irclog2html for #asterisk on 20061101

`00:01.43`	`backblue`	`gedia: yes`
`00:02.26`	`backblue`	`command_: you cant connect a T1 line to a regular network card, offcourse`
`00:02.28`	`[TK]D-Fender`	`command_ Depends what kind of router....`
`00:02.40`	`backblue`	`T1 it's not ethernet.`
`00:03.35`	`[TK]D-Fender`	`command_As for your question about why that tutorial didn't reference any cards, its because it deosn't REQUIRE any. Its a pure-voip solution`
`00:03.35`	`C6Vette`	`having weird things in show channels: http://pastebin.ca/231721`
`00:03.39`	`C6Vette`	`what would cause that?`
`00:07.21`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`00:08.30`	`AsteriskMonkey`	`any of the digium guys around?`
`00:08.39`	`file`	`perhaps, why?`
`00:08.54`	`AsteriskMonkey`	`Ah file.. i wanted to ask about the development boxes i saw on the digium website`
`00:09.01`	`file`	`the AADK?`
`00:09.01`	`AsteriskMonkey`	`they look like there a 4fxo unit`
`00:09.10`	`AsteriskMonkey`	`is that what its called?`
`00:09.19`	`AsteriskMonkey`	`what does it have 16megs ram or something?`
`00:09.21`	`file`	`Asterisk Appliance Developer Kit...`
`00:09.25`	`file`	`the full spec sheet is on the website`
`00:09.31`	`AsteriskMonkey`	`how much are they?`
`00:09.51`	`file`	`the unit itself is not being sold yet`
`00:10.20`	`AsteriskMonkey`	`shoot`
`00:10.27`	`AsteriskMonkey`	`i need 400 units :(`
`00:10.37`	`*** join/#asterisk Dovid (n=Dovid@85.159.160.207)`
`00:10.40`	`Dovid`	`~book`
`00:10.44`	`jbot`	`hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`00:10.48`	`backblue`	`omg, hoo needs 400 units of that? :o`
`00:11.01`	`*** join/#asterisk wacky_ (n=wacky@modemcable188.232-131-66.mc.videotron.ca)`
`00:11.04`	`AsteriskMonkey`	`file: if i order a large amount of them can I distriubte them?`
`00:11.06`	`backblue`	`who`
`00:11.12`	`file`	`I am not in sales.`
`00:11.14`	`wacky_`	`Hey, guys..`
`00:11.35`	`backblue`	`AsteriskMonkey: you have shipper products, with more hardware i think.`
`00:11.49`	`AsteriskMonkey`	`shipper?`
`00:12.09`	`backblue`	`ship?`
`00:12.27`	`AsteriskMonkey`	`backblue : " you have shipper products," what the heck does that mean?`
`00:12.40`	`backblue`	`sorry my english`
`00:12.57`	`wacky_`	`I recently upgraded from 1.1 (mdk) to 1.2.12, and I was wondering where the colors in the console went, and also the live details of which channel was doing what..`
`00:13.26`	`backblue`	`AsteriskMonkey: shipper, i tryed to say, not so expensive.`
`00:13.33`	`wacky_`	`is there a way to get that back ?`
`00:14.19`	`*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)`
`00:14.33`	`wacky_`	`do you guys still have colos in the console ?`
`00:14.52`	`AsteriskMonkey`	`na i wanna support digium, its not about cheaper its about helping an industry that supports yourself prosper`
`00:16.09`	`command_`	`[TK]D-Fender: whats wrong with pure vo-ip solutions?`
`00:16.44`	`[hC]`	`Anyone dicked around with using the acoustic echo suppression on polycom phones, or the built in QoS?`
`00:16.50`	`Dovid`	`TK: can u have a look at thos ?`
`00:16.51`	`Dovid`	`http://lists.digium.com/pipermail/asterisk-users/2006-October/171010.html`
`00:16.52`	`command_`	`in the end i should still be able to place outgoing calls to numbers across the world, im just paying the gateway servie for each call`
`00:17.17`	`backblue`	`AsteriskMonkey: well, maybe it's not just like you think.`
`00:17.48`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:19.17`	`Dovid`	`TK: can u look at the link ?`
`00:19.41`	`*** join/#asterisk CSD-199 (n=asdf@189.158.173.11)`
`00:19.44`	`CSD-199`	`hi`
`00:20.08`	`[TK]D-Fender`	`command_ : I never suggested there was anything wrong with it. You asked why that sample guide didn't sue a card. I answered it for you.`
`00:20.24`	`Dovid`	`hello CSD`
`00:21.08`	`[TK]D-Fender`	`Dovid : Bother are behind NAT?`
`00:21.21`	`CSD-199`	`i'm crazy connecting 2 asterisk servers, using IAX, but for some reasson, I can not do it. It display a mssg: exited non-zero`
`00:21.22`	`file`	`[TK]D-Fender: we can be heroes!`
`00:21.24`	`CSD-199`	`why?`
`00:21.37`	`Qwell`	`ugh`
`00:21.39`	`Dovid`	`yes`
`00:21.44`	`Dovid`	`both are behind NAT on the same lan`
`00:21.44`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`00:21.45`	`Qwell`	`file: You just made me want to hear a song :P`
`00:22.07`	`file`	`Qwell: I bet I'm listening to said song right now!`
`00:22.10`	`CSD-199`	`i'm crazy connecting 2 asterisk servers, using IAX, but for some reasson, I can not do it. It display a mssg: exited non-zero inside a lot of other messagess, they are routed in different lans`
`00:22.20`	`Qwell`	`file: quite unlikely`
`00:22.29`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`00:22.37`	`Qwell`	`however, mine MAY be a remix of yours, OR a remix of the remake of yours`
`00:22.43`	`file`	`o i c`
`00:22.47`	`C6Vette`	`having weird things in show channels: http://pastebin.ca/231721`
`00:22.57`	`Qwell`	`file: Philip Glass?`
`00:23.20`	`file`	`no!`
`00:23.22`	`[TK]D-Fender`	`Dovid : make sure each one uses a different SIP port.`
`00:23.26`	`Qwell`	`then?`
`00:23.30`	`Dovid`	`hmm`
`00:23.33`	`Dovid`	`where do i set that ?`
`00:23.42`	`[TK]D-Fender`	`file : Bowie for the win...`
`00:23.42`	`CSD-199`	`any can help? IAX config`
`00:23.43`	`[hC]`	`ugh. anyone who has suffered thru training FXO lines to eliminate echo, please identify yourself!`
`00:23.45`	`file`	`Qwell: file.`
`00:23.47`	`Qwell`	`:p`
`00:23.48`	`[TK]D-Fender`	`Dovid : sip.conf`
`00:23.48`	`Dovid`	`ur talking about 5060 and the other 5061 ?`
`00:23.55`	`Qwell`	`so, yeah, like I said... remix of the remake`
`00:24.01`	`Dovid`	`TK: i am using real time`
`00:24.01`	`[TK]D-Fender`	`Dovid : yes, those might be suggest ranges`
`00:24.19`	`[TK]D-Fender`	`Dovid : I only support FAKE time. Sorry`
`00:24.23`	`Dovid`	`lol`
`00:24.45`	`Dovid`	`but regardless if i am using real time or not its in sip.conf under the genreal settings ?`
`00:25.07`	`Dovid`	`or is it a setting on the individual sip account ?`
`00:26.02`	`[TK]D-Fender`	`Dovid : Not general, peer specific`
`00:26.30`	`Dovid`	`ok`
`00:27.25`	`Dovid`	`TK: what do i set ?`
`00:27.28`	`Dovid`	`port=5061 ?`
`00:27.59`	`mitcheloc`	`hello all, anyone experience with the polycom 480i? it's terminating calls with a # key...trying to disable it`
`00:28.44`	`AsteriskMonkey`	`look at the local dialplan in the phone`
`00:29.08`	`mitcheloc`	`it only says this: "X\|XX+*"`
`00:32.31`	`Dovid`	`tk: i set port to 5061 ?`
`00:32.40`	`AsteriskMonkey`	`look for a terminator digit option somewhere`
`00:33.13`	`mitcheloc`	`heh, it's right below that, called "Send Dialplan Terminator"`
`00:33.15`	`mitcheloc`	`it's unchecked!`
`00:36.29`	`[TK]D-Fender`	`mitcheloc : this is no Polycom 480i.......`
`00:37.20`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:37.20`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:38.21`	`AsteriskMonkey`	`sounds more like aastra`
`00:39.43`	`[TK]D-Fender`	`AsteriskMonkey : Hard to say which point he;s wrong on`
`00:40.47`	`*** join/#asterisk barttg (n=barttg@pool-70-20-23-115.bstnma.fios.verizon.net)`
`00:40.58`	`AsteriskMonkey`	`lol`
`00:41.16`	`barttg`	`im trying to use the M option to Dial but it doesnt seem to work`
`00:41.30`	`aptura`	`yea that is a aastra phone`
`00:42.55`	`Dovid`	`barttg: it ignores it completly ?`
`00:43.08`	`AsteriskMonkey`	`maybe the big honking aastra logo all over the phone can be easily misconstrued for polycom :P`
`00:43.09`	`AsteriskMonkey`	`lol`
`00:47.11`	`aptura`	`looks like my wifes bosses bussiness is getting that dreaded bussy line complaint from its customers.`
`00:47.24`	`Dovid`	`;)`
`00:47.33`	`aptura`	`if I am to provide a good solution might consider a sangoma`
`00:47.33`	`Dovid`	`Time to convert em to asterisk`
`00:47.33`	`lennard`	`wifes bosses business.. let me parse that`
`00:47.45`	`aptura`	`hehe`
`00:49.07`	`*** part/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net)`
`00:49.47`	`aptura`	`I do not think I would look too favorable if a digium card had irq or echo issues:) I would see him every time my teeth needed cleaning`
`00:50.03`	`Dovid`	`i have been real happy with sangoma`
`00:50.11`	`ManxPower`	`Dovid: you are in NJ?`
`00:50.16`	`Dovid`	`from NJ`
`00:50.19`	`Dovid`	`in Israel now`
`00:50.22`	`aptura`	`plus sangoma is a Canadian card.`
`00:50.33`	`aptura`	`I would not deal with customs issues.`
`00:50.42`	`ManxPower`	`Dovid: your post asking about PRIs "here" did not specify where "here" is`
`00:50.48`	`Dovid`	`ah ok`
`00:50.50`	`Dovid`	`lol`
`00:50.54`	`Dovid`	`I need it for NJ`
`00:51.04`	`ManxPower`	`aptura: you order from a USA distributer`
`00:51.10`	`Dovid`	`aptura: u can buy it here in the US. u need to order it from them`
`00:51.18`	`Dovid`	`www.telephonydepot.com`
`00:51.24`	`Dovid`	`www.voipsupply.com`
`00:51.26`	`aptura`	`ManxPower I am here in bc.`
`00:51.33`	`Dovid`	`they both sell in the US`
`00:51.34`	`Dovid`	`ah`
`00:51.41`	`Dovid`	`ManxPower: u sell PRI's ?`
`00:52.23`	`ManxPower`	`The only issue I have with Sangoma is 1) They may not support weird configurations and 2) not as many people use Sangmoa so community support will be less than Digium.`
`00:52.42`	`ManxPower`	`Dovid: no, but if you were in a place that the telco I use provided service I would recommend them`
`00:52.44`	`Dovid`	`i have heard that the digium response is try another mother board`
`00:53.02`	`Dovid`	`i had issues with a sangoma card and thier tech ssh'd in for 4 hours at no charge`
`00:53.16`	`mog`	`Dovid, that is not the case`
`00:53.35`	`mog`	`and if it was i apologize and people involved should be taken out and shot ^_^`
`00:53.37`	`ManxPower`	`Dovid: I can't say what Digium says. Both seem to have decent support`
`00:54.00`	`Dovid`	`mog: so i have heard. it never happend to me (and I have used some digium cards with no issues)`
`00:54.18`	`Dovid`	`ManxPower: I am in lakewood, NJ`
`00:54.29`	`ManxPower`	`But I HAVE had issues with various motherboards and Digium cards.`
`00:54.31`	`Dovid`	`u know anyone thats good in my area ?`
`00:54.34`	`[hC]`	`mog: release the g729 card now, kthx.`
`00:54.35`	`[hC]`	`:)`
`00:54.40`	`mog`	`lol`
`00:54.45`	`mog`	`im working on drivers as we speak`
`00:54.47`	`mog`	`im just tireds`
`00:54.56`	`[hC]`	`you just need some more jack in the box!`
`00:54.58`	`[hC]`	`:)`
`00:55.00`	`Dovid`	`mog: join the club`
`00:55.03`	`ManxPower`	`Almost all of these problems happen on production systems where the problem must be fixed RIGHT NOW DAMNIT and so I can't really do much troubleshooting`
`00:55.04`	`file`	`ha`
`00:55.05`	`mog`	`i need a nap`
`00:55.11`	`Dovid`	`and for stress get ur fav. pal, JACK !!!`
`00:55.11`	`aptura`	`ManxPower to bad you did not document which motherboards.`
`00:55.29`	`mog`	`i could use some jackin the box`
`00:55.38`	`Qwell`	`mog: let's go :p`
`00:55.40`	`mog`	`[hC], who were you at conference`
`00:55.45`	`Dovid`	`mog: i have a full bottle in case of emergency. come on over`
`00:55.46`	`mog`	`closest one is in nashville Qwell`
`00:55.51`	`Qwell`	`mog: let's go :p`
`00:55.51`	`mog`	`i dont think your wife would a ppreciate`
`00:56.07`	`mog`	`lol Dovid`
`00:56.15`	`mog`	`need jack in the box, not jack in a bottle`
`00:56.18`	`[hC]`	`mog: dayton`
`00:56.21`	`Dovid`	`hehe`
`00:56.27`	`ManxPower`	`aptura: for the Digium analog cards the motherboard does not seem to be the issue.`
`00:56.33`	`[hC]`	`mog: vancouver guy. i was pretty convinced that nobody knew my irc handle.`
`00:56.34`	`[hC]`	`hehe.`
`00:56.37`	`Dovid`	`mog: they work hand in hand`
`00:56.38`	`mog`	`yeah`
`00:56.39`	`file`	`Nashville is only 800km away`
`00:56.42`	`mog`	`i remember ya dayton`
`00:56.47`	`mog`	`ooh good point file`
`00:56.47`	`Qwell`	`[hC]: I knew :P`
`00:56.48`	`aptura`	`teliax voicepulse are really good known providers with a known good reputation who else can some one suggest.`
`00:56.50`	`mog`	`you want to meat us there`
`00:56.54`	`file`	`sure!`
`00:57.03`	`[hC]`	`Qwell: haha yep im sure you were one of the only ones, asside from juggie`
`00:57.04`	`Qwell`	`mog: let's give him a head start...`
`00:57.04`	`[hC]`	`:)`
`00:57.07`	`ManxPower`	`aptura: I suggest you not route calls over the internet.`
`00:57.12`	`Qwell`	`[hC]: heh, Russell probably did`
`00:57.24`	`aptura`	`Manx I know I have heard the complaints`
`00:57.25`	`Qwell`	`and blitzrage probably`
`00:57.29`	`ManxPower`	`You have NO control over those calls.`
`00:57.43`	`aptura`	`but office to office you do`
`00:57.50`	`aptura`	`to a point`
`00:58.05`	`mog`	`hmm i wonder if i could have jack in the box delivered`
`00:58.08`	`Qwell`	`[hC]: Why no cam hottie on your laptop this year?`
`00:58.17`	`Dovid`	`lol`
`01:02.12`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`01:02.18`	`Dovid`	`hi shmaltz`
`01:02.20`	`Dovid`	`can i pm u ?`
`01:02.24`	`shmaltz`	`sure`
`01:03.16`	`mitcheloc`	`aptura, heh, it is an Aastra, my mind was muddled, still can't figure out how to disable the wierd dialing though`
`01:04.05`	`aptura`	`I want to buy that phone. I think Canadians would go more for it because it looks like nortel phone and I believe it is a copy of a nortel phone.`
`01:04.33`	`mitcheloc`	`aptura, yes it looks very close, but i'd have to say i'm more impressed with the snom so far, it's got a sweet web interface, extremely programmable`
`01:04.44`	`mitcheloc`	`much better to dev asterisk solutions around`
`01:04.45`	`*** join/#asterisk RF_MIA (n=Administ@75.74.65.37)`
`01:04.53`	`mitcheloc`	`however i need to get the 480i up since we bought it`
`01:05.43`	`shmaltz`	`Dovid, do you have canreinvite=no for both?`
`01:06.01`	`Dovid`	`its real time - let me look`
`01:06.07`	`Dovid`	`(i know u hate real time )`
`01:06.14`	`shmaltz`	`exactly`
`01:06.21`	`shmaltz`	`what is the purpose of using realtime?`
`01:06.42`	`Dovid`	`thats how the orig. person set it up, i am just maintaining`
`01:07.09`	`shmaltz`	`oic`
`01:07.16`	`Dovid`	`canreinvite=yes for both`
`01:07.23`	`Dovid`	`it should be no for both ?`
`01:08.05`	`shmaltz`	`yes, make sure it's no for both`
`01:08.08`	`Dovid`	`basicly all phones behind nat should be set: canreinvite=no ?`
`01:08.21`	`Dovid`	`my bad`
`01:08.24`	`Dovid`	`one has no and the other has yes`
`01:08.32`	`shmaltz`	`Dovid, in most cases, unless you can realy predict where the phones will be relative to each other`
`01:08.45`	`shmaltz`	`which one has yes?`
`01:08.52`	`Dovid`	`all of them are in diffrent locations`
`01:08.56`	`Dovid`	`the snom 360`
`01:09.13`	`Dovid`	`they are both set to no now`
`01:09.17`	`[hC]`	`Qwell: you know, i talked to her about that and scorned her for not being available this year.`
`01:09.21`	`Dovid`	`i am gona have him reboot the phone`
`01:09.21`	`shmaltz`	`so the 300 is not at the same location as the 360?`
`01:09.27`	`Qwell`	`hah`
`01:09.31`	`shmaltz`	`no need to reboot the phone`
`01:09.32`	`[hC]`	`Qwell: i'll make sure to get 3 in a group video chat next year.`
`01:09.35`	`shmaltz`	`just reload asterisk`
`01:09.39`	`Qwell`	`[hC]: that works`
`01:09.49`	`Dovid`	`aslo when i do sip show peers they are both using the same port`
`01:10.05`	`Dovid`	`the 300 and 360 are on the same lan behind NAT going to a public server`
`01:10.16`	`Dovid`	`also*`
`01:10.20`	`shmaltz`	`interesting:`
`01:10.22`	`shmaltz`	`http://www.marketwatch.com/news/story/story.aspx?siteid=mktw&guid=%7B62DD9422-2892-4BB4-A8AD-9A6C227C77D8%7D`
`01:10.38`	`shmaltz`	`that is a problem, make sure they use different ports`
`01:10.39`	`[hC]`	`Qwell: heres an interesting question i never asked anyone. what phones do digium employees use?? :)`
`01:10.53`	`[hC]`	`Qwell: as their primary day to day phone, i mean.`
`01:11.00`	`Dovid`	`shmaltz: i just reloaded`
`01:11.05`	`Qwell`	`[hC]: we don't use phones`
`01:11.30`	`[hC]`	`Qwell: haha.. "phones suck, you people use phones?"`
`01:11.40`	`Dovid`	`lol`
`01:12.01`	`[hC]`	`Qwell: better answer would be "meridian" :)`
`01:12.03`	`Qwell`	`actually, we do have a standard.. you could figure it out pretty easily :D`
`01:12.32`	`[hC]`	`Qwell: if i were to guess it would be either polycom or snom, but..`
`01:13.20`	`Dovid`	`shmaltz: ?`
`01:14.14`	`shmaltz`	`Dovid, yes?`
`01:15.06`	`Dovid`	`both have canreinvite=no and the snom 300 still wont work`
`01:15.25`	`Dovid`	`TK: mentioned about setting in sip.conf for them to use diffrent ports ?`
`01:16.57`	`shmaltz`	`Dovid, it's a phone think, you should be able to set the port on the phone, if the phone dosen't support it, check with your NAT router to see if it would redirect incoming port numbers for you, if yes redirect 5061 to the 360, and 5062 to the 300`
`01:17.22`	`*** part/#asterisk RF_MIA (n=Administ@75.74.65.37)`
`01:17.26`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`01:18.06`	`[hC]`	`Qwell: so, by looking at the business edition, it seems to be that polycom and aastra are recommended :)`
`01:18.18`	`Dovid`	`so have nat forwarding`
`01:18.32`	`Dovid`	`5061 to one phone and 5062 to another ?`
`01:18.58`	`Dovid`	`and in the phone have one phone connect on 5061 and the other on 5062 ?`
`01:19.03`	`Dovid`	`or just try the NAT first ?`
`01:19.43`	`Dovid`	`bindport=5060/5061 ?`
`01:20.55`	`*** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com)`
`01:21.28`	`shmaltz`	`Dovid, don't change bindport`
`01:21.46`	`shmaltz`	`just change the NAT redirects, and in sip.conf for the 2 phones put a port value`
`01:21.46`	`Dovid`	`ok`
`01:21.50`	`Dovid`	`just NAT forwarding`
`01:21.58`	`Dovid`	`?`
`01:21.59`	`shmaltz`	`and the port in sip.conf`
`01:22.13`	`Dovid`	`using real time ARGHHHHHHHHHHHHHHHHHHHH`
`01:22.15`	`Dovid`	`i set port=`
`01:22.19`	`ManxPower`	`Dovid: you want some advice?`
`01:22.36`	`Dovid`	`however when it reloads it changes itself to a port in the 14k range`
`01:22.41`	`Dovid`	`shoot ManxPower`
`01:23.44`	`ManxPower`	`Asterisk's end of the UDP connection will always be port 5060. On the SIP client end of the connection will always be SOME PORT GREATER THAN 1024`
`01:23.58`	`ManxPower`	`but that client side port can be ANY port greater than 1024`
`01:24.14`	`Dovid`	`so how do i set it on the client side ? on the phone ?`
`01:24.47`	`ManxPower`	`what you see in "sip show peers" for the port is the REMOTE port.`
`01:25.04`	`Dovid`	`same port for both phones`
`01:25.05`	`ManxPower`	`Dovid: in general you should not care one bit about the port on the client side.`
`01:25.28`	`Dovid`	`10306/10306 65.8.232.115 D N 60414 OK (136 ms)`
`01:25.29`	`Dovid`	`10307/10307 65.8.232.115 D N 60414 OK (137 ms)`
`01:25.35`	`Dovid`	`i got that for sip show peers`
`01:25.38`	`ManxPower`	`Dovid: Are you doing port forwarding on the NAT router where the phones are located?`
`01:25.44`	`Dovid`	`no`
`01:25.47`	`ManxPower`	`good.`
`01:26.18`	`ManxPower`	`I assume those two peers are different phones behind the same NAT router and each phone has it's own internal LAN IP address?`
`01:26.26`	`Dovid`	`yes`
`01:26.40`	`ManxPower`	`Dovid: what phones? Make/Model?`
`01:26.49`	`Dovid`	`SNOM 300 and SNOM 360`
`01:26.59`	`ManxPower`	`Dovid: BTW, what you are seeing IS odd and not common.`
`01:27.13`	`ManxPower`	`Dovid: those phones have a web interface for configuration don't they?`
`01:27.57`	`Dovid`	`yes`
`01:28.00`	`ManxPower`	`I've never used SNOM, but the same basic ideas apply to most phones.`
`01:28.16`	`ManxPower`	`go int the web config for one of the phones and look at the SIP port listed THERE. What is it?`
`01:28.27`	`Dovid`	`5060 i believe`
`01:28.35`	`Dovid`	`i dont have access to them i can ask the client`
`01:28.36`	`ManxPower`	`Dovid: change it and see what happens.`
`01:28.41`	`Dovid`	`but i beleive 5060 for both`
`01:28.45`	`Dovid`	`one of them to 5061 ?`
`01:29.09`	`ManxPower`	`Dovid: really any random number, but 5061 is just as good of a number as 16384`
`01:29.16`	`Dovid`	`ok`
`01:29.27`	`ManxPower`	`BTW, do the phones have ANY NAT related settings enabled?`
`01:30.16`	`ManxPower`	`Nevermind. When you get access to the phones make sure any NAT, STUN, etc settings are disabled.`
`01:30.19`	`Dovid`	`not really`
`01:30.25`	`Dovid`	`how about ICE ?`
`01:30.30`	`ManxPower`	`nat=yes in asterisk's sip.conf should be all you need.`
`01:30.38`	`ManxPower`	`I have no idea what ICE is.`
`01:30.49`	`Dovid`	`let melog in to my soft phone`
`01:30.55`	`ManxPower`	`Other than Incredibly Cool Electronics`
`01:31.00`	`Dovid`	`lol`
`01:31.07`	`shmaltz`	`In Case of Emergency`
`01:31.12`	`shmaltz`	`~ ICE`
`01:31.14`	`jbot`	`hmm... ice is Intrusion Countermeasures Electronics. In Circut Emulator. Internal Compiler Error`
`01:31.15`	`Dovid`	`Offer ICE: on off`
`01:32.03`	`ManxPower`	`hold on`
`01:32.25`	`Dovid`	`okies`
`01:32.31`	`ManxPower`	`ICE should be off.`
`01:32.40`	`Dovid`	`its off`
`01:32.45`	`ManxPower`	`it's some SNOM NAT traversal thing according to a quick google search`
`01:33.22`	`*** join/#asterisk xanderp (n=xanderp@74.133.15.46)`
`01:33.35`	`*** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br)`
`01:33.50`	`Dovid`	`time to find my friend Jack`
`01:34.42`	`xanderp`	`could someone recommend a good 'comparison' site for comparing voip providers that are asterisk friendly for me please? or recommendations welcome too.`
`01:34.58`	`*** join/#asterisk lule (i=lule@host69.200-117-163.telecom.net.ar)`
`01:35.31`	`*** part/#asterisk lule (i=lule@host69.200-117-163.telecom.net.ar)`
