'd bet Polycom has the device at interop this year...i'll check it out then`
`01:55.22`	`Hymie`	`bmoraca: when's that?`
`01:55.24`	`bmoraca`	`[TK]D-Fender: take it from one who has used the M3...it's not a good phone.`
`01:55.25`	`siera08`	`anybody explain to me about AMPEXTENSIONS variable in amportal.conf?`
`01:56.07`	`[TK]D-Fender`	`siera08: You're in the wrong channell... try 3 to the left <-`
`01:56.08`	`bmoraca`	`[TK]D-Fender: sure, it's got a transfer button...but hitting *2 is just as easy...and analog phones aren't buggy as hell.`
`01:56.37`	`Hymie`	`bmoraca: well, most aren't buggy ;) that is, unless you buy some strange seimens phone`
`01:56.39`	`Hymie`	`:(`
`01:56.46`	`[TK]D-Fender`	`bmI've always gone ATA + cordless... which.. BTW... Uniden 500 series = uber crap (the cell-like flip phones)`
`01:57.27`	`siera08`	`[TK]D-Fender: what's the meaning?`
`01:57.39`	`bmoraca`	`[TK]D-Fender: the only Uniden phone's i've liked are their ruggedized ones. for general office, I use the Philips CD1500 phones...they have a MWI that works with Asterisk and a PAP2T`
`01:57.54`	`[TK]D-Fender`	`siera08: Go ask in #freepbx . it is not supported here`
`01:59.58`	`bmoraca`	`the only potentially cool thing about the M3 is that, sometimes, when the moon and planets are in a certain alignment and you're walking in a tub of cooking oil, you can roam from base station to base station with the handsets...usually doesn't work, though...and while a base station can "friend" up to 8 handsets, it can really only accomodate 3.`
`02:00.31`	`[TK]D-Fender`	`bmoraca: I tend to go for dual handset models because mine get abused. And by abused I mean brutally raped, Geneva-convention violated kinda things...`
`02:00.42`	`bmoraca`	`lol`
`02:01.17`	`[TK]D-Fender`	`bmoraca: And cheap enough that by the time they're done killing 2 handsets.. it still costs a ton less than 1 "ruggedized" model that would meet the undercarriage of a fork-lift based on karma alone.`
`02:01.18`	`frogonwheels`	`[TK]D-Fender: have you ever tried SMS through an ATA to a SMS -complient phone?`
`02:01.18`	`bmoraca`	`depends on the environment...engineering firm gets Philips...warehouse goons get the ruggedized Uniden...`
`02:01.30`	`frogonwheels`	`has anybody?`
`02:01.39`	`bmoraca`	`true that, [TK]D-Fender...`
`02:01.48`	`*** join/#asterisk MaliutaLap (n=biteme@203.171.192.150)`
`02:02.01`	`[TK]D-Fender`	`bmoraca: "Ruggedized" is just another word for "Destined to die a cruel and unusul death worthy of being in Final Destination 4..."`
`02:02.22`	`[TK]D-Fender`	`bmoraca: You know a funny things happened.... WHAMWHAMWHAMWHAMWHAMWHAMWHAMWHAM*`
`02:02.34`	`bmoraca`	`[TK]D-Fender: but i've found that people are willing to pay more for the illusion of a potentially better or longer lasting solution`
`02:02.51`	`[TK]D-Fender`	`bmoraca: I shoose what we buy, they get what I give :)`
`02:03.13`	`bmoraca`	`hence why the Snom M3 is ever sold...DECT IP phone must be better than analog+ATA, right?`
`02:03.51`	`bmoraca`	`[TK]D-Fender: yeah..if it was my company, that's how I'd do it...but then, i'm a salesman so i don't mind milking the customer.`
`02:04.11`	`[TK]D-Fender`	`bmoraca: Better still`
`02:04.44`	`[TK]D-Fender`	`bmoraca: Its sold on hope & hype... we keep saying its "better than WiFi"..`
`02:04.57`	`[TK]D-Fender`	`bm"shit looks pretty good.... when compared to CRAP`
`02:05.02`	`*** part/#asterisk Hymie (i=hymie@69.70.111.174)`
`02:06.15`	`lmadsen`	`I DON'T WANT TO MEET YOUR MOM!!!!! I JUST WANT!!!!!!!!!!!`
`02:06.40`	`bmoraca`	`[TK]D-Fender: well, i will give them that...the DECT IP phones ARE better than WiFi phones`
`02:06.52`	`[TK]D-Fender`	`lmadsen: ! ! !`
`02:09.21`	`lmadsen`	`http://www.albinoblacksheep.com/flash/bang`
`02:09.33`	`lmadsen`	`~!!!`
`02:09.34`	`jbot`	`rumour has it, !!! is BANG BANG BANG at http://www.starterupsteve.com/swf/Group_X_video.html`
`02:09.38`	`lmadsen`	`YES!`
`02:10.36`	`[TK]D-Fender`	`:D`
`02:12.14`	`Kobaz`	`yes yes`
`02:14.21`	`[gnubie]`	`waves.. gtg now.. thanks.. ;)`
`02:17.53`	`Micc`	`I've turned on CNAM lookups with my provider. Is there anything else I need to do for caller id name to be set when a call comes in? I'm using SIP btw.`
`02:20.30`	`[TK]D-Fender`	`Micc: No.`
`02:22.48`	`*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)`
`02:35.54`	`*** join/#asterisk keebler (n=keebler@h1.224.20.98.dynamic.ip.windstream.net)`
`02:36.39`	`keebler`	`can anyone tell me why asterisk is waiting 8 seconds before dialling the extension? What function in my dialplan sets the timer?`
`02:37.18`	`keebler`	`my only line in extension.conf is exten => _XXX,1,Dial(Sip/XXX${EXTEN})`
`02:37.23`	`frogonwheels`	`keebler: you using an ATA ?`
`02:37.32`	`frogonwheels`	`keebler: you sure it's not the timeout on your phone?`
`02:37.34`	`keebler`	`One ATA another SIP phone.`
`02:37.53`	`frogonwheels`	`.. as in you need to tell your ATA / SIP Phone when to send the numbers`
`02:38.05`	`keebler`	`Yeah...`
`02:38.17`	`keebler`	`Hmm, its using the defaut dialplan.`
`02:38.30`	`_ShrikE`	`does the 8 seconds occur before or after you see the dial appear in the cli?`
`02:39.05`	`keebler`	`brb... kids are awake`
`02:39.35`	`adr\|an`	`:)`
`02:42.56`	`*** join/#asterisk nix8n82 (n=nate@63.162.27.243)`
`02:44.21`	`bmoraca`	`keebler: you need to edit the dialplan of the ATA to add your particular three-digit structure to it, otherwise it defaults to like a 5 second timeout`
`02:44.25`	`Micc`	`Is it bad to reload extensions while you have a lot of calls, say like 50 calls going?`
`02:44.37`	`bmoraca`	`Micc: shouldn't affect it at all`
`02:45.10`	`Micc`	`bmoraca, trying to figure out how to make my dialplan dynamic without using realtime.`
`02:45.24`	`bmoraca`	`Micc: have fun with that, lol`
`02:45.29`	`keebler`	`bmoraca: Yeah. I'll look into the specific device dialplan structure, the Linksys SPA901 and ATA PAP2 both have the same dialplan.`
`02:45.36`	`_ShrikE`	`Micc: func_odbc`
`02:46.15`	`Micc`	`_ShrikE, does that just make a lookup once when the dialplan loads?`
`02:46.21`	`bmoraca`	`keebler: I'm not talking about which context they go to in asterisk, i'm talking about the specific dialplan in the phone/ata configuration...as in how it treats the characters load`
`02:46.23`	`bmoraca`	`Micc: no`
`02:46.29`	`keebler`	`_ShrikE: And to answer your question, its before it appears in the cli.`
`02:46.45`	`keebler`	`bmoraca: Thats what I was talking about too.`
`02:46.49`	`keebler`	`not my exten.conf`
`02:46.59`	`bmoraca`	`Micc: however, it will allow you to make dialplan changes in a database, but it will not dynamically reload them into asterisk. for tyhat, you have to issue a reload`
`02:47.06`	`bmoraca`	`keebler: gotcha. that needs adjusting.`
`02:47.14`	`keebler`	`on the LINE 1 tab, at the bottom. Both have the same dial plan.`
`02:47.16`	`Micc`	`bmoraca, yeah thats what I mean.`
`02:47.32`	`keebler`	`bmoraca: (9,[3469]11S0\|9,<:1408>[2-9]xxxxxx\|9,<:1>[2-9]xxxxxxxxxS0\|9,1[2-9]xxxxxxxxxS0\|9,011xx.\|9,xx.\|[1-8]xxx)`
`02:47.35`	`keebler`	`haha`
`02:47.46`	`bmoraca`	`keebler: what are your extensions?`
`02:47.57`	`keebler`	`100/200/300`
`02:48.02`	`_ShrikE`	`Micc: pretty much, it allow you to fill in the blanks in a dialplan with database queries.`
`02:48.04`	`keebler`	`but I just have it set up as wildcard`
`02:48.06`	`keebler`	`XXX`
`02:48.21`	`keebler`	`erm`
`02:48.26`	`bmoraca`	`keebler: try to keep them in the same range...101, 102, 103...it'll simplify things`
`02:48.30`	`[TK]D-Fender`	`Micc: reloading extensions has no impact on calls in progress`
`02:48.39`	`keebler`	`bmoraca: They will be on the producion.`
`02:48.41`	`keebler`	`tion`
`02:48.41`	`frogonwheels`	`keebler: I use 101 102 103.. 1001 1002 1003`
`02:48.47`	`keebler`	`....`
`02:48.52`	`keebler`	`As do I.`
`02:49.37`	`frogonwheels`	`keebler: Hre's mine (for Australia) (*xx\|0\|00\|000S0\|[2-9]xxxxxxxS0\|0[2378][2-9]xxxxxxxS0\|04[0-9]xxxxxxxS0\|1[358]00xxxxxxS0\|13[1-9]xxxS0\|10[1-9]S0\|100[1-9]S0\|70[0-9]S0\|xxxxxxxxxxxx.)`
`02:49.37`	`eppigy`	`I USE FIBONACCI PRIMES`
`02:49.39`	`keebler`	`Its how our panasonic PBXs have been setup since I got into PBXs`
`02:49.40`	`eppigy`	`WO`
`02:49.42`	`eppigy`	`WOW`
`02:50.02`	`bmoraca`	`keebler: you'll need to add a [1-9]xx\|9 in there. i'm not familiar with the sipura style dialplan, but it'd appear to me that you need (9,[3469]11S0\|9,[1-9]xx\|9,<:1408>[2-9]xxxxxx\|9,<:1>[2-9]xxxxxxxxxS0\|9,1[2-9]xxxxxxxxxS0\|9,011xx.\|9,xx.\|[1-8]xxx)`
`02:50.24`	`frogonwheels`	`keebler: 10[1-9]S0\|100[1-9]S0`
`02:50.37`	`bmoraca`	`frogonwheels: that looks like a PAP2T...he's using a PAP2...different config`
`02:50.38`	`keebler`	`I need to look on how to structure it because I have no freakin' clue what it all means.`
`02:50.45`	`keebler`	`acutally`
`02:50.46`	`keebler`	`sorry`
`02:50.51`	`keebler`	`I should have specified`
`02:50.54`	`keebler`	`PAP2T-NA`
`02:51.05`	`thehar`	`question ..`
`02:51.11`	`thehar`	`how would all of you handle overflow calls?`
`02:51.19`	`bmoraca`	`keebler: that dialplan doesn't look like any i've seen on a pap2t...should look more like what frogonwheels pasted...`
`02:51.20`	`frogonwheels`	`bmoraca: Thought they were similar except pap2 has option to send out the FXO port`
`02:51.34`	`thehar`	`say all 4 pris fill concurrently.. you have another pbx that has 3 pri avail on it.. you want to overflow into it.. DUNDi?`
`02:51.38`	`bmoraca`	`frogonwheels: one's sipura based and the other's newer...or something like that.`
`02:51.58`	`bmoraca`	`frogonwheels: not sure on the specifics`
`02:52.00`	`frogonwheels`	`bmoraca: so you are using 9 as your external extension?`
`02:52.02`	`keebler`	`Well my PAP2T-NA has the most current firmware.`
`02:52.15`	`[TK]D-Fender`	`thehar: SIP`
`02:52.15`	`keebler`	`You know, I don't even NEED an external extension.`
`02:52.25`	`thehar`	`just SIP?`
`02:52.34`	`frogonwheels`	`keebler: nope.`
`02:52.35`	`bmoraca`	`keebler: then just replace it all with [1-9]xx`
`02:52.39`	`thehar`	`could you do it with DUNDi?`
`02:52.44`	`thehar`	`or use SIP or IAX2`
`02:52.47`	`keebler`	`bmoraca: Thats it?`
`02:52.56`	`[TK]D-Fender`	`TheWho gives a grap about search for a route? You KNOW where to send it.`
`02:52.58`	`keebler`	`bmoraca: just dialplan: [1-9]xx ?`
`02:53.05`	`frogonwheels`	`keebler: I wouldn't`
`02:53.11`	`frogonwheels`	`keebler: you'll get your delay problem`
`02:53.24`	`keebler`	`Thats the one thing I need to eliminate.`
`02:53.26`	`frogonwheels`	`keebler: if you don't want the delay, you have to specify all the (common) possibilities`
`02:53.37`	`keebler`	`fiddle sticks`
`02:54.05`	`frogonwheels`	`keebler: ok - so how does it know when to stop?`
`02:54.32`	`*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)`
`02:54.32`	`*** mode/#asterisk [+o russellb] by ChanServ`
`02:54.40`	`keebler`	`what know when to stop? exten= => XXX,1,hangup() ?`
`02:54.53`	`frogonwheels`	`keebler: no.. the pap2t , not asterisk`
`02:54.53`	`keebler`	`erm xxx,2,hangup`
`02:54.57`	`bmoraca`	`frogonwheels: he is ONLY doing 3-digit dial to internal phones. he's not doing any external calls of any kind. he doesn't need provision for anything else`
`02:54.58`	`keebler`	`Oh`
`02:55.04`	`keebler`	`Yeah.`
`02:55.15`	`frogonwheels`	`oh. sorry. . well [1-9]xxS0`
`02:55.47`	`frogonwheels`	`you need the S0 to tell it to 'send as soon as you match that'`
`02:56.05`	`[TK]D-Fender`	`(x.T\|#x.T\|*x.T) <-- the only dialplan you'll ever need`
`02:57.32`	`keebler`	`Thanks frogonwheels and bmoraca, that worked great`
`02:58.00`	`keebler`	`I wish I weren't in such a hurry. :/ I really want to learn all this the right way.`
`03:00.58`	`eppigy`	`smoke purp by the pound`
`03:03.42`	`russellb`	`eppigy: wtf?`
`03:03.45`	`russellb`	`you say that all the time`
`03:03.49`	`russellb`	`just ... stop`
`03:03.59`	`*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu)`
`03:05.01`	`eppigy`	`=(`
`03:06.06`	`eppigy`	`i admire you russellb, which makes your admonition that much more hurtful`
`03:06.18`	`Micc`	`How can I create inbound calling limits?`
`03:06.26`	`keebler`	`Next question. What do I need to do to get my SPA9000 "PBX" to just act like a dummy ATA? I can set it just like my ATA, and it registers, but I can't, for some reason, get a dialtone.`
`03:06.33`	`Micc`	`So it would be busy when the 3rd person called lets say.`
`03:07.02`	`Micc`	`I have a virtual pri from vitelity, 4 channels but I want some did's to be busy after 2 calls.`
`03:07.54`	`bmoraca`	`Micc: this is why i route all my inbound/outbound through a different box than my PBXes are on`
`03:07.56`	`RoyK`	`Micc: show application busy`
`03:08.19`	`bmoraca`	`keebler: no idea...good luck :)`
`03:08.37`	`keebler`	`bmoraca: haha. I just want to keep it around for a spare ATA is all.`
`03:08.42`	`keebler`	`bmoraca: I won't use it in the field.`
`03:08.47`	`Micc`	`RoyK, thats great but how do I know what the current call count is? Maybe I can use variables to store the current call count.`
`03:08.50`	`RoyK`	`the asterisk dialplan isn't really well suited for scripting - better use agi with some language`
`03:08.58`	`keebler`	`I've got two more ATA's and some more hardware coming in tomorrow.`
`03:09.11`	`Micc`	`bmoraca, SER?`
`03:09.20`	`RoyK`	`Micc: there is a call counter thingie in asterisk, but it's buggy`
`03:09.31`	`Micc`	`RoyK, even in 1.6?`
`03:09.50`	`RoyK`	`haven't tried it with 1.6`
`03:09.51`	`Micc`	`bmoraca, what do you use for your first box or sbc?`
`03:10.08`	`Micc`	`I'm running 1.4.22 now anyways.`
`03:10.14`	`bmoraca`	`Micc: no, it's Asterisk...but a sip trunk can have call limits placed on it...so even though I may have 150 available inbound channels, i can restrict the SIP trunk between the gateway and the virtual PBX to 3 calls or whatever`
`03:10.27`	`[TK]D-Fender`	`russellb: He says all sorts of things all the time. Its Internet Tourette's ;)`
`03:10.40`	`Micc`	`bmoraca, Ah! thats a great idea.`
`03:11.33`	`edoceo`	`I'm trying to build 1.4.23 and seeing this error message: chan_dahdi.c:1029: error: 'DAHDI_TONE_DTMF_BASE' undeclared (first use in this function)`
`03:11.50`	`edoceo`	`And it won't compile :( Do I need dahdi libs somewhere?`
`03:12.06`	`bmoraca`	`Micc: if we get enough capacity, i'll probably convert it to a Cisco 2800 w/ call manager or something. i'm not aware of any DS3 interface cards for asterisk, lol`
`03:13.10`	`bmoraca`	`Micc: though Sangoma does make an 8port PRI card...so i'm good up to 24 PRIs. with FAS, that's a lot of channels. at that point, though, a DS3 is cheaper according to my provider`
`03:14.22`	`bmoraca`	`i just want to avoid having multiple trunk groups`
`03:15.07`	`bmoraca`	`well, back to the real world`
`03:15.53`	`edoceo`	`even if I configure using --without-dahdi it still fails :(`
`03:16.59`	`Qwell`	`edoceo: You get the compile error still when you use --without-dahdi?`
`03:17.38`	`Qwell`	`edoceo: also, what version of dahdi (or zaptel?) do you have installed?`
`03:18.18`	`frogonwheels`	`edoceo: you could move the chan_dahdi.c out of the directory - I think it builds all apps / channels etc in those directories.`
`03:20.06`	`edoceo`	`Turns out I had to deselect some stuffs with make menuselect`
`03:20.19`	`edoceo`	`My ebuild was not automatically configuring that part right`
`03:20.41`	`edoceo`	`Now I'm moving forward....horray!`
`03:20.59`	`*** join/#asterisk MaliutaLap (n=biteme@203.171.192.230)`
`03:21.01`	`Qwell`	`oh, ebuild. that explains a lot`
`03:21.18`	`edoceo`	`Yea - and one I wrote myself even....`
`03:21.37`	`edoceo`	`sips more wiskey and has another smoke`
`03:27.09`	`*** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv)`
`03:28.29`	`*** join/#asterisk trigeek38 (n=chatzill@ip72-196-100-30.ga.at.cox.net)`
`03:28.40`	`[TK]D-Fender`	`edoceo: Put. Down. The. Crack. Pipe.`
`03:29.09`	`Qwell`	`notifies JerJer about the blatant trademark infringement`
`03:29.23`	`[TK]D-Fender`	`(c) JerJer`
`03:29.29`	`edoceo`	`[TK]D-Fender: Hahaha`
`03:29.52`	`[TK]D-Fender`	`Qwell: I have an unlimited redistribution licence.`
`03:30.24`	`edoceo`	`After ./configure do I have to use make menuselect? make all seems to fail cause dahdi application is still selected even through I said --without-dahdi`
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`03:38.24`	`*** part/#asterisk RoyK (n=roy@ip-132-21-149-91.dialup.ice.no)`
`03:44.59`	`*** join/#asterisk joobie (n=joobie@mx01.anric.com.au)`
`03:45.37`	`joobie`	`Hi guys,.. im trying to buy a top quality phone.. that supports POE and has only basic functionality.. but excellent call quality.. ive been looking at the polycom 320.. is there any others i should look at or this is probably the best?`
`03:47.13`	`[TK]D-Fender`	`joobie: Linksys is a far better $ value in AU IIRC.`
`03:47.48`	`*** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net)`
`03:47.48`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`03:47.54`	`[TK]D-Fender`	`joobie: Polycom is nicer quality, but I'm not sure I would compare its value at the local premium`
