irclog2html for #asterisk on 20060704

`00:01.02`	`paolob-parroquia`	`Hi guys! How can I send on pstn a security code before the number to dial? i.e., in order to make long distance calls the telephone company gave me a code (*23456), after it I get a dial tone and I can dial 1xxxxxxx. How do I wait for the second dial tone after the code? thank you`
`00:02.36`	`znoG`	`maybe you can try the SendDTMF application`
`00:03.40`	`paolob-parroquia`	`znoG, will dial("SIP/*2345@pstn\|2) work?`
`00:07.15`	`Juggie`	`paolob, is your telco tdm or ip?`
`00:07.32`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`00:09.59`	`paolob-parroquia`	`Juggie, tdm I suppose (normal pstn`
`00:10.02`	`paolob-parroquia`	`)`
`00:10.31`	`nassy`	`is there a place i can go to for reviews (recommendations) for standalone hardware ethernet SIP phones. i am looking to use it at home but with the eventual goal of adding it to an office of about 150 users after i become familar with asterisk (and the phone)`
`00:10.33`	`paolob-parroquia`	`znoG, How do I tell sendDTMF where to send the dtmf tones?`
`00:10.37`	`Juggie`	`paolob-parroquia, what country?`
`00:10.45`	`paolob-parroquia`	`Juggie, Dominican Republic`
`00:10.55`	`Juggie`	`what would you normally dial for long distance?`
`00:11.02`	`Juggie`	`if you didnt have to dial the code`
`00:11.13`	`nassy`	`so far i am looking at grandstream gxp 2000`
`00:11.18`	`Juggie`	`are you talking local long distance (inside your country) or international?`
`00:12.44`	`paolob-parroquia`	`Juggie, in order to make a long distance call I must wait for the dial tone, dial *2345, then wait for the dial tone again, then dial 18097630026`
`00:13.06`	`Juggie`	`is the dialtone instant after the code?`
`00:13.19`	`Juggie`	`i mean its not a long w ait is it? just a second or two?`
`00:14.13`	`paolob-parroquia`	`Juggie, half a second`
`00:15.01`	`Juggie`	`ok, then all you do is Dial(whatever your zap device is/*2345www${EXTEN})`
`00:15.11`	`Juggie`	`www=1.5seconds (which should be tons)`
`00:15.15`	`Juggie`	`each w = 0.5`
`00:15.38`	`Klydal`	`anyone use FWDout?`
`00:15.41`	`paolob-parroquia`	`Juggie, let me try`
`00:16.05`	`Juggie`	`so, exten=> 1NXXNXXXXXX,1,Dial(Zap/g1/*2345www${EXTEN})`
`00:16.16`	`Juggie`	`replace g1 with whatever group or channel you have configured`
`00:17.09`	`paolob-parroquia`	`Juggie, it's a sip device (sipura spa3000)`
`00:17.36`	`Juggie`	`you said you were connected via pstn`
`00:18.04`	`paolob-parroquia`	`Juggie, :-( I was wrong, perhaps I hadn't understood the question`
`00:18.18`	`paolob-parroquia`	`Juggie, does your trick work with sip?`
`00:19.13`	`Juggie`	`paolob-parroquia, most likely no.`
`00:19.20`	`paolob-parroquia`	`Juggie, :-(`
`00:19.27`	`*** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)`
`00:21.18`	`Juggie`	`paolob-parroquia, how are you connected to the telco?`
`00:21.34`	`Juggie`	`because the fact you have a sip device doesnt matter`
`00:21.55`	`paolob-parroquia`	`Juggie, I am connecte to the spa3000 in the lan, the phone is with a sipura pap2`
`00:24.09`	`Juggie`	`the sipura pap2 is a ata device right? connected to the pstn?`
`00:24.38`	`anthm`	`isnt the sip version of juggies 'w' thing ... this.. Dial(SIP/foo@somehost,60,D(www${EXTEN}))`
`00:29.53`	`paolob-parroquia`	`Juggie, the pap2 is a ata two fxs device, the spa3000 is a one fxo one fxs`
`00:39.17`	`Juggie`	`try what anthm said`
`00:39.52`	`*** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au)`
`00:40.04`	`P-NuT`	`hi all.`
`00:40.53`	`P-NuT`	`I have an external IAX softphone extension, that I want to call the other sip phones in my house but it gives congestion.`
`00:41.08`	`P-NuT`	`I understand this is because the external IAX connectionis not authenticated?`
`00:41.17`	`P-NuT`	`How to I go about solving this?`
`00:51.34`	`*** join/#asterisk phigwork (n=phigan@71-209-135-101.phnx.qwest.net)`
`00:51.37`	`phigwork`	`hi guys`
`00:51.41`	`phigwork`	`I'm trying to compile 1.2.9.1`
`00:51.50`	`phigwork`	`it gives me a bunch of errors while working on pbx_dundi.c`
`00:52.03`	`phigwork`	`anyone else seen this?`