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`01:37.32`	`*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`01:37.43`	`[hC]`	`any sangoma a200d users alive?`
`01:38.20`	`*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)`
`01:39.12`	`shmaltz`	`[hC], not they were all killed in a masive avian virus release by Digium :P`
`01:39.32`	`mog`	`lol`
`01:39.39`	`[TK]D-Fender`	`shmaltz : Flock off ;)`
`01:40.12`	`[TK]D-Fender`	`mog : load chan_electronmicroscope.so`
`01:40.23`	`shmaltz`	`mog, were you vacinated?`
`01:41.14`	`mog`	`more than once`
`01:41.38`	`[TK]D-Fender`	`[hC] : What of it?`
`01:41.40`	`Dovid`	`lol`
`01:45.53`	`Dovid`	`i use myphonecompany.com for inbound and teliax.com for outbound`
`01:46.10`	`xanderp`	`quality/reliability is good for those?`
`01:46.14`	`[hC]`	`[TK]D-Fender: Ive got an a200d, and i have a client who claims they still have 'unreasonable' echo -- ive been on site a couple times and cant replicate it, but that doesnt mean its not there -- although, i have NO freakin clue where else to check.`
`01:46.25`	`[hC]`	`[TK]D-Fender: im running the latest beta10 drivers`
`01:46.26`	`Dovid`	`both great`
`01:46.38`	`Dovid`	`had issues with myphonecompany twice in 2 years`
`01:46.49`	`xanderp`	`david: have you had any experience with their support for any reason?`
`01:46.57`	`Dovid`	`once for 30 min dtmf didnt go thru and once they were down for 15 min`
`01:46.58`	`[TK]D-Fender`	`[hC] : I'd double-check your zapata config and your wanpipe config. have you run the Otasic diagnostic on it?`
`01:47.10`	`Dovid`	`myphonecompany's support isnt the best but u rarely need it`
`01:47.18`	`xanderp`	`is it in india?`
`01:47.20`	`Dovid`	`teliax's support is great. they answer thier phones fast`
`01:47.35`	`Dovid`	`except teliax i believe charges for inbound`
`01:47.40`	`Dovid`	`no they are both in the US`
`01:47.42`	`[hC]`	`[TK]D-Fender: ive gone thru a ton of stuff, making sure the EC is on on all the channels, which it claims it is.. ive gone over zapata.conf a billion times. ive tried adjusting the gains to be within the 50% mark on the vu meters when speaking, wanpipe config has HWEC=yes`
`01:47.49`	`Dovid`	`i also use voipjet - which is the cheapest but they have no support`
`01:47.51`	`[hC]`	`[TK]D-Fender: echocancel=yes is on`
`01:48.00`	`[hC]`	`[TK]D-Fender: It makes no sense to me.`
`01:48.37`	`xanderp`	`you have different dialplans for each provider based on what is cheapest for each i suppose?`
`01:48.50`	`ToyMan`	`David: both myphonecompany and teliax are using asterisk?`
`01:49.20`	`*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)`
`01:49.33`	`ToyMan`	`ah, never mind`
`01:49.56`	`ToyMan`	`just read xanderp's first post on the thread`
`01:50.17`	`Dovid`	`teliax uses asterisk`
`01:50.24`	`Dovid`	`myphonecompany dosent`
`01:50.42`	`Dovid`	`myphonecompany is a branch of xchange telecom`
`01:51.39`	`xanderp`	`david: myphonecompany provides ability for you to route calls to, or are you using hardware to adapt to analog lines with them?`
`01:52.21`	`xanderp`	`fxo/fxs conversion to a phone puck they provide?`
`01:52.41`	`Dovid`	`the calls come to asterisk and u route them the way u want`
`01:52.45`	`Dovid`	`it comes in via voip`
`01:53.26`	`xanderp`	`ok, i didn't understand the line you used earlier about 'myphonecompany doesnt' you mean they don't use asterisk on their side yes?`
`01:55.00`	`Dovid`	`they dont use asterisk`
`01:55.05`	`Dovid`	`but that dosent convern u`
`01:55.08`	`*** join/#asterisk sp00kz (n=ilubj00@pool-72-68-183-237.nycmny.east.verizon.net)`
`01:55.11`	`Dovid`	`any and all settings u do on ur end on asterisk`
`01:55.29`	`Dovid`	`as ling as they have voip (which they do) ur good to go`
`01:56.21`	`sp00kz`	`I just got in a Cisco 7970, but It came with the SIP 8 0 4 image, does anyone have the SIP70.8-0-3S one for use with asterisk?`
`01:56.28`	`xanderp`	`ok, i thought you meant they were not a 'bring your own hardware' provider.`
`01:57.23`	`xanderp`	`anyone have an opinion about sunrocket?`
`01:57.25`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net)`
`01:57.45`	`sp00kz`	`I`
`01:58.16`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net)`
`01:58.41`	`Dovid`	`no they are`
`01:58.47`	`Dovid`	`they have diffrent plans`
`01:59.00`	`Dovid`	`thier BYOD plans is called a MyDevicePlan`
`01:59.05`	`Dovid`	`its not on thier web site`
`01:59.09`	`Dovid`	`u have to email or cal them`
`01:59.33`	`*** join/#asterisk _DAW (n=_DAW@adsl-157-54-57.msy.bellsouth.net)`
`01:59.57`	`xanderp`	`k, thanks`
`02:01.22`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`02:02.16`	`_DAW`	`hello everyone`
`02:02.25`	`*** join/#asterisk Crashsys (n=kumba@office.crashsys.com)`
`02:03.13`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`02:03.44`	`Crashsys`	`When I try to modprobe wct4xxp, I get "wct4xxp: Unknown symbol release_firmware" and "wct4xxp: Unknown symbol request_firmware"... any idears?`
`02:10.28`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net)`
`02:13.47`	`ToyMan`	`so, trixbox is based on asterisk, yes?`
`02:14.02`	`ToyMan`	`just trying to figure out the basics here...`
`02:14.27`	`ToyMan`	`trixbox runs on win? only?`
`02:14.31`	`mog`	`trixbox contains asterisk`
`02:14.31`	`Strom_C`	`trixbox is a linux distribution which contains asterisk and a whole ton of other junk-i-mean-software`
`02:14.32`	`mog`	`no`
`02:15.19`	`ToyMan`	`ah, i see... looking at nerd vittles site, they're running a win box with vmware doing the linux`
`02:15.25`	`ToyMan`	`why, i wonder....`
`02:15.30`	`Strom_C`	`they like pain`
`02:15.34`	`Strom_C`	`apparently`
`02:15.34`	`ToyMan`	`heh`
`02:16.01`	`kph100`	`what happpens if the answer() is executed twice?`
`02:16.15`	`Strom_C`	`kph100: the second answer() does nothing if the channel is already answered`
`02:17.03`	`mog`	`answer queues an answer frame if one has already been sent it will most likely do nothing`
`02:17.07`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`02:17.08`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`02:17.36`	`kph100`	`but is better to only send the answer() only once?`
`02:18.07`	`[hC]`	`interesting. wireshark interprets huge huge jitter when dtmf keypresses occur`
`02:18.29`	`Strom_C`	`no, make sure you put answer() 100 times in a row just to be sure the channel is answered :)`
`02:18.35`	`[TK]D-Fender`	`kph100 : Doesn't amtter if you repeat it`
`02:18.38`	`DaveCanoe`	`found a nice big fat IAX bug today.`
`02:19.23`	`mog`	`ooh do tell DaveCanoe`
`02:19.50`	`DaveCanoe`	`Call is in progress as follows PSTN -> Cisco As5300 -> (sip) -> Asterisk A -> (iax) -> Asterisk B -> (sip) -> GS2000#1`
`02:20.53`	`DaveCanoe`	`asterisk B is a complete slave. Rule basically states "exten => _.,1,Dial(IAX2/AsteriskA/${EXTEN})" for the context of all registered phones.`
`02:22.06`	`DaveCanoe`	`Now. The following happens: GS2000#1 forwards to GS2000#2. This works. It creates new channels on the IAX trunk (both directions) and then they collapse ... just like they're supposed to.`
`02:23.10`	`DaveCanoe`	`Now the fun part. GS2000#2 forwards call to GS2000#3. GS2000#3 rings, but no sip RTP occurs on the lan. The call is lost. Always the 2nd forward, doesn't matter what phones are involved. Tested with polycom, too.`
`02:23.55`	`[hC]`	`sounds like you need to try disabling reinvite's in sip.conf and notransfer in iax.conf`
`02:24.18`	`DaveCanoe`	`If the rules on Asterisk B are extended to allow local dialing of the local sip phones, everything works (which is what we're doing), but this destroys the idea that Asterisk B could be a simple slave.`
`02:25.05`	`[hC]`	`IAX media transfers are not flawless`
`02:25.06`	`DaveCanoe`	`all four combinations of reinvite and notransfer tried.`
`02:25.33`	`DaveCanoe`	`... although it's worth noting that nothing tries to reinvite when the decisions are all made at asterisk A.`
`02:26.46`	`DaveCanoe`	`Anyways... I have some detailed logs warning of many dire things when this all goes on --- which I'd be happy to share. I'm not involved enough with asterisk to go any further.`
`02:30.09`	`ManxPower`	`DaveCanoe: if all else fails post a message to asterisk-users mailinglist`
`02:33.11`	`Crashsys`	`Any got any idears why wct4xxp tells me it has an unknown symbol "release_firmware"?`
`02:33.26`	`Ryanw`	`how do i debug MWI on a gXP2000 ?`
`02:33.39`	`Crashsys`	`It's not CRC CCITT... that's in the kernel...`
`02:38.17`	`DaveCanoe`	`Well... I've solved my problem for the momment. However, maybe there's a place to file a bug report?`
`02:39.00`	`Strom_C`	`bugs.digium.com maybe?`
`02:40.15`	`file`	`Crashsys: hotplug firmware loading is compiled as a module and the module is not loaded?`
`02:40.37`	`Crashsys`	`I dont think I had it compiled at all in the kernel`
`02:40.41`	`Ryanw`	`can someone fill me in on how MWI (Message Waiting Indicator) aka Voicemail light is meant to work please.`
`02:40.47`	`Crashsys`	`redoing the kernel to add it in`
`02:40.52`	`file`	`it won't be used unless it is detected in the kernel config`
`02:47.39`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.252)`
`02:50.23`	`*** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net)`
`02:54.55`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`02:54.56`	`*** mode/#asterisk [+o russellb] by ChanServ`
`02:57.55`	`*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-142-90.hsd1.fl.comcast.net)`
`03:00.41`	`*** part/#asterisk Corndawg_ (i=whoisit@c-66-176-142-90.hsd1.fl.comcast.net)`
`03:11.55`	`Ryanw`	`just a thought, its not a codec issue with allow/deny rules causing no rtp /`
`03:12.12`	`*** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net)`
`03:19.29`	`*** join/#asterisk rene1 (n=rene1@189.140.208.108)`
`03:19.39`	`rene1`	`hi`
`03:19.54`	`rene1`	`question: should iax peers report on REALTIME peers or is something that needs to be built`
`03:20.15`	`rene1`	`cuz sip peers do show realtime registered peers`
`03:25.46`	`*** join/#asterisk ComputerWarm (n=dan@h109.42.63.69.cable.ottr.cablerocket.net)`
`03:25.52`	`ComputerWarm`	`Hello all`
`03:26.19`	`ComputerWarm`	`question is there anyway to use the Bridge interface from Command line through the interface?`
`03:32.50`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`03:36.58`	`*** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net)`
`03:37.51`	`wacky_`	`Hey, guys.. where are the details in the console since * 1.2`
`03:39.27`	`*** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net)`
`03:40.25`	`wacky_`	`I used to see which extensions users were entering, but it's all gone`
`03:45.04`	`docelmo`	`Can the asterisk dialplan do regex to detect numeric patterns w/ no alpha?`
`03:45.58`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`03:49.19`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`03:52.59`	`*** join/#asterisk [shodan] (n=shodan@ip095.99-113-216.pppoe4.joliette.intermonde.net)`
`03:53.03`	`*** join/#asterisk bmg505 (n=leon@c1-252-12.rndf.isadsl.co.za)`
`03:54.31`	`*** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)`
`03:54.36`	`*** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net)`
`03:54.40`	`SomethingISODD`	`hello all/`
`03:54.43`	`[shodan]`	`what values of "FXO Port Impedance" should I try to get less echo on a SPA-3*02 ? (I don't get a lot of echo because the echocanceller works but the echo is still there and sometime there are glitches in the echocanceller)`
`03:55.45`	`[shodan]`	`I tried 600 (default) and 600+2.16uF but is there a chance one of the other value will work better (for a line from Bell Canada)`
`03:56.34`	`*** join/#asterisk BitBandit (n=polx@24-241-129-98.static.stgr.ut.charter.com)`
`03:56.41`	`*** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net)`
`03:57.14`	`SomethingISODD`	`Question is there anyway to get Conferncing working with out Zap hardware and no SCIC hardware`
`03:59.48`	`brookshire`	`SomethingISODD: ztdummy`
`04:00.18`	`docelmo`	`SomethingISODD Sangoma or Openvox hardware :)`
`04:01.44`	`SomethingISODD`	ztdummy won`t install i get a load of errors
`04:02.17`	`brookshire`	`are you using linux?`
`04:02.26`	`SomethingISODD`	`ya`
`04:02.32`	`brookshire`	`it should install`
`04:02.37`	`SomethingISODD`	`its an older system`
`04:02.44`	`SomethingISODD`	and i can`t really update it
`04:02.47`	`brookshire`	`what kernel version?`
`04:02.56`	`*** join/#asterisk bungalow (n=tasat-@c-24-7-62-81.hsd1.ca.comcast.net)`
`04:03.00`	`brookshire`	`uname -a`
`04:03.45`	`SomethingISODD`	`Linux 96151-www1 2.4.22-1.2199.nptlsmp #1 SMP Wed Aug 4 11:49:01 EDT 2004 i686 athlon i386 GNU/Linux`
`04:03.49`	`bungalow`	`Anyone know what the mix parameter represents in the manager monitor application? I see in voip-info that 1 doesn't seem to mix -- what does mix?`
`04:04.10`	`brookshire`	`something: zaptell will compile.. you just need to find the source for that kernel`
`04:04.25`	`SomethingISODD`	`oh ok, brookshire one other question for you.`
`04:04.42`	`SomethingISODD`	I need to get three way calling to work server side being my phone doesn`t support it
`04:05.00`	`SomethingISODD`	`is there anyway you can think of doing that other then modifing meetme`
`04:05.59`	`brookshire`	`flash hook?`
`04:06.05`	`brookshire`	`asterisk supports it :)`
`04:06.17`	`SomethingISODD`	`you mean flash the phone?`
`04:06.23`	`brookshire`	`yeah..`
`04:06.27`	`brookshire`	`at least my sip phone supports it`
`04:06.28`	`SomethingISODD`	`it hangs up the other call`
`04:06.29`	`brookshire`	`dunno why`
`04:06.30`	`brookshire`	`lol`
`04:06.47`	`brookshire`	`oh well :)`
`04:06.54`	`brookshire`	`file!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! ! !!!!!!!!!!!!!!!! ! ! !! !!!!!!!! !!`
`04:07.23`	`[shodan]`	`is there a good reason to force the users to prefix 9 to any outside call they make ?`
`04:07.51`	`brookshire`	`no`
`04:08.06`	`mog`	`conformity [shodan]`
`04:08.08`	`brookshire`	`except... you can make internal extensions match faster`
`04:08.11`	`mog`	`people can be very dumb`
`04:08.51`	`*** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro)`
`04:08.53`	`SomethingISODD`	`will the flash app work with SIP or just strictly with zap channels?`
`04:09.16`	`brookshire`	`SomethingISODD: it works with my sip phone.. it may not work with yours`
`04:09.30`	`brookshire`	`it doesn't work with my polycom, but it has two lines anyways`
`04:09.59`	`SomethingISODD`	brookshire ohh ok ya mine doesn`t oh well next idea lol
`04:10.01`	`brookshire`	`file: it does.. i'll show you :)`
`04:10.04`	`SomethingISODD`	`thanks brookshire for your time.`
`04:10.06`	`[shodan]`	`ah k , I thought there was some great pitfall that I was going to fall in if I let the users dial "transparently"`
`04:18.21`	`[TK]D-Fender`	`[shodan] : I have abolihed the "9" refix and even compensate the newly forced 10-digit dialing by prefixing 7-digit local #'s with the local area code transparently. there are many benifits to removing an outbound dial prefix`
`04:19.43`	`[TK]D-Fender`	`[shodan] : for instance imagine that incoming callerid from the PSTN arrives as 10 digits. Now if the callee misses the call , the phone will hold the caller's #. if you have an outbound prefix you would not be able to use a simple "callback" option on your phone. you'd have to edit the # first (assuming your phone even allows it).`
`04:21.53`	`[TK]D-Fender`	`Thus equating to a loss of effeciency. so the advatages are faster call-backs, faster outbound dialing (at least 1 less digit to dial), more natural feel. Also lends itself to having you set up your phone's local dial-plans to basically "just pass anything" to * after a 3s wait, or hitting "send" (as applicable)`
`04:23.24`	`*** join/#asterisk NDT (n=nunya@cpe-24-195-72-2.nycap.res.rr.com)`
`04:31.22`	`SomethingISODD`	`Question for anyone that can help I try to hit flash on my phone i get DTMF-relay event received: FLASH than a error saying sip_indicate: Don't know how to indicate condition 9 WARNING[8908]: channel.c:2070 ast_indicate: Unable to handle indication 9 for anyideas plz`
`04:38.39`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`04:40.13`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`04:41.34`	`*** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au)`
`04:47.57`	`*** join/#asterisk Alexphil (n=chatzill@daemon.aktel.kg)`
`04:48.21`	`Alexphil`	`Hi All!`
`04:49.42`	`Alexphil`	`I have a little troble with kernel 2.4.32 and compilling zaptel :(`
`04:50.39`	`Alexphil`	`zatel say next: Your kernel lacks CRC_CCIT support! , but in my .config file No this option...`
`04:50.55`	`Alexphil`	`Somebody, please help me :)`
`04:59.47`	`Alexphil`	`<PROTECTED>`
`05:00.28`	`Alexphil`	`<PROTECTED>`
`05:00.31`	`Alexphil`	`<PROTECTED>`
`05:00.37`	`Alexphil`	`<PROTECTED>`
`05:00.43`	`postel`	`stop it`
`05:01.19`	`postel`	`read this: http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#crc_ccitt`
`05:01.57`	`postel`	`Ask once and wait, dont PM ppl and dont repeat yourself. Its not that hard now, is it?`
`05:03.31`	`Alexphil`	`uytu`
`05:06.12`	`*** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net)`
`05:06.42`	`SomethingISODD`	`<PROTECTED>`
`05:07.54`	`aptura`	`Hard to believe but there is almost 600 voip prviders in the United States followed by Canada then russia`
`05:08.37`	`SomethingISODD`	`aptura not hard to believe USA is bigger then canada`
`05:09.25`	`Alexphil`	`I read this: http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#crc_ccitt, but in my config of kernel-2.4.32 no option about CRC-CCITT, I have only CRC32 functions zlib compression and decompression`
`05:13.56`	`*** join/#asterisk alerios (n=alerios@190.24.97.148)`
`05:17.30`	`[shodan]`	`can I do this somehow without asterisk itself having t.38 support ? analog Fax -> SPA-2002 -> * -> SPA-3002 -> PSTN -> another fax`
`05:19.37`	`Qwell`	`It has T.38 passthrough support`
`05:19.44`	`Qwell`	`well, in 1.4`
`05:21.51`	`[shodan]`	`does it need extra configuration ? or does it just work automagically ?`
`05:22.52`	`*** join/#asterisk thebleh33 (n=thebleh@ip68-224-138-154.lv.lv.cox.net)`
`05:23.32`	`*** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au)`
`05:25.30`	`*** join/#asterisk aadilismail (n=adilisma@202.38.55.114)`
`05:27.46`	`aadilismail`	`hi`
`05:27.58`	`saam4u`	`hi`
`05:28.03`	`saam4u`	`hi adil`
`05:28.22`	`aadilismail`	`hi`
`05:28.26`	`aadilismail`	`hi saam`
`05:29.05`	`[shodan]`	`hey Qwell , actually that works out of the box with 1.2.9 !??!?!?!`
`05:29.17`	`Qwell`	`T.38? No`
`05:30.19`	`Alexphil`	`Who can say to me what I should be do, for comile zaptel, with kernel 2.4, maybe USEflags or somthing else? I using Gentoo and kernel 2.4.32, and when I try emerge zaptel i receive error: Your kernel lacks CRC_CCIT support!, but in .config no this option, also no in make menuconfig, plrase, I little tired`
`05:30.20`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`05:31.56`	`[shodan]`	`Qwell, well I hooked it up like so "analog Fax -> SPA-2002 -> * -> SPA-3002 -> PSTN -> another fax" and I got an image , undistorted on "another fax" , dunno if it used T.38, probably not, but it "just worked"`
`05:32.19`	`Qwell`	`[shodan]: It likely used ulaw, which is definitely not recommended.`
`05:36.41`	`*** join/#asterisk tengulre (n=tengulre@221.11.5.182)`
`05:42.49`	`tengulre`	`hi,all`
`05:42.57`	`tengulre`	`anybody here?`
`05:49.24`	`*** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net)`
`06:01.55`	`*** join/#asterisk _Vile (i=_Vile@198.175.14.242)`
`06:03.11`	`*** join/#asterisk Alexphil (n=chatzill@daemon.aktel.kg)`
`06:05.02`	`*** join/#asterisk DoDaT69 (i=Tracer@66.240.110.115)`
`06:05.30`	`DoDaT69`	`anyone know of a good asterisk "appliance" for under 500 bux?`
`06:08.05`	`kph100`	`l`
`06:09.03`	`DoDaT69`	`yea?`
`06:18.30`	`*** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net)`
`06:20.16`	`[hC]`	`hm. i cant figure this out, what is double clicking on a users MWI icon in the flash operator panel supposed to do?`
`06:22.30`	`[hC]`	`it says it will "originate a call to voicemailmain" but - call to who??`
`06:24.37`	`*** join/#asterisk bmg505 (n=leon@c1-163-15.rndf.isadsl.co.za)`
`06:25.23`	`*** part/#asterisk Alexphil (n=chatzill@daemon.aktel.kg)`
`06:25.38`	`*** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com)`
`06:27.47`	`*** join/#asterisk VibroMax (n=phisto@83.229.70.154)`
`06:42.54`	`*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)`
`06:43.22`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`06:52.25`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-207-103.red.bezeqint.net)`
`06:53.38`	`*** join/#asterisk Crashsys (n=kumba@office.crashsys.com)`
`06:54.12`	`Crashsys`	`Is it possible to switch the Virtual console to TTY9 from rc.local? That way I always get the asterisk console up on the monitor...`
`06:59.58`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net)`
`07:00.02`	`*** join/#asterisk axscode (n=axscode@210.213.106.190)`
`07:01.32`	`saam4u`	`new to asterisk .. in extensions.conf what is hold command ?`
`07:06.22`	`*** join/#asterisk Alexphil (n=chatzill@daemon.aktel.kg)`
`07:08.39`	`Alexphil`	`Hi again, another question: Before I try use kernel 2.6 and all was good except that card digium te110 not work property, the indication was green, and I no have any alarms, but when i try do some call My D channel is go to down...`
`07:10.07`	`bsdfreak`	`biggup`