`03:50.14`	`Khratos`	`... going to sleep, see u later.`
`03:51.12`	`*** part/#asterisk Khratos (n=Khratos@190.166.129.54)`
`03:51.14`	`keebler`	`This is so frustrating. Its a waste of time I know, but I can't get regular line one working with the SPA9000. I just want to bypass the PBX function.`
`03:51.49`	`keebler`	`The Ext page is almost identical to a PAP2T.`
`03:56.40`	`keebler`	`How bad is the call quality with G729?`
`03:56.49`	`thehar`	`loooorrritab`
`03:59.05`	`joobie`	`$180AUD for a polycom 320 isnt bad tho fender`
`03:59.20`	`joobie`	`how much is the linksys equiv?`
`04:01.56`	`*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)`
`04:02.11`	`[TK]D-Fender`	`joobie: I don't know currently, I jsut had 2 AU clients who shopped around and from what I saw the difference was pretty wide`
`04:02.22`	`[TK]D-Fender`	`joobie: Compar against SPA-942 / 922`
`04:02.33`	`[TK]D-Fender`	`(seriously I'd splurge for the 942)`
`04:03.42`	`[TK]D-Fender`	`http://www.voipshop.com.au/IP-Handsets-Linksys/c22_101/p203/Linksys-SPA942/product_info.html`
`04:04.30`	`[TK]D-Fender`	`320 has no passthrough BTW.`
`04:04.39`	`[TK]D-Fender`	`joobie: A factor to consider`
`04:05.18`	`joobie`	`yaa`
`04:06.22`	`[TK]D-Fender`	`joobie: PLEASE shop around. You should have a number of sources bookmarked.`
`04:07.11`	`[TK]D-Fender`	`joobie: Polycom is a great phone, don't get me wrong, and if was a simple choice of which do I prefer, then it'd in hands down. But I can say the difference doesn't justify a large $ varience`
`04:07.12`	`joobie`	`i do`
`04:07.17`	`joobie`	`staticice.com.au is gold`
`04:07.19`	`joobie`	`and shopbot:P`
`04:07.54`	`joobie`	`942, cheapest is $153`
`04:08.00`	`joobie`	`320 polycom, cheapest is $180`
`04:08.08`	`[TK]D-Fender`	`joobie: Ok, so shop around and let us know the bottom line and we'll give you some opinions.`
`04:08.21`	`[TK]D-Fender`	`942 kills it at that range...`
`04:08.25`	`joobie`	`fender above functionality, i want call quality.. that to me is paramount.`
`04:08.33`	`joobie`	`yea.. i can see feature wise it's good`
`04:08.36`	`joobie`	`but call quality, duno..`
`04:08.43`	`joobie`	`feature wise it kills the 320 polycom.`
`04:08.47`	`*** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)`
`04:08.55`	`[TK]D-Fender`	`joobie: Polycom has an edge, but the 942 beats it on a lot of other factors, not just price.`
`04:09.09`	`[TK]D-Fender`	`joobie: It'd be hard for me to turn it down...`
`04:09.27`	`joobie`	`fender, if call quality was your only concern.. would it be the polycom 320?`
`04:09.43`	`joobie`	`drum roll`
`04:09.52`	`Carlos_PHX`	`Did someone say Linksys vs. Polycom?`
`04:09.56`	`joobie`	`lol`
`04:10.17`	`joobie`	`Carlos_PHX, linksys vs polycom for call quality comparison`
`04:10.21`	`[TK]D-Fender`	`joobie: on the SPA side : backlit display, passthrough port, 4 line-keys (little friendlier), easier to configure (less powerful on the fine-tuning though), and decently less expensive`
`04:10.22`	`Carlos_PHX`	`Same`
`04:10.27`	`Carlos_PHX`	`Except speaker`
`04:10.34`	`[TK]D-Fender`	`joobie: Polycom audio on the other hand....`
`04:10.40`	`joobie`	`except speaker?`
`04:10.50`	`Carlos_PHX`	`Speaker on the Polycom is WAY better.`
`04:10.59`	`joobie`	`ahh`
`04:11.01`	`[TK]D-Fender`	`Carlos_PHX: I couldn't in good conscience recommend the 320 over the 942 at that price...`
`04:11.14`	`Carlos_PHX`	`Yeah, I saw that.`
`04:11.17`	`joobie`	`fender, if call quality was the only consideration though...`
`04:11.17`	`Carlos_PHX`	`The 320...ugh`
`04:11.26`	`[TK]D-Fender`	`joobie: Polycom > All :)`
`04:11.29`	`Carlos_PHX`	`Since the 942 is $110...`
`04:11.35`	`[TK]D-Fender`	`Carlos_PHX: AUD$ <-`
`04:11.37`	`Carlos_PHX`	`And the 941 even less...`
`04:11.44`	`[TK]D-Fender`	`Carlos_PHX: and he's looking PoE`
`04:11.45`	`joobie`	`the issue is, the client is really really really sceptical about call quality.. the polycom 320 feature wise, he's happy with.. so my only goal is to ensure call quality is the best.`
`04:11.46`	`Carlos_PHX`	`Oh, so like $38357?`
`04:11.57`	`joobie`	`that's why feature wise, though it's better.. and price is less.. they are not concerns of the client..`
`04:12.11`	`[TK]D-Fender`	`joobie: they WILL be happy with Polycom. No argument whatsoever.`
`04:12.21`	`joobie`	`thanks fender`
`04:12.25`	`joobie`	`that says it all`
`04:12.30`	`Carlos_PHX`	`So here's the thing. Not one of our customers complained about Polycoms, but...`
`04:12.31`	`[TK]D-Fender`	`joobie: [23:07]<joobie>942, cheapest is $153`
`04:12.32`	`[TK]D-Fender`	`[23:08]<joobie>320 polycom, cheapest is $180`
`04:12.47`	`Carlos_PHX`	`Once we gave them a few Linksys, everyone wanted to replace the Polycoms with Linksys.`
`04:12.51`	`joobie`	`valid points about the linksys, definitely worth looking at for a standard deployment where features and call quality is balanced..`
`04:12.55`	`Carlos_PHX`	`Except one guy who does 4 hours/day on speaker.`
`04:13.23`	`joobie`	`why did they want to replace with the linksys?`
`04:13.27`	`*** join/#asterisk neurosys (n=vinix@c-71-196-8-216.hsd1.fl.comcast.net)`
`04:13.44`	`[TK]D-Fender`	`Carlos_PHX: I'd probably be OK on either for normal use... though I love the join/split / weight & audio on Polyc's`
`04:13.51`	`Carlos_PHX`	`Nicer to use, display, etc. Personally I replaced my own too.`
`04:14.18`	`Carlos_PHX`	`I have a Polycom 650 in one office but have an SPA525G coming to replace that.`
`04:14.20`	`[TK]D-Fender`	`Carlos_PHX: On a 430+ I'd remove the "nicer to use" point from Linksys :)`
`04:14.20`	`joobie`	`BTW, only diff between the 320 and 330 is the 330 can switch off to a PC ya? otherwise the same??`
`04:14.29`	`[TK]D-Fender`	`Carlos_PHX: ooohhh pricey`
`04:14.36`	`Carlos_PHX`	`[TK]D-Fender: I only report what the customers say...`
`04:14.39`	`[TK]D-Fender`	`525G= $`
`04:14.47`	`Carlos_PHX`	`$310`
`04:14.50`	`Carlos_PHX`	`Not cheap!`
`04:14.53`	`[TK]D-Fender`	`joobie: exactly the same otherwise`
`04:15.00`	`joobie`	`cheers`
`04:15.10`	`*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)`
`04:15.14`	`[TK]D-Fender`	`joobie: I usually sell the idea of 320's and tell them to invest the difference in wiring infrastructure`
`04:15.24`	`joobie`	`ya`
`04:15.30`	`[TK]D-Fender`	`Carlos_PHX: Distintly not cheap`
`04:15.33`	`Carlos_PHX`	`And PoE`
`04:15.33`	`joobie`	`that's what im doing here.. well they sorta went half-ass`
`04:15.38`	`*** join/#asterisk trigeek38 (n=chatzill@ip72-196-100-30.ga.at.cox.net)`
`04:15.45`	`joobie`	`they ran 2 cables to a pod of 4 desks.. so we have to put a POE switch at each desk`
`04:16.00`	`joobie`	`to power the 4 phones.. and uplink one cable for the telephone.. and the other cable has the data for the pcs`
`04:16.03`	`joobie`	`bit dodgy..`
`04:16.06`	`Carlos_PHX`	`No, but now if someone wants a wi-fi desk phone, I will have one. Assuming they don't suck as much as Linksys portable wi-fi phones do.`
`04:16.32`	`Carlos_PHX`	`joobie: Might as well not use PoE then.`
`04:17.13`	`joobie`	`i dont wnat to push the 4 PC's data through the same cat5 as the voip`
`04:17.32`	`joobie`	`ie. if they go and download a large file from the server.. it will max out the cat5`
`04:17.37`	`*** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl)`
`04:17.59`	`joobie`	`this way i have 1 x cat 5 guarenteed for 4 phones back to the switch..`
`04:18.15`	`joobie`	`so i'll need to switch that 1 x cat5 into 4.. need a switch for that, so may as well get a POE`
`04:18.21`	`joobie`	`that was my chain of thought..`
`04:19.51`	`*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.86)`
`04:22.16`	`trigeek38`	`Hi group`
`04:22.28`	`trigeek38`	`I have an ssh question`
`04:23.55`	`trigeek38`	`If I putty into my asterisk server and create a tunnel to 5060, should I be able to configure a softphone on that pc to connect through that tunnel?`
`04:24.32`	`trigeek38`	`or is there going to be NAT issues?`
`04:25.05`	`joobie`	`trying to bypass firewalls hey`
`04:25.36`	`trigeek38`	`something like it`
`04:26.08`	`joobie`	`I haven't tried it.. but my thoughts are; you'd be doing NAT so your * box would need to have nat support enabled.. the other thing is RTP, you'd need to look into that`
`04:26.11`	`trigeek38`	`I actually am looking for simple secure connection remotely`
`04:26.32`	`joobie`	`I've only played with SIP .. which to my understanding uses some UDP ports for RTP..`
`04:26.47`	`joobie`	`you'd need to tunnel those through too from your PC to the asterisk box`
`04:27.15`	`joobie`	`create a VPN and run the voip through the VPN`
`04:27.23`	`joobie`	`that'd be the best solution imhop`
`04:27.24`	`joobie`	`-p`
`04:28.01`	`rob0`	`-m pancakes!`
`04:30.09`	`[TK]D-Fender`	`trigeek38: Voip is UDP BTW.... make sure that's what you're tunneling.`
`04:30.15`	`trigeek38`	`yeah, I'll probably do that but I was looking for a lightweight solution on top of ssh and putty`
`04:30.39`	`[TK]D-Fender`	`trigeek38: expect pain (at best)`
`04:31.17`	`trigeek38`	`forgot about the UDP but I know putty can tunnel those too`
`04:31.42`	`[TK]D-Fender`	`trigeek38: And actually.... I really don't see how this is going to be possible... it hosts them off the connected server.... you don't control the ports * will pick... pretty FUBAR'd.`
`04:31.50`	`[TK]D-Fender`	`trigeek38: Go set something else up`
`04:32.14`	`[TK]D-Fender`	`hrm...`
`04:32.15`	`trigeek38`	`yep`
`04:33.10`	`joobie`	`I think you can add multiple ports to forward with putty though`
`04:33.24`	`joobie`	`so you could specify your range in the rtp conf and match this with the putty range`
`04:33.32`	`joobie`	`very messy setup`
`04:33.44`	`joobie`	`vpn is much cleaner if it's just security u want`
`04:34.12`	`joobie`	`or maybe look into stunnel...`
`04:34.19`	`trigeek38`	`yes!`
`04:34.19`	`joobie`	`still messy:OP`
`04:34.31`	`trigeek38`	`what was I thinking`
`04:35.14`	`joobie`	`ahh stunnel doesnt do UDP according to the opening speel on stunnel.org`
`04:35.17`	`joobie`	`interesting.`
`04:41.36`	`rob0`	`ssh cannot tunnel UDP, because ssh is TCP`
`04:42.25`	`rob0`	`openvpn would be my first choice, but that might cause some extra lag`
`04:43.51`	`trigeek38`	`looking into CIPE right ...`
`04:43.59`	`trigeek38`	`*now`
`04:45.46`	`[TK]D-Fender`	`CIPE : Last Update: Aug 03 2004`
`04:45.48`	`trigeek38`	`nevermind last updated in 2004`
`04:45.48`	`[TK]D-Fender`	`WOW`
`04:45.56`	`trigeek38`	`just saw that too`
`04:46.07`	`[TK]D-Fender`	`trigeek38: Are you reading a RH7 Bible"`
`04:46.10`	`[TK]D-Fender`	`right now? :)`
`04:46.23`	`trigeek38`	`no, why?`
`04:46.27`	`[TK]D-Fender`	`trigeek38: that's just about the last place CIPE was mentioned :P`
`04:46.35`	`[TK]D-Fender`	`trigeek38: OpenVPN <-`
`04:47.39`	`*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)`
`04:48.21`	`trigeek38`	`looking into it now, thanks. I just googled stunnel and asterisk and came across CIPE`
`04:48.24`	`*** join/#asterisk mattwj2002 (n=matt@c-71-63-163-89.hsd1.mn.comcast.net)`
`04:48.29`	`mattwj2002`	`hi guys`
`04:48.33`	`mattwj2002`	`I really could use some help`
`04:48.56`	`trigeek38`	`I'll look into openvpn, Thanks for your help`
`04:49.21`	`mattwj2002`	`my softphones don't appear to reach my asterisk server`
`04:49.33`	`mattwj2002`	`all my settings look right`
`04:49.45`	`[TK]D-Fender`	`mattwj2002: Extension cords are on the left.`
`04:49.53`	`mattwj2002`	`lol`
`04:49.57`	`mattwj2002`	`good one haha`
`04:50.10`	`mattwj2002`	`could I upload my configurations and you guys take a look?`
`04:50.25`	`*** join/#asterisk SlicerDicer (n=kvirc@69-92-107-4.cpe.cableone.net)`
`04:51.51`	`[TK]D-Fender`	`mattwj2002: pastebin away`
`04:52.14`	`mattwj2002`	`sweet`
`04:52.21`	`mattwj2002`	`I only have 3 files`
`04:52.30`	`mattwj2002`	`I'll put them in one pastebin`
`04:52.36`	`mattwj2002`	`http://pastebin.com/m5c8613e2`
`04:54.06`	`joobie`	`openvpn is not supported properly afaik`
`04:54.08`	`joobie`	`openswan owns.`
`04:55.15`	`joobie`	`hmm i think im wrong - their site looks pretty up to date now`
`04:56.31`	`mattwj2002`	`you guys see anything wrong?`
`04:56.38`	`mattwj2002`	`I don't have a module.conf`
`04:56.39`	`[TK]D-Fender`	`mattwj2002: Yup... looks like * 1.0 dialplan, and NO NAT configuration.`
`04:56.54`	`[TK]D-Fender`	`mattwj2002: So... where are your softphones?`
`04:57.03`	`mattwj2002`	`oh my other laptop`
`04:57.14`	`mattwj2002`	`so I need to add nat=yes ?`
`04:57.40`	`mattwj2002`	`*on my`
`04:58.41`	`[TK]D-Fender`	`mattwj2002: And WHERE is your laptop relative to your * server?`
`04:59.02`	`mattwj2002`	`my laptop and my asterisk server are both on my home network`
`04:59.18`	`mattwj2002`	`my laptop is on wireless and my asterisk server is hard wired`
`04:59.21`	`[TK]D-Fender`	`mattwj2002: When someone asks "Where do you live" answering "In my appartment" really isn't meaningful :p`
`04:59.36`	`[TK]D-Fender`	`mattwj2002: so they are on the same local subnet?`
`04:59.38`	`mattwj2002`	`lol`
`04:59.40`	`mattwj2002`	`yes`
`05:00.01`	`joobie`	`hey fender`
`05:00.04`	`[TK]D-Fender`	`mattwj2002: Ok, so * doesn't seem to see anything coming from the phone?`
`05:00.09`	`joobie`	`you are from AU ya?`
`05:00.14`	`*** join/#asterisk jameswf-home (n=james@unaffiliated/jameswf-home)`
`05:00.17`	`mattwj2002`	`correct`
`05:00.34`	`mattwj2002`	`it doesn't show the registeration or anything`
`05:00.36`	`[TK]D-Fender`	`joobie: Montreal, QC, Canada`
`05:00.42`	`joobie`	`oh`
`05:00.47`	`[TK]D-Fender`	`mattwj2002: Enable SIP DEBUG at CLI and watch for packets.`
`05:00.47`	`mattwj2002`	`I am getting time out on my softphone too`
`05:01.05`	`[TK]D-Fender`	`mattwj2002: and PB "iptables --list" from your server`
`05:01.20`	`trigeek38`	`maybe a dumb question but what about windows firewall`
`05:01.23`	`joobie`	`well, you may be interested anyway :P I just spoke to polycom australia, and they told me who their AU distributors are.. i signed up for an account with one of them and they sell the 320 phone for $217.19`
`05:01.30`	`joobie`	`more expensive than that 180$ place`
`05:01.32`	`joobie`	`heh`
`05:01.34`	`joobie`	`weird.`
`05:01.53`	`trigeek38`	`mattwj2002 - firewall issue?`
`05:02.08`	`mattwj2002`	`I don't think there is a firewall up`
`05:02.11`	`mattwj2002`	`let me check`
`05:02.18`	`[TK]D-Fender`	`trigeek38: Not dumb... only unconfirmed :)`
`05:02.37`	`*** join/#asterisk MaliutaLap (n=biteme@203.171.192.252)`
`05:02.43`	`[TK]D-Fender`	`joobie: OUCH .`
`05:02.54`	`mattwj2002`	`it is down`
`05:03.05`	`[TK]D-Fender`	`joobie: Holy shit, I could not do it personally... SPA-942 wins on too many other fronts.`
`05:03.24`	`mattwj2002`	`how do I turn on sip debug?`
`05:03.25`	`[TK]D-Fender`	`mattwj2002: please show us some backup...`
`05:03.34`	`[TK]D-Fender`	`mattwj2002: "sip debug on" should do it`
`05:03.37`	`joobie`	`yaa.. i read into features, it's 10 times better...`
`05:03.46`	`joobie`	`if only they had a hybrid:P`
`05:03.54`	`mattwj2002`	`no such command`
`05:04.07`	`[TK]D-Fender`	`joobie: No, I wouldn't say that, just that it has more simultaneous calls, passthrough, backlight and a lot cheaper`
`05:04.21`	`[TK]D-Fender`	`mattwj2002: "sip set debug on"`
`05:04.29`	`[TK]D-Fender`	`mattwj2002: what version are you on?`
`05:04.36`	`jaytee`	`nite everyone`
`05:04.42`	`[TK]D-Fender`	`jaytee: Later`
`05:04.58`	`mattwj2002`	`1.4.21.2`
`05:05.05`	`trigeek38`	`jaytee: later`
`05:05.07`	`joobie`	`fender, have u seen this phone - http://www.engadget.com/2009/01/09/openpeak-intros-atom-powered-proframe-voip-phone/`
`05:05.08`	`mattwj2002`	`the ubuntu package`
`05:05.15`	`joobie`	`doesnt have a price.. but it looks nuts.`
`05:05.22`	`joobie`	`that rips the linksys:P`
`05:05.29`	`trigeek38`	`mattwj2002: and what softphone`
`05:05.33`	`*** join/#asterisk siera08 (n=sosoriri@218.207.141.90)`
`05:05.36`	`[TK]D-Fender`	`mattwj2002: EW... and your dialplan and SIP setup says "1.0"`
`05:05.39`	`*** join/#asterisk johnakabean (n=none@pool-72-82-106-201.nrflva.east.verizon.net)`
`05:05.48`	`mattwj2002`	`x-lite in windows vista`
`05:05.59`	`johnakabean`	`AMD([initialSilence][\|greeting].... Is this milliseconds or seconds? I'm guessing Milliseconds.`
`05:06.06`	`mattwj2002`	`crap`
`05:06.08`	`*** join/#asterisk Gopaul (n=chatzill@59.97.121.82)`
`05:06.14`	`[TK]D-Fender`	`mattwj2002: Go check your windows side`
`05:06.18`	`mattwj2002`	`what changes do I need to make`
`05:06.29`	`johnakabean`	`matt, is your softphone breaking up and having problems...x-lite`
`05:06.33`	`mattwj2002`	`I tried it in linux and on a seperate computer running xp`
`05:06.36`	`[TK]D-Fender`	`mattwj2002: Go ask in #windows :)`
`05:07.02`	`johnakabean`	`mine started doing that recently; I have to close it and open it up again to make it stop...its not asterisk!`
`05:07.03`	`[TK]D-Fender`	`mattwj2002: So wheres the iptables dump? How about "netstat -an\|grep 5060" while you're at it....`
`05:07.03`	`mattwj2002`	`it doesn't appear to be a vista or even a laptop issue`
`05:07.17`	`mattwj2002`	`can we do the debug first`
`05:07.19`	`[TK]D-Fender`	`mattwj2002: NOTHING is apparent. You haven't shown us anything...`
`05:07.22`	`mattwj2002`	`?`