`00:52.52`	`dlynes_home`	`phigwork: make sure gcc is installed`
`00:53.19`	`dlynes_home`	`phigwork: also, what make tool are you using?`
`00:53.30`	`dlynes_home`	`phigwork: i think it's only made to work with gnu make`
`00:54.16`	`phigwork`	`<PROTECTED>`
`00:54.17`	`phigwork`	`GNU Make 3.80`
`00:54.29`	`phigwork`	`$ gcc --version`
`00:54.29`	`phigwork`	`gcc (GCC) 3.3.5 (Debian 1:3.3.5-13)`
`00:54.32`	`dlynes_home`	`what're the errors?`
`00:54.46`	`dlynes_home`	`the lack of a compiler was me being facetious :p`
`00:54.53`	`rob0`	`file: ping?`
`00:55.03`	`dlynes_home`	`rob0: pong?`
`00:55.03`	`rob0`	`I mean ls`
`00:55.05`	`phigwork`	`http://pastebin.ca/78340`
`00:55.09`	`syle`	`facetious wow people are still in the 80's hehe`
`00:55.09`	`phigwork`	`dlynes:`
`00:55.25`	`dlynes_home`	`syle: shenme?`
`00:55.42`	`syle`	`do you call people tools to heheh`
`00:55.59`	`rob0`	`dlynes_home: you're not an Asterlinker, are you?`
`00:56.15`	`dlynes_home`	`nopenopenopenope!!!`
`00:56.48`	`phigwork`	`dlynes_home: not workin?`
`00:57.08`	`dlynes_home`	`ummm`
`00:57.15`	`dlynes_home`	`It's a holiday today`
`00:57.17`	`rob0`	`I'm just wondering, for Asterlink, is there an advantage to SIP over IAX or vice versa?`
`00:57.21`	`dlynes_home`	`Saturday was Canada Day`
`00:57.26`	`dlynes_home`	`besides`
`00:57.29`	`dlynes_home`	`it's 6pm here`
`00:57.34`	`phigwork`	`dlynes_home: http://pastebin.ca/78340`
`00:57.42`	`phigwork`	`woah I can't believe I missed Canananana day`
`00:57.46`	`dlynes_home`	`phigwork: yeah...I got that`
`00:58.03`	`anthm`	`go with sip...`
`00:58.11`	`dlynes_home`	`phigwork: ummm....it's not obvious what's wrong?`
`00:58.13`	`phigwork`	`damn, that's the first Canananananadadada day I've missed boozin' and bbqin' in .. at least 10 years`
`00:58.24`	`phigwork`	`oh oops`
`00:58.36`	`phigwork`	`dlynes: i dunno, when i was looking at it in my terminal, i couldn't see it`
`00:58.43`	`phigwork`	`but now that i see it's the first freakin line i pasted in there`
`00:58.55`	`dlynes_home`	`There's only like 20 lines in the output saying you don't have zlib installed`
`00:58.58`	`rob0`	`anthm: thx`
`00:59.06`	`anthm`	`np`
`00:59.45`	`phigwork`	`hm, no i only see it in the first line :)`
`01:00.01`	`phigwork`	`i hate figuring out packages`
`01:00.16`	`phigwork`	`but if I install source then it messes everything else up :/`
`01:00.19`	`rob0`	`anthm: are you an Asterlinker, or a user, or just a fan of SIP in general? :)`
`01:01.44`	`anthm`	`I wrote the entire backend and most of the features in asterisk necessary to produce it.`
`01:03.57`	`*** join/#asterisk bernardovieira (n=bvieira@c911935d.static.bhz.virtua.com.br)`
`01:04.41`	`*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`01:09.51`	`*** join/#asterisk alephco1 (n=alephcom@host75.net14.mcsnet.ca)`
`01:14.40`	`*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-66.lsanca.fios.verizon.net)`
`01:19.15`	`*** part/#asterisk alephco1 (n=alephcom@host75.net14.mcsnet.ca)`
`01:24.12`	`*** join/#asterisk evilmnky (n=evilmnky@216-106-185-169.ds1-static.mia1.net.ststelecom.com)`
`01:36.18`	`*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)`
`01:46.44`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:49.14`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`01:51.29`	`*** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au)`
`01:51.54`	`Strom_C`	`happy 3rd of july! :)`
`01:54.18`	`*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)`
`01:55.39`	`P-NuT`	`Hi again all.`
`01:55.48`	`Strom_C`	`hello hello`
`01:58.54`	`*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)`
`02:01.30`	`littleball`	`hello, i am looking for solution based on asterisk server with connectivity to TV. any suggestion?`
`02:01.42`	`Strom_C`	`TV...as in television?`
`02:02.10`	`rob0`	`Tuvaalu?`
`02:02.18`	`littleball`	`yes Strom_C, i need to display something on TV`
`02:02.39`	`littleball`	`so, i assume that TV connector, and some vedio codec etc.....`
`02:02.40`	`rob0`	`Ah, my next guess was tetanus virus :(`
`02:02.50`	`littleball`	`i am not familiar with these thing and just started`