`07:10.48`	`Alexphil`	`the config of zaptel was next span 1,1,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 asterisk so channels but watch -n1 -t "cat /proc/zaptel/1" don't see any actives, where I have trouble?`
`07:11.55`	`*** join/#asterisk af_ (n=af@ip-172-163.sn1.eutelia.it)`
`07:18.09`	`axscode`	`~wiki`
`07:18.28`	`hads`	`~docs`
`07:18.38`	`jbot`	`somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`07:18.53`	`axscode`	`there.. thanks hadz`
`07:18.57`	`hads`	`:)`
`07:19.52`	`Alexphil`	`very funny.. before I come Here, I already read almost this, and dont find any sollution///`
`07:19.52`	`axscode`	`gurus and masters docs`
`07:21.53`	`Alexphil`	`I configure all, how wrote in manual, but digium car see other side like phisical device, but logical not undersand, I trying othe coding and framing parameters...`
`07:22.53`	`Alexphil`	`and another parameters coding and framing not working`
`07:23.15`	`Alexphil`	`any ideas?`
`07:25.20`	`Alexphil`	`who can me say, is it critical for zaptel libpri and asterisk what kernel I use, 2.4 or 2.6?`
`07:26.13`	`hads`	`Understand you I cannot.`
`07:26.40`	`Alexphil`	`Sorry for my English`
`07:27.03`	`mog`	`doesnt matter Alexphil`
`07:27.34`	`*** join/#asterisk apardo (n=apardo@87.217.147.18)`
`07:27.50`	`Alexphil`	`what abot x64 or x32 systems?`
`07:28.10`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`07:28.22`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`07:28.27`	`Juggie`	`Alexphil, no.`
`07:28.30`	`Juggie`	`use whatever you want :)`
`07:28.34`	`hads`	`You can use either 2.4 or 2.6 if that is what you are asking.`
`07:28.40`	`Alexphil`	`thank you!`
`07:28.41`	`axscode`	`hads: you sounds like master yoda.`
`07:29.35`	`Alexphil`	`where then I have trouble?`
`07:30.58`	`mog`	`juggie!`
`07:31.11`	`Juggie`	`ogorman!`
`07:34.44`	`axscode`	`Alexphil: use the newer kernel.. 2.6`
`07:35.52`	`axscode`	`Alexphil: install libri, then zaptel, then asterisk.... but before that.. check for the depencies it needs.... then. make sure that your TE110 Digium card is in the right PCI slot... considering the power voltage of the PCI and your digium card.`
`08:06.35`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`08:06.35`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) Join #freepbx for freepbx/trixbox support.`
`08:17.50`	`*** join/#asterisk svenna_ (n=svenna@p548D2FC5.dip0.t-ipconnect.de)`
`08:23.51`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`08:24.55`	`*** join/#asterisk prttp (i=Ftv@187.Red-81-44-146.dynamicIP.rima-tde.net)`
`08:31.46`	`*** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro)`
`08:45.33`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com)`
`08:47.21`	`*** join/#asterisk dzh (n=dzh@eth1.rt001a.cxnet.dk)`
`08:48.03`	`FuriousGeorge`	`can both asterisk and ser listen on udp 10000-20000 or is that not allowed?`
`08:48.39`	`Juggie`	`guess.`
`08:56.40`	`FuriousGeorge`	`Juggie: i guess no`
`08:57.07`	`Juggie`	`:)`
`08:59.19`	`axscode`	`lol`
`08:59.54`	`axscode`	`anyone offering SIP termination here?`
`09:00.00`	`FuriousGeorge`	`Juggie: you are supposed to tell me if i guessed correctly`
`09:00.11`	`Juggie`	`FuriousGeorge, of course the answer is no`
`09:00.17`	`Juggie`	`each program needs to keep track of the udp ports in use`
`09:00.22`	`Juggie`	`and of course if they share a range they cannot`
`09:02.53`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`09:06.06`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`09:06.30`	`FuriousGeorge`	`i cant for the life of me figure out why nat is messing up my media stream on this particular server. i set it up the same way i always do. forwarded 5060, 8000, 10k-20k udp, set externip to my *.gotdns.org account, and set locallan...`
`09:06.39`	`FuriousGeorge`	`the only difference is im using 1.4beta here`
`09:09.49`	`Juggie`	`rtp debug/sip debug`
`09:13.32`	`FuriousGeorge`	`im looking at it now, here it is....`
`09:16.42`	`FuriousGeorge`	`Juggie: did you get the pm?`
`09:19.22`	`*** join/#asterisk postel_ (n=jp@wikimedia/Postel)`
`09:22.06`	`[shodan]`	`if I call a number with a SPA-3002 and that number is busy , how can the user tell asterisk to hangup and redial ? because the SPA-3002 does not appear to return a BUSY dialstatus when you hit a busy line :\ maybe a feature ? user hits certain keys and * hangup and redial`
`09:22.16`	`FuriousGeorge`	`and rtp bug keeps saying Sent RTP packet to 10.0.0.100:9170 (type 00, seq 011016, ts 001280, len 000160)`
`09:22.25`	`FuriousGeorge`	`should that be 10.0.0.100?`
`09:23.47`	`FuriousGeorge`	`thats the local address of the remote client, which shouldnt be there because i set externip, right?`
`09:24.08`	`FuriousGeorge`	`could this have something to do with the fact im using 1.4?`
`09:26.34`	`*** join/#asterisk inspired (n=mikael@85.221.7.59)`
`09:29.46`	`FuriousGeorge`	`sigh`
`09:31.07`	`Juggie`	`set nat=yes`
`09:32.52`	`FuriousGeorge`	`Juggie: i did`
`09:33.01`	`Juggie`	`then sometihng is amiss`
`09:33.06`	`FuriousGeorge`	`Juggie: i notice 1.4 has changed a bunch of stuff`
`09:33.16`	`Juggie`	`try a fresh 1.4 config file.`
`09:33.24`	`Juggie`	`make sure externip is set, make sure localnet is set.`
`09:33.51`	`FuriousGeorge`	`they are. ive done this a bunch of times. it always works, but for now. did you see the pm i sent you with the sip debug ouput?`
`09:34.04`	`FuriousGeorge`	`and rtpdebug says its sending to 10.0.0.10`
`09:34.17`	`FuriousGeorge`	`which is the local address of the remote client`
`09:35.28`	`SheriF_SpacE`	`how can i adjust buffer in sip ?`
`09:35.42`	`SheriF_SpacE`	`tos options ? or there is ab buffer option ?`
`09:35.46`	`FuriousGeorge`	`sip.conf has a jtterbuffer setting`
`09:35.49`	`FuriousGeorge`	`and tos`
`09:36.23`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`09:36.39`	`Juggie`	`FuriousGeorge, does sip list peers show nat N for the client?`
`09:37.19`	`FuriousGeorge`	`yeah, it does`
`09:37.22`	`FuriousGeorge`	`Juggie:`
`09:40.02`	`*** join/#asterisk Modcuts (n=bob@lan.proporta.com)`
`09:41.21`	`SheriF_SpacE`	`FuriousGeorge: i didn't find jtterbuffer in voip-info ?`
`09:43.11`	`FuriousGeorge`	`SheriF_SpacE: maybe thats only in iax.conf`
`09:43.54`	`FuriousGeorge`	`yeah its not in sip.conf`
`09:44.05`	`FuriousGeorge`	`clients often have settings for the jitterbuffer though`
`09:44.33`	`FuriousGeorge`	`Juggie: udp 5060,5061,8000,and10k-20k, right?`
`09:44.36`	`*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)`
`09:44.41`	`SheriF_SpacE`	`FuriousGeorge: what if my client is another asterisk server :D getting complicated :D`
`09:45.11`	`FuriousGeorge`	`complication seems to be the flavor of the day`
`09:45.23`	`axscode`	`do it really NEED a STUN server .. does it really helps!?`
`09:45.25`	`*** join/#asterisk aadilismail (n=adilisma@202.38.55.114)`
`09:45.35`	`aadilismail`	`hi`
`09:45.56`	`FuriousGeorge`	`axscode: except for today i have always been able to get clients working with nat on both ends without one`
`09:46.36`	`SheriF_SpacE`	`FuriousGeorge: trye ;-)`
`09:47.25`	`axscode`	`what i mean if it really helps enhance the network? or is it a need..`
`09:47.44`	`FuriousGeorge`	`axscode: dunno`
`09:49.09`	`*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)`
`09:57.19`	`FuriousGeorge`	`Juggie: do you know if rtp debug should say this: Sent RTP packet to 10.0.0.100:9170...`
`09:57.26`	`FuriousGeorge`	`10.0.0.100 is the remote clients local ip`
`09:58.39`	`FuriousGeorge`	`i just answered my own question`
`09:58.44`	`FuriousGeorge`	`it should absolutely not do that`
`09:58.47`	`FuriousGeorge`	`wtf`
`10:01.45`	`[hC]`	`so this is interesting`
`10:02.08`	`[hC]`	`using sangomas echo cancellation tools i have learned that echo on this line exceeds 1024ms`
`10:02.12`	`[hC]`	`who the hell fixes that! heh`
`10:05.04`	`SheriF_SpacE`	`what good value for maxjbuffer?`
`10:06.04`	`SheriF_SpacE`	`and maxexectbuffer`
`10:06.05`	`*** join/#asterisk The_Gus (n=The_Gus@81.200.133.119)`
`10:06.11`	`The_Gus`	`morn all :-)`
`10:10.41`	`*** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie)`
`10:13.29`	`RoyK`	`<PROTECTED>`
`10:14.43`	`FuriousGeorge`	`which is the option that adjusts the address rtp is sent to for nat? isnt that externip?`
`10:14.51`	`FuriousGeorge`	`for sip`
`10:15.46`	`FuriousGeorge`	`SheriF_SpacE: i just noticed 1.4 has more jitter buffer options in sip.conf`
`10:16.15`	`FuriousGeorge`	`i guess ill try downgrading to 1.2.x and seeing if that fixes this ridiculous nat issue`
`10:16.15`	`SheriF_SpacE`	`FuriousGeorge: i'm using 1.2`
`10:16.24`	`SheriF_SpacE`	`FuriousGeorge: cool :)`
`10:16.34`	`The_Gus`	`<PROTECTED>`
`10:16.59`	`The_Gus`	`which directory should I create with gsm files in order to make asterisk reading the message in the language I wish (italian)?`
`10:17.19`	`FuriousGeorge`	`forget downgrading i dont have the patience for it right now`
`10:17.25`	`FuriousGeorge`	`grrr`
`10:22.01`	`*** join/#asterisk blueneon (i=hfklows@dsl-146-28-12.telkomadsl.co.za)`
`10:22.51`	`blueneon`	`for some reason my TDM400 keeps cutting out spontaniously and all that is heard is a very loud white noise / hissing and the line is made dead (no calls can call in or out as the card seems to hold the line open)`
`10:22.55`	`blueneon`	`any ideas?`
`10:23.57`	`*** join/#asterisk pif (n=ldm@zenon.apartia.fr)`
`10:25.50`	`FuriousGeorge`	`blueneon: cron asterisk to restart`
`10:25.57`	`FuriousGeorge`	`what zaptel drivers are u using?`
`10:26.00`	`FuriousGeorge`	`i used to get that problem`
`10:26.15`	`blueneon`	`even if i restart asterisk its still dead`
`10:26.21`	`blueneon`	`i have to shut down the box and reboot to fix`
`10:26.47`	`blueneon`	`i forgot the zaptel version how can i check`
`10:27.18`	`FuriousGeorge`	`no, i mean restart asterisk daily to prevent it from happening`
`10:27.28`	`blueneon`	`zaptel-1.2.10`
`10:29.09`	`SheriF_SpacE`	`blueneon: i used to have this problem 1 i made a ground to the box`
`10:29.20`	`SheriF_SpacE`	`2 i make sure that i'm loading the correct modules`
`10:29.21`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com)`
`10:29.42`	`SheriF_SpacE`	`blueneon: ur card tdm400p has any FXS modules?`
`10:30.20`	*** join/#asterisk blueneon` (i=hfklows@dsl-146-28-12.telkomadsl.co.za)
`10:30.29`	blueneon`	`bah and now its seems its killing me dsl too`
`10:30.30`	blueneon`	`hehe`
`10:30.37`	blueneon`	`dont know if u got my last message`
`10:31.01`	blueneon`	`but basically asterisk has been restarted a few times yet the prob persists`
`10:31.59`	`*** part/#asterisk rkr245 (n=ravi@cw.callsat-telecom.com)`
`10:36.13`	blueneon`	`:( help?`
`10:38.28`	`axscode`	`anyone tried playing a video on asterisk? or its only pass through?`
`10:41.02`	`The_Gus`	`I can't fix a outisbusy problem : outbound calls to toll free 800. works, any other number like 32. for cellular phones gives 404 on both sip trunks I use. I've no other dial plan than "."`
`10:41.08`	`The_Gus`	`Any suggestions?`
`10:46.24`	`*** join/#asterisk NDT (n=nunya@cpe-24-195-72-2.nycap.res.rr.com)`
`10:47.04`	`SheriF_SpacE`	`blueneon: ground the asterisk , and make sure u have the correct module running .. ur card has FXS ports?`
`10:47.41`	`NDT`	`Trying to use sip realtime still LOL... Anyone see anything wrong here? http://pastebin.ca/232460 Keep getting Username/auth name mismatch but put the same info in sip.conf xlite registers fine...`
`10:50.12`	`NDT`	`mysql,mydatabase,sipbuddies <--is what it really is...now extensions realtime works fine`
`10:50.17`	`SheriF_SpacE`	`NDT:make sure ur res config is fine`
`10:51.09`	`NDT`	`it is because you can see the output of >> realtime load sipusers name 1000`
`10:51.15`	`NDT`	`in that pastebin`
`10:51.39`	`NDT`	`And the fact that extensions is running from the same db`
`10:52.22`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`10:52.29`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
`10:52.38`	`hi365`	`Hi!`
`10:52.41`	`NDT`	`driving me nuts cause it just doesn't make any sense heh`
`10:52.58`	`hi365`	`tzafrir: ma nishmah?`
`10:54.07`	`hi365`	`anyone using 1.1.1.14 om a grandstream 2000?`
`10:54.44`	`pif`	`retch`
`10:58.53`	`*** join/#asterisk hello2007 (n=test@mail.splendor.net)`
`11:03.22`	`hello2007`	`i m trying to configure ad digium card with 4 fxo mudules:`
`11:03.23`	`hello2007`	`i configured`
`11:03.23`	`hello2007`	`Zaptel.conf`
`11:03.23`	`hello2007`	`fxsks=1-4`
`11:03.23`	`hello2007`	`Zapata.conf`
`11:03.23`	`hello2007`	`signalling = fxs_ks`
`11:03.25`	`hello2007`	`context=splendorout`
`11:03.27`	`hello2007`	`group=2`
`11:03.29`	`hello2007`	`channel => 1-4`
`11:03.48`	`hello2007`	`but i m getting when i reload`
`11:03.51`	`hello2007`	`Nov 1 13:56:45 ERROR[21798]: chan_zap.c:10305 setup_zap: Unable to reconfigure channel '1'`
`11:03.52`	`hello2007`	`Nov 1 13:56:45 WARNING[21798]: chan_zap.c:11067 reload: Reload of chan_zap.so is unsuccessful!`
`11:04.05`	`hello2007`	`what did i do wrong?`
`11:06.26`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`11:19.41`	`*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`11:20.33`	`*** join/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it)`
`11:20.38`	`*** part/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it)`
`11:21.55`	`*** join/#asterisk ftexcom (n=ftexcom@169.Red-88-2-81.staticIP.rima-tde.net)`
`11:26.06`	`hello2007`	`nothing is wrong i should have stopped and restrat asterisk for changes to takes effect`
`11:29.44`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`11:33.10`	`aadilismail`	`hi`
`11:35.47`	`puzzled`	`morning`
`11:37.31`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
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`11:44.11`	`*** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr)`
`11:46.03`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`11:46.32`	`BurtyB`	`well i found a crappy solution to my problem of a second call cutting off the first.... VPN around the shitty router`
`11:48.04`	`HarryR`	`Ohh, which router was that?`
`11:48.32`	`HarryR`	`(brand, model etc.)`
`11:48.43`	`BurtyB`	`its a bethere one ... sec shudder`
`11:48.55`	`BurtyB`	`SpeedTouch 780`
`11:51.12`	`HarryR`	`ahah, never heard of any problems with that model before, but I love it when broken implementations cause more problems than they could solve :)`
`11:51.30`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`11:52.05`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
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`12:02.54`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`12:02.54`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) Join #freepbx for freepbx/trixbox support.`
`12:03.40`	`*** join/#asterisk jontow (i=jontow@hijacked.us)`
`12:05.47`	`*** join/#asterisk lorinc (n=ang@caracas-1359.adsl.interware.hu)`
`12:07.01`	`SheriF_SpacE`	`anyone knows how to benchmark asterisk sserver ?`
`12:08.41`	`*** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br)`
`12:11.13`	`*** join/#asterisk blueneon (i=hfklows@dsl-146-28-12.telkomadsl.co.za)`
`12:12.41`	`*** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br)`
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`12:16.32`	`*** join/#asterisk ambriento (n=ambrient@201-27-80-188.dsl.telesp.net.br)`
`12:17.59`	`blueneon`	`Im getting this error: zt_handle_event: Didn't finish Caller-ID spill. Cancelling.`
`12:18.08`	`blueneon`	`and caller id isnt getting sent to my internal lines`
`12:18.10`	`blueneon`	`any idea?`
`12:19.15`	`mfroes`	`can anyone tell what's the temporary greeting on voicemail is ???? the sound file is not used anywhere on the code besides the part to record it`
`12:19.19`	`*** join/#asterisk aadilismail (n=adilisma@202.38.55.114)`
`12:19.22`	`aadilismail`	`hi`
`12:29.12`	`SheriF_SpacE`	`i want to stop asterisk listen on dundi port even i don't have dundi.conf`
`12:29.42`	`SheriF_SpacE`	`mfroes: sorry can u ask in anothe way ?`
`12:32.14`	`mfroes`	`SheriF_SpacE: where is used the temporary greeting ?? i have oppened the source code of voicemail(.c) and checked the name of the file where it is recorded. tried to find where it would be used again. but it is not used on any other part of voicemail.c`
`12:33.23`	`SheriF_SpacE`	`mfroes: what ur trying to do ?`
`12:33.27`	`*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)`
`12:34.19`	`mfroes`	`SheriF_SpacE: figure out what temporary greeting is for`
`12:39.40`	`SheriF_SpacE`	`mfroes: oh ic hmm no i don't think i can help`
`12:39.45`	`*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)`
`12:40.17`	`*** join/#asterisk evilstoy (i=HydraIRC@83-131-0-96.adsl.net.t-com.hr)`
`12:43.40`	`blueneon`	`I cant for the life of me figure out why im getting Didn't finish Caller-ID spill. Cancelling.`
`12:45.31`	`*** join/#asterisk Skymarshal (n=Skymarsc@195.141.78.162)`
`12:46.16`	`SheriF_SpacE`	`how to stop dundi ? unload what?`
`12:47.21`	`puzzled`	`in modules.conf add noload => pbx_dundi.so or unload it from the cli`
`12:48.32`	`puzzled`	`blueneon: analog line? maybe add a small wait before you answer to give * more time to receive the callerid`
`12:49.09`	`SheriF_SpacE`	`puzzled: thx :)`
`12:51.12`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`12:51.44`	`Skymarshal`	`I am having troubles to call into ISDN in Switzerland. I am using a Junghanns QuadBRI and the same configuration doesn't have a problem in Germany. Incoming calls (from ISDN) are working fine but I can't call out. Is there a configuration difference between Germany and Switzerland ISDN wise?`
`12:53.01`	`blueneon`	`puzzled how?`
`12:53.15`	`blueneon`	`i put Wait(5) before Answer(), same message`
`12:55.01`	`*** join/#asterisk Bat2001 (n=Bat2001@62-213-205-18.colo.kangaroot.net)`
`12:56.43`	`*** join/#asterisk fbnts (n=root@mail.vidicom.co.uk)`
`12:59.11`	`Bat2001`	`Hi all - Does anyone know why Voicemail would work internally but not for an external call over a SIP trunk`
`12:59.58`	`fbnts`	`Hi everyone, I have recently bought 3 Cisco 7940 phones. (Used to have a 7960 a while back) but its currently set to Cisco Call Manager not SIP.`
`13:00.21`	`fbnts`	`No problem I thought, I have found the Cisco upgrade page and have the latest firmware however it won't load it.`
`13:00.41`	`fbnts`	`Apparantly I need to upgrade to SIP 7.1 first and then up to SIP 8`
`13:00.45`	`saam4u`	`hi`
`13:01.03`	`fbnts`	`but the cisco site only lists the latest version. Any idea where I can get the old version from?`
`13:01.13`	`aadilismail`	`hi`
`13:09.35`	`*** join/#asterisk lorinc (n=ang@caracas-3779.adsl.interware.hu)`
`13:09.52`	`*** join/#asterisk RoyK (n=roy@213.160.242.90)`
`13:14.52`	`*** join/#asterisk inspired (n=mikael@85.221.7.59)`
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`13:38.45`	`tzafrir`	`I've just extracted zaptel 1.4-beta2 from tag. When I run ./configure; make if fails to build as nothing tries to run ./configure in menuselect/mxml`
`13:39.22`	`tzafrir`	`Now I personally think that menuselect should not be a requirement for the default target. But I still expect the default target to work...`
`13:40.01`	`tzafrir`	`What am I missing? Who is supposed to call ./configure in mxml?`
`13:41.10`	`saam4u`	`need help .. new to asterisk " want to dial digits and get response from asterisk ??`
`13:41.48`	`puzzled`	`saam4u: download or buy the Asterisk book and read that first`
`13:43.09`	`SheriF_SpacE`	`tzafrir: default targets works`
`13:43.12`	`*** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it)`
`13:43.27`	`Sasch`	`can help me with zaptel source and my kernel ??`
`13:43.48`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`13:43.56`	`tzafrir`	`Sasch, we can try, if you'll be more specific`
`13:44.03`	`Sasch`	`ok <tzafrir>`
`13:44.05`	`SheriF_SpacE`	`saam4u: the head lines are 1- crete sip account in asterisk 2- adjust dialplan for this extention , 3- get a sip softpone and test.`
`13:44.24`	`Sasch`	`i install my ubuntu with kernel Linux neo 2.6.15-27-server`
`13:44.28`	`SheriF_SpacE`	`tzafrir: i can't try now but ihad 1.4 compiled few days ago`
`13:44.55`	`Sasch`	`<tzafrir> i download linux-headers-2.6.15-27-server`
`13:45.01`	`SheriF_SpacE`	`Sasch: u need kerlen-source and kernle headers`
`13:45.22`	`Sasch`	`beacase when i lunch make linux26`
`13:45.24`	`Sasch`	`all work`
`13:45.25`	`tzafrir`	no. linux-headers-`uname -r` is good enough. Actually: better
`13:45.29`	`*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-166-226.buff.east.verizon.net)`
`13:45.37`	`SuPrSluG`	`hello`
`13:45.44`	`Sasch`	`but when i lunch make install return`
`13:45.49`	`*** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)`
`13:45.52`	`Sasch`	`Unknown kernel build version requested... exiting.`
`13:45.55`	`*** join/#asterisk prttp (i=achi@187.Red-81-44-146.dynamicIP.rima-tde.net)`
`13:46.09`	`tzafrir`	(for Ubuntu, or Debian Etch/sid. For Sarge: kernel-headers--`uname -r` )
`13:46.29`	`tzafrir`	Sasch, apt-get install linux-headers-`uname -r`
`13:46.40`	`tzafrir`	`maybe you got the version wrong`
`13:47.05`	`saam4u`	`want to dial digits through user agent like 44234xnxnx and 4323233 and want both different response from asterisk like .. "busy" or "congestion" or hold etc... ??`
`13:47.19`	`Sasch`	`<tzafrir> i install it in /usr/src/ and i link /usr/src/linux to it`
`13:47.23`	`Sasch`	`but not work`
`13:47.29`	`Sasch`	`when i lunch make install return`
`13:47.46`	`Sasch`	`[: 66: ==: unexpected operator`