`05:08.14`	`johnakabean`	`Its a windows update applied recently that started causing the problem...it does it in windows 7 beta too`
`05:09.00`	`*** part/#asterisk gloin (i=me@unaffiliated/gloin)`
`05:09.03`	`johnakabean`	`That big flaw in windows that allows hackers to compromise the remote procedure call service in windows`
`05:09.28`	`johnakabean`	`anyway,hey fender. AMD([initialSilence][\|greeting] is this in milliseconds or seconds?`
`05:09.36`	`[TK]D-Fender`	`johnakabean: No idea`
`05:09.52`	`johnakabean`	`Waitforsilence is milliseconds so I was guessing its thorough throughout`
`05:10.00`	`johnakabean`	`to be milliseconds`
`05:10.52`	`[TK]D-Fender`	`johnakabean: Source would confirm that`
`05:11.00`	`trigeek38`	`mattwj2002: for shi*tz and giggles, try zoiper or sjphone`
`05:11.02`	`[TK]D-Fender`	`johnakabean: I see 4 digits all around`
`05:11.09`	`[TK]D-Fender`	`~zoiper`
`05:11.09`	`jbot`	`[~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com`
`05:11.48`	`mattwj2002`	`okay I'll give it a try`
`05:12.44`	`frogonwheels`	`mattwj2002: argh Telstra.. they have such lovely people answering the phone. .. and they're still useless.`
`05:13.04`	`mattwj2002`	`Telstra?`
`05:13.17`	`frogonwheels`	`ah sorry - thought you were from Oz`
`05:13.23`	`frogonwheels`	`my ad`
`05:13.24`	`frogonwheels`	`bad`
`05:14.11`	`[TK]D-Fender`	`frogonwheels: I hear nothing but ill of them`
`05:14.30`	`[TK]D-Fender`	`frogonwheels: My clients were contantly frustrated by their lack of CID support, etc`
`05:14.37`	`[TK]D-Fender`	`frogonwheels: CDS issues, etc`
`05:14.44`	`*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-8d1de12135c604e5)`
`05:14.45`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)`
`05:14.48`	`frogonwheels`	`[TK]D-Fender: they are evil. and their evil twins @ bigpond are worse`
`05:15.00`	`frogonwheels`	`(being their ISP arm)`
`05:15.29`	`mattwj2002`	`I just had an idea`
`05:15.32`	`frogonwheels`	`[TK]D-Fender: but if you want a new land-line installed -you have to go through them.`
`05:15.39`	`mattwj2002`	`I wonder if Skype is causing me problems`
`05:15.44`	`[TK]D-Fender`	`frogonwheels: AU Telecom gets the shaft and the Socialist (s//fasciast) tendencies are just flat-out scary`
`05:15.47`	`johnakabean`	`digging up source`
`05:15.54`	`[TK]D-Fender`	`mattwj2002: No`
`05:16.03`	`[TK]D-Fender`	`mattmport-wise`
`05:16.14`	`mattwj2002`	`okay`
`05:16.25`	`[TK]D-Fender`	`johnakabean: I already did... looks like MS`
`05:16.50`	`mattwj2002`	`let me try using the sample configs`
`05:17.01`	`mattwj2002`	`and see if those work`
`05:18.14`	`[TK]D-Fender`	`mattwj2002: Nope`
`05:18.44`	`mattwj2002`	`nope?`
`05:18.45`	`[TK]D-Fender`	`mattwj2002: if you enabled SIP debug on your * machine and you don't see PACKETS then configs aren't involved much.`
`05:19.19`	`mattwj2002`	`the syntax you gave me for sip debug didn't work`
`05:19.53`	`[TK]D-Fender`	`[19:21]<[TK]D-Fender>Don't worry about traffic if your car won't even start <-`
`05:20.05`	`*** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-250-236.dsl.pltn13.sbcglobal.net)`
`05:20.11`	`[TK]D-Fender`	`mattwj2002: sip debug on`
`05:20.37`	`mattwj2002`	`I have no module.conf`
`05:20.44`	`mattwj2002`	`could this be causing the problem?`
`05:20.51`	`[TK]D-Fender`	`mattwj2002: Could be.`
`05:21.03`	`[TK]D-Fender`	`mattwj2002: in the case that NO modules are loading.`
`05:21.46`	`mattwj2002`	`that worked`
`05:21.46`	`mattwj2002`	`:D`
`05:22.03`	`mattwj2002`	`a round of virtual beers for all those that helped`
`05:22.05`	`mattwj2002`	`:D`
`05:23.15`	`trigeek38`	`burp`
`05:24.26`	`mattwj2002`	`now the next challenge`
`05:24.30`	`*** join/#asterisk bmoraca (n=bmoraca@adsl-75-12-126-173.dsl.skt2ca.sbcglobal.net)`
`05:24.32`	`mattwj2002`	`skype to sip`
`05:30.07`	`*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)`
`05:31.48`	`*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)`
`05:38.53`	`*** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:1d38:54e9:2689:a280)`
`05:45.24`	`edoceo`	`I'm specifying some callerid values in my .call files that don't seem to be passed along - issue at my provider?`
`05:46.57`	`frogonwheels`	`edoceo: you're expecting the provider to pass on the callerid values?`
`05:47.13`	`frogonwheels`	`edoceo: I suspect many providers would shut that down as a security risk.`
`05:47.28`	`edoceo`	`Hmm - I've seen it with some (odd I guess)`
`05:47.51`	`frogonwheels`	`edoceo: possibly some providers allow you to have a callerid set?`
`05:48.03`	`frogonwheels`	`edoceo: if I were a provider, that's the way I'd handle it.`
`05:48.05`	`edoceo`	`Yes - they would pass the info from * alon`
`05:48.22`	`frogonwheels`	`edoceo: - no I mean they would apply whatever they had set at _their_ end`
`05:48.23`	`edoceo`	`Some require it to be set in their "web portal"`
`05:48.25`	`johnakabean`	`Anyone know where to get NVlinedetect for asterisk 1.4`
`05:48.28`	`edoceo`	`Yes -`
`05:48.31`	`frogonwheels`	`edoceo: yep.`
`05:48.45`	`edoceo`	`Hmm - diamondcard will pass whatever you tell them`
`05:48.54`	`frogonwheels`	`Otherwise you could set your callerid name and/or number to be 000 or 911 or whatever you wanted to`
`05:50.41`	`edoceo`	`This page says Teliax will pass it along (but right now they done) http://www.voip-info.org/wiki/view/TelIAX`
`05:50.46`	`edoceo`	`s/done/dont/`
`05:51.08`	`bmoraca`	`edoceo: most telcos force your caller id to be your pilot number. some don't. some doe.`
`05:51.08`	`frogonwheels`	`hey.. clever jbot`
`05:51.08`	`edoceo`	`thinks that jbot is cute`
`05:51.16`	`bmoraca`	`wow...jbot understands sed`
`05:51.35`	`frogonwheels`	`s/[jk]bot/mybot/`
`05:51.44`	`frogonwheels`	`:)`
`05:51.58`	`frogonwheels`	`s/:)/:(/`
`05:52.03`	`frogonwheels`	`lol`
`05:52.04`	`bmoraca`	`lol`
`05:52.13`	`bmoraca`	`s/lol/rofl`
`05:52.19`	`bmoraca`	`hrm`
`05:52.25`	`frogonwheels`	`need to terminate it`
`05:52.29`	`bmoraca`	`ah`
`05:52.31`	`bmoraca`	`pedant`
`05:52.33`	`frogonwheels`	`^need^have^`
`05:52.51`	`frogonwheels`	`x en de pedant`
`05:52.55`	`frogonwheels`	`~x en de pedant`
`05:53.09`	`frogonwheels`	`:)`
`05:53.39`	`frogonwheels`	`should really play with the jbot in a private channel I guess.`
`05:53.41`	`bmoraca`	`i don't remember ever having to terminate...unless i was using an option like g or something`
`05:54.03`	`frogonwheels`	`bmoraca: well you should really. but no, you generally don't have to.`
`05:54.36`	`drmessano`	`frogonwheels, you're cool`
`05:54.41`	`drmessano`	`s/cool/a douche`
`05:54.42`	`bmoraca`	`i haven't used sed in like 5 years...`
`05:54.48`	`drmessano`	`:(`
`05:54.49`	`drmessano`	`Didnt work`
`05:54.52`	`drmessano`	`Sorry`
`05:54.55`	`frogonwheels`	`drmessano: .. you need to terminate it`
`05:55.00`	`drmessano`	`Duh`
`05:55.05`	`drmessano`	`frogonwheels, you're cool`
`05:55.09`	`drmessano`	`s/cool/a douche/`
`05:55.13`	`frogonwheels`	`that's better`
`05:55.14`	`drmessano`	`YAY!!!`
`05:55.41`	`drmessano`	`sorry, you could just as easily have been anyone else`
`05:55.46`	`drmessano`	`Nothing personal`
`05:55.53`	`drmessano`	`heh`
`05:55.59`	`*** part/#asterisk ABom9 (n=adamirc@cpe-67-246-182-12.buffalo.res.rr.com)`
`05:56.00`	`frogonwheels`	`shrugs.`
`05:56.19`	`frogonwheels`	`I've probably been called worse by yourself and [TK]D-Fender ..`
`05:57.02`	`frogonwheels`	`when I was being particularly annoyingly newbish.`
`05:57.15`	`[TK]D-Fender`	`frogonwheels: I don't normally resort to petty name calling :)`
`05:57.23`	`drmessano`	`I doubt it`
`05:57.46`	`frogonwheels`	`ok. maybe not from drmessano.`
`05:57.48`	`drmessano`	`I'll imply the hell out of you being a dumbass, newb, neo-dweebie douchbag, but never SAY IT`
`05:58.05`	`drmessano`	`heh`
`05:58.18`	`drmessano`	`It's not my style`
`05:58.32`	`*** join/#asterisk robwafle (i=robwafle@c-67-167-204-1.hsd1.il.comcast.net)`
`05:58.41`	`robwafle`	`hello`
`05:58.42`	`drmessano`	`Unless someone is headed for a Darwin aware`
`05:58.44`	`drmessano`	`Unless someone is headed for a Darwin award`
`05:58.55`	`drmessano`	`Like "I just installed AsteriskWin32"`
`05:58.58`	`robwafle`	`how is everyone tonight?`
`05:59.02`	`drmessano`	`Then I may insult their mom`
`05:59.18`	`robwafle`	`lol drmessano`
`05:59.23`	`frogonwheels`	`hm.. yeah. I installed that for fun once so that I could muck around with it.`
`05:59.37`	`frogonwheels`	`needless to say it wasn't.`
`05:59.43`	`robwafle`	`I just installed trixbox for the first time`
`05:59.44`	`frogonwheels`	`(fun)`
`05:59.55`	`[TK]D-Fender`	`drmessano: http://www.motivatedphotos.com/?id=1340`
`05:59.56`	`drmessano`	`I love Windows.. I admin it all day long.. Needless to say, there's some things that dont go well together...`
`06:00.06`	`drmessano`	`I like Peanut Butter, I love Shrimp.. but yeah`
`06:00.20`	`[TK]D-Fender`	`robwafle: "not supported here"`
`06:00.48`	`bmoraca`	`[TK]D-Fender: lol. you'd probably like forumwarz INCIT.`
`06:00.51`	`drmessano`	`stabs robwafle`
`06:00.52`	`robwafle`	`TrixBox is pretty nice.. seems the current beta of AsteriskNow has FreePBX now, but it was incomplete.`
`06:00.56`	`robwafle`	`lol`
`06:00.57`	`[TK]D-Fender`	`drmessano: I love szcechuan... they go GREAT together`
`06:01.05`	`robwafle`	`I am running Asterisk too`
`06:01.11`	`joobie`	`hey guys is there a way to send a CID to say "use private" ?`
`06:01.27`	`robwafle`	`I have two servers setup to get experience with both`
`06:01.33`	`drmessano`	`stabs robwafle repeatedly, OJ Style.. /////\\////////////////\////`
`06:01.34`	`[TK]D-Fender`	`joobie: "core show application setcallerpres"`
`06:01.36`	`robwafle`	`LOL`
`06:01.46`	`robwafle`	`please don't hate ! :)`
`06:01.54`	`drmessano`	`goes off to find the real killer`
`06:02.12`	`[TK]D-Fender`	`ghost writes for drmessano "If I did it....."`
`06:02.18`	`drmessano`	`HAHAH`
`06:02.18`	`robwafle`	`at least I never tried the Win32 version`
`06:02.30`	`[TK]D-Fender`	`hi-5's drmessano`
`06:02.32`	`drmessano`	`Now that... was ... OJFAIL`
`06:02.34`	`joobie`	`fender, per extension?`
`06:02.42`	`joobie`	`i use the sip.conf atm to set specific CID..`
`06:02.52`	`drmessano`	`"I didnt do IT!!.. But man, if I did.. lemmetellya"`
`06:02.57`	`[TK]D-Fender`	`joobie: before you dial`
`06:03.17`	`joobie`	`ahhh`
`06:03.20`	`joobie`	`cheers`
`06:03.21`	`*** join/#asterisk mchou (n=mchou@unaffiliated/mchou)`
`06:06.46`	`johnakabean`	`Fender, know of an application that detects zapateller tones?`
`06:07.24`	`[TK]D-Fender`	`johnakabean: IIRC they get picked up as A-D`
`06:08.20`	`frogonwheels`	`Is there an easy way of doing a background sayUnixTime ?`
`06:09.07`	`[TK]D-Fender`	`frogonwheels: Doesn't look like`
`06:09.17`	`johnakabean`	`uhmm that picks up busy tones; i'm looking in regards of disconnected number tones`
`06:09.22`	`johnakabean`	`like Zapateller`
`06:09.34`	`johnakabean`	`Nvlinedetect is too old for 1.4`
`06:09.39`	`johnakabean`	`its included in 1.6 though`
`06:09.41`	`johnakabean`	`sigh`
`06:09.52`	`frogonwheels`	`[TK]D-Fender: what I figured. possibly the agi version backgrounds. but don't really want to get into that if I don't have to.`
`06:09.58`	`johnakabean`	`The source is too old per say`
`06:10.10`	`[TK]D-Fender`	`frogonwheels: nope`
`06:10.34`	`*** join/#asterisk smokebowl (i=bowlburn@ip68-225-77-246.no.no.cox.net)`
`06:10.54`	`[TK]D-Fender`	`frogonwheels: people keep thinking AGI is magic. The apps you call are EXACTLY the same as normal dialplan. Only this you can do is stream straight audio while doing other stuff.`
`06:11.09`	`joobie`	`Thanks fender, it works a charm.`
`06:11.53`	`[TK]D-Fender`	`frogonwheels: You could create your entire time phrasing script to pick out the files tos tream, but boy thats a lot of trouble`
`06:11.53`	`[TK]D-Fender`	`joobie: np`
`06:11.53`	`frogonwheels`	`[TK]D-Fender: yeah - I know it's not magic - but saw some doco that said say unixtime backgrounded - but it seemed weird they'd be different.`
`06:11.59`	`frogonwheels`	`[TK]D-Fender: yeah - I've done that for sayNumber - but that's easy.`
`06:12.14`	`bmoraca`	`frogonwheels: use Cut() and parse it yourself, lol`
`06:13.10`	`frogonwheels`	`bmoraca: oh wow - that sounds like SOOO much fun. :)`
`06:13.36`	`bmoraca`	`frogonwheels: nothing worth doing is easy :P`
`06:13.56`	`bmoraca`	`frogonwheels: except that I can't think of a more worthless thing than a speaking clock in your phonesystem`
`06:14.18`	`frogonwheels`	`bmoraca: I've got a speaking clock :) .. but it's not for that.`
`06:14.29`	`frogonwheels`	`bmoraca: it's for saying the time of a missed call.`
`06:15.56`	`bmoraca`	`frogonwheels: uhh...just make them listen to the voicemail envelope or look at the missed calls directory on their phone`
`06:16.04`	`*** join/#asterisk MaliutaLap (n=biteme@203.171.192.223)`
`06:17.04`	`frogonwheels`	`bmoraca: ok ... missed call /last call / last received call - so you can dial back .. but I've had an idea.`
`06:17.33`	`frogonwheels`	`bmoraca: the answer is not to play back the time. have an option to do it..`
`06:17.50`	`*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-194-200.lns10.mel6.internode.on.net)`
`06:18.12`	`frogonwheels`	`bmoraca: and I did it b4 I had a phone that _handled_ missed calls.`
`06:18.21`	`bmoraca`	`frogonwheels: i don't understand the point of the application. virtually all IP desk phones can already do that and have an option to dial`
`06:18.39`	`bmoraca`	`in fact, most analog phones with caller ID can do it too`
`06:18.54`	`bmoraca`	`in any case`
`06:19.05`	`frogonwheels`	`bmoraca: 'cause its designed for a home system .. and I didn't _have_ a phone with callerid.`
`06:19.17`	`bmoraca`	`it's nyquil time...have fun all`
`06:19.22`	`bmoraca`	`frogonwheels: ahh, gotcha`
`06:20.46`	`*** join/#asterisk drfreeze (n=Jim@207.191.114.82)`
`06:20.49`	`drfreeze`	`Hello`
`06:21.32`	`drfreeze`	`Anyone know of an example of how I can allow specific callers to dial an internal extension?`
`06:23.05`	`edoceo`	`drfreeze: put those specific callers in a specific context that permits connection to extensions`
`06:23.22`	`frogonwheels`	`drfreeze: exten=_X./04111111,1,DISA(000\|mycontext)`
`06:23.48`	`frogonwheels`	`drfreeze: look at DISA`
`06:24.02`	`[TK]D-Fender`	`What does DISA have to do with that?`
`06:24.11`	`[TK]D-Fender`	`CRAZY talk`
`06:24.46`	`frogonwheels`	`[TK]D-Fender: erm.. presumably he's talking about somebody calling from outside on a specific phone.. and wanting to dial an internal extension...`
`06:25.02`	`drfreeze`	`frogonwheels: yes`
`06:25.30`	`[TK]D-Fender`	`You don't need DISA for this...`
`06:25.34`	`frogonwheels`	`drfreeze: there are other solutions -`
`06:25.37`	`*** join/#asterisk robwafle (i=robwafle@c-67-167-204-1.hsd1.il.comcast.net)`
`06:25.37`	`[TK]D-Fender`	`totally nuts`
`06:25.51`	`[TK]D-Fender`	`drfreeze: pastebin what you've got`
`06:25.59`	`robwafle`	`so... what did I miss?`
`06:26.31`	`drfreeze`	`[TK]D-Fender: I have a set of internal extensions 6xx and 5xx`
`06:26.57`	`frogonwheels`	`drfreeze: you could also put them into an IVR and include your extensions into that.`
`06:26.59`	`[TK]D-Fender`	`drfreeze: pastebin....`
`06:27.03`	`drfreeze`	`When calls from 5551234 come in, they need to be able to connect to a 5xx or 6xx number`
`06:27.23`	`[TK]D-Fender`	`drmessano: put them in a context that includes one that contains those then`
`06:27.33`	`drfreeze`	`[TK]D-Fender: sorry, nothing to paste but the 2 line description above`
`06:27.51`	`*** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com)`
`06:28.08`	`[TK]D-Fender`	`drmessano: put them in a context that includes one that contains those then <---`
`06:28.59`	`frogonwheels`	`s/drmessano/drfreeze/`
`06:29.08`	`[TK]D-Fender`	`that too`
`06:31.05`	`*** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-250-236.dsl.pltn13.sbcglobal.net)`
`06:31.12`	`drfreeze`	`something like: exten => s,1,Dial(Zap/tgExternalPtP//${CALLERIDNUM})`
`06:31.29`	`drfreeze`	`whatever tgExternalPtP means`
`06:31.52`	`[TK]D-Fender`	`WTF?`
`06:33.05`	`*** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)`
`06:33.28`	`frogonwheels`	`drfreeze: http://pastebin.com/d4d289c8a`
`06:34.39`	`frogonwheels`	`sorry.. there's an 'extern' in there that should be 'exten'`
`06:35.01`	`drfreeze`	`frogonwheels: thx. will try that`
`06:35.19`	`drfreeze`	`what does the /0411111... do?`
`06:35.39`	`frogonwheels`	`that is a shortcut to check for callerid(num) matching 0411111 or whatever you want there`
`06:36.12`	`drfreeze`	`ok`
`06:36.21`	`frogonwheels`	`drfreeze: so there should be a line after that saying exten => _X,1,Dial(SIP/myexten1&SIP/myexten2) or howeveryou do it now.`
`06:36.30`	`frogonwheels`	`_X.`
`06:51.07`	`*** join/#asterisk ABom9 (n=Abom9@cpe-67-246-182-12.buffalo.res.rr.com)`
`06:53.22`	`*** part/#asterisk ABom9 (n=Abom9@cpe-67-246-182-12.buffalo.res.rr.com)`
`06:53.38`	`*** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe)`
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`06:56.31`	`johnakabean`	`what happened to asterisk 1.4.31.2?`
`06:56.47`	`johnakabean`	`what happened to asterisk 1.4.31.1?`