`13:47.46`	`Sasch`	`Unknown kernel build version requested... exiting.`
`13:47.46`	`Sasch`	`make: *** [install] Error 1`
`13:47.57`	`tzafrir`	`Though Ubuntu have a bad habbit of breaking the ABI on each kernel release. This means you'll have to rebuild the zaptel-modules package often`
`13:48.31`	`tzafrir`	`Sasch, no, you don't need the symlink /usr/src/linux . Chances are they'll only cause confusion`
`13:48.56`	`tzafrir`	The kernel-headers packages contain the proper symlink from /lib/modules/`uname -r`/build
`13:48.56`	`Sasch`	`a ok`
`13:49.48`	`*** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-202.usadatanet.com)`
`13:51.08`	`Sasch`	`<tzafrir> where is the better distro of linux to run asterisk ??`
`13:51.15`	`ctaloi`	`Hey all, anyone around? I've got some questions about the best way to use MOH`
`13:51.32`	`ctaloi`	`tzafrir - I use CentOS`
`13:51.35`	`ctaloi`	`works for me`
`13:51.51`	`tzafrir`	`ctaloi, I use Debian. Works For Me`
`13:52.06`	`florz`	`ctaloi: Nope, there is noone around, in particular noone who could answer any questions.`
`13:52.12`	`ctaloi`	`yup debian works well for asterisk too`
`13:52.16`	`ctaloi`	`ha, that's too bad`
`13:52.31`	`ctaloi`	`trying to figure out the best way to do MOH on a larger system`
`13:53.39`	`Sasch`	`<tzafrir> i erase /usr/src/linux`
`13:53.43`	`Sasch`	`but not work`
`13:53.51`	`Sasch`	`when i lunch make install return`
`13:53.56`	`Sasch`	`Unknown kernel build version requested... exiting.`
`13:53.56`	`Sasch`	`make: *** [install] Error 1`
`13:54.06`	`tzafrir`	`Sasch, but do you have the right linux headers package?`
`13:54.14`	`Sasch`	`o yes`
`13:54.15`	`tzafrir`	Sasch, apt-get install linux-headers-`uname -r`
`13:54.22`	`Sasch`	i lunch apt-get install linux-headers-`uname -r`
`13:56.34`	`SheriF_SpacE`	`Sasch: aptitude install linux-headers linux-source`
`13:57.06`	`Sasch`	`when i lunch make linux26`
`13:57.12`	`Sasch`	`return a lot of warning`
`13:57.13`	`Sasch`	`include/asm/mpspec_def.h:78: warning: �packed� attribute ignored for field of type �unsigned char[6]�`
`13:57.44`	`*** join/#asterisk zotz (n=zotz@24.244.133.107)`
`13:57.59`	`Sasch`	`<SheriF_SpacE> i lunch it`
`13:58.22`	`SheriF_SpacE`	`Sasch: just make`
`13:58.32`	`SheriF_SpacE`	`and the warning is okay`
`14:01.22`	`b11d\|bbl`	`morning lads`
`14:05.21`	`*** join/#asterisk crosslimits (n=adilisma@202.38.55.114)`
`14:05.24`	`crosslimits`	`hi`
`14:06.40`	`b11d`	`hi`
`14:06.42`	`*** join/#asterisk alerios (n=alerios@190.24.97.148)`
`14:07.17`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:09.48`	`*** join/#asterisk sp00kz (n=ilubj00@141.155.224.162)`
`14:10.48`	`*** join/#asterisk skirmisha (n=vk@213.144.225.179)`
`14:10.54`	`skirmisha`	`hello guys`
`14:11.02`	`skirmisha`	`have a quick question`
`14:12.54`	`b11d`	`so.. ask it then?`
`14:14.17`	`skirmisha`	`let me figure it out`
`14:14.27`	`skirmisha`	`in chan_sip.c`
`14:14.46`	`skirmisha`	`i want to write a rule which must be checked every minute`
`14:15.01`	`skirmisha`	`is that possible and if so where exactly should i put it`
`14:15.02`	`b11d`	`no idea`
`14:15.34`	`skirmisha`	`anyone else who knows?`
`14:20.39`	`b11d`	`good luck with that one`
`14:21.39`	`sp00kz`	`anyone use a cisco 7970 color touch phone using SIP?`
`14:21.45`	`DrukenHME`	`you want to physically change chan_sip.c?`
`14:21.52`	`DrukenHME`	`go into #asterisk-dev`
`14:22.40`	`jhiver`	`hi all`
`14:22.50`	`jhiver`	`I have something strange going on with AGI`
`14:22.58`	`jhiver`	`on the asterisk CLI i see:`
`14:23.02`	`jhiver`	`Nov 1 14:31:21 WARNING[30554]: res_agi.c:1101 handle_exec: Could not find application (EXEC)`
`14:23.54`	`*** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net)`
`14:25.41`	`*** join/#asterisk kiddy (n=kiddy@59.93.32.241)`
`14:25.42`	`*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`14:25.45`	`b11d`	`my god sasch has a horrible quit message`
`14:26.18`	`DrukenHME`	`i'm assuming it's supposed to be a austin powers thing....`
`14:26.23`	`DrukenHME`	`failed miserably....`
`14:26.24`	`b11d`	`possibly`
`14:26.52`	`kiddy`	`I cannot make or recieve call on broadvoice connection , Its saying "the number you have dialed is not inserted" while I make a call to the bv connection`
`14:27.17`	`DrukenHME`	`inserted?`
`14:27.43`	`kiddy`	`and "All circuits are busy now " when I make a call from that number`
`14:28.01`	`sp00kz`	`anyone use a cisco 7970 color touch phone using SIP?`
`14:28.36`	`DrukenHME`	`kiddy: did it work before?`
`14:28.44`	`kiddy`	`yes`
`14:28.52`	`DrukenHME`	`maybe bv is having issues?`
`14:28.56`	`*** join/#asterisk prttp (i=achi@187.Red-81-44-146.dynamicIP.rima-tde.net)`
`14:29.22`	`kiddy`	`no I have a teliax connection too . It also have same issue`
`14:29.36`	`kiddy`	`"Dial failed due to CHANUNAVAIL"`
`14:29.52`	`DrukenHME`	`have you restarted asterisk lately?`
`14:30.04`	`kiddy`	`no`
`14:30.27`	`b11d`	`sp00kz... i havent, but was looking at those.. whats the deal with them?`
`14:30.37`	`DrukenHME`	`maybe ya should try that.. hehehe i've had that fix stupid things...`
`14:30.43`	`kiddy`	`first there was a issue with the registration problem to the broadvoice and teliax connection`
`14:30.47`	`mfroes`	`can someone tell me why at least once a day my client's asterisk crashes and i have to reboot`
`14:30.51`	`sp00kz`	`the newest firmware doesnt work with asterisk, i need 1 version older, 8.0.3`
`14:30.58`	`b11d`	`does it core dump mfroes?`
`14:30.59`	`sp00kz`	`8.0.4 is broken, fails to register`
`14:31.09`	`kiddy`	`when I reinstalled asterisk and rebooted the server it works for the registration problem`
`14:31.19`	`kiddy`	`now I cannot make a call or recieve a call`
`14:31.33`	`b11d`	`ahh that sucks`
`14:32.09`	`kiddy`	`DrukenHME : can you help me to understand why this is happening ?`
`14:32.12`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`14:32.12`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:32.23`	`sp00kz`	`7960/7940's work like a champ`
`14:32.32`	`b11d`	`yeah i've got some 7940's`
`14:32.36`	`Qwell`	`except SIP sucks :p`
`14:32.44`	`b11d`	`why?`
`14:32.44`	`Qwell`	`yeah, you heard me`
`14:32.55`	`Qwell`	`skinny is just way cooler`
`14:32.57`	`sp00kz`	`I miss some of the features of sccp`
`14:33.24`	`b11d`	`im all about string..`
`14:33.28`	`sp00kz`	`but i need the 8-0-3 firmware for my 7970 :(`
`14:33.30`	`b11d`	`fuck this voip shit.. im rolling out tin cans`
`14:33.34`	`DrukenHME`	`kiddy: your registering to them ?`
`14:34.09`	`b11d`	`can you hear me?`
`14:34.35`	`DrukenHME`	`nope... must be one way audio... check the rtp :)`
`14:34.40`	`b11d`	`hahaha`
`14:34.56`	`[TK]D-Fender`	`Qwell : SIP sucks? BTW.... hows that chan_skinny botnet of your yours coming along? ;)`
`14:35.02`	`kiddy`	`DrukenHME : Yes`
`14:35.14`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`14:35.17`	`Qwell`	`[TK]D-Fender: got me - I didn't write that code`
`14:35.23`	`b11d`	`hey TK.. i've got those "buddy" lists going.. thanks for that advice.`
`14:35.30`	`[TK]D-Fender`	`b11d: np`
`14:35.53`	`b11d`	`one question though.. how do you easily manage all those config files on the provisioning server?`
`14:36.09`	`b11d`	`im going to have like 250+ phones.. and theres like 3+ files per phone already.. its a nighmare to wade though!`
`14:36.21`	`Qwell`	`b11d: SEPDefault.cnf.xml, heh`
`14:36.37`	`b11d`	`?`
`14:37.22`	`b11d`	`oh yeah`
`14:38.12`	`kiddy`	`DrukenHME:what this means "Dial failed due to CHANUNAVAIL" ?`
`14:39.25`	`mfroes`	`b11d: what do you mean core dump ???`
`14:40.09`	`b11d`	`you said * was crashing every day.. is it dumping a .core file?`
`14:40.12`	`kiddy`	`DrukenHME:also I get "fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf"`
`14:40.13`	`b11d`	`or is it hanging? or what`
`14:40.27`	`*** join/#asterisk quidpro (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com)`
`14:40.43`	`mfroes`	`nope ... i need it right ?`
`14:40.52`	`b11d`	`ok.. lets go back to the beginning.`
`14:40.58`	`b11d`	`what is it doing to make you reboot it?`
`14:40.59`	`*** join/#asterisk TGW (n=toronto@gw.gibbs.iasl.com)`
`14:41.51`	`mfroes`	`it simply stops responding ... no one can dial .. receive .. auteticate`
`14:42.07`	`b11d`	`and the system is useable at the console? its not frozen or anything?`
`14:42.12`	`b11d`	`asterisk is still "running" ?`
`14:42.45`	`TGW`	`Hi all, I have a quick question that I'm hoping might be a simple answer. I have a * box running 1.2, and I have 2 Cisco AS5300 as my Media Gateways, if I take a DID and Dail it directly to an exten it work fine, however if I take a call to a menu and then have the user dial say ext 210, the calling party doesn't hear ringing all the time they hear what I would say is broken ringing almost like jitter....and its not consistant, some times it gets 1 ring`
`14:42.48`	`TGW`	`any ideas?`
`14:43.00`	`mfroes`	`if i try to start by asterisk -r ,..... it cant`
`14:43.08`	`mfroes`	`it is still running`
`14:43.28`	`b11d`	`hmm.. I take it you're at the console, or are you SSHing into the box?`
`14:43.49`	`DrukenHME`	`kiddy: i don't know about the localprefix thing... however, the dial failed do to chanunavail means just that, channel unavailable...`
`14:43.54`	`mfroes`	`sshing`
`14:44.04`	`b11d`	`so the system is still responsive.. its just that asterisk hangs.`
`14:44.07`	`DrukenHME`	`what are you using? SIP/IAX/SKINNY ?`
`14:44.16`	`b11d`	`what OS and what version of Asterisk?`
`14:44.26`	`quidpro`	`Anybody here used CAGI for AGI C development?`
`14:44.34`	`mfroes`	`fedora core 5, asterisk-1.2.12.1`
`14:44.51`	`b11d`	`and when did this problem start?\`
`14:45.22`	`mfroes`	`it always had ... like .. a week without rebooting .. it crashes`
`14:45.23`	`b11d`	`did you do any upgrades or anything just before the problem arose?`
`14:45.39`	`mfroes`	`nope .. no upgrades`
`14:46.12`	`mfroes`	`even when i used 1.2.10 ....`
`14:46.20`	`b11d`	`hmm`
`14:46.32`	`b11d`	`i'll say the ususal "upgrade to the latest"`
`14:46.53`	`b11d`	`but.. i dunno on that one`
`14:47.08`	`mfroes`	`that what i have been told ... then i upgrated to 1.2.12.1 and still crashing`
`14:47.14`	`b11d`	`1.2.13 is out`
`14:47.39`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`14:47.39`	`*** mode/#asterisk [+o mog] by ChanServ`
`14:47.51`	`b11d`	`does it crash at the same time? or when a specific number of people have registered against it?`
`14:48.22`	`b11d`	`what about network traffic? are you seeing any odd stuff? are you being attacked?`
`14:48.38`	`b11d`	`i guess i really dont know though`
`14:48.42`	`b11d`	`i should stop speculating :)`
`14:49.33`	`b11d`	`do you kill asterisk and then restart it, or do you actually need to reboot the server before asterisk begins working again?`
`14:52.01`	`*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)`
`14:52.49`	`*** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu)`
`14:54.49`	`b11d`	`anyone here ever use a vg224 with asterisk?`
`14:55.42`	`b11d`	`as soon as i get this beast working, im posting documentation to voip-info..`
`14:56.46`	`TGW`	`Is there anyway to tell if asterisk is using the ztdummy for a timing source?`
`14:56.55`	`*** join/#asterisk dasenjo (n=dasenjo@63.245.86.137)`
`14:57.57`	`AndyCap`	`TGW: some zap show or show zap command?`
`14:58.50`	`iulius`	`lsmod(8) will tell you which kernel modules are loaded, one of which would be ztdummy`
`14:59.26`	`iulius`	`To find out if you have a timing source of any kind, you could try dialing into a meetme conference`
`14:59.48`	`TGW`	`I see the ztdummy loaded in lsmod, and if I do a zap show it does show ztdummy and meetme does work I have 5 people in it`
`14:59.53`	`TGW`	`so I guess that answers my question`
`15:00.09`	`TGW`	`I'm still trying to figure out why ringing sounds choppy when the call goes out my media gateways`
`15:00.23`	`[TK]D-Fender`	`TGW: load chan_impiricalevidence.so`
`15:01.08`	`*** join/#asterisk lorinc (n=ang@caracas-3779.adsl.interware.hu)`
`15:01.56`	`*** join/#asterisk jarg (n=jarg@200.56.225.61)`
`15:02.19`	`DrukenHME`	`life as a fish has got to suck... never sleep, always swimming, and absolutly no variety in food, unless i turn canible and chew on my brother`
`15:02.56`	`Ox0F0-0FF`	`And not enough brain to think about this shit... so no worries for the fishies :P`
`15:04.16`	`DrukenHME`	`not enough brains to pinch the loaf OF shit...`
`15:04.42`	`SplasPood`	`Fish sleep, I think...`
`15:05.02`	`DrukenHME`	`if they do, they must sleep swim... hehehe`
`15:05.41`	`TGW`	`TK D-Fender: is that a stock module or one that I need to find and compile in ?`
`15:05.55`	`SplasPood`	`You can walk without thinking about it, I presume fish could swim the same`
`15:06.03`	`sp00kz`	`I need SIP70.8-0-3S firmware for the 7970 :(`
`15:06.47`	`drfreeze`	`hi`
`15:07.08`	`drfreeze`	`I just noticed that my cdr-csv log files are not recording outgoing calls`
`15:07.20`	`drfreeze`	`Do I need to enable this?`
`15:08.35`	`SplasPood`	`It should be...`
`15:08.43`	`*** join/#asterisk viler (i=1000@ip-70-228.telesat.com.co)`
`15:09.13`	`SplasPood`	`drfreeze: you're getting incoming ones in there?`
`15:09.16`	`*** join/#asterisk davdunc (n=dduncan@bwext.kpimdp.com)`
`15:09.34`	`*** join/#asterisk Sedorox (n=brandon@smartserv/cna/Sedorox)`
`15:10.14`	`drfreeze`	`yes,`
`15:10.24`	`drfreeze`	`SplasPood: incoming and internal`
`15:10.29`	`Sedorox`	`Sweet, my school is teaching a voip class, and concentrating on asterisk :)`
`15:10.42`	`SplasPood`	`and theres... NOTHING when making an outgoing call?`
`15:10.49`	`Sedorox`	`school==college`
`15:10.49`	`SplasPood`	`and you're not using NoCDR() anywhere?`
`15:11.03`	`quidpro`	`Anybody done any AGI programming in C?`
`15:11.14`	`mfroes`	`b11d: i kill it and restart only asterisk`
`15:12.04`	`b11d`	`ohh`
`15:12.06`	`b11d`	`hmm`
`15:12.08`	`drfreeze`	`SplasPood: don't think so`
`15:12.11`	`b11d`	`yeah thats bizzare man`
`15:12.30`	`drfreeze`	`SplasPood: nowhere in /etc/asterisk/*`
`15:12.42`	`*** join/#asterisk Modcuts (n=bob@lan.proporta.com)`
`15:13.07`	`hello2007`	`if i want to sell asterisk and giving support on it is it legal?`
`15:13.37`	`DrukenHME`	`you cannot sell asterisk, however, you can GIVE asterisk and charge to support it...`
`15:13.41`	`SplasPood`	`drfreeze: thats a bit.. odd...`
`15:13.56`	`SplasPood`	`drfreeze: anything odd looking towards the end of the call on console`
`15:13.59`	`SplasPood`	`or in your logs..`
`15:14.09`	`hello2007`	`so i give asterisk freely and i charge only on th support?`
`15:14.20`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`15:14.42`	`quidpro`	`Or you could sell it on the ahrdware and mark up the hardware. :)`
`15:14.56`	`drfreeze`	`SplasPood: setting verbose = 30 should show calls being made,right?`
`15:15.06`	`SplasPood`	`heh yea`
`15:15.30`	`SplasPood`	`I gotta run in a couple min`
`15:15.55`	`hello2007`	`quidpro: what do you mean by sell it on the hardware and markup the hardware?`
`15:16.09`	`drfreeze`	`SplasPood: K`
`15:16.59`	`SplasPood`	`hello2007: you sell an EXPENSIVE computer`
`15:17.05`	`SplasPood`	`hello2007: that happens to be setup as a PBX`
`15:17.39`	`SplasPood`	`Then... You make them bend over on service contracts for each time they need a change/service`
`15:17.42`	`SplasPood`	`thats how its done`
`15:17.53`	`SplasPood`	`thats how all the big boys do it`
`15:17.58`	`SplasPood`	`:P`
`15:18.53`	`hello2007`	`SplasPood:and is this legal`
`15:18.58`	`hello2007`	`?`
`15:20.01`	`SplasPood`	`yer selling yer support and setup of a piece of software on SUPER DUPER hardware`
`15:20.06`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`15:20.07`	`*** mode/#asterisk [+o russellb] by ChanServ`
`15:20.19`	`davdunc`	`hello2007, have you read the GPL?`
`15:20.26`	`SplasPood`	`gpl?`
`15:21.09`	`*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`15:21.13`	`davdunc`	`SplasPood, GNU General Public License!`
`15:21.14`	`hello2007`	`yes but i didn t really understand much,it s a little bit complicated`
`15:21.30`	`davdunc`	`You can't sell GPL software.`
`15:21.32`	`RaYmAn-Bx`	`it's perfectly legal for anyone to sell asterisk, as long as they still provide source.`
`15:21.37`	`RaYmAn-Bx`	`davdunc: wrong`
`15:21.40`	`RaYmAn-Bx`	`you can sell it just fine`
`15:21.43`	`davdunc`	`<PROTECTED>`
`15:21.45`	`RaYmAn-Bx`	`if you can get someone to buy it`
`15:22.01`	`SplasPood`	`davdunc: yea I know, I was being funny.. or at least I thought I was, and thats all that matters ;)`
`15:22.08`	`davdunc`	`:)`
`15:22.09`	`RaYmAn-Bx`	`would be stupif if you couldn't sell GPL software.`
`15:22.21`	`davdunc`	`RaYmAn-Bx, well. . . you can't.`
`15:22.31`	`SplasPood`	`You can charge a nominal fee for the media :P`
`15:22.34`	`RaYmAn-Bx`	`davdunc: are you entirely sure? Because I'm pretty sure you are wrong.`
`15:22.38`	`RaYmAn-Bx`	`hmm`
`15:22.49`	`RaYmAn-Bx`	`good point, possibly`
`15:22.58`	`SplasPood`	`anyway`
`15:23.03`	`davdunc`	`RaYmAn-Bx, I am 100% certain that if you want to sell, you need to publish under a different license.`
`15:23.11`	`davdunc`	`Like mysql`
`15:23.16`	`hello2007`	`so i don t really need to sell it installed on a hardware`
`15:23.41`	`davdunc`	`hello2007, just sell your installation and configuration time -- and the hardware.`
`15:23.49`	`RaYmAn-Bx`	`davdunc: http://www.gnu.org/philosophy/selling.html`
`15:24.14`	`*** join/#asterisk nayyares (n=Nayyar@202.163.91.46)`
`15:24.33`	`RaYmAn-Bx`	`according to that you can pretty much charge whatever you want`
`15:24.42`	`davdunc`	`yea. "Distributing free software for a fee"`
`15:24.58`	`*** join/#asterisk wepy (n=airuike@216-158-7-169.dca.net)`
`15:25.00`	`hello2007`	`can i change the logo?`
`15:25.03`	`jojuba`	`exit`
`15:25.08`	`*** part/#asterisk jojuba (n=Nayyar@202.163.91.46)`
`15:25.22`	`RaYmAn-Bx`	`davdunc: http://www.gnu.org/philosophy/selling.html#TOCHighOrLowFeesAndGPL read that bit`
`15:25.25`	`wepy`	`hello, i am considering using VOIP to make free soft-phone to soft-phone calls internationally...`
`15:25.28`	`RaYmAn-Bx`	`it says there is no limit`
`15:25.46`	`wepy`	`however, the two computers with the softphones are behind all NATs, etc..`
`15:25.46`	`davdunc`	`There is no limit on charging for distribution.`
`15:25.51`	`RaYmAn-Bx`	`you can sell it for whatever you want, but you can't charge for giving them the source if they ask, except for the nominal fee of course`
`15:26.07`	`wepy`	`is there any easy way to get this stuff working, i heard I might want to use some sort of SIP proxy perhaps...`
`15:26.09`	`RaYmAn-Bx`	`davdunc: same thing, essentially.`
`15:26.20`	`davdunc`	`right. and you must make the original source available to the customer.`
`15:26.41`	`davdunc`	`that is what got Linksys in trouble. They just released the binaries.`
`15:27.07`	`RaYmAn-Bx`	`sure`
`15:27.30`	`RaYmAn-Bx`	`that's exactly what I said. You can charge whatever you want for it as long as you provide source on request.`
`15:27.40`	`teknoprep`	`anyone have some nice business class ring tones for voip phones?`
`15:27.43`	`*** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com)`
`15:27.54`	`teknoprep`	`any format is fine .. i can just convert to wav`
`15:28.08`	`RaYmAn-Bx`	`I can't quite tell whether you are required to provide it to people who you haven't distributed binaries to`
`15:28.12`	`RaYmAn-Bx`	`probably not`
`15:28.20`	`davdunc`	`They altered the original code and did not make the source available. So hello2007 can change the logo, but he can't embed it in a binary distribution and not make the source with modifications available.`
`15:28.57`	`davdunc`	`please excuse my run-on sentence.`
`15:29.24`	`wepy`	`or.. are there any good web sites where i could find some SIP + NAT solutions?`
`15:29.28`	`RaYmAn-Bx`	`admittedly I didn't read the entire discussion, but did he indicate that he wouldn't provide sourcecode by request?`
`15:30.06`	`hello2007`	`davdunc:what do u mean by can't embed it in a binary distribution and not make the source with modifications available.`
`15:30.49`	`hello2007`	`changing the logo is considered source modification?`
`15:30.57`	`davdunc`	`hello2007, you can't alter the source, compile it, and then distribute the binaries without making your source available upon request.`
`15:31.21`	`davdunc`	`hello2007, if you change any part of the distribution, you have altered the source.`
`15:31.59`	`davdunc`	`Look at CentOs.org for a perfect example.`
`15:32.01`	`hello2007`	`but if changed the logo and made the source availbale is it doable?`
`15:32.01`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`15:32.15`	`davdunc`	`hello2007, yes.`
`15:32.35`	`RaYmAn-Bx`	`hello2007: you can change whatever you want as long as you provide the modified source code on request..well..you can't remove original copyrights and licenses of course`
`15:33.17`	`davdunc`	`RaYmAn-Bx, thanks for the clarification.`
`15:33.55`	`RaYmAn-Bx`	`davdunc: I can't tell whether you are putting me down or actually thanking me.`
`15:34.12`	`davdunc`	`Oh. Sorry. I am thanking you.`
`15:34.17`	`RaYmAn-Bx`	`okay, heh`
`15:34.26`	`davdunc`	`That was an important distinction.`
`15:34.34`	`*** join/#asterisk Trakkasure (n=Trakk@adsl-225-102-164.mia.bellsouth.net)`
`15:34.38`	`*** part/#asterisk wepy (n=airuike@216-158-7-169.dca.net)`
`15:34.51`	`RaYmAn-Bx`	`I spend too much time in channels where people put insults into every sentence more or less ;)`
`15:36.01`	`hello2007`	`ok ,thanks guys`
`15:36.10`	`davdunc`	`RaYmAn-Bx, heh. I understand.`
`15:36.42`	`*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.111.Dial1.SanJose1.Level3.net)`
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`15:53.30`	`b11d`	`dont pick at it`
`15:53.35`	`b11d`	`it'll get infected`
`15:53.36`	`PupenoR`	`I have an iax2 links that sudenly went unreachable and I can't find what's going on. The same computers have other iax2 links from and to other computers. With tcpdump I can see that there's iax2 traffic between the two computers. Any ideas how to trobleshoot this ?`
`15:53.36`	`HarryR`	`anybody interested in memcache support for asterisk...`
`15:54.04`	`HarryR`	`got a memcache dialplan/function working, now I'm just trying to work out how I can do some cool clustering things with it`
`15:54.21`	`b11d`	`neato`
`15:54.48`	`HarryR`	`gotta talk to our sip guys to see if things like sip failover are feasable`
`15:55.20`	`HarryR`	`(e.g. if I just replace all the astdb stuff with memcache... hehe fun fun:)`
`15:56.17`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:03.19`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net)`
`16:03.20`	`*** join/#asterisk rene1 (n=rene1@189.140.231.131)`
`16:03.26`	`rene1`	`hey all`
`16:04.12`	`b11d`	`hey rene`
`16:04.45`	`rene1`	`i have been working with sip peers happily but now i have to work with remote extensions. iax peers shine there,, i have them working in realtime mode... what it bothers me that iaxpeers does not seem to report on realtime iax peers, where the old sip show peers does... is this functionality missing in iax peers?`
`16:05.07`	`b11d`	`no idea..`
`16:06.14`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`16:09.37`	`*** join/#asterisk Heliocentric (n=anon@rrcs-65-34-29-126.se.biz.rr.com)`
`16:10.46`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
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`16:12.09`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.89)`
`16:12.41`	`PupenoR`	`Does Asterisk support IPv6 ? can it stablish IAX2 connections over IPv6 ?`
`16:13.04`	`mog`	`not today`
`16:13.12`	`mog`	`work is being done on it`
`16:13.17`	`HarryR`	`PupenoR, although people are throwing money at it: http://www.voip-info.org/wiki-Asterisk+bounty+IPv6`
`16:14.26`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com)`
`16:14.29`	`Sedorox`	`v6?`
`16:14.54`	`Sedorox`	`iirc, unless its changed recently, asterisk doesn't support v6 yet`
`16:15.23`	`b11d`	`I could drop v6 support into asterisk in like, ten minutes.`
`16:15.31`	`b11d`	`what'll I do with the other 50 minutes and 23 hours in the day?`
`16:15.35`	`b11d`	`thats the real question`
`16:15.43`	`b11d`	`ok thats a lie..`
`16:15.50`	`Sedorox`	`yes :p`
`16:16.50`	`Sedorox`	`oo there is a v6 patch`
`16:16.55`	`rene1`	`i am being silly asterisk just needs a rtcachefriends param to support realtime peers`
`16:16.57`	`Sedorox`	`I'll have to look into it`
`16:17.37`	`*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)`
`16:20.14`	`*** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net)`
`16:20.52`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:21.58`	`GaVak`	`Is there a way to affect how a TRANSFER call behaves? When my users transfer calls to eachother, it doesn't display the callerID of the original caller.`
`16:22.37`	`ManxPower`	`GaVak: blind transfer will show the original callerid. supervised/conference transfers will show transferer callerid unless you use a super secret Dial option.`
`16:23.12`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:23.13`	`*** mode/#asterisk [+o mog] by ChanServ`
`16:23.28`	`*** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu)`
`16:23.37`	`GaVak`	`I'm using incoming ZAP channels for the calls, and Polycom 501 SIP phones. The behavior now is that it shows the caller ID (Extension) of the person who transferred the call.`
`16:23.37`	`*** join/#asterisk pdtwork (n=ptinsley@209.12.249.243)`
`16:23.38`	`WildPikachu`	`hi guys`
`16:23.51`	`ManxPower`	`GaVak: Correct.`
`16:24.01`	`WildPikachu`	`can I add something like member => ZAP/xxxxxx/xx to my queues.conf file?`
`16:24.14`	`ManxPower`	`Want to know the super secret option? "show application dial" in the Asterisk CLI`
`16:24.28`	`ManxPower`	`WildPikachu: I think so, but why?`
`16:24.40`	`WildPikachu`	`ManxPower, to direct my office queue to my cellphone :)`
`16:26.08`	`*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)`
`16:26.43`	`GaVak`	`lol, ok I see the listing of dial commands.`
`16:27.28`	`WildPikachu`	`pefect!!!`
`16:27.32`	`WildPikachu`	`it works ManxPower`
`16:27.40`	`GaVak`	`grin Reading.`
`16:27.58`	`*** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net)`
`16:31.30`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`16:31.30`	`*** mode/#asterisk [+o anthm] by ChanServ`
`16:33.54`	`GaVak`	`I'm speculating that I need to set the phones up as 'screened'?`
`16:34.26`	`tzafrir`	`anyway, I reverted back to using a tarball of zaptel 1.4.0-beta2 and all was well. Maybe something in my checkout was bogus. Though I could not find any difference`
`16:35.11`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.168.211)`
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`16:37.28`	`*** part/#asterisk rene1 (n=rene1@189.140.231.131)`
`16:41.15`	`Heliocentric`	`Does anyone know if "supervision" is the same as "signalling"?`
`16:42.15`	`*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)`
`16:42.28`	`Heliocentric`	`My telco says we are using "wink start" signaling. They also say we are using "supervision t1 em"`
`16:43.23`	`*** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it)`
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`17:02.14`	`wepy`	`any good free softphones for windows that support STUN servers?`
`17:03.48`	`Modcuts`	`x-lite`
`17:04.28`	`wepy`	`so allegedly, if i setup my own stun server, all this SIP nonsense should just kind of work like magic?`
`17:05.11`	`AndyCap`	`wepy: or use a public one?`
`17:05.39`	`wepy`	`how does it work though?`
`17:05.47`	`Modcuts`	`indeed`
`17:06.52`	`wepy`	`as i understand it, SIP clients will open random ports and the other side has no way of knowing what they are, but even with a 3rd party on the outside telling the connecting client the proper ports, the firewalls will still be a problem right?`
`17:08.07`	`*** join/#asterisk javar (n=javar@69.79.134.24)`
`17:08.28`	`*** join/#asterisk jero (n=jerou@savoirfairelinux.net)`
`17:08.31`	`jero`	`hi`
`17:08.53`	`*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)`
`17:09.34`	`*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)`
`17:09.50`	`wepy`	`i mean, it's just really damn hard to make a phone call since everyone has a firewall..`
`17:09.53`	`syzygyBSD`	`any reasons why asterisk would stop writing the CDR log?`
`17:10.01`	`MRH2`	`Hi is there any way to check that the 1.2 build I am runninng is a 'make dont optimize' one`
`17:10.22`	`syzygyBSD`	`possibly old problems...`
`17:11.20`	`MRH2`	`(i built it ages ago and now have a crash'n trace)`
`17:11.53`	`docelmo`	`MRH2 if you just did make.. then you get the symbols.. You have to modify the makefile to remove the portion that keeps it from being optimized`
`17:12.01`	`ManxPower`	`syzygyBSD: logging partition full?`
`17:12.19`	`docelmo`	`syzygyBSD yes.. logger is unloaded, turned off, or partition full?`
`17:12.20`	`MRH2`	`yeah i can't remember which one i did`
`17:12.49`	`ManxPower`	`syzygyBSD: also maybe you rotated the logs OUTSIDE of Asterisk. Just do a 'asterisk -rx "logger rotate"'`
`17:13.02`	`syzygyBSD`	`partition was never full, I have a monitor checking and it sends me a message if it ever gets over 85%`
`17:13.20`	`syzygyBSD`	`the logger worked fine when I restarted asterisk`
`17:13.45`	`syzygyBSD`	`the CDR log was never rotated, it continued appending on restart`
`17:14.06`	`syzygyBSD`	`I am probably just going to say it was because it was only asterisk 1.0.10`
`17:14.27`	`*** join/#asterisk kilobit2001 (n=locid@206-248-134-147.dsl.teksavvy.com)`
`17:17.31`	`syzygyBSD`	`it was restarted 15 minutes before it started having the problem`
`17:18.52`	`*** join/#asterisk geoffl (n=geoff@gjctech.plus.com)`
`17:19.46`	`teknoprep`	`did voicepulse just go down?`
`17:19.47`	`FuriousGeorge`	`hey all. remote sip clients arent working here. the usual nat issues, rtp is not being sent to the clients externip, but instead to his local lan address, which does nor resolve obviously from where asterisk is`
`17:19.51`	`aptura`	`just looked in voip-info and the x100p was revived so what did mark do sell the rights of this card? thay claim its not a knock off but the original.`
`17:20.05`	`FuriousGeorge`	`the funny thing is though, i set up, externip, localnet, and the forewarded ports just like i always do?`
`17:20.25`	`aptura`	`http://www.x100p.com/`
`17:20.34`	`aptura`	`guess there is still a market for it.`
`17:22.14`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`17:22.26`	`syzygyBSD`	`FuriousGeorge: depending, the packets may say one thing and do another.`
`17:23.11`	`geoffl`	`Hi - I have a problem with Idefisk IAX client on Mac OS-X. Locally, it works just fine and also registers with Asterisk from the Internet. I can call that client and hold a two-way conversation. However, every time I dial from the Mac client, Asterisk rejects the call. The client is in the same context both when local and when remote. Any ideas?`
`17:23.19`	`FuriousGeorge`	`syzygyBSD: well i know the packets arent being sent to the right address cuz i dont have audio`
`17:24.09`	`FuriousGeorge`	`i thought nat=yes is supposed to rewrite the return address for the rtp stream in the sip header`
`17:24.26`	`FuriousGeorge`	`basically, i set this up the way i always do, the only difference being that this is 1.4 here`
`17:24.30`	`syzygyBSD`	`well, you said you had nat setup right`
`17:24.44`	`FuriousGeorge`	`yeah`
`17:24.49`	`syzygyBSD`	`I have heard other people having problems with nat in 1.4, I don't know for sure though`
`17:24.49`	`FuriousGeorge`	`no`
`17:24.55`	`FuriousGeorge`	`lol`
`17:25.05`	`syzygyBSD`	`I only run 1.2 or lower`
`17:25.20`	`FuriousGeorge`	`i meant that i have always been able to set up * with nat correctly, but with this 1.4beta 3 my efforts are being thwarted`
`17:26.00`	`file`	`have you posted your relevant configuration so people can see, and a sip debug?`
`17:26.30`	`FuriousGeorge`	`file: i can do it again, brb`
`17:28.48`	`FuriousGeorge`	`file: rtp debug seems to demonstrate the problem best. Sent RTP packet to 10.0.0.100:9170 <-- on every other server with the same client, the clients external ip goes here, and rtp is not lost`
`17:29.20`	`FuriousGeorge`	`and sip debug also fails to mention the clients external ip, but has plenty of instances where its looking for 10.0.0.100`
`17:30.39`	`*** join/#asterisk djflux (n=djflux@mm.shermfin.com)`
`17:31.24`	`file`	`I can't diagnose anything until I see configuration, and sip debug`
`17:31.32`	`file`	`and potentially know more about what the network setup is like`
`17:31.55`	`FuriousGeorge`	`file: you got it, brb`
`17:33.43`	`syzygyBSD`	`ya.. is just the client behind nat, or server and client?`
`17:34.50`	`*** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net)`
`17:39.03`	`FuriousGeorge`	`syzygyBSD: both, but like i said i have it working now with half a dozen server accross the state of nj`
`17:40.01`	`syzygyBSD`	`I have always had the most trouble when both were behind nat, if either one is it works just fine with nat=1 but when both are, even if the ports are forwarded, it takes more`
`17:41.00`	`FuriousGeorge`	`specifically ive found that it takes setting locallan and externip, that usually does it for me`
`17:41.19`	`file`	`nat=yes for remote clients behind NAT, externip and localnet for when you are behind NAT with port forwarding`
`17:41.23`	`*** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it)`
`17:41.25`	`Qwell[]`	`FuriousGeorge: localnet`
`17:42.01`	`FuriousGeorge`	`Qwell[]: yeah thats what i meant`
`17:42.48`	`*** part/#asterisk geoffl (n=geoff@gjctech.plus.com)`
`17:42.53`	`hoobastooba`	`i am still having intermittent issues where calls coming into my queue, answered by a queue member are not audible to the queue member, but the caller calls back and says that they could hear the queue member. I have done the nat=no and canreinvite=no check hardware issues like irq and hyper threading.`
`17:43.21`	`FuriousGeorge`	`ok here it is http://pastebin.ca/232956`
`17:43.34`	`FuriousGeorge`	`file: syzygyBSD: i hope that helps`
`17:43.54`	`file`	`externhost`
`17:44.26`	`file`	`specifying a host to externip is... silly`
`17:44.43`	`file`	`curious though if it would work`
`17:45.02`	`Qwell[]`	`file: It would not`
`17:45.05`	`FuriousGeorge`	`file: it has always worked for me`
`17:45.11`	`Qwell[]`	`s/would/should/`
`17:45.11`	`Qwell[]`	`:D`
`17:45.19`	`file`	`actually it would work`
`17:45.32`	`Qwell[]`	`but externhost does a refresh, yes?`
`17:45.48`	`file`	`yes`
`17:46.00`	`FuriousGeorge`	`good to know. it says in the comments to use with care`
`17:46.13`	`FuriousGeorge`	`i can try it, maybe it no longer works with externip and 1.4`
`17:46.35`	`file`	`have you tried explicitly putting in an IP instead of a dynamic host? if it works, then you've isolated the cause`
`17:47.03`	`file`	`HA`
`17:47.06`	`MRH2`	`if the clients are bhind nat they might have nat specific features - i know polycom does`
`17:47.07`	`file`	`I know why.`
`17:47.21`	`file`	`your box is resolving your dynamic host as 127.0.0.1 and putting that into the SDP`
`17:47.29`	`file`	`it's probably in /etc/hosts`
`17:48.07`	`FuriousGeorge`	`file: i did make a change to that file`
`17:48.17`	`FuriousGeorge`	`involving 127.0.0.1`
`17:55.03`	`FuriousGeorge`	`file: no, i made that file look like all my working server's and no go. the difference here is that i run an xmpp server which needs some trickeration in that file, but i untrickerated and no change`
`17:55.15`	`file`	`pastebin /etc/hosts`
`17:55.23`	`file`	`and an nslookup of your dyndns host`
`17:57.16`	`FuriousGeorge`	`file: hosts is simple: like all the other boxes all its got is 127.0.0.1 localhost`
`17:57.24`	`FuriousGeorge`	`and that entry for ipv6`
`17:57.33`	`b11d`	`anyone here ever use a vg224 with asterisk?`
`17:58.09`	`*** join/#asterisk heison (n=heison@ns.somanetworks.com)`
`17:59.58`	`FuriousGeorge`	`file: i didnt know there was an nslookup port for linux. its not in my portage.. does it have another name for *nix?`
`18:00.10`	`file`	`try dig`
`18:00.35`	`b11d`	`dig bwnz`
`18:02.06`	`FuriousGeorge`	`file: htdig maybe?`
`18:02.24`	`BurtyB`	`FuriousGeorge I'd expect it to come with bind or bind-tools`
`18:02.44`	`quidpro`	`Anybody done any AGI programming in C?`
`18:02.47`	`HarryR`	`FuriousGeorge, nslookup came from the unix world :) (e.g. the version which comes with windows came from BSD)`
`18:03.00`	`*** join/#asterisk davidcsi (n=davidcsi@190.39.152.92)`
`18:03.13`	`FuriousGeorge`	`HarryR: my bad :)`
`18:03.20`	`FuriousGeorge`	`BurtyB: that;s it. bind tools`
`18:03.20`	`quidpro`	`Windows "tracert" is an ugly version of it's cousins`
`18:03.26`	`*** part/#asterisk davidcsi (n=davidcsi@190.39.152.92)`
`18:03.45`	`FuriousGeorge`	`is there a port of command.com`
`18:03.46`	`FuriousGeorge`	`j/k`
`18:03.48`	`b11d`	`hahaha`
`18:03.52`	`*** join/#asterisk tuxd00d (n=tuxinato@128.187.157.18)`
`18:03.57`	`HarryR`	`FuriousGeorge, /bin/sh ..`
`18:04.06`	`b11d`	`kldload /boot/modules/command.com.so`
`18:04.18`	`HarryR`	`it's much older, much more functional big brother`
`18:05.04`	`quidpro`	`What is really sad.. is kids these days aren't learning the nuances of command line interfaces... unless they are freaks liek us adn discover Linux/BSD`
`18:05.16`	`b11d`	`yeah its so true`
`18:05.36`	`pjz`	`enh; any kid with enough curiousity to care what goes on behind the icons finds out soon enough`
`18:05.42`	`b11d`	`Microsoft WAS going to start shipping a unix-like shell with Vista but it got pulled.. like everything else.`
`18:05.58`	`quidpro`	`like even the quickest kid on a mouse is no challenge for the speed of command line when doing file move/delete/rename`
`18:06.08`	`HarryR`	`b11d, yeah Monad actually looked really cool`
`18:06.08`	`pjz`	`kind of like the kids who are curious about cars learn about cylinders transmissions and etc`
`18:06.12`	`b11d`	`yeah it idd`
`18:06.15`	`b11d`	`idd = did`
`18:06.24`	`quidpro`	`b11d: It would have been a door that onec opened, they could not stop`
`18:06.26`	`pjz`	`I thought monad was getting shipped with excahnge2007 or something crazy like that`
`18:06.34`	`b11d`	`oh really? i hadnt heard that`
`18:06.40`	`HarryR`	`aha nice product bundling :)`
`18:06.48`	`b11d`	`i understood it was so rife with security holes that they had no choice but to shut down the project.`
`18:06.52`	`b11d`	`a real first for MS. :)(`
`18:06.53`	`HarryR`	`you can still download it off ... a website anyway`
`18:07.45`	`b11d`	`im coverting to DOS 3.1 on all my boxes in the back`
`18:08.12`	`b11d`	`hehe`
`18:08.53`	`FuriousGeorge`	`file: http://pastebin.ca/232984 ns lookup is at the top. does that look right to you?`
`18:09.55`	`FuriousGeorge`	`im gonna try it on my other working boxen to see what they say`
`18:10.00`	`file`	`it's resolving to your internal IP`
`18:11.43`	`*** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com)`
`18:12.39`	`HarryR`	`Can somebody take a look at: http://www.midnight-labs.org/asterisk/func_memcache.c`
`18:13.28`	`b11d`	`what a mess that is`
`18:13.29`	`b11d`	`:)`
`18:13.31`	`FuriousGeorge`	`file: yeah., i see that when running nslookup on another box it resolves to the external ip`
`18:13.34`	`FuriousGeorge`	`what gives with that`
`18:14.32`	`FuriousGeorge`	`i think this is something ive done in my firewall`
`18:16.51`	`FuriousGeorge`	`file: you da man :) how did you figure out it was getting it from some hosts file? just an educated guess?`
`18:16.54`	`*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`18:17.02`	`file`	`sure.`
`18:17.27`	`FuriousGeorge`	`fine, be coy but thanks anyway :)`
`18:19.33`	`*** join/#asterisk RoyK (n=roy@ti211310a080-6219.bb.online.no)`
`18:25.13`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`18:28.28`	`*** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net)`
`18:30.05`	`b11d`	`dammit`
`18:30.16`	`b11d`	`im getting one-way audio from a Polycom 501, through asterisk, into a vg224, and to an analog phone..`
`18:30.19`	`b11d`	`and i cant figure this one out.`
`18:30.30`	`b11d`	`the poly is phone A and the analog is phone B`
`18:30.40`	`b11d`	`B can call A and A can call B with no problems`
`18:30.50`	`b11d`	`you can hear A on B, but A cannot hear B.`
`18:30.52`	`b11d`	`arg`
`18:31.54`	`FuriousGeorge`	`b11d: nat?`
`18:34.23`	`ManxPower`	`1-way audio is frequently a NAT, routing, or SIP translation issue.`
`18:36.27`	`ManxPower`	`I assume you set no service sip fixup 5060`
`18:37.25`	`*** part/#asterisk ftexcom (n=ftexcom@169.Red-88-2-81.staticIP.rima-tde.net)`
`18:40.39`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
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`18:42.51`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-207-103.red.bezeqint.net)`
`18:45.01`	`*** join/#asterisk ellisdee (n=ellisdee@69.15.174.114)`
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`18:50.44`	`*** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com)`
`18:56.44`	`*** join/#asterisk Crescendo (n=Master@adsl-072-151-080-148.sip.rmo.bellsouth.net)`
`18:56.50`	`Crescendo`	`How do I transfer a call from a phone?`
`18:57.37`	`*** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com)`
`18:57.38`	`*** join/#asterisk haggai (n=halls@credativ.bcnadsl.com)`
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`19:02.11`	`syzygyBSD`	`lol @ Crescendo`
`19:02.18`	`syzygyBSD`	`depends on how you have it setup`
`19:02.27`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`19:02.32`	`Crescendo`	`Mrm, I'm running trixbox.`
`19:02.46`	`Crescendo`	`Assume I get a call, I want to transfer it to another extension.`
`19:02.47`	`syzygyBSD`	`mrm, never used that, nor will i`
`19:02.50`	`Crescendo`	`Lol.`
`19:02.58`	`Crescendo`	`How SHOULD it work?`
`19:03.23`	`syzygyBSD`	`well it SHOULD work so you can set it up so you can transfer the call`
`19:03.33`	`syzygyBSD`	`adding a t to the Dial command`
`19:03.58`	`syzygyBSD`	`try just dialing #exten to transfer to`
`19:05.00`	`ManxPower`	`Crescendo: you press the TRANSFER button on your phone.`
`19:05.29`	`syzygyBSD`	`huh, one of the two phones on my desk has that...`
`19:05.30`	`Crescendo`	`ManxPower, particularly, client independent.`
`19:05.58`	`ManxPower`	`Crescendo: Hmm? Hpw you use a phone is ALWAYS different for different vendors.`
`19:06.12`	`Crescendo`	`Transferring a call isn't handled serverside?`
`19:06.28`	`Crescendo`	`I can't dial a feature code to transfer a call... :(`
`19:06.32`	`syzygyBSD`	`ohh, both of them do`
`19:06.41`	`*** join/#asterisk cwedwards (n=chatzill@209-254-24-50.ip.mcleodusa.net)`
`19:06.45`	`ManxPower`	`Crescendo: It is not SUPPOSED to be handled server side. Asterisk has a server side transfer feature that is an ugly hack and should not be used unless you have no other choice.`
`19:06.51`	`syzygyBSD`	`Crescendo: try #`