`06:57.07`	`*** part/#asterisk drepan (n=pandre@apcdns2.autopage.co.za)`
`06:59.37`	`drmessano`	`Not out yet`
`06:59.56`	`drmessano`	`--> /topic`
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`07:49.23`	`*** join/#asterisk ajmcello (n=ajmcello@75.151.111.233)`
`07:50.58`	`ajmcello`	`i'm having a problem with nat and asterisk. it seems as though asterisk is seeing my phones internal ip address of 192.168.1.104 instead of my comcast ip.`
`07:51.09`	`ajmcello`	`i have nat=1 in sip.conf and NAT is set to yes on my phone`
`07:51.12`	`ajmcello`	`any idea what is happening?`
`07:51.51`	`*** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis)`
`07:54.47`	`*** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:1d38:54e9:2689:a280)`
`07:56.22`	`frogonwheels`	`ajmcello: nat=yes would be what I expect..`
`07:56.48`	`frogonwheels`	`ajmcello: Does the phone have an 'externalip' setting?`
`07:57.03`	`ajmcello`	`let me try that`
`07:57.10`	`ajmcello`	`it works fine in asterisk 1.4`
`07:57.12`	`frogonwheels`	`ajmcello: presumably your asterisk box and your phone are not on the same private network.`
`07:57.15`	`ajmcello`	`im using 1.6.1`
`07:57.28`	`ajmcello`	`frogon correct, im at home using comcast and the server is at work`
`07:57.49`	`ajmcello`	`the phone does have an external ip and im trying to figure out how to set it using the config file..`
`07:58.19`	`frogonwheels`	`ajmcello: hopefully that's a static external ip?`
`07:58.59`	`frogonwheels`	`ajmcello: I haven't touched 1.6 - so I may not be of help. .. however..`
`07:59.32`	`frogonwheels`	`ajmcello: I'd collect together a pastebin of configs and logs with stuff obfuscated if need be.`
`07:59.40`	`ajmcello`	`ok`
`08:00.15`	`ajmcello`	`i set nat_address=myip on my phone`
`08:00.20`	`ajmcello`	`and i think that did the trick`
`08:00.35`	`ajmcello`	`weird thing about is that it worked for about 3 minutes and then went unreachable and never came back....`
`08:08.28`	`*** join/#asterisk unasi7 (n=unasi7@62-2-119-222.static.cablecom.ch)`
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`08:30.19`	`*** join/#asterisk chosig (n=gunnar@the.spacewaster.net)`
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`08:31.36`	`chosig`	`is it possible to use asterisk in with softphones (if that's the right word, use the computer as a phone)?`
`08:33.28`	`*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)`
`08:34.49`	`viq`	`yes, they are called softphones, and yes, you can use them with asterisk`
`08:37.49`	`chosig`	`great :)`
`08:38.06`	`chosig`	`feels likes out in deep water...`
`08:40.24`	`frogonwheels`	`chosig: life raft perhaps?`
`08:40.27`	`frogonwheels`	`~thebook`
`08:40.27`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`08:44.16`	`frogonwheels`	`~tell chosig about thebook`
`08:44.25`	`Thiago_Lima`	`Good Morning... Any idea what may be happening in these log errors? http://pastebin.com/m57bc33f5`
`08:44.36`	`frogonwheels`	`jbot, tell chosig about thebook`
`08:44.54`	`frogonwheels`	`ah.. sorry chosig`
`08:47.53`	`chosig`	`no worries :)`
`08:48.16`	`chosig`	`a friend asked me to set up a pbx for hes (small) company`
`08:49.41`	`chosig`	`btw, the html link jbot sends for the book is broken`
`08:51.37`	`frogonwheels`	`chosig: nope - not here it isn't.`
`08:52.04`	`chosig`	`frogonwheels: strange, have to try later i don't get a response`
`08:58.32`	`*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)`
`09:00.17`	`*** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br)`
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`09:03.39`	`synthetiq`	`what would cause chan_sip to hang dead an not do anything with incoming sip messages`
`09:06.08`	`*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)`
`09:08.06`	`oej`	`synthetiq: Severe DNS issues`
`09:10.29`	`synthetiq`	`even if i specify the ip in the registe message?`
`09:10.32`	`synthetiq`	`register`
`09:12.10`	`synthetiq`	`dns seems to be fine`
`09:17.57`	`mattwj2002`	`hi guys`
`09:18.09`	`mattwj2002`	`I have been trying to connect asterisk and skype all night`
`09:18.12`	`mattwj2002`	`with no luck`
`09:18.13`	`mattwj2002`	`:(`
`09:19.22`	`mattwj2002`	`any suggestions?`
`09:19.55`	`frogonwheels`	`~skype`
`09:19.55`	`jbot`	`[~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option.`
`09:20.13`	`frogonwheels`	`~SkypeForAsterisk`
`09:20.14`	`jbot`	`[~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details.`
`09:21.04`	`mattwj2002`	`any other suggestions?`
`09:21.40`	`mattwj2002`	`would an fxs connected to a skype phone adapter work?`
`09:22.41`	`*** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it)`
`09:29.12`	`frogonwheels`	`mattwj2002: hadn't thought of that - yeah should be able to...`
`09:29.46`	`frogonwheels`	`mattwj2002: thought don't know how you dial users...`
`09:29.55`	`mattwj2002`	`yeah`
`09:29.57`	`mattwj2002`	`hmmm`
`09:30.45`	`mattwj2002`	`maybe I'll just stick with Asterisk as a toy and buy service for it`
`09:31.02`	`mattwj2002`	`I'll use Skype to call my parents because it is cheap`
`09:31.18`	`frogonwheels`	`mattwj2002: my ISP provides my Voip - and it works great..`
`09:31.27`	`frogonwheels`	`mattwj2002: I've not used skype hardly since.`
`09:31.28`	`*** join/#asterisk virtualme123 (n=chatzill@fentech.gotadsl.co.uk)`
`09:31.35`	`mattwj2002`	`nice`
`09:31.56`	`angryuser`	`mattwj2002: asterisk <> skype i a stotal crap if you want my opinion`
`09:32.00`	`frogonwheels`	`mattwj2002: don't have a landline even (all over adsl2+)`
`09:32.27`	`angryuser`	`mattwj2002: quality like you talk from the toilet`
`09:32.36`	`frogonwheels`	`angryuser: I think the only good thing about such a gateway would be to call skype users you know...`
`09:32.54`	`mattwj2002`	`what is total crap angryuser?`
`09:33.00`	`virtualme123`	`I got some warnings in my Asterisk log while on some calls to say that the 'chan_iax2.c had Max retries exceeded to host', does anyone know what this could mean?`
`09:33.11`	`angryuser`	`frogonwheels: no, the only good thing it to let OTHERS call you`
`09:33.40`	`frogonwheels`	`sure.`
`09:34.41`	`edoceo`	`my pbx_spool is not accepting the caller ID in the file - however my carrier says they do support me passing caller id - seems I can't get pbx_spool to set caller ID before dialing out`
`09:37.19`	`frogonwheels`	`edoceo: have you tried making your call file drop you into a context with some variables set to set your callerid - just to test things out?`
`09:37.34`	`*** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com)`
`09:37.38`	`edoceo`	`That call file drops into context _after_ the call answers :(`
`09:37.49`	`edoceo`	`I don't know how to get it to do stuff preDial`
`09:38.19`	`frogonwheels`	`edoceo: which phone are you dialing with the call file?`
`09:38.25`	`edoceo`	`My mobile`
`09:38.51`	`frogonwheels`	`edoceo: and where's the other end connecting to?`
`09:39.08`	`edoceo`	`It's calling and * is just playing a file to me`
`09:39.13`	`edoceo`	`"hello-world"`
`09:40.10`	`frogonwheels`	`edoceo: ok - and you are using Callerid: Name<number> (or whichever way it is)`
`09:40.20`	`edoceo`	`correct`
`09:41.00`	`edoceo`	`I can see the NoOps in my context - after connection - but can't see where in the dialplan it goes before that - seems default-=> trunk but I can't tell`
`09:41.12`	`frogonwheels`	`edoceo: ok - just to test, how about reverse it...`
`09:41.42`	`frogonwheels`	`edoceo: call a Local/number@context and get it to set the callerid and Dial(SIP/yourmobile)`
`09:42.08`	`edoceo`	`I don't know how to make call from CLI`
`09:42.22`	`edoceo`	`I don't have any devices (SIP or otherwise) connected to this *`
`09:42.24`	`frogonwheels`	`Using a call file.`
`09:42.53`	`frogonwheels`	`you know you can Dial(Local/) ?`
`09:43.08`	`edoceo`	`Hmm, so...`
`09:43.19`	`frogonwheels`	`edoceo: though really, what you've done should work.`
`09:43.58`	`frogonwheels`	`I've had callerid set from a call file work.`
`09:44.29`	`frogonwheels`	`edoceo: here's an idea - connect a SIP softphone to your * and TEST it.`
`09:44.29`	`edoceo`	`So I had this: Channel: IAX2/teliax/12223334444`
`09:44.54`	`edoceo`	`And switch to this: Channel: Local/2065551213@spool`
`09:45.10`	`edoceo`	`And now the stuff in my * console looks more like I want`
`09:45.23`	`*** join/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch)`
`09:45.35`	`frogonwheels`	`sure`
`09:47.23`	`*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)`
`09:47.38`	`mattwj2002`	`well an iax provider sure is easier than skype`
`09:47.45`	`mattwj2002`	`:)`
`09:48.05`	`*** join/#asterisk asim- (n=asim@62.121.28.195)`
`09:48.57`	`asim-`	`Hey guys I'm getting reports of echo and crackling from users on my asterisk iax/sip server. Are there any basic config settings to help me resolve this?`
`09:54.58`	`mattwj2002`	`I got incoming and outgoing calls`
`09:54.59`	`mattwj2002`	`:D`
`09:55.20`	`mattwj2002`	`~800`
`09:55.26`	`edoceo`	`I'm using the right syntax for a caller id: "edoceo <2065551212>"`
`09:55.29`	`mattwj2002`	`~tollfree`
`09:55.49`	`edoceo`	`And I explicitly set callerid before Dialing out my Trunk (IAX2/teliax)'`
`09:56.11`	`edoceo`	`However it's not set when displayed at the called party end of the line :(`
`10:03.19`	`mattwj2002`	`hey guys`
`10:03.34`	`mattwj2002`	`has freeworlddialup changed their homepage?`
`10:05.44`	`mchou`	`lol`
`10:06.00`	`mchou`	`mattwj2002: you still use that crap?`
`10:06.46`	`mattwj2002`	`I thought it would be good for free toll free`
`10:07.06`	`mattwj2002`	`oh nevermind`
`10:07.10`	`mattwj2002`	`they are charging now`
`10:09.35`	`mchou`	`their voice quality was never good to begin with`
`10:10.09`	`mattwj2002`	`any recommendations for free toll free?`
`10:10.19`	`mchou`	`there are better options for toll free termination`
`10:11.23`	`mchou`	`callwithus.com has free toll free term`
`10:12.14`	`mchou`	`it's generally way more stable than tollfreegateway.com`
`10:12.19`	`mattwj2002`	`oh really`
`10:12.25`	`mattwj2002`	`that is my service provider`
`10:12.30`	`mattwj2002`	`I didn't know that`
`10:12.30`	`mattwj2002`	`:d`
`10:14.11`	`mattwj2002`	`no they charge for tollfree`
`10:14.14`	`mattwj2002`	`I just checked`
`10:14.44`	`mchou`	`umm, I get thru with no problems, and I'm not even a customer.....`
`10:15.33`	`mattwj2002`	`I am getting charged 0.0020 usd`
`10:15.44`	`mattwj2002`	`which is 2/10 of a cent`
`10:15.52`	`mchou`	`dude, I'm not even a customer.....`
`10:16.04`	`mchou`	`I dont get charged anything using them`
`10:16.18`	`mattwj2002`	`hmmm`
`10:16.23`	`mchou`	`I dont know what crack pipes you smoking`
`10:17.17`	`mchou`	`they dont even have my demographic info. all they have (perhaps) is my ip address and the 800 numbers that I call`
`10:17.48`	`mchou`	`so they cant exactly be "charging" me`
`10:20.32`	`mattwj2002`	`I don't know how your doing it....but according to my CDR they charged me`
`10:20.34`	`mattwj2002`	`let me try again`
`10:22.49`	`mattwj2002`	`yeah they are definitely charging me`
`10:29.11`	`*** join/#asterisk whynotwhy (n=elastixr@196.211.34.2)`
`10:29.28`	`whynotwhy`	`hi there what does ztdummy do?`
`10:30.44`	`frogonwheels`	`whynotwhy: asterisk requires a timing channel for various operation - including keeping the meetme in sync.`
`10:30.55`	`frogonwheels`	`whynotwhy: it gets this from the Zaptel drivers.`
`10:31.16`	`frogonwheels`	`whynotwhy: if you don't _have_ any ZAP channels, then you can use ztdummy to provide the timing.`
`10:31.19`	`*** join/#asterisk cjk (n=cjk@vodsl-9733.vo.lu)`
`10:31.25`	`frogonwheels`	`~ztdummy`
`10:31.26`	`jbot`	`hmm... ztdummy is a driver that interacts with zaptel to provide a timing source to Asterisk. On 2.4.x kernals, timing is obtained from a UHCI USB controller. It will not work with OHCI controllers. On 2.6.0 and later kernels, the timing is provided by the kernel, thus no hardware is required at all.`
`10:31.34`	`frogonwheels`	`huh that'd be easier.`
`10:31.51`	`cjk`	`hi, i just got a digium dual pri card and i want to put one port into NT mode but i can not find this in the docu. any idea?`
`10:35.34`	`*** part/#asterisk ddl (i=erikw@suiko.acc.umu.se)`
`10:35.53`	`mattwj2002`	`mchou`
`10:36.02`	`mattwj2002`	`can you provide me with how you have it setup?`
`10:36.08`	`mattwj2002`	`for toll free`
`10:36.20`	`mattwj2002`	`I think I have to send it through to tf.callwithus.com`
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`10:39.09`	`mort_gib`	`Hi, I need to create a chan_dahdi.conf with one Digium B410P (Pri card), are the any sample configs??`
`10:39.52`	`kaldemar`	`cjk: there is no NT mode in PRI, but put the other end into CPE mode and the other into NET mode. it's done with the signalling parameter in the config.`
`10:40.51`	`nomad_cz`	`Hi. Is there some comparision of asterisk 1.4 vs asterisk 1.6 ? I do not know which one to choose :/`
`10:41.12`	`kaldemar`	`mort_gib: configs/chan_dahdi.conf.sample in source packages`
`10:41.17`	`cjk`	`kaldemar, ok, so not jumbers or modules parameters... great`
`10:42.56`	`mort_gib`	`kaldemar I have the chan_dahdi.conf (sample) But it's not making a lot of sense to me...`
`10:43.19`	`kaldemar`	`mort_gib: what is unclear?`
`10:43.46`	`kaldemar`	`do you have a concrete problem?`
`10:44.54`	`mort_gib`	`Yeah, I have the card installed, dahdi_scan show the card and the spans, so hardware is installed fine, module loaded`
`10:45.17`	`mort_gib`	`I need to create the chan_dahdi.conf so I can start using it... Last bit`
`10:46.45`	`mort_gib`	`My actual problem is like this...`
`10:47.09`	`mort_gib`	`I have clients in Spain, using a Sangoma A500 card (BRI) and it works really well`
`10:47.12`	`*** join/#asterisk inam (n=root@116.71.215.72)`
`10:47.15`	`*** join/#asterisk tamseel (n=IceChat7@116.71.215.72)`
`10:47.23`	`tamseel`	`hi buddiez`
`10:47.36`	`tamseel`	`i need help`
`10:47.40`	`mort_gib`	`I have clients in Gibraltar that uses the same setup, but they are having some calls dropping`
`10:48.02`	`tamseel`	`i have installed a 64 bit sentos 5.2 on my machine`
`10:48.20`	`tamseel`	`is there some issues`
`10:48.48`	`tamseel`	`about 64bit or should have to migrate to 32 bit`
`10:49.16`	`mort_gib`	`Local telco is unhepful (Surprise) so I'm testing with a Digium card to see if that makes any change`
`10:49.25`	`mort_gib`	`Grasping for straws`
`10:51.15`	`tamseel`	`hellllllllllllooooooooooooooo`
`10:51.32`	`tamseel`	`i need help for 64bit sentos 5.2`
`10:52.04`	`*** join/#asterisk ultrav1olet (n=telnet@94.180.4.213)`
`10:52.07`	`*** part/#asterisk mattwj2002 (n=matt@c-71-63-163-89.hsd1.mn.comcast.net)`
`10:52.21`	`kaldemar`	`tamseel: do you just need a tap on the back or do you have a problem?`
`10:52.24`	`tamseel`	`(h)`
`10:52.30`	`ultrav1olet`	`How can I call the second zap channel of a different asterisk server?`
`10:52.53`	`ultrav1olet`	`without creating an extension`
`10:53.08`	`tamseel`	`well i am configuring new asterisk server`
`10:53.37`	`tamseel`	`so i am now installing 64bit sentos version 5.2 on my i7 server`
`10:53.38`	`ultrav1olet`	`[Jan 29 15:51:20] WARNING[20400]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'login:pass@IP' (cause 66 - Channel not implemented)`
`10:53.42`	`kaldemar`	`mort_gib: i haven't used dahdi nor newer digium BRI cards but with other cards and zaptel it was pretty straight forward the same as with PRI. only that signalling was e.g. bri_cpe`
`10:53.51`	`tamseel`	`so i am not familiar with 64 bit sentos`
`10:54.17`	`kaldemar`	`ultrav1olet: you can't dial anything without creating an extension. show your dialtring, there's something wrong with it.`
`10:54.18`	`*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903)`
`10:54.33`	`frogonwheels`	`ultrav1olet: You probably need ZAP2/`
`10:54.39`	`tamseel`	`so i am asking that is this operating system will create problems in future`
`10:55.31`	`ultrav1olet`	`Dial(iax2/user:pass@192.168.0.1/Zap2,30,r) doesn't work either`
`10:55.57`	`mort_gib`	`kaldemar: Hmmmm`
`10:57.17`	`ultrav1olet`	`if I call a zap/2 extension a wh*re, do you think the owner of that phone will be offended?`
`10:57.38`	`tamseel`	`so what should i do now`
`10:58.39`	`tamseel`	`any body is aware of any known issues with asterisk on 64bit CentOS`
`10:58.44`	`kaldemar`	`ultrav1olet: depends on whether she/he is a wh*re and fine with it. show your dialstring.`
`10:59.33`	`kaldemar`	`frogonwheels: forgetting Zap2/ isn't his problem, you can't pass "login:pass@IP" to a zap channel. and besides, it would be "Zap/2"`
`11:01.39`	`kaldemar`	`ultrav1olet: you can't dial a channel like that, you have to make an extension in the other end that dials Zap/2.`
`11:01.58`	`*** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br)`
`11:02.12`	`ultrav1olet`	`kaldemar: got that, trying to figure out how to implement that`
`11:03.20`	`kaldemar`	`ultrav1olet: in the originating end: Dial(iax2/user:pass@192.168.0.1/123,30,r) - in the receiving end you put exten => 123,1,Dial(Zap/2) in the context where the call lands.`
`11:04.10`	`ultrav1olet`	`kaldemar: I've just done everything like that but it doesn't seem to work`