`19:07.00`	`syzygyBSD`	`if trixbox doesnt' suck # would work`
`19:07.05`	`syzygyBSD`	`oh wait...`
`19:07.16`	`ManxPower`	`That would be called "DTMF Transfer" See features.conf`
`19:07.17`	`cwedwards`	`Can anyone suggest a reason why I'd be able to receive phonecalls with a WTD400P but not make calls?`
`19:07.31`	`syzygyBSD`	`cwedwards: you arn't registered`
`19:07.32`	`Qwell[]`	`cwedwards: bad/invalid context`
`19:07.37`	`syzygyBSD`	`arn't in the right context`
`19:07.39`	`Qwell[]`	`syzygyBSD: zap :p`
`19:07.45`	`Qwell[]`	`No registration`
`19:07.45`	`ManxPower`	`cwedwards: not familiar with the WTD400P. Perhaps you mean TDM400P`
`19:07.50`	`*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`19:07.59`	`cwedwards`	`Syzygy: Thanks but they're analog lines, there is no registering`
`19:08.06`	`cwedwards`	`invalid context? Hrm..`
`19:08.07`	`ManxPower`	`and registration problems would prevent you from receiving calls.`
`19:08.18`	`cwedwards`	`Sorry, TDM400P yes. I keep thinking Wildcat`
`19:08.57`	`syzygyBSD`	`wait.. you can't make calls?`
`19:09.16`	`syzygyBSD`	`dtmf tones arn't correctly configured for your zap line...`
`19:09.17`	`cwedwards`	`nope, can't make calls. I can receive calls, but not make them.`
`19:09.26`	`cwedwards`	`hrm`
`19:09.47`	`ManxPower`	`cwedwards: do you get dialtone?`
`19:09.53`	`syzygyBSD`	`your provider hates you...`
`19:10.15`	`ManxPower`	`sounds like a context problem to me. put your /etc/asterisk/zaptel on pastebin.ca`
`19:10.21`	`cwedwards`	`Yes. If I hook up my PE810 directly to the line I can get DT and make calls`
`19:10.23`	`ManxPower`	`als your extensions.conf there too`
`19:10.37`	`syzygyBSD`	`ManxPower: context would be for incomming, not outgoing`
`19:10.55`	`cwedwards`	`Do you mean Zapata.conf?`
`19:11.02`	`syzygyBSD`	`dial(zap/1/mynumber) should work from any context`
`19:11.03`	`ManxPower`	`syzygyBSD: it could be.`
`19:11.18`	`ManxPower`	`syzygyBSD: no, it should work from the context the FXS port is in.`
`19:11.22`	`syzygyBSD`	`cwedwards: are you trying to dial 10 digit or 7 digit numbers?`
`19:11.27`	`ManxPower`	`if there is no correct dial in that context then it won't work`
`19:11.28`	`cwedwards`	`10 digit`
`19:11.32`	`cwedwards`	`I haven't tried 7.`
`19:11.40`	`ellisdee`	`config a did`
`19:11.42`	`ManxPower`	`cwedwards: yes zaptel.conf`
`19:11.51`	`syzygyBSD`	`ManxPower: he already said he can recieve calls`
`19:12.17`	`ellisdee`	`what kinda errors you getting`
`19:12.20`	`ellisdee`	`when making a call out`
`19:12.22`	`ellisdee`	`from the cli`
`19:12.22`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`19:12.23`	`cwedwards`	`I don't have a zaptel.conf, I have zapata.conf. That's where you configure the channels and groups, correct?`
`19:12.35`	`ellisdee`	`in debug mode`
`19:12.35`	`ManxPower`	`cwedwards: paste any zap type of config`
`19:12.36`	`syzygyBSD`	`cwedwards: /etc/zaptel.conf`
`19:12.40`	`cwedwards`	`I don't know how to make a call out from the CLI`
`19:12.49`	`ManxPower`	`and yes, I meant /etc/asterisk/zapata.conf`
`19:12.56`	`syzygyBSD`	`just watch the cli as you call from another phone`
`19:13.10`	`cwedwards`	`Got it Syz, just a sec for pastebin`
`19:13.28`	`aptura`	`cnn news web milestone 100 million web sites :)`
`19:13.29`	`ManxPower`	`cwedwards: I hope you are not using Trixbox or AMP or any GUI like that or I will be forced to bitchslap you into next week.`
`19:14.03`	`ellisdee`	`cwedwards, can you make local calls ?`
`19:14.11`	`cwedwards`	`http://pastebin.ca/233102`
`19:14.28`	`cwedwards`	`Ellisdee: No idea. I'm from out of town so when I'm here I don't have anything but LD cellphones to test`
`19:14.46`	`ManxPower`	`cwedwards: what ports have phones lines on them and which ports have phones on them?`
`19:14.48`	`cwedwards`	`Manx: Bitchslap away. Everything works but this`
`19:15.01`	`cwedwards`	`no phones on the zaptel cards, only 6 analog lines`
`19:15.04`	`cwedwards`	`two cards`
`19:15.35`	`*** join/#asterisk RoyK (n=roy@ti211310a080-6219.bb.online.no)`
`19:15.36`	`ManxPower`	`cwedwards: looks like you have two cards with only FXO ports on them. No FXS cards`
`19:15.38`	`ManxPower`	`So no phones`
`19:15.55`	`cwedwards`	`SIP phones`
`19:15.58`	`ManxPower`	`cwedwards: I also need your /etc/asterisk/zapata.conf`
`19:16.04`	`cwedwards`	`Manx: Just a sec`
`19:17.14`	`cwedwards`	`http://pastebin.ca/233108`
`19:17.31`	`*** join/#asterisk strobhen (n=thomas@rrcs-71-41-120-142.se.biz.rr.com)`
`19:17.35`	`*** part/#asterisk strobhen (n=thomas@rrcs-71-41-120-142.se.biz.rr.com)`
`19:17.56`	`b11d`	`ManxPower.. what was that you said?`
`19:18.04`	`b11d`	`[12:36] <ManxPower> 1-way audio is frequently a NAT, routing, or SIP translation issue.`
`19:18.04`	`b11d`	`[12:38] * file is now known as directory`
`19:18.05`	`b11d`	`[12:38] <ManxPower> I assume you set no service sip fixup 5060`
`19:18.18`	`b11d`	`im not traversing a NAT gateway, but yeah.. i think its a SIP issue.`
`19:18.20`	`*** join/#asterisk chapeaurouge (n=chapeaur@84.5.71.244)`
`19:18.24`	`b11d`	`no service sip fixup 5060 though eh?`
`19:18.24`	`ManxPower`	`cwedwards: yup, it looks like a context problem for your SIP devices.`
`19:18.34`	`ManxPower`	`what is the context= line for your SIP devices.`
`19:18.50`	`ManxPower`	`b11d: I don't remember the EXACT command but it is close to what I gave you`
`19:19.09`	`b11d`	`hmm`
`19:19.15`	`b11d`	`there is no "service sip" on this vg224`
`19:19.22`	`ManxPower`	`no service ? would be a start`
`19:19.26`	`b11d`	`i did that already`
`19:19.29`	`b11d`	`sip isnt listed..`
`19:19.44`	`ManxPower`	`ah, the IOS you are using does not have that command then and yuo don't have to worry about it.`
`19:20.00`	`ManxPower`	`IIRC it is something specific to the FW loads of IOS`
`19:20.05`	`b11d`	`ahh`
`19:20.14`	`b11d`	`hmm.. dammit :)`
`19:20.20`	`cwedwards`	`I have to admit I don't know where to look for that`
`19:20.25`	`b11d`	`this one is driving me nuts.. I guess i'll have to call the TAC.`
`19:20.29`	`*** join/#asterisk alerios (n=alerios@190.24.97.148)`
`19:20.31`	`ManxPower`	`cwedwards: the context= lines in sip.conf shoudl point to a [whatever] context in extensions.conf that have your Dial(... lines`
`19:20.46`	`*** join/#asterisk brif8 (n=brif8@trumpetsoftruth.com)`
`19:20.49`	`ManxPower`	`cwedwards: /etc/asterisk/sip.conf`
`19:20.58`	`cwedwards`	`context = from-sip-external`
`19:21.53`	`hmmhesays`	`that screams freepbx`
`19:22.08`	`cwedwards`	`well, yes`
`19:22.09`	`ManxPower`	`cwedwards: and does the [from-sip-internal] section of extensions.conf have a Dial line that would let you call where you want to call?`
`19:22.13`	`b11d`	`its definatly a sip issue of some kind.. like, when my SIP phone calls the analog phone, the analog rings.. but the SIP doesnt generate a ring in the earpeice.`
`19:22.20`	`b11d`	`back to the manuals`
`19:22.22`	`cwedwards`	`I'm not familiar enough with Linux to be able to do this without the GUI crap`
`19:22.38`	`ManxPower`	`cwedwards: and we are not familiar enough with GUI crap to help you with it.`
`19:22.39`	`cwedwards`	`I can call between extensions`
`19:22.46`	`cwedwards`	`Manx: I understand.`
`19:22.53`	`cwedwards`	`I was just asking for suggestions of what to look at`
`19:23.31`	`cwedwards`	`I appreciate your time and effort`
`19:23.50`	`ManxPower`	`cwedwards: when a call comes in from the device the call will be sent to the [from-sip-internal] section of extensions.conf and match an exten => line. IF it does not match it won't work`
`19:24.33`	`ManxPower`	`it sounds like you have exten => lines in [from-sip-internal] section of extensions.conf to match extension patterns but not real telephone number patterns.`
`19:24.37`	`cwedwards`	`Calls coming in work fine`
`19:24.40`	`cwedwards`	`calls going out don't`
`19:24.44`	`ManxPower`	`but you REALLY should be on #freepbx just like it says in the /topic.`
`19:25.20`	`cwedwards`	`understood`
`19:25.20`	`ManxPower`	`cwedwards: calls come into Asterisk from the PSTN and from the SIP phones.`
`19:25.20`	`ManxPower`	`I understand your calls coming in from the PSTN are working. Calls coming in from SIP devices are not.`
`19:25.23`	`FuriousGeorge`	`i got something of a problem here. the same settings in /etc/hosts/ that make my xmpp server work right break remote sip`
`19:25.23`	`brif8`	`Hi all, anyone using Clipcomm CP-101 phones ?? My first one registers the 2nd one fails sip show peers has it listed but the phone does not accept or make calls and the dial pad is locked up. The only diff I can see if firmware 1.1.60?? Anyone had this ???`
`19:25.41`	`cwedwards`	`Then why is it I can call other extensions?`
`19:25.50`	`cwedwards`	`I'll take my questions to #freepbx`
`19:26.10`	`cwedwards`	`As a final request, does anyone here offer paid support, even if it's a trixbox install?`
`19:26.27`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`19:27.08`	`ManxPower`	`cwedwards: not I. It would take me 4 times as long just to figure out the incredible complexity of trixbox`
`19:27.41`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`19:28.14`	`cwedwards`	`that's wierd, trixbox seems like it's just an overlay for Asterisk to me, like Windows 3.1 on top of DOS 5.`
`19:28.54`	`b11d`	`i dislike the name trixbox.. i cant seem to get over it either :(`
`19:29.12`	`b11d`	`trix are for kids dammit`
`19:30.08`	`*** join/#asterisk soylent (n=soylent@ppl.webex.com)`
`19:32.10`	`*** part/#asterisk Zak (n=thebleh@ip68-224-138-154.lv.lv.cox.net)`
`19:33.07`	`FuriousGeorge`	`this is a little over my head: if i have the line 127.0.0.1 myhost.dynu.com localhost.localdomain in /etc/hosts it breaks remote sip`
`19:33.17`	`FuriousGeorge`	`the problem is that i need it for xmpp`
`19:34.09`	`FuriousGeorge`	`should i just use a different dynamic url service for sip and xmpp? but why would i need to do that... surely there is some way to get this working`
`19:37.39`	`Marshall16`	`Where Can I Buy IAX Serivce WITH DiDS?`
`19:37.47`	`hmmhesays`	`you think thsoe people in the herpes commercial actually have herpes?`
`19:37.53`	`HarryR`	`Marshall16, VoIPtalk.org :)`
`19:37.53`	`Marshall16`	`Where Can I Buy IAX Serivce WITH DiDS I Need To Call From Any NAME And Number?`
`19:38.00`	`Marshall16`	`any name?`
`19:38.06`	`hmmhesays`	`many places depending on your location`
`19:38.13`	`Marshall16`	`usa`
`19:38.14`	`HarryR`	`any name, you should be able to, but you cant set your own CLID`
`19:38.14`	`Marshall16`	`ohio`
`19:38.24`	`Marshall16`	`CLID?`
`19:38.28`	`HarryR`	`Caller Id`
`19:38.31`	`Marshall16`	`why?`
`19:38.33`	`Marshall16`	`lol`
`19:38.43`	`hmmhesays`	`vitelity`
`19:38.45`	`hmmhesays`	`broadvoice`
`19:38.49`	`hmmhesays`	`I use vitelity for my did's`
`19:38.52`	`hmmhesays`	`they work pretty well`
`19:38.59`	`HarryR`	`Because if you were allowed to set your own caller id, you could pretend to be the whitehouse and call up the press office and give them fake news`
`19:39.08`	`Marshall16`	`so when i call from someones house that has there name and number show up a listed number`
`19:39.14`	`Marshall16`	`if i call from there number`
`19:39.19`	`Marshall16`	`will it show up`
`19:39.23`	`Marshall16`	`there name on callerid`
`19:39.26`	`*** part/#asterisk alerios (n=alerios@190.24.97.148)`
`19:39.50`	`hmmhesays`	`sometimes there is`
`19:39.56`	`*** part/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`19:40.28`	`Marshall16`	`i need that`
`19:41.29`	`DarKnesS_WolF`	`Marshall16: u need to read about LookUpCIDname as i think`
`19:41.31`	`hmmhesays`	`not all providers have a name`
`19:41.40`	`DarKnesS_WolF`	`Marshall16: u want to see the name instade of the phone number?`
`19:41.56`	`DarKnesS_WolF`	`hmmhesays: in egypt none has name and we don't have sip prov. and also we suck :D`
`19:42.05`	`soylent`	`hey all, is there a best practice or somewhere I could go to read up on, for how to route extensions between asterisk boxes? I want to be able to have extension 4000 in different locations but from location 1 dial say location id 2+4000 to reach someone at another asterisk that is also at extension 4000`
`19:42.11`	`soylent`	`does that sound crazy?`
`19:43.44`	`soylent`	`I've gotten it working by trunking them together (using freepbx) into the from-internal, which seems to work, unless the same extension sits on both sides.`
`19:44.06`	`soylent`	`so is this a freepbx question only or would it be relevant here too?`
`19:45.28`	`xheliox`	`Does anyone have any thoughts as to why one-way audio would be occuring on 911 calls over a PRI? All other calls work flawlessly.`
`19:45.53`	`*** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br)`
`19:46.25`	`*** join/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`19:48.25`	`quidpro`	`xheliox: DUnno... but that's bad.`
`19:49.02`	`hmmhesays`	`xheliox: that is really odd`
`19:49.15`	`aptura`	`did the 911 operator hear you or them?`
`19:49.15`	`*** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com)`
`19:49.19`	`hmmhesays`	`do you have 911 set up for a different route in your ata?`
`19:49.23`	`aptura`	`or did you hear them`
`19:49.27`	`*** join/#asterisk TheBleh (n=thebleh@ip68-224-138-154.lv.lv.cox.net)`
`19:49.31`	`xheliox`	`quidpro: Thanks for the insight. ;)`
`19:49.42`	`xheliox`	`We can hear the 911 operator, but they can't hear us.`
`19:50.06`	`aptura`	`mmm thats really bad that warrents a dispatch of at least one police car.`
`19:50.12`	`xheliox`	`Admittedly, it does only happen with this one Telco EdgeGate FXS gateway piece of shi....`
`19:50.25`	`xheliox`	`aptura: We've coordinated with the PSAP for testing.`
`19:50.30`	`aptura`	`okay`
`19:50.32`	`xheliox`	`No 5-0 coming here just yet.`
`19:50.51`	`mfroes`	`voicemail fowarding is not working .. can anyone help me out ???? i have defined the extension but it still dont work`
`19:50.55`	`mfroes`	`any ideas?`
`19:51.37`	`xheliox`	`I had an idea, but peer just ruined it for him.`
`19:52.36`	`*** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br)`
`19:53.48`	`*** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com)`
`19:59.09`	`hmmhesays`	`~seen [tk]d-fender`
`19:59.15`	`jbot`	`[tk]d-fender <n=joe@MTRLPQ02-1177745839.sdsl.bell.ca> was last seen on IRC in channel #asterisk, 4h 58m 52s ago, saying: 'TGW: load chan_impiricalevidence.so'.`
`20:00.29`	`*** join/#asterisk crochat (i=crochat@84-74-145-66.dclient.hispeed.ch)`
`20:00.29`	`*** join/#asterisk TexasJay (n=me@ns1.accu-com.com)`
`20:06.38`	`*** join/#asterisk soylent (n=soylent@ppl.webex.com)`
`20:07.28`	`b11d`	`one-way audio problems lick balls`
`20:08.18`	`hmmhesays`	`yep`
`20:08.19`	`hmmhesays`	`sure do`
`20:08.22`	`aptura`	`if it was that easy then there would be no techs or consultants to call on`
`20:08.24`	`aptura`	`:)`
`20:08.35`	`hmmhesays`	`so b11d: I'm switching to t.38 and SER for faxing`
`20:08.41`	`hmmhesays`	`what a freaking b1atch`
`20:09.07`	`aptura`	`b11b i had the hardest time long ago trying to figure out my one way audio. Found out it was a antiquated firewall that and some sip.conf [general] config information missing.`
`20:10.20`	`TexasJay`	`does anyone have any good suggestions for scripts to log in and out of queues?`
`20:10.21`	`hmmhesays`	`wiki wiki wild wild`
`20:10.37`	`TexasJay`	`I'm sure there are some good ones out there, and I don't want to reinvent the wheel. ;)`
`20:11.05`	`aptura`	`good what TexasJay`
`20:11.21`	`TexasJay`	`Queue log in/out scripts`
`20:11.40`	`TexasJay`	`And by 'script' I mean "exten => blah blah" lines.`
`20:12.31`	`hmmhesays`	`just write one`
`20:12.38`	`hmmhesays`	`simple enough`
`20:12.44`	`b11d`	`no way..`
`20:12.52`	`b11d`	`why are you switching?`
`20:12.55`	`b11d`	`just couldnt hack it out eh?`
`20:12.56`	`b11d`	`:)`
`20:13.15`	`TexasJay`	`Ideally, I'd like this: Dial an extension, enter a usercode/password, log that person into the queue at that phone, and log out anyone who may have been logged in at that phone.`
`20:13.17`	`b11d`	`aptura.. well, this isnt passing over a firewall, and the asterisk side sip stuff seems to be fine.`
`20:13.26`	`b11d`	`the problem is getting the voice BACK from the analog phone, across the vg224, and to the asterisk box.`
`20:14.25`	`b11d`	`i just captured all the sip debug info from the vg224.. time to go over it. I'll have to compare it against what i've got on the * box too.`
`20:14.41`	`b11d`	`it doesnt seem to be showing any errors though.. everything looks OK.`
`20:14.45`	`b11d`	`but its obviously not.`
`20:15.03`	`b11d`	`for example, when my SIP phone calls the analog on the vg224, it rings the analog phone. But the SIP phone doesnt generate the ring in the ear peice.`
`20:15.11`	`b11d`	`so.. yeah.. theres something up with SIP at this point.`
`20:15.19`	`b11d`	`maybe I should say fuck it and go to MGCP?`
`20:17.10`	`kilobit2001`	`is it necessary to have the 'u' in front of mailbaox in voicemail command?`
`20:17.24`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`20:17.43`	`*** join/#asterisk tbic (n=tbic@protious.fciautomation.com)`
`20:21.09`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`20:23.22`	`*** join/#asterisk TGW (n=toronto@gw.gibbs.iasl.com)`
`20:24.20`	`TGW`	`is there any easy way to restrict the number of calls that can be sent to any sip device? For example if I have a peer that is going to have an * box running as well, can I from my side only send 4 calls and once there signal congestion?`
`20:25.31`	`*** join/#asterisk brif8 (n=brif8@67.78.24.178)`
`20:25.43`	`CunningPike`	`kilobit2001: Either u or b - and the better way is to have is after as in Voicemail(${EXTEN}@default\|u)`
`20:25.54`	`CunningPike`	`s/have is/have it/`
`20:26.32`	`brif8`	`So is anyone using Clipcomm CP-101 phones ?? My first one registers the 2nd one fails The only diff I can see if firmware 1.1.60?? Anyone have sucess with the CP-101 ?`
`20:26.35`	`CunningPike`	`TGW: Take a look at channel groups`
`20:27.02`	`TGW`	`thanks`
`20:32.00`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`20:34.05`	`xheliox`	`quidpro, hmmhesays, aptura --- fixed the 911 problem. Had to check "Send RTP on 183 Session Progress" on the SIP device. Strange that it only affected 911 though.`
`20:37.30`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`20:38.57`	`drfreeze`	`Anyone know how to keep the volume of Polycom 501 phone at the level set by the user?`
`20:39.05`	`drfreeze`	`Our phones keep resetting after every call`
`20:39.55`	`quidpro`	`xheliox: Glad you got the problem licked... that is definitely something you don't want a "work around" for.`
`20:41.08`	`xheliox`	`quidpro: Indeed. And this is why you do 911 testing -- I learned my lesson after the 2nd or 3rd old lady was found dead in their apartments clinging to a phone.`
`20:41.40`	`ManxPower`	`drfreeze: it is a config option. See the Polycom Admin manual. You can't set it via the Web GUI`
`20:42.18`	`drfreeze`	`ManxPower: you mean the invisible polycom admin manual. :) Seriously, never seen it.`
`20:42.24`	`*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)`
`20:42.30`	`*** join/#asterisk evilstoy (i=HydraIRC@83-131-0-96.adsl.net.t-com.hr)`
`20:42.38`	`drfreeze`	`ManxPower: do I just browse to each phone and set the phone volume?`
`20:42.43`	`kilobit2001`	`anyone knows where I can find a description for the fields in asterisk realtime voicemail.`
`20:45.34`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`20:45.37`	`*** join/#asterisk svenna_ (n=svenna@p548D2FC5.dip0.t-ipconnect.de)`
`20:46.39`	`NDT`	`Qwell: You here?`
`20:47.20`	`*** join/#asterisk fbnts (n=thomas@www.multiplix.co.uk)`
`20:47.51`	`*** join/#asterisk Zhad (n=tom@cpc1-sout6-0-0-cust691.sot3.cable.ntl.com)`
`20:47.54`	`ManxPower`	`drfreeze: would you like a copy of it?`
`20:48.14`	`drfreeze`	`ManxPower: yes, thx`
`20:49.03`	`fbnts`	`Hey, Just configuring an inbound IAX call to ring 3 handsets. Its all calling fine but when the handset presses hold the caller gets nothing. but if I call out and press hold it plays music on hold. Is there any way to make it play music on hold on incoming calls?`
`20:49.39`	`Zhad`	`what type of handset is it?`
`20:49.46`	`fbnts`	`Cisco 7940`
`20:50.27`	`*** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br)`
`20:50.46`	`ManxPower`	`drfreeze: http://www.polycom.com/common/pw_item_show_doc/0,1276,4349,00.pdf`
`20:50.57`	`ManxPower`	`and http://www.polycom.com/common/pw_item_show_doc/0,1276,4922,00.pdf`
`20:51.03`	`ManxPower`	`these links should not require a login`
`20:51.08`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.168.211)`
`20:51.18`	`Zhad`	`what is the debug output when you put the call on hold?`
`20:51.48`	`*** part/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br)`
`20:52.49`	`*** join/#asterisk knobenheimer_lap (n=brettkoe@c-71-205-29-151.hsd1.mi.comcast.net)`
`20:52.57`	`fbnts`	`nothing in the asterisk panel once the phone has picked up the call. When I press Hold the phone flashes the call coming in but nothing shows in console`
`20:54.12`	`Zhad`	`you don't get a Call placed on hold message? (Assuming the debug is set to at least 3).`
`20:54.30`	`kilobit2001`	`how can I change the voicemail greeting.. from vm-intro`
`20:54.38`	`fbnts`	`nope I dont - I have connected to the console with asterisk -rvvvvvvvvvvvvvvvvvvvvvvvv`
`20:54.59`	`Zhad`	`is this with the SIP firmware?`
`20:55.29`	`b11d`	`ONE WAY VOICE PROBLEMS ARE CREATED BY SATAN HIMSELF`