`11:05.14`	`*** part/#asterisk Avelino (n=Avelino@mail.paterno.com.br)`
`11:05.23`	`kaldemar`	`ultrav1olet: doubt it. show your iax.conf's and extensions.conf's.`
`11:05.25`	`*** join/#asterisk siera08 (n=sosoriri@218.207.141.90)`
`11:05.57`	`*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)`
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`11:19.13`	`*** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56)`
`11:19.48`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`11:23.26`	`frogonwheels`	`s/ZAP2/IAX2/`
`11:23.35`	`frogonwheels`	`but way too late :)`
`11:23.44`	`frogonwheels`	`and probably wrong anyway.`
`11:24.34`	`kaldemar`	`right on that one, but it wasn't the only problem.`
`11:24.47`	`frogonwheels`	`sure.`
`11:25.48`	`mort_gib`	`Trying to load chan_dahdi.conf I get unknown signalling method bri_cpe`
`11:25.55`	`*** join/#asterisk jksM (i=jks@193.189.93.254)`
`11:26.06`	`mort_gib`	`What is the right signalling method for BRI in UK??`
`11:26.20`	`*** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net)`
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`11:31.46`	`kaldemar`	`mort_gib: have you built libpri?`
`11:32.01`	`*** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56)`
`11:32.08`	`mort_gib`	`Do I need libpri for bri??`
`11:32.58`	`kaldemar`	`yes`
`11:35.01`	`*** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com)`
`11:35.32`	`*** join/#asterisk defswork (n=andy@195.58.91.66)`
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`11:40.47`	`mort_gib`	`libpri is installed, was installed`
`11:43.40`	`mort_gib`	`dahdi is not accepting signaling bri_cpe`
`11:43.44`	`defswork`	`I've got a site, E1, Sangoma Pri, that occasionally starts ringing engaged. No errors on wanpipe card, outgoing calls work, and nothing registering anywhere that I can see on incoming calls`
`11:44.28`	`kaldemar`	`mort_gib: asterisk needs to be built with libpri support also, are you sure you compiled them in the right order?`
`11:44.34`	`*** join/#asterisk t3rr1c (n=chatzill@78-105-115-225.zone3.bethere.co.uk)`
`11:44.56`	`mort_gib`	`I can try one more time, libpri-dahdi-asterisk -Right??`
`11:45.05`	`defswork`	`I'm suspecting external issue, but rebooting the box makes it work - although not immediately - appears as if the exchange suddenly realises they are there`
`11:45.36`	`t3rr1c`	`Hi guys. Hoping you can help. I am looking for a list of all the syntax for Lumenvox so I can get the $50 lite kit working. Any ideas where I need to look? Counldn't find on Lumenvox's site`
`11:46.15`	`t3rr1c`	`I am struggling with the names for the already present grammer (the only thing that I am struggling with currently)`
`11:47.30`	`kaldemar`	`mort_gib: dahdi-libpri-asterisk`
`11:47.34`	`whynotwhy`	`hi there i have another question if i may, i have 4 sip trunks witch if the one is busy i want it to overflow to the other ans so on an so on,, the problem is that i can not use ${DIALSTATUS} to route to the second line for i can make more that one call per sip trunk but i want to force it to go to second sip trunk,, ZAP is easy for u got groups,, is there a way to limit the calls over a sip trunk and maybe get a dialstaus of busy then i can route it t`
`11:47.50`	`whynotwhy`	`sorry that was a mouth full, any help welcome`
`11:50.06`	`mort_gib`	`Woud I need misdn??`
`11:51.05`	`frogonwheels`	`whynotwhy: er.. there's a function somewhere to get at how many lines in use for a trunk..`
`11:52.52`	`*** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net)`
`11:53.08`	`jpmcallister`	`whynotwhy: have you tried using call-limit=1 at the sip peer definition`
`11:53.09`	`whynotwhy`	`: frogonwheels: do you know what is called`
`11:53.28`	`frogonwheels`	`whynotwhy: nope.`
`11:53.41`	`frogonwheels`	`jpmcallister: does that help with overflowing?`
`11:54.53`	`whynotwhy`	`thx will give it a go : jpmcallister`
`11:54.57`	`jpmcallister`	`whynotwhy: it should allow a sip peer to receibe only one call.`
`11:55.52`	`kaldemar`	`whynotwhy: take a look at dialplan functions GROUP and GROUP_COUNT`
`11:57.34`	`t3rr1c`	`I am working with Lumenvox. Does anyone have any documentation of how to program this with an Asterisk server?`
`11:59.12`	`*** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk)`
`12:00.37`	`*** join/#asterisk awk (n=awk@security.web.za)`
`12:00.53`	`awk`	`hi, what is the best way to monitor asterisk remotly, eg: pri down, etc.. anything other than nagios or zabbix`
`12:02.18`	`*** join/#asterisk edoceo (n=edoceo@c-98-247-254-241.hsd1.wa.comcast.net)`
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`12:06.17`	`t3rr1c`	`Awk: how remotely? on the LAN or from a totally different location?`
`12:08.01`	`t3rr1c`	`awk: If you for instance use a windows os and want to view all things that are going on through the asterisk box you can use putty (I am using this anyway) via a LAN alternatively if you are somewhere else use the outward facing IP address and forward port 22 to the Asterisk PBX box on the inside of the network`
`12:08.41`	`whynotwhy`	`jpmcallister: thx where did u find that i've been googling and reading anything on sip for 2 days. call-limit=1 and u rule!!!!`
`12:09.48`	`kaldemar`	`whynotwhy: call-limit was deprecated in 1.6 and will be removed in the next release, keep that in mind.`
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`12:13.46`	`jpmcallister`	`whynotwhy: I think I found that on the documention that comes with asterisk source code.`
`12:13.56`	`awk`	`t3rr1c: I need to monitor 500+ pbx's.. need reports the second a pri goes down, etc.`
`12:14.32`	`*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)`
`12:17.46`	`t3rr1c`	`awk: cant help you with that sadly other than suggesting writing some code that monitors the output from admin of each server or something. I am fairly new to all of this and only currently have 2 asterisk units. I have hit a wall trying to make a custom IVR with voice recognition (using Lumenvox) on a Asterisk based trixbox package. I have the package working just have run out of code...`
`12:17.48`	`t3rr1c`	`...examples so am not fully aware of how the grammer feature works in it correctly (as I dont have a list of code to assist other than example projects)`
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`12:31.16`	`virtualme123`	`When you use the manager API to place calls you get a response back as to whether it is successfull or it failed. Are there some cases where the response can be delayed and in the mean time the call get placed?`
`12:34.58`	`*** part/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch)`
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`12:40.36`	`*** part/#asterisk ultrav1olet (n=telnet@94.180.4.213)`
`12:41.20`	`virtualme123`	`Anyone used the Manager API to place calls?`
`12:43.28`	`jpmcallister`	`awk: Asterisk has support to SNMP. Maybe you can use that and a monitor like Nagios to constantly monitor asterisk`
`12:43.37`	`*** join/#asterisk HeMan (n=jimmy@193.12.106.19)`
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`12:44.33`	`awk`	`jpmcallister looks like the best option i have`
`12:44.46`	`jpmcallister`	`awk: Maybe it's possible to configure asterisk to send traps through its snmp agent`
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`13:17.12`	`vncsnvs`	`a2billing or astbill?`
`13:17.57`	`vncsnvs`	`userlist \| grep a`
`13:18.01`	`vncsnvs`	`lol`
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`13:22.44`	`*** join/#asterisk Khratos (n=khratos@190.166.103.146)`
`13:23.31`	`Khratos`	`Good morning`
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`13:41.13`	`whynotwhy`	`hello`
`13:41.21`	`whynotwhy`	`people her rock`
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`13:41.35`	`whynotwhy`	`HERE`
`13:56.23`	`ruben23`	`hi`
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`13:57.45`	`*** mode/#asterisk [+o russellb] by ChanServ`
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`14:00.38`	`*** join/#asterisk Cheetah (n=Cheetah@main-gw.bense.de)`
`14:00.43`	`Cheetah`	`hey guys`
`14:00.58`	`Cheetah`	`we've got an asterisk server with a bunch of Snom 360s in our company`
`14:01.26`	`Cheetah`	`and lots of our workers are enableing "redirect after timeout" or "always redirect"`
`14:01.36`	`Cheetah`	`is there a way to disable that feature for an incoming call?`
`14:06.35`	`*** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com)`
`14:06.46`	`*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)`
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`14:13.41`	`Khratos`	`Maybe not, because that's something that is being set directly on the phone`
`14:13.51`	`Khratos`	`I have had that incident too`
`14:14.53`	`*** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com)`
`14:15.36`	`kaldemar`	`Cheetah: sure there are ways to prevent that in the asterisk side, but they're agly as hell. do it on the phone side.`
`14:16.26`	`Khratos`	`On Asterisk side you would have to deal with headers`
`14:16.59`	`Cheetah`	`hmm`
`14:17.14`	`Cheetah`	`we have a number of phones that take incoming calls`
`14:17.38`	`Cheetah`	`i figure that if I put those into a queue and use ringall, asterisk would not redirect a call if a phone tells it to?`
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`14:18.54`	`kyper`	`hi. quick question regarding using asterisk and a spa400`
`14:20.10`	`kyper`	`I'm trying to dial #21# out to the spa400 on one of the lines. THe problem is that the spa400 stop dialing when it gets to a #. Asterisk is actually sending the correct number via sip (SIP/SPA400/#21#)`
`14:20.15`	`kyper`	`any ideas?`
`14:20.42`	`Khratos`	`Cheetah, I have not tested that yet, but I think that the redirection will occur anyways`
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`14:21.53`	`*** join/#asterisk awannabe (n=brad@ip70-162-201-97.ph.ph.cox.net)`
`14:22.29`	`awannabe`	`hello all, on cisco 79xx phones the media ports should be mapped to the same setting asterisk is mapped to, correct?`
`14:25.51`	`*** join/#asterisk clintc (n=clintc@n128-227-185-136.xlate.ufl.edu)`
`14:27.56`	`Khratos`	`Are you facing audio problems awannabe ?`
`14:28.37`	`awannabe`	`Khratos, the phones work most of the time, but they do get some jitter. everyone else on the same proxy has no call quality issues, and we are connected via a P2P T1 to the carrier`
`14:29.15`	`awannabe`	`so it appears its not the connecton or the carrier. It just like random, only thing else I can think of is the RTP is wrong and causing spurastic problems`
`14:30.58`	`Khratos`	`Is the problem present when using other phones?`
`14:31.15`	`Khratos`	`Or xLites softphones (just to be sure) ?`
`14:31.45`	`awannabe`	`Polycom's seem to be fine, the customer says they "all" do it, but you know how that goes. from what I have seen its only this cisco'`
`14:32.26`	`awannabe`	`I have faxes that are working fine over the same circuit, so the carriers and all that seem fine, using straight up ulaw (on a GXW4024 gateway)`
`14:35.02`	`Khratos`	`uuumm... Then it would be reasonable to make changes on Cisco port settings`
`14:35.48`	`awannabe`	`in the configs I never set the ports, and then i remember that asterisk doesnt use the 16xxx to 32xxx by default for RTP ports`
`14:36.03`	`awannabe`	`I havent setup any Cisco's in years, so I kind of spaced that, heh`
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`14:38.55`	`Khratos`	`I think that asterisk uses a different range than the one you mentioned, let me confirm`
`14:39.13`	`Khratos`	`Yes, on rtp.conf I see`
`14:39.16`	`Khratos`	`rtpstart=10000`
`14:39.17`	`Khratos`	`rtpend=20000`
`14:39.24`	`mort_gib`	`kaldemar: Are you still here??`
`14:39.36`	`awannabe`	`Khratos, yeah default is 10000 to 20000`
`14:39.38`	`Khratos`	`At least on 1.4.X`
`14:39.45`	`awannabe`	`yeah, thats what made me think of that`
`14:40.56`	`Khratos`	`Maybe the 16000 - 32000 range sometimes does not match with the 10000 - 20000 that Asterisk uses, and the jitter problem is present (just a theory)`
`14:41.13`	`Khratos`	`But accoring to Murphy's Law, it can happend`
`14:41.28`	`Khratos`	`And surely will happend to the Client's phone a lot of time`
`14:41.50`	`eppigy`	`hello`
`14:43.00`	`mort_gib`	`THe correct install for Asterisk is Dahdi-linux - Dahdi-tools libpri Asterisk -RIght??`
`14:43.35`	`*** join/#asterisk axisys (n=axisys@155.70.141.45)`
`14:43.52`	`awannabe`	`Khratos, yeah, thats what im hoping, gonna try and see if it helps at all`
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`14:45.06`	`Khratos`	`Yes, give it a try. I know what does it feels when a client generalizes the problem and says that it 'allways' happend`
`14:45.17`	`awannabe`	`yeah, gotta love that, makes troubleshoot IMPOSSIBLE`
`14:45.27`	`awannabe`	`just like they said their fax didnt work, have to let it ring more then once lol`
`14:46.30`	`Khratos`	`haha, I thought that clients on my country were the only ones that behaves like that`
`14:46.53`	`stix_`	`I am trying to make snom-phones update its settings from a global URL. Can anyone give me a hint on how my Asterisk can respond with an URL when a phone sends a SIP SUBSCRIBE?`
`14:49.51`	`awannabe`	`Khratos, heck no! that's one thing I think we ALL have in common`
`14:50.45`	`mort_gib`	`I need a little help to install a B410P Digium card`
`14:51.40`	`tzafrir_laptop`	`mort_gib, dahdi?`
`14:52.06`	`virtualme123`	`Does anyone know if this is still a bug - http://bugs.digium.com/view.php?id=8286 because I believe I had this problem recently ...`
`14:52.13`	`mort_gib`	`Dahdi installs fine, I install libpri after Dahdi but I get "unknown signalling method 'bri_cpe'" When I try to load Dahdi`
`14:52.34`	`tzafrir_laptop`	`what version of asterisk and libpri?`
`14:52.37`	`mort_gib`	`tzafrir_laptop: I have seen your posts`
`14:52.49`	`mort_gib`	`latest, libpri 1.4.7`
`14:53.14`	`tzafrir_laptop`	`seems chan_dahdi was built without libpri support`
`14:53.32`	`mort_gib`	`Dahdi-linux-complete (2.1.0.3+2.1.0.2)`
`14:53.33`	`tzafrir_laptop`	`ldd /usr/lib/asterisk/modules/chan_dahdi.so \| grep libpri`
`14:53.53`	`virtualme123`	`It suggests that channels can get stuck in an error loop if there is a network read error ...`
`14:54.53`	`mort_gib`	`libpri.so.1.4 => /usr/lib/libpri.so.1.4 (0x00c49000)`
`14:55.13`	`mort_gib`	`I tried it all over again a few times`
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`14:55.55`	`awannabe`	`Khratos, ahh yes, default ports are 16384 to 32766 on the Cisco. yeah that could be a big problem`
`14:56.40`	`mort_gib`	`tzafrir_laptop: I would really appreciate help on this one....`
`14:57.14`	`Khratos`	`wow`
`14:57.54`	`Khratos`	`You could increase the asterisk ending rtpport limit`
`14:58.07`	`Khratos`	`So it includes the ones that phone uses`
`14:58.30`	`Khratos`	`And you don't have to touch phones settings`
`14:58.53`	`tzafrir_laptop`	`mort_gib, hmm.. asterisk 1.4 or 1.6 ?`
`14:59.38`	`mort_gib`	`Asterisk 1.4.22`
`15:01.23`	`tzafrir_laptop`	`hmm... I'm not sure if it is supported in 1.4.22 (that is: before 1.6.0)`
`15:03.26`	`mort_gib`	`So I should go for Zaptel`
`15:03.45`	`mort_gib`	`I get Dahdi show channels in i1.4.22`
`15:03.59`	`mort_gib`	`but only if Dahdi loads`
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`15:06.01`	`stix_`	`How do I provide this configuration on my Asterisk: http://wiki.snom.com/wiki/index.php/Settings/pnp_config ?`
`15:06.46`	`jaytee`	`changing something if features.conf requires a full restart of asterisk?`
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`15:32.05`	`lirakis`	`hey everyone`
`15:33.32`	`lirakis`	`ive got a peer with a dynamic IP, so i've set up the peer host as the FQDN and i've enabled refresh lookups dnsmgr.conf - however asterisk seems to never refresh the ip of the peer`
`15:34.28`	`*** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com)`
`15:34.29`	`lirakis`	`ive read that "chan_sip doesnt support dns refresh" ... but i dont know if this is still the case, and if so - is there any way to get asterisk to not cache the dns lookup of the peers fqdn`
`15:35.12`	`oej`	`Not really.`
`15:35.27`	`oej`	`Why don't you let the peer register?`
`15:35.38`	`*** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net)`
`15:37.01`	`lirakis`	`oej: .. the peer is for outbound termination (aka carrier) ...`
`15:37.05`	`*** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-f4ec3e596b05b1ed)`
`15:38.34`	`lirakis`	`oej: there are actually multiple termination points (ip's) and the company has you terminate to a domain, which uses dns to randomize which gateway you go to`
`15:39.13`	`lirakis`	`oej: so ... id like for asterisk to stop caching the dns result on startup, and instead refresh it every once in a while`
`15:39.19`	`oej`	`A carrier on dynamic IP? That's very strange`
`15:39.32`	`lirakis`	`oej: its not really dynamic as ive described`
`15:39.35`	`Chesther`	`Sounds more like round-robin than dynamic`
`15:39.37`	`oej`	`I got that now...`
`15:39.56`	`oej`	`Well, according to SIP we should stay at the first choice until it fails.`