`20:55.30`	`Zhad`	`or sccp`
`20:55.30`	`b11d`	`ARGH`
`20:55.32`	`fbnts`	`yep V8.2`
`20:55.40`	`Zhad`	`weird`
`20:56.01`	`fbnts`	`Also transfer and blind xfer dont work`
`20:56.12`	`Zhad`	`is asterisk still in the media path?`
`20:56.15`	`xheliox`	`b11d: I'm pretty sure they weren't. But I wouldn't surprised if he advocates typing in all caps. :)`
`20:56.23`	`b11d`	`:)`
`20:56.43`	`fbnts`	`yep. Asterisk is bridging the IAX inbound call and the SIP phone`
`20:57.28`	`b11d`	`maybe I should give MGCP a try instead of using SIP on this..`
`20:57.39`	`Zhad`	`:-)`
`20:58.47`	`Zhad`	`atm. I'm using cheap (�30) PA1688 handsets and attended xfer works fine as does MoH, transfers don't really work with the IAX firmware though.`
`20:59.26`	`hmmhesays`	`bah my reinvites aren't working`
`20:59.55`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`21:02.05`	`ManxPower`	`b11d: MGCP has no support for ANY NAT and no authentication either.`
`21:02.16`	`b11d`	`im not traversing any NAT shit so thats no problem`
`21:02.18`	`fbnts`	`Just checked by doing an outbound call to my mobile and Hold, transfer and blind transfer all work!?`
`21:02.22`	`b11d`	`and I dont care about auth at this point`
`21:02.26`	`b11d`	`not for my analog phones`
`21:02.31`	`b11d`	`I just want it to work!`
`21:02.53`	`ManxPower`	`b11d: you have a problem you don't know about that is causing this.`
`21:04.50`	`b11d`	`yes`
`21:04.59`	`b11d`	`all I know is that its likely sip related`
`21:05.06`	`b11d`	`but i've been pouring over the debug output and cant find any issues..`
`21:07.19`	`kilobit2001`	`is there a way to change the playback file vm-intro in voicemail to something else?`
`21:07.21`	`b11d`	`do you secretly know what my problem is ManxPower? dont hold out on me man`
`21:08.04`	`drfreeze`	`ManxPower: What is the diff between those two docs?`
`21:08.09`	`ManxPower`	`b11d: wgat us tge Cisco model again?`
`21:08.19`	`b11d`	`vg-224`
`21:08.25`	`ManxPower`	`b11d: I have no idea WHAT the problem is. Most people do not have this problem.`
`21:08.29`	`b11d`	`oh :/`
`21:08.34`	`b11d`	`thanks anyway though`
`21:09.01`	`b11d`	`calling from an analog phone to an analog phone (both on the vg224) works fine.`
`21:09.18`	`b11d`	`i'll just have to keep pouring over the debug output.. maybe i'll give mgcp a try too.`
`21:09.32`	`ManxPower`	`b11d: I assume you are using this because it's all you had on hand and someone is too cheap to buy something good?`
`21:09.41`	`b11d`	`no`
`21:09.47`	`b11d`	`i bought it because I was told it was the best`
`21:10.03`	`b11d`	`whats good?`
`21:10.09`	`ManxPower`	`By whom? I can't even find any mention of "VG-224" on the Asterisk mailing list.`
`21:10.13`	`NDT`	`Anyone familiar with realtime sip...can you take a look at this: http://pastebin.ca/233256 tell me what you think could be causing this?`
`21:10.16`	`*** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com)`
`21:10.46`	`ManxPower`	`b11d: If you want 24 analog phones, I recommend a Digium or Sangoma 1-Port T-1 card and an Adtran TA-750.`
`21:11.01`	`b11d`	`I dont recall by whom`
`21:11.14`	`b11d`	`I was looking at that exact setup actually Manx.`
`21:11.19`	`b11d`	`since i've already got the t1 cards.`
`21:11.24`	`b11d`	`or, t1 ports i mean to say.`
`21:11.36`	`b11d`	`but.. here I am with the vg-224.`
`21:12.03`	`b11d`	`you can believe i'll be posting everythign I know about the vg224 when im done with this..`
`21:12.35`	`ManxPower`	`b11d: nod I don't know of anyone that has used those boxes.`
`21:12.39`	`b11d`	`weak :)`
`21:12.47`	`b11d`	`damn me and my limited experience`
`21:12.54`	`b11d`	`oh well.. live and learn.`
`21:12.55`	`ManxPower`	`An Adtran TA6750 is like $300 on eBay 8-)`
`21:13.08`	`b11d`	`yeah well i kinda also wanted the smartnet coverage for the rest of my lan :)`
`21:13.13`	`b11d`	`just for the software that is`
`21:13.54`	`ManxPower`	`b11d: I won't say that you got "bad advice", but I can say people that want 24 analog ports don't usually use a Cisco analog gateway with Asterisk.`
`21:14.17`	`b11d`	`so i'm learning :)`
`21:14.36`	`b11d`	`well.. im going to try mgcp, and if that fails, i'll call the TAC, and if that fails.. the adtran it is.`
`21:14.54`	`ManxPower`	`I have a TA-750 in storage with 8xFXO and 12xFXS. I'm just waiting until I can afford a Sangoma card 8-)`
`21:15.05`	`b11d`	`and i've got 3 a104d cards`
`21:15.14`	`ManxPower`	`b11d: I hate you.`
`21:15.15`	`ManxPower`	`8-)`
`21:15.19`	`b11d`	`:)`
`21:15.29`	`b11d`	`those sangoma cards are great though..`
`21:16.32`	`b11d`	`I wish I didnt have so many god damn vultures circling above me..`
`21:16.35`	`b11d`	`Qwest is just waiting for me to fuck this up`
`21:16.41`	`b11d`	`so they can come in with their big dollar solutions`
`21:16.43`	`ManxPower`	`b11d: as cheap as Adtrans are on eBay you can just buy a spare.`
`21:16.55`	`b11d`	`yeah at that price.. buy two spares :)`
`21:17.16`	`ManxPower`	`b11d: I assume you have the latest IOS for the VG224?`
`21:17.18`	`*** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw)`
`21:17.24`	`b11d`	`yeah`
`21:17.26`	`b11d`	`12.4(4)`
`21:17.37`	`b11d`	`once i get the smartnet registered i can verify that`
`21:17.44`	`b11d`	`...it's the latest`
`21:17.56`	`ManxPower`	`have you tried the 12.3 version?`
`21:18.09`	`ManxPower`	`BTW, if you have ANY software access on smartnet you should be able to get most any IOS`
`21:18.28`	`b11d`	`yeah im just waiting on that..`
`21:18.34`	`b11d`	`no.. i dont have the 12.3 available yet.`
`21:18.41`	`*** join/#asterisk qdk (n=qdk@hotel-levor-fo.b.astral.ro)`
`21:18.42`	`b11d`	`available to me that is`
`21:19.31`	`*** join/#asterisk ast_freak\|Laptop (n=jesse@h69-130-161-87.69-130.unk.tds.net)`
`21:21.20`	`*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)`
`21:23.17`	`*** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com)`
`21:23.26`	`b11d`	`I am seeing a 404 Not Found error in my sip debug info from the vg224 to the pbx.`
`21:23.31`	`b11d`	`only one.. and no other details..`
`21:24.17`	`ManxPower`	`b11d: That pretty much tells me what the problem is.`
`21:24.26`	`b11d`	`oh really?`
`21:24.37`	`b11d`	`elaborate please :)`
`21:24.39`	`ManxPower`	`b11d: put your sip.conf on pastebin.ca (only [general] and the entry for the vg224)`
`21:24.43`	`ManxPower`	`also /msg me`
`21:25.05`	`b11d`	`i actually dont have a [general] section in sip.conf :/`
`21:25.07`	`b11d`	`wow.`
`21:25.17`	`aptura`	`Manx is there a site that talks about debug in depth?`
`21:25.27`	`aptura`	`wooops`
`21:25.39`	`aptura`	`thats a good part of your one way voice problem`
`21:26.56`	`ManxPower`	`b11d: <dry tone> that might be part of the problem</dry tone>`
`21:27.01`	`b11d`	`hmm.. i guess i never thought about it because everything else works`
`21:27.04`	`b11d`	`sip to sip works fine`
`21:27.29`	`ManxPower`	`b11d: Cisco gateways don't send out authentication details when connecting.`
`21:27.48`	`ManxPower`	`so you have to carefully craft your sip.conf`
`21:28.00`	`ManxPower`	`(well cisco ROUTERS don't at least)`
`21:28.12`	`b11d`	`im not trying to do any authentication`
`21:28.21`	`b11d`	`and the 5454 sucessfully registers with asterisk`
`21:28.24`	`*** join/#asterisk TexasJay (n=me@ns1.accu-com.com)`
`21:28.29`	`ManxPower`	`b11d: put the info on pastebin.ca and I'll fix it for you if I can`
`21:28.40`	`b11d`	`i dont have anything to paste there is the thing :)`
`21:28.56`	`b11d`	`i'll spend some time on this [general] section stuff`
`21:29.00`	`b11d`	`right now`
`21:29.04`	`ManxPower`	`b11d: if you don't have an entry for the GV224 and you don't have a [general] then you will get a 404`
`21:29.19`	`*** join/#asterisk Un1x (n=x@CPE006323108591-CM00407b87dc81.cpe.net.cable.rogers.com)`
`21:29.22`	`ManxPower`	`404 means "not found", which usually means the call is not falling into the context= you are expecting it to.`
`21:29.24`	`b11d`	`i dont have an entry for the vg224, but I have one for one of the numbers ON the vg224.`
`21:29.27`	`TexasJay`	`Hi guys, I'm having some problems with * calling php scripts. I have the script in /var/lib/asterisk/agi-bin/, I have the shebang line correct, PHP is installed and working, but I don't think my script is being called properly`
`21:29.41`	`Un1x`	`someone suggest a good processor where i can easily run at least 3 concurrent g729 calls`
`21:29.46`	`TexasJay`	`If I have an "echo" statement in my php script, should it show what I echo'd in the CLI?`
`21:29.50`	`b11d`	`I can successfully reach a SIP phone from an analog phone on the vg224..`
`21:29.53`	`b11d`	`its just the audio issue`
`21:30.01`	`Zhad`	`Un1x> anything that you can buy nowadays`
`21:30.16`	`Un1x`	`zhad u can still buy semrpn 2200+`
`21:30.24`	`Un1x`	`those wont run 3 g729 calls`
`21:30.33`	`Zhad`	`They should do`
`21:30.40`	`drfreeze`	`TexasJay: why don't you think it is being called correctly?`
`21:30.44`	`Zhad`	`the payload isn't that great`
`21:30.45`	`*** join/#asterisk syzygyBSD_ (n=chatzill@poplar.matraex.com)`
`21:30.52`	`Un1x`	`Zhad who says that`
`21:30.57`	`b11d`	`and does that explain why I can hear voice from the SIP on the analog, but not vice versa?`
`21:31.00`	`Un1x`	`g.729 is processor intensive`
`21:31.07`	`b11d`	`but.. i'll go get into this [general] stuff`
`21:31.10`	`b11d`	`and investigate further.`
`21:31.20`	`TexasJay`	`drfreeze: Well, I'm expecting some sort of output on the CLI. Am I wrong in thinking an echo statement in my PHP script would echo in the CLI?`
`21:31.23`	`Zhad`	`well, lets put it this way, if the folks at aredfox can manage to get it working on a Z80`
`21:31.25`	`ManxPower`	`b11d: I don't know, but I do know that if things are not set up right it can cause all sorts of weird stuff.`
`21:31.44`	`b11d`	`yeah thats for sure.`
`21:31.48`	`Zhad`	`Un1x> It is processor intensive by comparison, but not nearly that much.`
`21:31.50`	`b11d`	`i appreciate the help and everything`
`21:31.57`	`fbnts`	`Zhad: Thanks for the help earlier I have done some more googling and I noticed Mark Spencer talking about the problem but its from 2002 and don't seem to be able to find any update. Does he hang around on here often?`
`21:32.00`	`Un1x`	`well Zhad i asked whats a good cpu too :P`
`21:32.07`	`ManxPower`	`b11d: what is the IP of the vg224 and what codec are you wanting to use?`
`21:32.21`	`Un1x`	`i was wondering if a AMD X2 3800+ socket am2`
`21:32.23`	`Un1x`	`can handle it`
`21:32.23`	`b11d`	`10.0.5.5 and g711ulaw`
`21:32.35`	`b11d`	`connecting to an * box on 10.0.5.11`
`21:32.38`	`b11d`	`on the same switch`
`21:32.53`	`drfreeze`	`TexasJay: no. asterisk is grabbing the echo statements`
`21:33.16`	`drfreeze`	`write to a file or watch the console that asterisk was launched from`
`21:33.16`	`*** join/#asterisk syzygyBSD_ (n=chatzill@poplar.matraex.com)`
`21:33.18`	`Zhad`	`Un1x> and I was claiming that a Sempron 2200+ could`
`21:33.31`	`TexasJay`	`drfreeze: Ok. I'm still new to this whole * thing. How can I have * output what my php script echoes/returns?`
`21:33.36`	`drfreeze`	`problem is asterisk -r will not show the debug (echo) statements. Only the original console`
`21:33.58`	`drfreeze`	`TexasJay: use original console, or write to a file`
`21:34.12`	`ManxPower`	`[general]`
`21:34.12`	`ManxPower`	`disallow=all`
`21:34.12`	`ManxPower`	`allow=ulaw`
`21:34.12`	`ManxPower`	`context=theextensionsconfcontextyouwant`
`21:34.39`	`ManxPower`	`if the incoming call does not match a [section] then it will hit the context in [general] in sip.conf`
`21:34.50`	`TexasJay`	`drfreeze: it's possible to transfer info between php and asterisk, correct?`
`21:34.53`	`b11d`	`ohh ok`
`21:34.57`	`drfreeze`	`TexasJay: yes`
`21:35.38`	`drfreeze`	`TexasJay: asterisk talks to scripts with commandline arguments and parameter/value pairs`
`21:35.48`	`ManxPower`	`THAT is why I normally put context=INVALID in [general] I don't want random people connecting to my server.`
`21:35.52`	`docelmo`	`Transfer what between php and asterisk?`
`21:35.54`	`drfreeze`	`if you want your script to control asterisk, look at AMI`
`21:36.02`	`docelmo`	`You mean run AGI's from asterisk in PHP?`
`21:36.04`	`TexasJay`	`Ok. I guess I need to check into it more then. Can I nest a php call (that outputs info) in an asterisk command that sets a variable?`
`21:36.30`	`ManxPower`	`TexasJay: yes, you need to use an AGI`
`21:36.39`	`drfreeze`	`TexasJay: don't see a problem with that`
`21:36.59`	`TexasJay`	`Could you give me a test syntax?`
`21:37.00`	`docelmo`	`Use PHPAGI its a decient framework for Asterisk and PHP`
`21:37.29`	`docelmo`	`And you can use $agi->setvar("VAR","DATA"); to set your variable`
`21:37.36`	`b11d`	`im going to go try some of this stuff out`
`21:37.36`	`b11d`	`brb`
`21:40.48`	`TexasJay`	`so, how would I finish this: exten => 1234,1,Set(foo=`
`21:41.10`	`docelmo`	`BAR)`
`21:41.22`	`TexasJay`	`lol`
`21:41.28`	`TexasJay`	`I meant with calling a PHP script :)`
`21:41.48`	`docelmo`	`You would do 1234,1,AGI(bla.agi)`
`21:41.49`	`hads`	`You wouldn't, you'd start a new line.`
`21:41.59`	`hads`	`Yeah, what docelmo said.`
`21:42.24`	`docelmo`	`I am one of the core developers of phpagi.. :)`
`21:42.25`	`TexasJay`	`ok, then how would you capture what bla.agi returns?`
`21:42.42`	`docelmo`	`what do you mean capture`
`21:42.49`	`TexasJay`	`sorry for being such a dunce. usually I pick up on stuff rather quickly, but this whole Asterisk thing has my head reeling`
`21:42.51`	`hads`	`< docelmo> And you can use $agi->setvar("VAR","DATA"); to set your variable`
`21:43.12`	`docelmo`	`You should have been in my Asterisk101 talk @ astricon`
`21:43.13`	`docelmo`	`:P`
`21:47.29`	`hmmhesays`	`god reinvites can be a bitch`
`21:47.30`	`*** join/#asterisk flujan (n=chmencac@internet.nube.com.br)`
`21:47.55`	`ManxPower`	`hmmhesays: only if you have nat 8-)`
`21:47.56`	`flujan`	`hi guys... how can I calculate the jitter in sip channels?`
`21:47.57`	`TexasJay`	`probably should have been. My bosses forced me into doing Asterisk and I'm having a helluva time grasping it all.`
`21:48.08`	`flujan`	`there is now sip show netstats... :)`
`21:48.11`	`ManxPower`	`~book`
`21:48.12`	`jbot`	`[book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`21:48.17`	`hmmhesays`	`ManxPower we built this network, no nat, but the reinvites were failing because of a preferred codec issue`
`21:48.25`	`ManxPower`	`TexasJay: Asterisk isn't really a PBX. It's a toolkit for building your own PBX`
`21:48.45`	`ManxPower`	`hmmhesays: that's why I never allow= more than one codec`
`21:48.45`	`hmmhesays`	`GRE tunnels galore`
`21:48.47`	`TexasJay`	`See? Even my terminology is off. ;)`
`21:49.02`	`flujan`	`I am having problems with the jitter in my network`
`21:49.15`	`flujan`	`I am using moziax a iax2 softphone`
`21:49.32`	`flujan`	`I want to try it with x-lite to see if it is a softphone related issue.`
`21:49.46`	`*** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se)`
`21:49.55`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`21:50.11`	`ManxPower`	`TexasJay: Everyone CALLS Asterisk a PBX. It's not. It's a telephony toolkit 8-)`
`21:50.29`	`aptura`	`I am getting a SIP/2.0 404 Not Found`
`21:50.40`	`flujan`	`hi ManxPower`
`21:50.55`	`*** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`21:51.00`	`TexasJay`	`Ok, so I'm using Asterisk to create my own little PBX in-office. Better? :P`
`21:51.07`	`aptura`	`and the contect is right. Testing a sip cleint and using my external ip address to pass though and back into network`
`21:51.16`	`*** join/#asterisk syzygyBSD_ (n=chatzill@poplar.matraex.com)`
`21:52.52`	`ManxPower`	`TexasJay: the terms are not really important.`
`21:53.14`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`21:53.35`	`TexasJay`	`docelmo: Looking at the PHPAGI site, can't seem to find install docs. Where should I be looking?`
`21:54.38`	`aptura`	`ManxPower what do you think may be my case`
`21:56.20`	`TexasJay`	`Nevermind, docelmo. Didn't realize it merely needs to be included.`
`21:56.58`	`*** join/#asterisk knobenheimer_lap (n=brettkoe@c-71-205-29-151.hsd1.mi.comcast.net)`
`21:57.05`	`Zodiacal`	`anyone know how to remove ztdummy?`
`21:57.10`	`Zodiacal`	`i have a digium card i don't needs it`
`21:58.23`	`*** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do)`
`21:59.01`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`21:59.41`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`21:59.58`	`aptura`	`just rmmod it`
`22:01.18`	`drfreeze`	`Anyone know where I can get the latest firmware upgrades for Polycom 501 and 301 phones?`
`22:01.39`	`Un1x`	`anyone know whats a good processor`
`22:01.45`	`Un1x`	`to run 3 g729 calls at once`
`22:02.01`	`Qwell[]`	`Un1x: anything made after 1992`
`22:03.14`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`22:03.44`	`aptura`	`drfreeze either the vendor that you purchace the phone from if thay will help you with that or you become a vendor of the phones. That requires you have a web presence.`
`22:03.45`	`NDT`	`Hey Qwell you have any ideas on this? http://pastebin.ca/233256`
`22:04.01`	`*** join/#asterisk p1p (i=j@64.200.16.100)`
`22:04.03`	`Zodiacal`	`aptura is there a way to have it not load tho?`
`22:04.38`	`aptura`	`why would you not want it to load`
`22:05.03`	`Zhad`	`Un1x> Believe us, you should be able to do it on anything better than roughly a PIII-600`
`22:05.05`	`ManxPower`	`drfreeze: older firmwares are available on polycom's site.`
`22:05.11`	`Zodiacal`	`cuz my digium card can do the timing better, so i've read`
`22:05.16`	`Zhad`	`comfortably`
`22:05.23`	`Zhad`	`uncomfortably on slower machines`
`22:05.36`	`p1p`	`Anyone familiar with quintum gateways?`
`22:06.31`	`b11d`	`im familiar with quantum leap`
`22:06.32`	`b11d`	`and sliding`
`22:06.35`	`b11d`	`but thats about it`
`22:07.00`	`p1p`	`yeah same`
`22:07.06`	`p1p`	`thats why im having problems :0`
`22:07.09`	`b11d`	`:)`
`22:07.22`	`b11d`	`just leap into the next universe and it'll be solved.. eventually`
`22:07.56`	`p1p`	`I wish I could! It would be better than having a client that is perpetually angry that their phone system doesnt work quite rgiht ;(`
`22:08.01`	`b11d`	`oh right.. we only deal with solutions founded in reality here.`
`22:08.01`	`b11d`	`damn`
`22:08.34`	`*** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189)`
`22:08.51`	`Zhad`	`Un1x> lets express it another way, Beijing RedFox Communication Technology manage to encrypt a G.729 call with an iNtel 8051 and an ADSP2181 (8-bit processor and DSP).`
`22:09.36`	`*** join/#asterisk zotz (n=zotz@24.244.133.107)`
`22:09.53`	`paolob-parroquia`	`Hi guys! I have to asterisk server communicating with iax2 protocol. Since yesterday, iax2 doesn't work any more, I think it was after the last debian (unstable) upgrade, now when trying to communicate, the asterisk I'm calling from says "chan_iax2.c:6907 socket_read: Rejected connect attempt from 10.152.58.9, who was trying to reach '600@entrantes'". What is the reason? What is happening?`
`22:10.26`	`paolob-parroquia`	`The asterisk I'm calling to says: "chan_iax2.c:6997 socket_read: Call rejected by 10.152.58.1: No authority found"`
`22:10.35`	`paolob-parroquia`	`Anyone could help me? thank you!`
`22:10.35`	`Zhad`	`Un1x> so encrypting 3 calls (presumably 6 streams) is trivial`
`22:10.58`	`*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)`
`22:11.14`	`*** join/#asterisk ac3c (n=chatzill@wuser32-shapiro.umnet.umich.edu)`
`22:12.43`	`ManxPower`	`paolob-parroquia: I can't think of ANY reason a software update would cause that.`
`22:13.09`	`paolob-parroquia`	`ManxPower, but I haven't changed anything in iax configuration!`
`22:13.13`	`ManxPower`	`the first message you posted indicated an auth or context issue, the second one would indicate an auth issue`
`22:13.45`	`Nivex`	`paolob-parroquia: are you using RSA keys or just straight passwords?`
`22:14.01`	`paolob-parroquia`	`Nivex, how do I check it?`
`22:14.12`	`Nivex`	`if you don't know, you're probably just using passwords.`
`22:14.20`	`Nivex`	`scratch my idea.`
`22:14.48`	`*** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it)`
`22:14.56`	`paolob-parroquia`	`Nivex, I'm using passwords`
`22:15.00`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`22:15.14`	`Zhad`	`that's one thing that went wrong when I was briefly using 1.4b3 (before going back to 1.4b2), my IAX trunk which was RSA encrypted stopped working.`
`22:15.59`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`22:17.10`	`paolob-parroquia`	`but the "no authority found" message what does it refer to?`
`22:19.39`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`22:20.08`	`brif8`	`Who is the best suport for Clipcomm in the USA. I've tried KR not sucess. I have a CP101 failing to correctly register. I'm sure it's a firmware issue, as one works and one fails.`
`22:20.28`	`TexasJay`	`docelmo, you mind if I send you a private message?`
`22:20.43`	`docelmo`	`Go ahead everyone else does.. :P`
`22:21.49`	`*** join/#asterisk sp00kz (n=ilubj00@maverick.specialai.com)`
`22:22.06`	`sp00kz`	`Does anyone know if its possible to turn off the Message Waiting Indicator for a user?`
`22:22.26`	`sp00kz`	`Not their voicemail, but 1 phones MWI`
`22:22.33`	`b11d`	`just remove mailbox = in sip.conf ?`
`22:22.38`	`b11d`	`what kind of phone?`
`22:22.43`	`sp00kz`	`cisco`
`22:22.43`	`[TK]D-Fender`	`sp00kz : Yes. In the the device setup you had to tell * which mailbox to onitor. Simply REMOVE that line and presto.`
`22:22.48`	`sp00kz`	`7970`
`22:22.55`	`sp00kz`	`I want to leave a voicemail though`
`22:24.55`	`[TK]D-Fender`	`sp00kz : meaning?`
`22:24.56`	`ManxPower`	`sp00kz: mailbox= only specifies the message waiting indicator, it has nothing to do with voicemail other than that`
`22:25.11`	`[TK]D-Fender`	`sp00kz (what ManxPower said)`
`22:25.17`	`ManxPower`	`it could have just as easily been called mwi=`
`22:25.52`	`[TK]D-Fender`	`ManxPower : Actually.... thats less characters so technically it'd have been EASIER (not to mention clearer)`
`22:27.48`	`*** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net)`
`22:27.51`	`*** join/#asterisk alexhopper (n=a27386@resnet-71-34.mta.ca)`
`22:28.38`	`Ryanw`	`has anyone written functionality to create a new conference room through the keypad TTS the randomly generated pin numbers?`
`22:29.19`	`*** join/#asterisk SwK (n=Silik0nJ@208.44.30.242)`
`22:29.59`	`sp00kz`	`thanks, i'll try it again`
`22:30.23`	`FuriousGeorge`	`is there any reason to worry about forwarding 10000 ports for RTP?`
`22:30.51`	`FuriousGeorge`	`my roomate just stammered something about firewall performance`
`22:31.30`	`ManxPower`	`FuriousGeorge: you can expect to use two RTP ports per call.`
`22:32.03`	`ManxPower`	`and make sure to update rtp.conf to match what your are forwarding`
`22:32.04`	`FuriousGeorge`	`ManxPower: but does not filtering traffic slow down the firewall?`
`22:32.23`	`ManxPower`	`FuriousGeorge: port forwarding is not filtering.`
`22:32.36`	`ManxPower`	`But I suppose it would depend on the firewall.`
`22:32.53`	`ManxPower`	`I forwarded 16 ports on my Cisco box. Did not seem to massivly slow it donw.`
`22:32.54`	`FuriousGeorge`	`linux/iptables based`
`22:33.14`	`ManxPower`	`FuriousGeorge: I would not worry about it unless you machine is massivly underpowered.`
`22:33.18`	`paolob-parroquia`	`Guys, in the user configuration of iax.conf, I have a "deny=0.0.0.0/0.0.0.0" line before the permit statement. On a 1.2.7 asterisk it doesn't affect calls, while in 1.2.13 it does: in 1.2.13 putting a deny=0.0.0.0/0.0.0.0 before the permit=otherhost doesn't permit iax2 connections, but on the 1.2.7 asterisk that's not true. Anyone could tell me why?`
`22:33.48`	`ManxPower`	`paolob-parroquia: file it as a bug on bugs.digium.com`
`22:34.05`	`Qwell[]`	`When you say "otherhost", do you mean a hostname, or an IP?`
`22:34.25`	`paolob-parroquia`	`ManxPower, but what is wright? the 1.2.13 behaviour or the 1.2.7 one?`
`22:34.30`	`b11d`	`hey manx..`
`22:34.36`	`paolob-parroquia`	`Qwell, a hostname`
`22:34.42`	`b11d`	`i've got some additional info.. it might shed some light.. dunno`
`22:34.44`	`Qwell[]`	`Then the new behavior is correct.`
`22:34.51`	`*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)`
`22:34.58`	`ManxPower`	`paolob-parroquia: I'm pretty sure you'll find out if you file a bug.`
`22:35.00`	`Qwell[]`	`permit/deny are meant only to work on IPs, and it was recently fixed`
`22:35.27`	`b11d`	`in the sip "handshake" im seeing us 0x4 (ulaw) peer - audio 0x4 (ulaw) combined 0x4 (ulaw)`
`22:35.27`	`b11d`	`but non-codec capabilities.. there is a mismatch`
`22:35.32`	`b11d`	`us 0x1 telephone event - peer 0x0 (nothing)`
`22:35.33`	`ManxPower`	`b11d: aha!`
`22:35.36`	`b11d`	`this explains it`
`22:35.42`	`ManxPower`	`b11d: gimme a min`
`22:35.44`	`b11d`	`ok`
`22:36.05`	`ManxPower`	`b11d: actually that should not cause one-way-audio problems, but it should be fixed or DTMF won't work and who knows, it might cause your problem`
`22:36.09`	`paolob-parroquia`	`Qwell, do you mean that I can't use a hostname, I must use a IP?`
`22:36.18`	`b11d`	`i suspect it at least causes my lack-of-ringing problem`
`22:36.32`	`Qwell[]`	`paolob-parroquia: pretty sure that was the case, yes`
`22:38.15`	`NDT`	`Can anyone maybe confirm or deny if realtime sip is working in 1.2.12.1 and 1.2.13? http://pastebin.ca/233256`
`22:38.56`	`NDT`	`Can't figure out why in a static sip.conf file xlite registers fine...add it to realtime get username/auth failures`
`22:39.08`	`b11d`	`I think DTMF is working though Manx.. because I can reach a sip phone from the analog..`
`22:41.51`	`ManxPower`	`b11d: the ringback problem is a known issue.`
`22:42.04`	`b11d`	`where did you find that out?`
`22:42.07`	`ManxPower`	`it's just that so few people use Asterisk with Cisco gateways I'm having trouble finding a post with the solution on it.`
`22:42.12`	`b11d`	`ahh`
`22:42.13`	`b11d`	`yeah`
`22:42.40`	`ManxPower`	`b11d: I just happen to know that there is a cisco setting that fixes the ringback issue. After like 4 years in the asterisk community you pick up little things like that`
`22:43.00`	`b11d`	`yeah.. its why I sit in this channel all day long :)`
`22:44.28`	`NDT`	`I guess I am going to have to write a cron to generate a static file and reload sip in intervals...cause this realtime just won't work`
`22:45.57`	`ManxPower`	`b11d: still checking`
`22:47.15`	`b11d`	`it's appreciated`
`22:47.19`	`ManxPower`	`b11d: what is the event stuff (non codec) capabilities`
`22:47.32`	`b11d`	`hang on a sec`
`22:48.10`	`b11d`	`Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)`
`22:48.21`	`b11d`	`when I call another sip phone, it shows`
`22:48.39`	`b11d`	`Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone event), combined - 0x1 (telephone-event)`
`22:51.19`	`ManxPower`	`b11d: in your dial-peer put " dtmf-relay rtp-nte"`
`22:51.25`	`b11d`	`ok`
`22:51.26`	`ManxPower`	`on the cisco`
`22:51.45`	`ManxPower`	`I don't know if this would apply to the VG224 but it does to other IOS based routers.`
`22:53.04`	`b11d`	`brb`
`22:53.35`	`ManxPower`	`b11d: also try one "ringback alert-no-PI" in the dial-peer`
`22:56.34`	`b11d`	`that first option has been set.. no dice..`
`22:56.38`	`b11d`	`and I lack that 2nd option`
`22:56.53`	`b11d`	`oh wait`
`22:56.53`	`b11d`	`no..`
`22:56.56`	`b11d`	`it did fix it`
`22:57.00`	`b11d`	`but it didnt "fix" anything :)`
`22:57.08`	`b11d`	`non-codec shows 0x1 telephone-event now`
`22:57.11`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`22:57.52`	`ManxPower`	`b11d: it is progress.`
`22:58.04`	`ManxPower`	`the ringback should be fixed by the ringback line`
`22:59.07`	`b11d`	`hmm.. ok.. let me check again`
`22:59.14`	`b11d`	`and yes.. it is progress!`
`22:59.28`	`hoobastooba`	`quick question for you all... when upgrading firmware version on polycom phones, do i just have to replace my sip.cfg file?`
`23:00.20`	`b11d`	`no.. im not seeing that ringback command anywhere.`
`23:00.25`	`ManxPower`	`hoobastooba: no, you replace the sip.ld`
`23:00.28`	`b11d`	`not in voice-port, dial-peer, or even just in the root of configure t.`
`23:00.40`	`b11d`	`nor in the sip-ua`
`23:00.41`	`ManxPower`	`b11d: must be a difference in IOS stuff`
`23:00.47`	`b11d`	`yeah..`
`23:00.50`	`kronic`	`anyone able to give me a hand troubleshooting circular call distribution/queueing and member progression?`
`23:00.51`	`hoobastooba`	`that is it?`
`23:00.56`	`*** part/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do)`
`23:01.04`	`hoobastooba`	`sip.ld? i do not have to do anything else?`
`23:01.08`	`ManxPower`	`b11d: you are at the point where you need a real sip debug on pastebin`
`23:01.11`	`*** part/#asterisk BurtyB (n=chris@alxa.camelnetwork.com)`
`23:01.33`	`ManxPower`	`hoobastooba: if you want to upgrade the sip firmware you replace the sip.ld file on your FTP/tftp server.`
`23:01.48`	`CunningPike`	`hoobastooba: The general recommendation is to replace sip.ld and then replace or merge your sip.cfg (and ipmid.cfg if you have one) as there are often changes to the files caused by changes in sip.ld`
`23:01.50`	`b11d`	`i pasted what syslog had..`
`23:01.50`	`ManxPower`	`OLD firmware versions (more than a year old) you had to do something special if you were using tftp`
`23:02.12`	`hoobastooba`	`ok`
`23:02.19`	`hoobastooba`	`thanks for the edutorial`
`23:02.21`	`ManxPower`	`b11d: The same area that had the codec info and stuff like that.`
`23:02.21`	`CunningPike`	`hoobastooba: For example, 1.6.7 changed the audio levels - lots of people who didn't use the new sip.cfg had feedback issues on speakerphone etc`
`23:02.29`	`b11d`	`yeah.. let me try to snag it`
`23:02.51`	`hoobastooba`	`thanks all`
`23:02.55`	`ManxPower`	`the stuff I saw earlier on pastebin from you were more ASTERISK debug, rather than sip protocol debug`
`23:02.57`	`b11d`	`ah ha!`
`23:02.59`	`*** join/#asterisk backblue (n=moo@87-196-3-207.net.novis.pt)`
`23:03.17`	`b11d`	`its "tone ringback alert-no-PI"`
`23:03.20`	`b11d`	`one sec..`
`23:03.50`	`ManxPower`	`b11d: also you have the CISCO log to the syslog server too, just be sure to edit /etc/sysconfig/syslog and add -r to the parameters line then "service syslog restart"`
`23:04.10`	`ManxPower`	`and set logging 1.2.3.4 in the cisco where 1.2.3.4 is the ip address of your asterisk server.`
`23:04.42`	`b11d`	`im already doing that manx.`
`23:04.53`	`b11d`	`ive been dumping the cisco side all this time to my syslog box.`
`23:06.39`	`b11d`	`well.. that tone ringback stuff didnt seem to work :/`
`23:06.48`	`b11d`	`let me get that syslog data..`
`23:06.48`	`b11d`	`brb`
`23:06.51`	`ManxPower`	`b11d: usually it's the lazy or non-technical ones that have these sorts of weird problems.`
`23:07.00`	`ManxPower`	`You are neither`
`23:07.27`	`b11d`	`thanks :)`
`23:08.04`	`NDT`	`Anyone at all in here now using realtime sip? Not realtime static conf...but realtime sip and extensions?`
`23:08.06`	`*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`23:08.40`	`ManxPower`	`b11d: My advice at this point is to ask on asterisk-users mailing list.`
`23:08.54`	`b11d`	`thats cool.. good advice..`
`23:09.34`	`b11d`	`this syslog output from the box is like.. 580 pages..`
`23:09.43`	`b11d`	`im going to delete it.. and get a fresh capture.`
`23:10.32`	`ManxPower`	`nod`
`23:11.00`	`ManxPower`	`I've started a nice strong drink, so expect my typing to start suffering.`
`23:12.47`	`*** join/#asterisk nmsclera (n=chatzill@207.114.199.107)`
`23:13.08`	`b11d`	`what are you drinking Manx?`
`23:14.05`	`b11d`	`this is a fresh capture from the cisco side..`
`23:14.06`	`b11d`	`http://pastebin.ca/233449`
`23:14.25`	`nmsclera`	`Query -- say you have a set of extensions in a "main_menu" context, given extension => 1 and say, extension => 111. If setting the digit timeout in that context to 5 seconds, how can you make the first "1" the user presses not recognized as a match so that a user may finish hitting "111" and still be valid?`
`23:14.28`	`*** join/#asterisk fiber0pti (n=John@207.114.199.107)`
`23:16.31`	`nmsclera`	`Basically, I'm trying to have a menu with "press 1 for apples, 2 for pears, 3 for oranges -- or if you know the extension of the party you're trying to reach, you can press the extension at any time" and have a user extension, of say "222" not match the "pear" menu option`
`23:17.53`	`Un1x`	`anyone know whats a good processor`
`23:17.54`	`Un1x`	`to run 3 g729 calls at once`
`23:18.04`	`kronic`	`nmsclera: WaitExten?`
`23:18.07`	`*** join/#asterisk ma-requin (n=youssef@dyn10.tcom.ou.edu)`
`23:18.16`	`b11d`	`Un1x.. didnt you just get done talking about that?`
`23:18.26`	`b11d`	`wasnt the answer basically anything made after 2004?`
`23:18.33`	`Qwell[]`	`1992`
`23:18.43`	`b11d`	`hah.. yeah i bet a 386 will handle it.`
`23:18.44`	`b11d`	`:)`
`23:18.59`	`ma-requin`	`do I really have to have extra hardware other than PC and wireless card to call PSTN ??`
`23:19.07`	`b11d`	`well dont just buy the oldest, cheapest, you can find.. otherwise you wont have luck finding replacement parts when the time comes.`
`23:19.48`	`ManxPower`	`b11d: you don't have any firewall or packet filtering on the Asterisk box, right?`
`23:20.21`	`*** join/#asterisk icallandy (n=andy@cpe-74-72-215-143.nyc.res.rr.com)`
`23:20.24`	`ManxPower`	`nmsclera: don't have an option 2`
`23:20.44`	`ManxPower`	`you don't want overlapping options`
`23:21.09`	`ManxPower`	`What I do is "if you know the extension of the person you are trying to reach, dial 1" and that sends them to a prompt to dial the extensions`
`23:21.11`	`icallandy`	`Anyone who's available during the daytime for some part-time work, please PM me`
`23:21.13`	`Un1x`	`hmm u sure 386 will handle 3 calls encoded in g.729`
`23:21.19`	`Un1x`	`coz i was told g.729 was processor intensive`
`23:21.32`	`Un1x`	`and something like a AMD X2 4800+ or p4 + was good`
`23:21.42`	`Un1x`	`but youre saying 386 will handle it..`
`23:21.55`	`b11d`	`Manx.. I do have ipfw, but its set to allow all traffic.. nothign gets denied.`
`23:21.58`	`icallandy`	`Un1x: You probably don't need something that powerful, but a 386 might be pushing it.`
`23:21.58`	`ManxPower`	`I had a Pentium that did not do 1 G729 call.`
`23:22.35`	`b11d`	`no one runs 386's except satellites in orbit and the space shuttle..`
`23:22.43`	`b11d`	`only cause they have been radiation-hardened`
`23:23.03`	`b11d`	`and that nerd in the back of your classroom`
`23:23.06`	`b11d`	`which is probably me`
`23:23.07`	`b11d`	`:)`
`23:23.11`	`b11d`	`I still have a VAX at home :)`
`23:23.32`	`Un1x`	`so b11d are you saying a AMD X2 4800+ is good enough to run stable with 3 calls >?`
`23:23.33`	`ManxPower`	`b11d: I don't see anything obviously wrong`
`23:23.41`	`*** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com)`
`23:23.43`	`b11d`	`yeah.. neither do i.. you solved all the obvious issues :)`
`23:23.58`	`b11d`	`well.. i'll probably call it a day and hit this up tomorrow morning.`
`23:24.04`	`b11d`	`i'll try the IOS downgrade..`
`23:24.19`	`b11d`	`thanks again for the help.. its just a great comfort to even have someone to talk about this stuff with`
`23:24.21`	`ManxPower`	`b11d: BTW so we fixed the 404s and the ringback issue and the dtmf event issue?`
`23:24.28`	`b11d`	`oh.. well.. no`
`23:24.35`	`b11d`	`we fixed the 404 and the non-codec issue`
`23:24.38`	`b11d`	`but it still doesnt ring back..`
`23:24.42`	`ManxPower`	`ah, OK.`
`23:25.02`	`ManxPower`	`b11d: look at the progress options in sip.conf.exmaple and play around with them. I doubt it will help`
`23:25.24`	`b11d`	`yeah, i was also looking at fooling with the "progress_ind" settings in the dial-peer options`
`23:25.33`	`b11d`	`but i am not sure which values mean what..`
`23:25.42`	`ManxPower`	`b11d: yes, play around with those first.`
`23:25.45`	`b11d`	`like.. progress_ind setup enable`
`23:25.49`	`b11d`	`like.. 1 2 or 3.`
`23:25.51`	`b11d`	`whats up with that ;)`
`23:25.55`	`ManxPower`	`b11d: progress tones are the ring, bust, etc.`
`23:26.18`	`b11d`	`ahh.. i just want to verify that 1 means X and 2 means Y and 3 means B`
`23:26.19`	`b11d`	`etc..`
`23:26.24`	`Un1x`	`Comon someone please just suggest some god damn good cpu to use lol`
`23:26.33`	`b11d`	`go get Cray`
`23:26.46`	`b11d`	`Un1x.. get a modern, dual core, AMD cpu..`
`23:26.52`	`b11d`	`done.`
`23:26.52`	`ManxPower`	`Un1x: try it on whatever PC you have right now.`
`23:27.05`	`Un1x`	`i have a AMD Sempron 2200+`
`23:27.07`	`Un1x`	`lol`
`23:27.31`	`ManxPower`	`Un1x: I have no idea what that is other than it's from AMD. Does it run at 2.2 Ghz?`
`23:27.43`	`Un1x`	`Manxpower yes`
`23:27.50`	`b11d`	`it has virtually no on-chip cache, IIRC.`
`23:27.58`	`ManxPower`	`Un1x: that should do 3 G729 calls`
`23:28.01`	`b11d`	`the sempron is like the duron, or the celeron.. iirc.`
`23:28.11`	`[TK]D-Fender`	`Un1x : Should do plenty more than that.`
`23:28.17`	`b11d`	`stupid processor names.. bring back the x86 line..`
`23:28.20`	`ManxPower`	`I stopped keeping track of CPUs after the Celeron`
`23:28.22`	`b11d`	`i want my 986 dammit`
`23:28.27`	`ManxPower`	`They are all fast enough for me.`
`23:28.32`	`Un1x`	`so why was everyone in the day saying 729 was soo processor intensive when its really not?`
`23:28.35`	`[TK]D-Fender`	`b11d : I want my 976.....`
`23:28.41`	`b11d`	`haha`
`23:28.44`	`b11d`	`you and your porno`
`23:28.46`	`Un1x`	`cool imma try g.729 on tihs in a bit :P`
`23:29.08`	`b11d`	`well guys.. everything is GREATLY appreciated, as usual.`
`23:29.11`	`b11d`	`thanks again for all the assistance`
`23:29.14`	`[TK]D-Fender`	`Un1x : It is when you cap out a dozen or so channels.... just try and see ok?`
`23:29.17`	`b11d`	`i shall see you on the morrow.`
`23:29.26`	`b11d\|bbl`	`goodnight`
`23:29.41`	`[TK]D-Fender`	`night`
`23:30.54`	`Un1x`	`ok`
`23:31.05`	`Un1x`	`thanks d-fender`
`23:31.33`	`Un1x`	`018`
`23:35.56`	`[TK]D-Fender`	`Un1x :018?`
`23:35.57`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`23:36.17`	`*** join/#asterisk adorah (n=admin@84.94.124.170.cable.012.net.il)`
`23:36.59`	`carrar`	`hi [TK]D-Fender`
`23:38.31`	`[TK]D-Fender`	`carrar : y0`
`23:40.40`	`Un1x`	`d-fender mistake :P`
`23:40.53`	`[TK]D-Fender`	`Un1x : ?`
`23:41.21`	`Ryanw`	`how can a user record a custom voicemail greeting?`
`23:41.29`	`b11d\|bbl`	`mailbox options`
`23:41.31`	`b11d\|bbl`	`dammit im gone`
`23:41.32`	`b11d\|bbl`	`:)`
`23:41.47`	`[TK]D-Fender`	`Ryanw : in voicemailmain you can do all the recordings (busy/unavail,temp, and name for the Directory)`
`23:41.58`	`*** join/#asterisk javar (n=javar@69.79.134.24)`
`23:42.00`	`b11d\|bbl`	`ok for real this time.. im leaving`
`23:42.05`	`[TK]D-Fender`	`b11d\|bbl : Denial... its not just a river in Egypt ;)`
`23:42.05`	`b11d\|bbl`	`goodnight all`
`23:42.10`	`b11d\|bbl`	`haha I know I know!`
`23:42.56`	`Ryanw`	`thanks, 0 for mailbox options, i was too impatient to listen to all the prompts on the first menu`
`23:43.10`	`b11d\|bbl`	`those "advanced options" is confusing to most at first..`
`23:43.13`	`b11d\|bbl`	`wow..`
`23:43.16`	`[TK]D-Fender`	`Ryanw : Or read the WIKI, the BOOK, etc....`
`23:43.16`	`b11d\|bbl`	`WHY AM I STILL HERE!`
`23:43.25`	`b11d\|bbl`	`...must.. go.. home..`
`23:43.35`	`[TK]D-Fender`	`b11d\|bbl : unload chan_persist.so`
`23:43.36`	`kronic`	`http://pastebin.ca/233489 anyone able to tell me why if NOANSWER is return for member 00001 it doesn't progress to 00002`
`23:43.39`	`b11d\|bbl`	`:)`
`23:43.52`	`b11d\|bbl`	`ok.. no bs.. im out the door.. ttyl`
`23:45.15`	`Ryanw`	`kronic, just a guess because you have retry set to 0`
`23:45.23`	`NDT`	`rmmod b11d\|bbl`
`23:45.33`	`NDT`	`lol`
`23:46.24`	`b11d\|bbl`	`yeah im still here`
`23:46.25`	`b11d\|bbl`	`god`
`23:46.26`	`b11d\|bbl`	`that sucks`
`23:46.28`	`b11d\|bbl`	`:)`
`23:46.35`	`*** join/#asterisk [Outcast] (n=bill@222-154-61-84.jetstream.xtra.co.nz)`
`23:46.40`	`NDT`	`your like a kernel module that won't unload\`
`23:46.42`	`NDT`	`heh`
`23:46.47`	`b11d\|bbl`	`i know!`
`23:46.52`	`b11d\|bbl`	`ok.. theres only one way out.`
`23:46.54`	`b11d\|bbl`	`ttyl all`
`23:46.55`	`b11d\|bbl`	`for real`
`23:46.59`	`NDT`	`hehe later`
`23:47.08`	`*** join/#asterisk dexteruk (n=charper@adsl-68-90-35-43.dsl.hstntx.swbell.net)`
`23:47.25`	`kronic`	`Ryanw: ok`
`23:48.20`	`Ryanw`	`kronic, its trying the queue using the rrmemory strategy, timing out and then not trying a 2nd time. if you have ringall set retry = 0, rrmemory retry = numberofagents`
`23:49.21`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`23:49.51`	`dexteruk`	`I have calls comming into a location A and calls on 281-727-2323 gotos to location B at location B i see the caller ID but i no longer see the 281-727-2323... How can i pass that number to the B as well as the caller ID`
`23:49.58`	`kronic`	`Ryanw: no difference`
`23:50.15`	`dexteruk`	`the calls are arriving as SIP`
`23:50.50`	`[TK]D-Fender`	`kronic : You seem to have assigned different penalties to each of your members and because the first one isn't being kicked out the call never tries the 2nd penalty level.`
`23:50.56`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`23:51.25`	`kronic`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+config+queues.conf`
`23:51.28`	`kronic`	`see the bottem, that's why`
`23:51.54`	`[TK]D-Fender`	`kronic : What part?`
`23:52.04`	`kronic`	`about rrmemory and roundrobin`
`23:52.14`	`kronic`	`regard circular call distribution`
`23:52.21`	`kronic`	`s/regard/regarding/`
`23:53.29`	`[TK]D-Fender`	`kronic : I might just as soon remove that penalty if I were you.`
`23:53.34`	`dexteruk`	`Can anyone help. I have SIP calls arriving at my main office and i want to pass those calls via the IAX trunks to the remote office but keeping both the called number and the caller ID intact?`
`23:54.28`	`[TK]D-Fender`	`kronic : and that paragraphs' methodology seems to refer more to roundrobin, not RRMEMORY`
`23:54.55`	`kronic`	`[TK]D-Fender: I've asked this question a fair few times on this channel, and someone suggested that`
`23:56.31`	`[TK]D-Fender`	`kronic : try and take out that priority`
`23:56.38`	`kronic`	`I have, that works`
`23:56.57`	`kronic`	`but I want the call behavour/direction to be the same each time, eg start at 00001 and progress`
`23:57.43`	`Ryanw`	`kronic, does increasing the timeout of the queue help ?`
`23:57.59`	`dexteruk`	`<PROTECTED>`
`23:58.21`	`[TK]D-Fender`	`kronic : then it sounds like you want roundrobin all by itself.`
`23:58.24`	`Ryanw`	`kronic if you want 00001 to be first all the time then why just not use the stragerty rr instread of rrmemory ?`
`23:59.12`	`Zodiacal`	`anyone if there�s a way to turn off echo cancellation for a specific call?`
`23:59.48`	`kronic`	`Ryanw, [TK]D-Fender: cheers, I'll take a look`

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