`15:40.11`	`lirakis`	`oej: and it does fail...`
`15:40.16`	`lirakis`	`oej: but asterisk doesnt update`
`15:40.41`	`lirakis`	`oej: which is why i want it to refresh to dns every once in a while`
`15:40.44`	`*** join/#asterisk seanmh (n=johndoe@216.31.95.99)`
`15:41.32`	`oej`	`Well, there has been work in trunk and some of the 1.6 releases to make that better`
`15:42.46`	`lirakis`	`oej: hmm okay... do you have any info on where that refresh happens? im suprised its chan specific`
`15:42.54`	`lirakis`	`oej: i mean where in the source`
`15:43.18`	`oej`	`No, not really. Sorry.`
`15:43.39`	`oej`	`It's some magic in the dnsmanager that kevin worked with at a stay at an airport and some stuff in chan_sip very recently`
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`15:46.25`	`lirakis`	`oej: yeah i just saw dns.c and dnsmgr.c in main/ .... ill poke around in chansip ... not sure if i can do anything... but it sure would be nice to be able to do this`
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`15:48.44`	`beherit-`	`i have two * registrar with dundi on it, if the user 1001 is registered to * server 1 and something happen to register 1 is their a way to for the user 1001 to still make a call using the secondary *?`
`15:49.00`	`beherit-`	`!`
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`15:49.06`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
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`15:50.23`	`FabiOne`	`hi all`
`15:53.00`	`ajohnson`	`Hmmm found an Asterisk crash, but I'm not sure I want to admit to finding it`
`15:53.32`	`ajohnson`	`When you send a manager redirect action with a channel and extrachannel set to the same channel name and leave out a priority... Asterisk crashed`
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`15:56.31`	`*** join/#asterisk bmoraca (n=bmoraca@209.60.253.58)`
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`15:57.52`	`merlin8282`	`Hi`
`15:59.01`	`merlin8282`	`I got a problem again with my asterisk installation, in fact ISDN installation. I have a QuadBRI ISDN PCI card in my * server, 2 ports in TE mode and the 2 other in NT mode.`
`15:59.30`	`merlin8282`	`All works, but ISDN : i have one ISDN phone, and it doesn't have any current.`
`15:59.47`	`merlin8282`	`The phone works, i plugged it directly to the line, it's ok.*`
`16:00.33`	`merlin8282`	`I tried this one : plug the ISDN line into the TE port : the LED goes green, but the port on which the phone is keeps red.`
`16:00.49`	`merlin8282`	`(NT port, for the phone)`
`16:01.00`	`merlin8282`	`Anybody has an idea ?`
`16:04.41`	`kyper`	`hi. anyone any good with spa400?`
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`16:06.10`	`*** join/#asterisk manxpower (n=Administ@router.asteriasgi.com)`
`16:07.01`	`merlin8282`	`Is the Junghanns card supposed to give (so to say "transfer") the power to the NT ports when a line is connected to a TE port ?`
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`16:19.00`	`thansen`	`I'm looking for some help with chan_mobile...I'm running mobile search and getting.. All Bluetooth adapters are in use at this time. ...is this good?`
`16:19.25`	`rue_work`	`hmm cant help there`
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`16:21.35`	`rue_work`	`merlin8282, never worked with that card before, is the channeltype comming up in asterisk when you say core show channeltypes`
`16:23.08`	`merlin8282`	`rue_work: it should be this one : Zap DAHDI Telephony Driver w/PRI no yes no`
`16:23.14`	`merlin8282`	`although it's a BRI line`
`16:24.57`	`rue_work`	`ok`
`16:25.22`	`rue_work`	`your sure system.conf is right?`
`16:26.13`	`merlin8282`	`what is system.conf ? Don't see such file in /etc/asterisk/`
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`16:28.28`	`keebler`	`bmoraca: Had another WIFI theory/question.`
`16:28.29`	`merlin8282`	`The * server is running well, with FreePBX, its analog phones and its SIP phones. The only problem is this ISDN phone (got only one).`
`16:29.00`	`keebler`	`bmoraca: What if I did like you originally said, did a Canopy system then just had Clients + ATA's.`
`16:29.18`	`keebler`	`Or would the latency be too high?`
`16:29.35`	`bmoraca`	`keebler: probably not any higher than 802.11`
`16:29.59`	`keebler`	`hmm.`
`16:30.00`	`bmoraca`	`keebler: however, that solution is overkill. you're looking at $15k per oil rig under a canopy system.`
`16:30.13`	`keebler`	`bmoraca: I know. Haha.`
`16:30.22`	`keebler`	`bmoraca: Just wondering if it would work.`
`16:30.25`	`bmoraca`	`keebler: when i made that recommendation, i misunderstood your requirements`
`16:30.42`	`*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)`
`16:31.17`	`keebler`	`bmoraca: Well. What about a pseudocanopy? Using just 802.11a hardware?`
`16:31.25`	`bmoraca`	`keebler: your best solution is Cisco Aironet. 1300 series for 2.4ghz or 1400 series for 5.8ghz. 1300 is what i would go with, personally. point to multipoint bridges with their high-gain omni antenna will get you ~.21 miles.`
`16:31.49`	`keebler`	`hmm`
`16:32.00`	`Sargun`	`Does anyone know of any VoIP carriers that do custom phone numbers?`
`16:32.38`	`*** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com)`
`16:32.39`	`awannabe`	`Sargun, custom?`
`16:33.13`	`Sargun`	`Like, not in the carrier's standard phone space...`
`16:33.15`	`Sargun`	`etc..`
`16:33.33`	`awannabe`	`like a vanity number?`
`16:33.46`	`awannabe`	`like 123-SIPLOVE`
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`16:38.20`	`mort_gib`	`I'm trying to get a Digium B410P card to work`
`16:38.51`	`mort_gib`	`But I'm having loads of issues with Dahdi, not loading bri_cpe signalling`
`16:39.24`	`mort_gib`	`I HAVE installed libpri so what do I do short of starting all over with the OS again??`
`16:39.44`	`mort_gib`	`Or is the problem that the cards are not yet plugged in??`
`16:39.58`	`kyper`	`is there a way to escape the first # so that # is dialled on the trunk?`
`16:40.24`	`beherit-`	`i have two * registrar with dundi on it, if the user 1001 is registered to * server 1 and something happen to register 1 is their a way to for the user 1001 to still make a call using the secondary *?`
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`16:46.55`	`Sargun`	`awannabe, basically.`
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`16:47.37`	`awannabe`	`Sargun, got a ideal of the number?`
`16:48.34`	`Sargun`	`I know the number I want`
`16:48.39`	`Sargun`	`it wasn't allocated before`
`16:53.15`	`*** join/#asterisk riksta (n=rick@office.encompassmedia.co.uk)`
`16:57.22`	`keebler`	`This might sound really naive, but is there any "Asterisk Certification/Training" classes?`
`16:57.45`	`russellb`	`yup`
`16:57.50`	`russellb`	`grabs a link`
`16:57.59`	`Sargun`	`keebler, http://www.digium.com/en/training/`
`16:58.06`	`russellb`	`keebler: http://www.digium.com/en/training/`
`16:58.07`	`russellb`	`awww`
`16:58.08`	`russellb`	`i lose`
`16:58.15`	`Sargun`	`LOSER.`
`16:58.19`	`russellb`	`:-(`
`16:58.53`	`*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)`
`16:59.28`	`*** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-1-150.w90-56.abo.wanadoo.fr)`
`16:59.47`	`*** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net)`
`17:08.45`	`keebler`	`Anyone order parts from www.wlanparts.com?`
`17:11.56`	`*** join/#asterisk Defraz (n=T0tal@72-24-26-22.cpe.cableone.net)`
`17:21.02`	`*** join/#asterisk nix8n82 (n=nate@63.162.27.243)`
`17:26.42`	`*** join/#asterisk rbd_ (n=rbd@rrcs-96-10-27-206.se.biz.rr.com)`
`17:26.58`	`rbd_`	`does format_mp3 give asterisk mp3 playback support for AGI's STREAM FILE command?`
`17:29.55`	`russellb`	`yes`
`17:30.06`	`rbd_`	`sweet, thanks`
`17:30.31`	`*** join/#asterisk n3hxs (n=HAMming@63.68.135.4)`
`17:42.01`	`*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)`
`17:43.49`	`*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)`
`17:55.04`	`*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)`
`18:00.11`	`*** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net)`
`18:01.53`	`*** join/#asterisk killown (n=Yamato@unaffiliated/killown)`
`18:11.29`	`*** join/#asterisk oej (n=olle@ns.webway.se)`
`18:15.25`	`*** join/#asterisk StephenF (n=none@198.144.201.106)`
`18:16.10`	`StephenF`	`Is there a way to warn a caller when recording a voicemail that the message length is nearing the limit, or atleast explain the limit has been reached instead of just hanging up?`
`18:28.01`	`*** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis)`
`18:31.15`	*** join/#asterisk Pro` (n=matt@dyn34108.demon.co.uk)
`18:33.27`	`rue_work`	`hmm so merlin didn't know about system.conf in /etc/dahdi`
`18:34.59`	`*** join/#asterisk dandan (n=dandan@yarde-GW.customer.alter.net)`
`18:35.04`	`dandan`	`hello`
`18:35.30`	`dandan`	`hey guys, does anyone have a TE405p for sale?`
`18:35.36`	`dandan`	`The Revision A card`
`18:35.39`	`dandan`	`must be Rev A.`
`18:39.09`	`edoceo`	`I have TDM400P`
`18:40.01`	`dandan`	`awesome, but no, thanks`
`18:40.23`	`dandan`	`I will pay good money for any number of those, must be Rev A 405p, please ask around`
`18:41.32`	`hardwire`	`Anybody have User-1/2 rings on SPA-94x? I can set the Alert-Info header to choose any ring except those two.`
`18:41.33`	`rwaite`	`what would cause a call going through an iax provider to "reach" the pstn destination, but no audio to be there?`
`18:41.34`	`hardwire`	`it's annoying`
`18:44.08`	`manxpower`	`rwaite: NAT`
`18:45.23`	`manxpower`	`hardwire: can you set those rings in the config for the SPA and do they work?`
`18:45.58`	`hardwire`	`manxpower: you have to submit a URL to the phone to set the ringtone`
`18:46.02`	`hardwire`	`which it sucks from a tftp server`
`18:46.14`	`manxpower`	`hardwire: I bet the ringtone does not work at all`
`18:46.15`	`hardwire`	`it set's it into the phone nicely.. but grr.. you can't seem to use it on demand.`
`18:46.21`	`hardwire`	`manxpower: it works fine.`
`18:46.29`	`manxpower`	`that was my question`
`18:46.43`	`manxpower`	`you're not doing something stupid like using quotes are you?`
`18:46.44`	`hardwire`	`my sleep level is low late.y`
`18:46.52`	`thehar`	`hardwire: mmhm`
`18:46.56`	`thehar`	`anyone have success with asterfax?`
`18:46.59`	`hardwire`	`manxpower: I can set it to any other ringtone`
`18:47.08`	`keebler`	`Okay. I've got two bridges same subnet. A phone on each bridge. with the asterisk server on the host AP. I can call each phone, but I cannot get any audio. If I move both of the phones to one client bridge and cut out the second client bridge, I can talk between phones. Any idea as to why I'm not getting audio on separate client bridges?`
`18:47.09`	`manxpower`	`hardwire: get some sleep`
`18:47.13`	`hardwire`	`manxpower: no`
`18:47.38`	`hardwire`	`anyways, so you've not done it?`
`18:48.10`	`manxpower`	`I was just trying to eliminate any obvious issues`
`18:48.50`	`hardwire`	`wprd`
`18:48.51`	`hardwire`	`word`
`18:49.03`	`hardwire`	`gah! linksys changed their site layout!`
`18:51.13`	`hardwire`	`nice .. fanc new firmware for spa-942`
`18:51.32`	`edoceo`	`What was the code to strip MSD from my EXTEN? ${EXTEN:1-} ??`
`18:51.43`	`hardwire`	`s/-//`
`18:51.54`	`hardwire`	`no... I was correcting edoceo`
`18:52.04`	`hardwire`	`don't assume jbot! you know what happens when you assume!`
`18:52.23`	`edoceo`	`hardwire: ${EXTEN:1} will give MSD or strip MSD?`
`18:52.26`	`hardwire`	`s/assume!/ass-out-of-u-and-me!/`
`18:52.30`	*** part/#asterisk Pro` (n=matt@dyn34108.demon.co.uk)
`18:52.40`	`edoceo`	`hahah`
`18:52.45`	`hardwire`	`edoceo: there's only one way to find out :)`
`18:52.55`	`hardwire`	`including lots of docs :)`
`18:52.55`	`edoceo`	`finds out`
`18:53.19`	`edoceo`	`but then I'd have to google search for at least three minutes!`
`18:53.43`	`hardwire`	`http://www.voip-info.org/wiki/view/Asterisk+variables#Substrings`
`18:53.53`	`hardwire`	`shaves a few minutes off`
`18:54.02`	`hardwire`	`edoceo: place your hand on voip-info.org.`
`18:54.09`	`edoceo`	`See - folks like you make it so easy for me to slack!`
`18:54.17`	`hardwire`	`Do you solemnly swear to use this wiki to your advantage and learn it's ways?`
`18:54.44`	`edoceo`	`I'm on voip-info all the time - just lazy - thanks!`
`18:54.57`	`edoceo`	`buys hardwire virtual beer`
`18:55.14`	`edoceo`	`Hmm - is virtual beer like VB (Victoria Bitter)?`
`18:56.14`	`*** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net)`
`18:57.18`	`hardwire`	`edoceo: virtual beer is useless to me`
`18:57.30`	`*** join/#asterisk oej (n=olle@ns.webway.se)`
`19:10.38`	`*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)`
`19:15.58`	`*** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net)`
`19:19.33`	`*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net)`
`19:21.21`	`*** join/#asterisk RoyK (n=roy@ip-192-59-149-91.dialup.ice.no)`
`19:22.06`	`RoyK`	`hi. my fs box sends an options message (OPTIONS sip:sip.radiomeloy.no;transport=udp SIP/2.0) to an asterisk box, and the asterisk box answers 404`
`19:22.10`	`RoyK`	`http://pastebin.com/m54a7863`
`19:22.13`	`RoyK`	`any idea why?`
`19:23.44`	`*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)`
`19:24.23`	`citats`	`RoyK: without looking too deep it looks like the OPTIONS are sent to sip:sip.radiomeloy.no instead of sip:royk@sip.radiomeloy.no`
`19:25.57`	`*** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net)`
`19:27.29`	`RoyK`	`citats: it seems to be trying to pass the options package onto the dialplan`
`19:28.23`	`[TK]D-Fender`	`RoyK: * responds to OPTIONS (use for Qualify NAT keep-alive).`
`19:28.37`	`[TK]D-Fender`	`RoyK: the fact it answers is important, not the response itself`
`19:28.57`	`RoyK`	`well, here it answers 404`
`19:29.08`	`[TK]D-Fender`	`RoyK: thats jsut what * does`
`19:29.22`	`[TK]D-Fender`	`RoyK: it isn't a "negative" response per-se`
`19:29.25`	`RoyK`	`it tries to find stdsip,s,1`
`19:29.35`	`RoyK`	`to look there for something`
`19:29.47`	`RoyK`	`404 is negative - not found`
`19:29.57`	`RoyK`	`as opposed to 200`
`19:30.18`	`[TK]D-Fender`	`RoyK: Feel free to write a patch for chan_sip to change the response :)`
`19:30.41`	`RoyK`	`I was just wondering, really. I'm done spending time in that codebase`
`19:30.42`	`RoyK`	`thanks`
`19:30.43`	`*** part/#asterisk RoyK (n=roy@ip-192-59-149-91.dialup.ice.no)`
`19:32.20`	`*** join/#asterisk path_ (n=path@188-103-21-190.adsl.terra.cl)`
`19:34.37`	`dlewis`	`[TK]D-Fender: I'm going to install OSLEC. I currently have zaptel 1.4.3 and it'll be tough to install the current version of dahdi... Are there any major changes that would make it mandatory to install dahdi in order to get OSLEC to work properly?`
`19:34.37`	`*** join/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy)`
`19:35.20`	`jaytee`	`this sounds like a 180 degree twist from rue_mohr`
`19:35.30`	`*** join/#asterisk killown (n=Yamato@unaffiliated/killown)`
`19:36.04`	`*** join/#asterisk Micho123 (n=mcho123@77.42.179.31)`
`19:36.30`	`Micho123`	`Hi all, I'm getting the following notice when trying to send a FAX..rtp.c: Unknown RTP codec 100 received from 'GW_IP'`
`19:36.49`	`*** join/#asterisk dfkl (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net)`
`19:36.53`	`dfkl`	`hi there`
`19:37.27`	`*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)`
`19:37.27`	`*** mode/#asterisk [+o russellb] by ChanServ`
`19:37.56`	`punkgode`	`hello...maybe someone can help me on this. Can a call queue be transfered by a local channel agent to another queue?`
`19:38.11`	`punkgode`	`The local agent its not a real agent, just a point in a dialplan that checks a few things and sends the call to another queue.`
`19:43.09`	`dfkl`	`i just got a small question, i try to configure " dial plan " on my linksys pap2 ATA i live in France to make a local call i do this 0122334455 from my linksys ATA and to make a international i have to 00442334455667 for example`
`19:44.25`	`dfkl`	`now i want to shift to london with my linksys ata and i don't to do each 00442334455667 from london to make a local`
`19:45.35`	`dfkl`	`what dialplan i have to configure on my ATA to use simply compose 02334455667 from UK to make a local call ?`
`19:45.45`	`*** join/#asterisk killown (n=Yamato@unaffiliated/killown)`
`19:45.59`	`*** join/#asterisk fexy (n=fexy@208.3.217.29)`
`19:46.42`	`*** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it)`
`19:47.00`	`dfkl`	`thx to cooporate`
`19:47.05`	`fexy`	`I'm running asterisk with chan-sccp-b v3 (for realtime support). I want to find the most elegant way to add an entry in to my mysql server when a phone attempts to register with the server.`
`19:47.08`	`fexy`	`Any thoughts?`
`19:47.45`	`fexy`	`I could poll a file in /var/log looking for the SEPs and add them to mysql when I find them, but that seems a bit hackish`
`19:48.11`	`fexy`	`Or if someone has done this already I would be very interested in that. No need to reinvent the wheel! :D`
`19:49.11`	`dfkl`	`hello`
`19:49.17`	`dfkl`	`is there anyone here ?`
`19:49.40`	`Corydon76-dig`	`nope`
`19:50.06`	`Gido-E`	`no`
`19:50.21`	`dfkl`	`i can't understand the notion of the dialplan`
`19:50.31`	`Corydon76-dig`	`~thebook`
`19:50.32`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`19:50.47`	`dfkl`	`read a lot not understand`
`19:50.57`	`dfkl`	`need pratic i m here`
`19:51.00`	`*** join/#asterisk JerJer (n=PhatJ@24-236-207-64.dhcp.aldl.mi.charter.com)`
`19:51.01`	`dfkl`	`need pratice i m here`
`19:51.03`	`Corydon76-dig`	`dfkl: If you pick up a phone and dial a number, what happens?`
`19:51.36`	`dfkl`	`i join the end user`
`19:51.39`	`kaldemar`	`i still can't tell if he's trying to configure asterisk or just the linksys ATA.`
`19:51.46`	`Corydon76-dig`	`dfkl: without the dialplan, nothing happens`
`19:51.58`	`Corydon76-dig`	`dfkl: dialplan is what determines what happens`
`19:52.40`	`*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)`
`19:52.40`	`*** mode/#asterisk [+o russellb] by ChanServ`
`19:52.47`	`dfkl`	`yes that's right`
`19:52.57`	`dfkl`	`now what i just trying to do is`
`19:52.59`	`*** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw)`
`19:53.04`	`*** join/#asterisk hmmhesays (n=hmmhesay@97-114-181-38.farg.qwest.net)`
`19:53.11`	`dfkl`	`i just register with an sip provider`
`19:53.24`	`dfkl`	`and i m trying my ata`
`19:53.35`	`dfkl`	`and i m trying to configure my ata`
`19:53.58`	`dfkl`	`as i said before`
`19:54.49`	`dfkl`	`from france if i dial 0122334455 i able to make to make a local`
`19:54.51`	`dfkl`	`call`
`19:54.55`	`JerJer`	`anyone know of a way to change the caller*id of a call that is being transfered to another exten?`
`19:55.16`	`Gido-E`	`JerJer change CALLERID(num)`
`19:55.16`	`bkw_`	`JerJer: attended transfers?`
`19:55.21`	`JerJer`	`they want the calling party number of the call being transfered`
`19:55.31`	`JerJer`	`not the exten transferring the call`
`19:55.41`	`JerJer`	`bkw_: both types would be nice`
`19:55.49`	`dfkl`	`if i try to do same thing from london for example to make a local call 02334455667 for example not`
`19:55.52`	`dfkl`	`working`
`19:55.52`	`bkw_`	`thats a fun one...`
`19:56.02`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)`
`19:56.10`	`JerJer`	`Gido-E: i know that, but explain how you do it for just transfers ? : )`
`19:56.16`	`dfkl`	`i have to do 00442334455667`
`19:57.03`	`kaldemar`	`my all time favorite paratemer in asterisk is/was "useincomingcalleridonzaptransfer".`
`19:57.41`	`dfkl`	`what i want to do is : i simply want to compose 02334455667 as make a local from london`
`19:57.42`	`Gido-E`	`before transfer set an VAR?`
`19:57.49`	`*** join/#asterisk killown (n=Yamato@unaffiliated/killown)`
`19:57.52`	`dfkl`	`that"s all`
`19:58.17`	`dfkl`	`so what dial plan i have to put on my ata ?`
`19:58.39`	`*** join/#asterisk jsolis (n=jimmy@190.41.153.85)`
`19:58.49`	`manxpower`	`my fave was always randomlydisconnectmycalls=yes It's an alias for callprogress=yes`
`19:59.55`	`rob0`	`anyone here from Digium training? Have y'all considered offering self-study+IRC courses? And then the student could make an appointment for the hands-on testing and certification.`
`20:00.56`	`Qwell`	`rob0: I do believe you can take just the dCAP by itself`
`20:01.42`	`*** join/#asterisk tomcontr3 (n=gcontrer@186-85-20-190.adsl.terra.cl)`
`20:01.52`	`rob0`	`yeah, but I think I could benefit from the advanced course`
`20:02.01`	`Qwell`	`then take the advanced course. heh`
`20:02.16`	`tomcontr3`	`hi... Im having some problems with a SIP trunk that I hired.`
`20:02.17`	`tomcontr3`	`here is the log`
`20:02.37`	`Gido-E`	`tomcontr3 :-)`
`20:02.37`	`rob0`	`4 days in HSV, would rather do 3 days at home and one day at HSV. :)`
`20:02.40`	`tomcontr3`	`http://pastebin.ca/1322073`
`20:03.05`	`rob0`	`Besides, I've never been very impressed with the classroom education model.`
`20:03.33`	`rob0`	`And $3k is a bunch of bux for an unemployed trailer trash bum.`
`20:04.31`	`jaytee`	`the Advanced Course wasn't bad`
`20:04.54`	`*** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net)`
`20:04.58`	`jaytee`	`they just need to add more "Advanced" stuff to it.`
`20:05.30`	`dlewis`	`Qwell: I'm going to install OSLEC. I currently have zaptel 1.4.3 and it'll be tough to install the current version of dahdi... Are there any major changes that would make it mandatory to install dahdi in order to get OSLEC to work properly?`
`20:05.51`	`tomcontr3`	`any idea?`
`20:05.52`	`kaldemar`	`tomcontr3: [Jan 29 16:52:56] NOTICE[6558] chan_sip.c: No compatible codecs, not accepting this offer!`
`20:06.02`	`dfkl`	`jklyuio`
`20:06.13`	`*** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net)`
`20:06.14`	`tomcontr3`	`yep, I read that... but what codecs... I mean I hace the G729`
`20:06.17`	`tomcontr3`	`have`
`20:06.39`	`tomcontr3`	`and the G723`
`20:07.15`	`kaldemar`	`tomcontr3: the caller tells in the SDP that he speaks G.729 and G.723. on the other hand, "Capabilities: us - 0x0 (nothing)". have you allowed codecs in sip.conf?`
`20:07.41`	`tomcontr3`	`allow=g729`
`20:08.41`	`jaytee`	`and on the phone too, not just in sip.conf`
`20:09.57`	`tomcontr3`	`It has also support`
`20:10.54`	`kaldemar`	`i don't see that one reaching the phone yet. pastebin your sip.conf so you'll get an extra pair of eyes.`
`20:11.32`	`tomcontr3`	`ok`
`20:15.39`	*** join/#asterisk becks` (n=sdfgsfdg@169-244.104-92.cust.bluewin.ch)
`20:15.51`	`tomcontr3`	`http://pastebin.ca/1322084`
`20:16.33`	becks`	`hi, is it possible that when i call a number, asterisk first sends some DTMF and then sends a re-invite so i'm connected to that number?`
`20:16.47`	`kaldemar`	`tomcontr3: you have a disallow=all under [Redvoice1]. remove it.`
`20:17.28`	`jaytee`	`tomcontr3, yes disallow always comes BEFORE an allow= statement`
`20:17.41`	`jaytee`	`you're crippling your phone in sip.conf`
`20:18.01`	`jaytee`	`now might be a good time to review some documentation?`
`20:18.04`	`jaytee`	`~book`
`20:18.05`	`jbot`	`hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook`
`20:19.43`	`tomcontr3`	`but the only codec that the provider allow me to use is the G729... thats why I use disallow all`
`20:20.35`	`kaldemar`	`yes, but by using disallow=all BELOW allow=g729, you also disallow g729. codec lines are read top-down.`
`20:23.19`	`jaytee`	`which if you read the book it's spelled out pretty clearly`
`20:25.44`	`tomcontr3`	`I see... thanks!`
`20:25.46`	`kaldemar`	`tomcontr3: it doesn't matter if you allow more than just one codec, the codec negotiation in the call setup takes care of choosing a codec that both have. and besides, in the invite the provider tells that you also can use G.723.`
`20:27.45`	`*** join/#asterisk Micho123 (n=mcho123@77.42.150.82)`
`20:30.43`	`beek`	`afternoon jaytee`
`20:30.59`	`jaytee`	`hi beek`
`20:31.41`	`Micho123`	`Hi alll, I'm trying to send a FAX thru asterisk..I got the following error...rtp.c: Unknown RTP codec 100 received from 'GW address'...I tried to add the following to rtp.c and compile Asterisk [100] = {1, AST_FORMAT_H100},...Got the following error during make...http://pastebin.com/d56fc1963...any advice is really appreciated`
`20:32.46`	`Micho123`	`soory the erroris located on http://pastebin.com/d42354770`
`20:33.15`	`*** join/#asterisk boynas (n=garyflor@wsip-98-190-136-194.ph.ph.cox.net)`
`20:33.18`	`Micho123`	`anybode face such issue befor?`
`20:34.14`	`boynas`	`I was looking for something like QueueMetrics to get calling reports. Anybody knows a open source replacement or alternative to this?`
`20:40.57`	`JerJer`	`anyone see why this isn't workin? exten => s.n,GotoIf($[${ISNULL(${BLINDTRANSFER})}]?retrieve,1)`
`20:41.14`	`JerJer`	`i print a noop and then that goto:`
`20:41.24`	`manxpower`	`yes. you have an extra .`
`20:41.24`	`JerJer`	`<PROTECTED>`
`20:41.24`	`JerJer`	`<PROTECTED>`
`20:41.28`	`manxpower`	`between the s and the n`
`20:42.10`	`manxpower`	`JerJer: try ${LEN} instead. it may be empty no null`
`20:42.11`	`JerJer`	`copy/paste error in here (i typed the everything up until the Gotoif...`
`20:42.14`	`manxpower`	`not null, that is`
`20:43.16`	`Micho123`	`manxpower, [100] = {1, AST_FORMAT_H100} is supported by asterisk?`
`20:43.40`	`manxpower`	`GotoIf(${LEN(${BLINDTRANSFER})} = 0]?retrieve1)`
`20:43.47`	`manxpower`	`Micho123: never heard of it`
`20:44.14`	`kaldemar`	`manxpower: yours is missing $[`
`20:44.24`	`manxpower`	`kaldemar: goo catch`
`20:44.26`	`twisted`	`haha, YEAH MANXPOWER`
`20:44.26`	`Micho123`	`manxpower, I guess [100] = {1, AST_FORMAT_H263}...This could work when using T.38?`
`20:44.32`	`kaldemar`	`and a comma, but maybe that's irrelevant with JerJer.`
`20:44.59`	`manxpower`	`Micho123: I know nothing about T.38 or codec 100`
`20:45.19`	`twisted`	`t.38 doesn't use h263`
`20:45.22`	`twisted`	`h263 is video`
`20:46.03`	`Micho123`	`manxpower, I'm getting the following error when trying to send a FAX...rtp.c: Unknown RTP codec 100 received from 'GW address'`
`20:46.11`	`Micho123`	`manxpower, what do you suggest?`
`20:46.12`	`twisted`	`and generally speaking, in SDP, codec mapping of 100 is an alternate dtmf`
`20:46.24`	`twisted`	`micho: get a sip debug and look at the sdp message`
`20:46.31`	`JerJer`	`twisted: woot woot`
`20:46.34`	`twisted`	`sup jerjer`
`20:46.34`	`kaldemar`	`Micho123: http://www.asteriskguru.com/tutorials/unknown_codec_received.html`
`20:47.18`	`JerJer`	`twisted: keeping a customer happy :)`
`20:47.37`	`twisted`	`jerjer: hehe, cool`
`20:48.05`	`manxpower`	`Micho123: I suggest you try to find someone that can help you.`
`20:48.25`	`manxpower`	`Rather than just assuming I know everything in the universe about Asterisk`
`20:48.28`	`Micho123`	`manxpower, WhT ABOUT YOU?:)`
`20:48.53`	`*** join/#asterisk Cheetah (n=Cheetah@main-gw.bense.de)`
`20:48.58`	`twisted`	`haha`
`20:49.16`	`jjshoe`	`manxpower get your root password reset?`
`20:49.19`	`manxpower`	`I know QUITE a bit about the features I USE.`
`20:49.31`	`manxpower`	`jjshoe: yup. boot to single user mode`
`20:49.42`	`manxpower`	`I don't know much about the features I don't use.`
`20:50.02`	`*** join/#asterisk ^conner (n=conner@rma.ifa.hawaii.edu)`
`20:50.23`	`^conner`	`Hi Foks, is it possible to completely disable all voicemail? I can't see to figure out a global config option`
`20:50.31`	`manxpower`	`I'm starting to learn some AEL2 now.`
`20:50.43`	`manxpower`	`^conner: GUIs are not supported here.`
`20:50.56`	`^conner`	`manxpower, that's fortunate as I'm not using one`
`20:51.07`	`^conner`	`manxpower, however there doesn't seem to be an option in voicemail.conf`
`20:51.08`	`twisted`	`likes AEL`
`20:51.10`	`manxpower`	`^conner: then remove all instances of "Voicemail" from your dialplan`
`20:51.13`	`twisted`	`haven't played with AEL2 yet`
`20:51.22`	`^conner`	`manxpower, well that's the issue, there aren't any`
`20:51.46`	`citats`	`^conner: add noload => app_voicemail.so to your modules.conf`
`20:51.58`	`manxpower`	`twisted: Not not as easy as I thought to convert from what I call "stone skipping programming" i.e. use Gotos to "the right way" i.e. use other methods.`
`20:52.05`	`^conner`	`manxpower, I have some polycom 501 phones and it seems like after some many rings they refuse the call and somehow it's ending up in voicemail although there is no way to reach it via the dailplan`
`20:52.09`	`twisted`	`manxpower: hehe`
`20:52.18`	`twisted`	`manxpower: you should see my old standard dialplan lol`
`20:52.20`	`^conner`	`citats, i'll give that a try, thanks`
`20:52.35`	`dandan`	`hey guys, does anyone have a TE405p for sale? Rev. A? Thank you. :)`
`20:52.39`	`citats`	`^conner: though if your calls are ending up in voicemail then somewhere you must have it in your dialplan`
`20:52.51`	`manxpower`	`^conner: watch the Asterisk CLI`
`20:53.18`	`kaldemar`	`dandan: stay away from those if possible. go for rev c or d if you want old cards.`
`20:53.56`	`^conner`	`citats, grep -i voicemail extensions.conf`
`20:53.56`	`dandan`	`kaldemar: I know it, but I have about 12 *, 0.8 or pre-1 with an old zaptel that I can't touch`
`20:53.59`	`^conner`	`citats, returns nothing`
`20:54.02`	`dandan`	`I need those very cards...`
`20:54.10`	`^conner`	`citats, so there is clearly some default behavior at play here`
`20:54.13`	`Qwell`	`dandan: ...upgrade`
`20:54.23`	`manxpower`	`^conner: No. There. Isn't.`
`20:54.32`	`dandan`	`Qwell: unfortunately, that is for tomorrow, for today - I need those cards...`
`20:54.36`	`manxpower`	`^conner: Asterisk does NOTHING unless you configure it to.`
`20:54.38`	`citats`	`^conner: maybe you include some other config file in your extensions.conf?`
`20:54.42`	`kaldemar`	`dandan: oh, with such old asterisks they do work. but with newer ones you might experience funny hangs.`
`20:54.43`	`manxpower`	`^conner: something else is going on`
`20:54.44`	`dandan`	`I am even willing to buy the newest rev. and swap them with rev. a`
`20:54.54`	`^conner`	`manxpower, I know something else is going on ;) I just don't know what`
`20:54.54`	`dandan`	`if anyone has 'em`
`20:55.12`	`^conner`	`can sip devices request Voicemail specifically?`
`20:55.26`	`Qwell`	`dandan: This is precisely why you should be upgrading.`
`20:55.27`	`dandan`	`manxpower: is 1234 still a default extension in [demo]? :)`
`20:55.34`	`manxpower`	`^conner: yes, but it won't work if it's not in the dialplan`
`20:55.50`	`dandan`	`Qwell: Grandfathered systems, can't do it right this moment, but in the next 12-18 months - for sure`
`20:55.56`	`^conner`	`manxpower, it's most definately not in the dialplan`
`20:56.10`	`Qwell`	`You should have upgraded several YEARS ago.`
`20:56.18`	`^conner`	`manxpower, or in any macro's invoked from the dialplan`
`20:56.28`	`dandan`	`Qwell: THEY, not me :)`
`20:56.33`	`^conner`	`manxpower, nor is "Voicemail(" used anywhere under /etc/asterisk`
`20:56.57`	`dandan`	`^conner: call, and use pastebin to show us what is going on on the CLI`
`20:57.03`	`citats`	`^conner: like manxpower said I'd check the console output`
`20:57.13`	`^conner`	`ok, i'm gonna try that now`
`20:57.20`	`^conner`	`any particular debug needed?`
`20:57.26`	`dandan`	`Qwell: I am getting desperate, as I said: I am willing to trade a brand spanking new card for a Rev. A...`
`20:57.38`	`dandan`	`^conner: connect with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvr`
`20:57.41`	`*** join/#asterisk telecos (n=sergio@204.166.219.87.dynamic.jazztel.es)`
`20:57.42`	`citats`	`^conner: just verbose at least 4 or so`
`20:57.44`	`dandan`	`and call in`
`21:00.34`	`manxpower`	`dandan: I've not used the default config is 5 years`
`21:00.48`	`*** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)`
`21:00.50`	`manxpower`	`dandan: ask on the asterisk-users mailing list`
`21:03.17`	`bmoraca`	`does verbose above 5 actually change what you can see on the console without enabling any other feature-specific debugging? i don't think i've ever noticed a difference.`
`21:03.41`	`dandan`	`manxpower: lol, no need`
`21:04.20`	`dandan`	`bmoraca: i think 6 is max`
`21:04.30`	`dandan`	`although I haven't checked the source in a looong time`
`21:04.36`	`*** join/#asterisk aatmaa_ (i=aatma@118.103.237.101)`
`21:04.57`	`*** join/#asterisk UQlev (n=kvirc@91.184.220.73)`
`21:05.05`	`manxpower`	`I meant the Rev A card`
`21:05.14`	`dandan`	`manxpower: ah!`
`21:05.16`	`dandan`	`will do`
`21:05.17`	`dandan`	`thx`
`21:05.26`	`dandan`	`(did ebay, Clist and linkedin already)...`
`21:06.05`	`manxpower`	`dandan: I expect there would be many people wanting to get rid of Rev A cards.`
`21:06.17`	`*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)`
`21:06.18`	`manxpower`	`Digium doesn't have a great history of getting hardware right the first time.`
`21:06.43`	`dandan`	`manxpower: the funny thing is they have not a single Rev. A card in stock!`
`21:06.50`	`dandan`	`I was sold the last one they had...`
`21:07.00`	`manxpower`	`dandan: I meant the used marke`
`21:07.02`	`manxpower`	`t`
`21:07.25`	`^conner`	`manxpower, citats http://pastebin.com/m19d75291`
`21:07.27`	`dandan`	`right: neither used, nor for reference, not even techies...`
`21:07.38`	`*** join/#asterisk Micho123 (n=mcho123@77.42.150.82)`
`21:08.15`	`manxpower`	`^conner: whatever device is connected to Zap/1 is pretending to be a voicemail system`
`21:08.31`	`manxpower`	`I have a vintage T400P`
`21:08.49`	`kyper`	`hi`
`21:08.58`	`dandan`	`need quad-T card... not the quad FXO/S...`
`21:09.13`	`kyper`	`I have a questions regarding a SPA400 and asterisk. anyone familiar with both?`
`21:09.24`	`manxpower`	`T400P is a T-1`
`21:09.25`	`^conner`	`manxpower, is there some sort of default for zaptel devices then?`
`21:09.27`	`manxpower`	`TDM400P is analog`
`21:09.31`	`outtolunc`	`dandan, the T400P is a 4 port t1/pri card`
`21:09.37`	`manxpower`	`^conner: what is plugged into zap/1`
`21:09.49`	`^conner`	`manxpower, $10 analog phone from walmart`
`21:09.53`	`*** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com)`
`21:09.59`	`dandan`	`manxpower: oh!....`
`21:10.06`	`manxpower`	`^conner: and zap/4 is the telco?`
`21:10.08`	`dandan`	`hm, are you able to send me a pic of it?`
`21:10.21`	`^conner`	`manxpower, ya, that's the pots line, this is a tiny remote site`
`21:10.23`	`manxpower`	`dandan: It locks up about once per week.`
`21:10.32`	`manxpower`	`^conner: you have the ports reversed.`
`21:10.37`	`manxpower`	`port 1 is the telco`
`21:10.47`	`*** part/#asterisk dfkl (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net)`
`21:10.54`	`*** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com)`
`21:11.12`	`^conner`	`manxpower, uh, no I don't`
`21:11.17`	`manxpower`	`The top port is port 1`
`21:11.18`	`dandan`	`manxpower: keep it for me, I will try asterisk-users, if that fails, we will talk business`
`21:11.28`	`manxpower`	`dandan: I didn't say I wanted to sell it.`
`21:11.35`	`dandan`	`trade it?`
`21:11.36`	`dandan`	`:0`
`21:11.37`	`manxpower`	`I was given the card because it was flaky`
`21:11.37`	`dandan`	`:)`
`21:11.56`	`^conner`	`manxpower, i would think it rather matters more where the fxo and fxs modules are installed on the card`
`21:12.08`	`jaytee`	`flaky is good but only when talking about a croissant`
`21:12.13`	`^conner`	`manxpower, and seeing as you can actually dial into the system and iit's been working for 2 years`
`21:12.25`	`manxpower`	`^conner: not really. If you plug the wrong line into the wrong port you just blow the ports and have to buy new ones.`
`21:12.45`	`^conner`	`manxpower, the only problem is that you end up in voicemail if nobody answers`
`21:12.52`	`rwaite`	`what could cause an iax call to answer but then be silent?`
`21:12.57`	`citats`	`^conner: your pastebin says app_voicemail. so if something is playing a voicemail announcement it isnt asterisk`
`21:13.06`	`boynas`	`I was looking for something like QueueMetrics to get calling reports. Anybody knows a open source replacement or alternative to this?`
`21:13.07`	`citats`	`^conner: er app_voicemail is unloaded`
`21:13.25`	`manxpower`	`rwaite: NAT`
`21:13.35`	`manxpower`	`^conner: not on the asterisk system you don't`
`21:13.41`	`jaytee`	`or a deaf mute on the other end`
`21:14.21`	`rwaite`	`manxpower: i thought iax didnt have that issue?`
`21:14.24`	`manxpower`	`^conner: BTW, in the future run at verbosity 3 when you do pastebina`
`21:14.39`	`manxpower`	`rwaite: any time you run a server behind nat you have to forward a port.`
`21:14.48`	`rwaite`	`i am, thought`
`21:14.51`	`rwaite`	`though*`
`21:14.53`	`^conner`	`ok, maybe somebody changed the phone to something with voicemail`
`21:14.56`	`manxpower`	`"In this house we obey the laws of thermodynamics!`
`21:14.57`	`^conner`	`let me call the remote site`
`21:15.28`	`jaytee`	`wonders what you hear if you call the home phone of Marcel Marceau`
`21:15.36`	`Sargun`	`SS7 Question -> 1) How does switch A know to send the signals to switch be for subscriber B? 2) How are the routing tables for the STPs built? ( http://www.iec.org/online/tutorials/ss7/topic07.asp )`
`21:15.43`	`manxpower`	`rwaite: IAX2 has fewer/easier to fix NAT issues than SIP, but you still have to obey the laws of NAT`
`21:16.28`	`kyper`	`does anyony know how to get a number like #21# to be dialed on a FXO port?`
`21:16.54`	`kyper`	`im sending sip/PSTN/#21# where PSTN is my ATA gateway`
`21:17.24`	`^conner`	`manxpower, citats LOL - you were right, somebody had purchased a fancy panasonic cordless phone`
`21:17.37`	`manxpower`	`^conner: we usually are.`
`21:18.01`	`twisted`	`it's martini time`
`21:18.04`	`rwaite`	`manxpower: but the port is forwarded? that's my major misunderstanding, it seems to be forwarding fine but only some calls will "connect" and then be silent and hang up`
`21:18.07`	`citats`	`kyper: look at the D option for app_dial`
`21:18.11`	`rwaite`	`maybe the router is just screwing up the nat`
`21:18.22`	`rwaite`	`or maybe its my provider, i dont know today sucks.`
`21:18.23`	`manxpower`	`rwaite: or firewall`
`21:18.46`	`rwaite`	`manxpower: if it were firewall i would expect it to happen consistently`
`21:19.28`	`kyper`	`citats: im sending it to a SPA400 device.`
`21:20.10`	`citats`	`kyper: and your using Dial to send the call there right?`
`21:21.26`	`kyper`	`citats: im using trixbox. A custom extension that's dialing sip/PSTN/#21# so i assume it's using dial`
`21:21.57`	`jaytee`	`Trixbox? LEPER!!! OUTCAST!!! UNCLEAN!!!!`
`21:22.18`	`manxpower`	`A GUI? Think of the children!`
`21:22.27`	`jaytee`	`no one was spared`
`21:22.38`	`kyper`	`im just fixing a friends business pbx and it runs trixbox`
`21:22.51`	`jaytee`	`friends don't let friends......`
`21:22.55`	`citats`	`kyper: so look into the D option to dial. I couldn't tell you how to get the options onto it on trixbox though`
`21:23.11`	`jaytee`	`shoe horn, some duct tape and prayer?`
`21:23.44`	`citats`	`jaytee: might need some bungee cord too`
`21:23.57`	`citats`	`or at least some rubber bands`
`21:24.21`	`jaytee`	`and a left handed phillips screwdriver`
`21:24.24`	`kyper`	`citats: so it's the D option for Dial() ?`
`21:24.40`	`Kobaz`	`rwaite: check your rtp port range, i had a problem where the rtp port range was 1000-1500, but the firewall was only allowing 1000-1050`
`21:24.48`	`*** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk)`
`21:24.51`	`Kobaz`	`rwaite: so some calls would work, and others would get silence`
`21:25.32`	`^conner`	`manxpower, citats thanks guys`
`21:26.49`	`rwaite`	`i thought iax didnt use rtp?`
`21:28.02`	`*** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net)`
`21:28.03`	`kyper`	`citats: so Dial(sip/PSTN, 20,D(#21#))`
`21:28.12`	`kyper`	`citats: is that the correct syntax?`
`21:28.51`	`Kobaz`	`rwaite: oh if it's iax then, just make sure the iax ports are open`
`21:29.12`	`rwaite`	`heh... thats what i thought...`
`21:30.17`	`thansen`	`anyone have some pointer on how to get chan_mobile SMS working with an android phone?`
`21:30.45`	`thansen`	`I've got a voice connection, I just want SMS functionality`
`21:31.27`	`rwaite`	`so iax passes the audio through 4569 too?`
`21:32.30`	`kyper`	`citats: I seem to have gotten somewhere. Thank you :)`
`21:33.31`	`Kobaz`	`rwaite: yeah`
`21:33.34`	`*** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net)`
`21:33.40`	`rwaite`	`this is confusing`
`21:36.27`	`*** join/#asterisk mik3 (n=mike@c-67-175-50-184.hsd1.il.comcast.net)`
`21:44.49`	`*** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com)`
`21:45.31`	`CrashSys`	`Anyone know what's involved to get asterisk 1.2 to install with zaptel 1.4`
`21:46.14`	`tzafrir_laptop`	`Rebuild of Asterisk`
`21:46.28`	`CrashSys`	`zap show status doesn't list anything`
`21:46.42`	`tzafrir_laptop`	`zaptel 1.4 is binary incompatible but source-level compatible with zaptel 1.2`
`21:47.16`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`21:55.13`	`citats`	`kyper: glad I could help. sorry I had to step away to put the munchkin down for a nap`
`21:59.24`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`22:04.51`	`*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)`
`22:06.14`	`*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)`
`22:10.32`	`CrashSys`	`well damn, that sucks, i'm having a problem with 1.4 not setting the channel to hung-up and causing my AGI's to become defunct`
`22:15.31`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`22:15.31`	`*** mode/#asterisk [+o denon] by ChanServ`
`22:17.51`	`*** join/#asterisk HermesNeto (n=HermesNe@189-94-11-208.3g.claro.net.br)`
`22:33.34`	`adr\|an`	`hi there`
`22:33.45`	`*** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net)`
`22:40.47`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)`
`22:43.28`	`eppigy`	`hello [TK]D-Fender`
`22:43.45`	`[TK]D-Fender`	`eppigy: you are dave`
`22:44.11`	`manxpower`	`Never EVER add extra spaces to Asterisk config files`
`22:48.55`	`*** join/#asterisk Greek-Boy (n=greek@41.222.89.77)`
`22:49.13`	`Greek-Boy`	`Has anybody successfully setup asterfax? Does it require iaxmodem to function?`
`22:54.38`	`*** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110)`
`22:57.06`	`*** join/#asterisk voxter (n=voxter@76.77.95.2)`
`22:57.59`	`voxter`	`So, pressing '' kills audio in a call on this one box I have, even though I changed the disconnect code from to *0. What gives?`
`23:01.02`	`manxpower`	`Greek-Boy: Do you have H or h as the Dial options`
`23:01.15`	`manxpower`	`sorry, that was for voxter`
`23:01.32`	`voxter`	`negative.`
`23:01.43`	`voxter`	`it doesnt hang up, it just stops passing audio`
`23:04.06`	`*** join/#asterisk michaely (n=Mike@207.114.199.107)`
`23:07.00`	`adr\|an`	`hi there`
`23:07.04`	`adr\|an`	`i just instaled the asterisk`
`23:07.06`	`adr\|an`	`and`
`23:07.09`	`adr\|an`	`i got :`
`23:07.09`	`adr\|an`	`[Jan 29 11:50:52] WARNING[11777]: pbx_spool.c:476 scan_thread: Unable to stat /var/spool/asterisk/outgoing`
`23:07.18`	`adr\|an`	`what is wrong ?`
`23:07.24`	`michaely`	`I'm writing an application that utilizes the AMI. I noticed that when sending a channel to the parking lot via a blind transfer and the ParkAndAnnounce command the "from" field is reported on the AMI event, Parked Cal,l as null. Is there aye way i can get this information to report correctly by some type of dial plan modification?`
`23:08.21`	`voxter`	`adr\|an: mkdir /var/spool/asterisk/outgoing`
`23:09.23`	`*** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk)`
`23:17.22`	`adr\|an`	`voxter : thank you`
`23:19.54`	`Greek-Boy`	`how does one check which modules are FXS and which are FXO through the console?`
`23:20.16`	`*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)`
`23:22.41`	`*** join/#asterisk stream (n=stream@72.22.21.62)`
`23:23.03`	`stream`	`what is the differece between asterisk and trixbox`
`23:23.31`	`frogonwheels`	`~trixbox`
`23:23.33`	`jbot`	`it has been said that trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it.`
`23:23.45`	`stream`	`sweet`
`23:23.48`	`frogonwheels`	`~FreePBG`
`23:23.51`	`frogonwheels`	`~FreePBX`
`23:23.51`	`jbot`	`[~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`
`23:24.16`	`frogonwheels`	`pats the jbot.`
`23:26.02`	`*** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp)`
`23:27.50`	`Deeewayne`	`~beer`
`23:27.51`	`jbot`	`ACTION has disconnected (Read error: 99 (Connection reset by beer))`
`23:28.09`	`frogonwheels`	`very funny`
`23:28.58`	`*** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis)`
`23:29.43`	`stream`	`i'm trying to setup asterisk for the first time, been reading the manul.. on page 137 :) i hve a SIP trunk and a Soft phone`
`23:30.25`	`Qwell`	`~die`
`23:30.26`	`jbot`	`ACTION takes two shots to the head and crumples to the ground, lifeless.`
`23:30.56`	`Deeewayne`	`he doesn't seem to like beer`
`23:32.03`	`frogonwheels`	`stream: going good so far?`
`23:32.23`	`stream`	`so far so good.. trying to get the SIP trunks working for inbound/outbound calling..`
`23:32.53`	`stream`	`can I put a register => line in sip.conf?`
`23:33.05`	`frogonwheels`	`stream: That's exactly what I was going to make sure you have done.`
`23:33.12`	`frogonwheels`	`stream: yep - that's where it goes.`
`23:33.16`	`stream`	`ok`
`23:33.18`	`stream`	`at the top?`
`23:33.51`	`frogonwheels`	`stream: in the [general] section - which should be at the top.`
`23:33.55`	`stream`	`ok`
`23:34.38`	`stream`	`what ports do i have to forward in the firewall. only 5060? udp`
`23:35.07`	`frogonwheels`	`stream: nope. you possibly gotta forward the RTP stream as well`
`23:35.40`	`frogonwheels`	`stream: unless (I think) you've got the proper SIP/RTP masquerading in the firewall..`
`23:35.44`	`frogonwheels`	`sias`
`23:35.49`	`frogonwheels`	`suck -it and see`
`23:36.04`	`stream`	`that works ok b/c i have another VOIP pbx that works with this sip provider`
`23:36.50`	`frogonwheels`	`stream: cool.`
`23:36.51`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`23:36.53`	`*** join/#asterisk watchy2 (n=watchy@76.196.98.139)`
`23:36.59`	`stream`	`its not opensource though`
`23:37.10`	`watchy2`	`looks like we are gonna put in about 50 of those cyberdata paging speakers`
`23:37.16`	`watchy2`	`they seem pretty neat`
`23:38.38`	`stream`	`should i setup my provider as SIP or IAX`
`23:38.59`	`frogonwheels`	`if it's a SIP provider, then it's SIP.`
`23:39.06`	`stream`	`they support both`
`23:39.21`	`frogonwheels`	`I think IAX is better then.`
`23:39.24`	`watchy2`	`i think iax would be best`
`23:39.32`	`*** join/#asterisk incoherence_ (n=gnucrack@98.108.208.63)`
`23:39.32`	`frogonwheels`	`stream: apart from anything, it plays nice with firewalls.`
`23:39.47`	`stream`	`so the config would be different ?`
`23:39.51`	`watchy2`	`yes`
`23:40.57`	`*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)`
`23:41.34`	`stream`	`ok so how do i setup my softphone to dial out of the sip trunk`
`23:42.22`	`watchy2`	`setup your softphone to connect to *`
`23:42.23`	`incoherence_`	`question for any/all: i'm running asterisk on a i386-compatible with a sis7019 audio ctrl/dsp (only have driver for oss, not alsa). i can do SIP calls just fine, audio works etc. i can play sounds outside of asterisk through /dev/dsp with mpg123. but using console dial to an extension that just plays tt-monkeys, there's no audio. any ideas/pointers?`
`23:42.31`	`watchy2`	`and on outgoing calls use your iax provider`
`23:42.42`	`stream`	`not sure how to config that just yet.. brb`
`23:43.20`	`frogonwheels`	`stream: Get your softphone to connect to * - make a dummy context that just plays a sound file.`
`23:43.29`	`stream`	`softphone is connected`
`23:43.30`	`stream`	`[Jan 29 18:43:03] NOTICE[29209]: chan_iax2.c:8647 socket_process: Rejected connect attempt from 192.168.5.20, request '8007672775@phones' does not exist`
`23:43.32`	`frogonwheels`	`stream: there are examples in the default extensions.conf`
`23:43.42`	`*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)`
`23:43.56`	`stream`	`i set`
`23:43.57`	`stream`	`[outgoing]`
`23:43.58`	`stream`	`exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)`
`23:44.20`	`drfreeze`	`Hello`
`23:44.25`	`manxpower`	`stream: that is for outgoing calls, not incoming calls`
`23:44.39`	`manxpower`	`(the exten line)`
`23:44.40`	`stream`	`iim working on outgoing first`
`23:44.47`	`manxpower`	`the message you pasted is for phone -> asterisk`
`23:45.18`	`manxpower`	`the phone has a context=phones in it's sip.conf entry and you don't have anything that matches 8007672775`
`23:45.34`	`drfreeze`	`I'm trying to debug the following: exten => s,n,_X./8885551212,1,Goto(...), because I can't get it to match the callers number`
`23:46.02`	`drfreeze`	`I am printing out the CALLERID(num) and CALLERID(all) and it looks like it should match`
`23:46.05`	`manxpower`	`drfreeze: All extensions start with a priority 1`
`23:46.16`	`drfreeze`	`manxpower: hmm, ok. let me try that`
`23:46.29`	`frogonwheels`	`drfreeze: or pastebin the entire context.`
`23:46.35`	`manxpower`	`extensions that do not start with a priority one are ignored`
`23:47.04`	`frogonwheels`	`drfreeze: presumably you have s,1,NOOP($CALLERID(num)) s,n,_X./8885551212,1,Goto()`
`23:47.40`	`frogonwheels`	`drfreeze: presumably you have s,_X.,1,NOOP($CALLERID(num)) s,_X./8885551212,n,Goto()`
`23:47.47`	`frogonwheels`	`drfreeze: actually look at that line - it's screwed.`
`23:48.14`	`frogonwheels`	`drfreeze: it's soooo confused.`
`23:48.31`	`frogonwheels`	`exten => _X./8885551212,1,Goto(...)`
`23:48.50`	`frogonwheels`	`drfreeze: is _that what you mean?`
`23:49.01`	`stream`	`how do i know what extension my client got`
`23:49.04`	`stream`	`softphone client`
`23:49.19`	`frogonwheels`	`stream: it doesn't "Get" an extension.`
`23:49.43`	`frogonwheels`	`stream: you add an extension into your local dialplan context that calls the phone given a particular extension`
`23:49.55`	`frogonwheels`	`stream: work through some more examples in the book.`
`23:51.27`	`stream`	`looks like the firewall is setup right b/c the SIP calls are coming into the system`
`23:51.32`	`drfreeze`	`frogonwheels: this is what I get: http://pastie.textmate.org/private/xfb4seyvu8r9x0g2h3m20g`
`23:52.28`	`stream`	`[Jan 29 18:52:05] NOTICE[29219]: chan_sip.c:16968 handle_request_invite: Failed to authenticate user "4126933356" <sip:4126933356@64.2.142.13>;tag=as69b8de80`
`23:52.31`	`frogonwheels`	`drfreeze: arghh.. you're coming in on an 's' extension. not expecting that.`
`23:52.49`	`frogonwheels`	`exten => s/8885551212,1,Goto(...)`
`23:53.33`	`frogonwheels`	`exten => s/8885551212,n,Goto(...)`
`23:53.49`	`frogonwheels`	`actually - otherwise that line will replace the Answer() when it matches.`
`23:54.01`	`frogonwheels`	`.. and you won't get your NoOps happening.`
`23:54.57`	`stream`	`ok i'm stuck`
`23:55.26`	`stream`	`i dont see where i bind my softphone to use the SIP trunk for outbound calls`
`23:55.49`	`*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`23:55.49`	`*** mode/#asterisk [+o lmadsen] by ChanServ`
`23:56.06`	`*** part/#asterisk michaely (n=Mike@207.114.199.107)`
`23:56.19`	`lmadsen`	`Asterisk 1.6.1-rc1 is now available for testing! http://www.asterisk.org/node/48563`
`23:56.34`	`drfreeze`	`frogonwheels: ok. working on the next step now`
`23:58.57`	`*** join/#asterisk grantm (n=grant@68.142.138.4)`

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