`02:02.53`	`Strom_C`	`littleball: well, use AGI and AMI to write a program that integrates with both asterisk and the television`
`02:03.17`	`Strom_C`	`what are you writing? videophones or something? :)`
`02:03.18`	`rob0`	`The OS and/or GUI will handle any display issues, not Asterisk.`
`02:03.20`	`littleball`	`Strom_C, how to integrate to television?`
`02:03.31`	`Strom_C`	`littleball: beats me`
`02:03.37`	`Strom_C`	`littleball: integrating with asterisk is a cinch`
`02:03.51`	`littleball`	`Strom_C, simple advertisement`
`02:03.52`	`Strom_C`	`littleball: integrating with television is beyond the scope of this discussion :)`
`02:03.57`	`littleball`	`to be shown on TC`
`02:04.01`	`littleball`	`TV`
`02:04.08`	`littleball`	`ok`
`02:04.21`	`*** join/#asterisk iq\|mobile (n=iq@unaffiliated/iq)`
`02:04.43`	`Strom_C`	`however, if you wanted to pay me consulting fees, I'm quite sure I could find a solution for you :)`
`02:05.17`	`littleball`	`hehe ,Strom_C`
`02:05.17`	`rob0`	`I couldn't, but I could surely prolong the problem!`
`02:08.25`	`*** join/#asterisk stormfr (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net)`
`02:15.24`	`anonymouz666`	`Does anyone know if CALLERID works on X100P?`
`02:15.42`	`anonymouz666`	`DTMF`
`02:16.13`	`[TK]D-Fender`	`anonymouz666 : Sometimes. So close are worse than others`
`02:16.27`	`[TK]D-Fender`	`anonymouz666 : And it depends on regional support as well`
`02:17.34`	`anonymouz666`	`same thing on TDM cards? or no?`
`02:17.46`	`Strom_C`	`man, what the hell is with the X100P? Didn't Digium discontinue them ages ago? :)`
`02:17.52`	`anonymouz666`	`yeah`
`02:17.56`	`anonymouz666`	`but I still have one.`
`02:17.59`	`anonymouz666`	`and works.`
`02:18.23`	`anonymouz666`	`My only problem is callerid`
`02:18.25`	`Strom_C`	`More people come in here with x100p problems than TDM400 problems...`
`02:18.30`	`anonymouz666`	`asterisk never shows it`
`02:18.37`	`Strom_C`	`anonymouz666: there is a callerid setting`
`02:18.39`	`drray`	`it depends on your line`
`02:18.48`	`anonymouz666`	`yes, there is a callerid setting.`
`02:18.51`	`[TK]D-Fender`	`anonymouz666 : pastebin your zapata.cong. Also where are you located?`
`02:18.57`	`anonymouz666`	`Brazil`
`02:19.02`	`Strom_C`	`cidsignaling`
`02:19.03`	`Strom_C`	`try`
`02:19.14`	`Strom_C`	`cidsignalling=dtmf`
`02:19.54`	`anonymouz666`	`that does not work.`
`02:19.58`	`anonymouz666`	`already tried this.`
`02:20.09`	`[TK]D-Fender`	`DTMF CID? eek...`
`02:20.17`	`Strom_C`	`does the dtmf come in after a polarity reversal or after the ring?`
`02:20.41`	`anonymouz666`	`polarity reversal.`
`02:20.53`	`Strom_C`	`what does adding cidstart=polarity do?`
`02:23.23`	`anonymouz666`	`well, I am not sure. I think that CID comes in the second ring. So....`
`02:23.39`	`Strom_C`	`why not try it and tell me what happens`
`02:24.38`	`anonymouz666`	`I don't have access right now. but that config can be test quickly tomorrow. if that doesn't work do you have any suggestion or no?`
`02:25.41`	`Strom_C`	`anonymouz666: oh hell, dont pull this "I want you to help me troubleshoot but I don't have access to the system" nonsense. Come back on IRC when you're sitting in front of the console`
`02:26.07`	`anonymouz666`	`:(`
`02:26.36`	`Strom_C`	`at that point, I'm sure we will be able to help you`
`02:27.42`	`anonymouz666`	`you already did a great favor to me. I am sure that will work tomorrow. If not, I can switch to a TDM card and test again.`
`02:44.11`	`anonymouz666`	`Strom_C`
`02:44.21`	`Strom_C`	`yes`
`02:44.42`	`anonymouz666`	`changing the cidsignalling I should do a reload on chan_zap or I must reload the whole pbx`
`02:44.58`	`anonymouz666`	`stop now and safe_asterisk again?`
`02:45.01`	`Strom_C`	`probably best to restart asterisk`
`02:45.05`	`Strom_C`	`just do "restart now"`
`02:45.53`	`anonymouz666`	`ok`
`02:46.38`	`file`	`Strom_C: Strommy Boy!`
`02:46.52`	`Strom_C`	`file: FILEFILE!`
`02:47.40`	`file`	`eep`
`02:47.42`	`file`	`ungood`
`02:52.22`	`JunK-Y`	`hey mr file!`
`02:52.29`	`file`	`hola!`
`02:53.26`	`*** join/#asterisk TheCops (i=nobody@got.securebinary.com)`
`02:53.36`	`JunK-Y`	`whats up?`
`02:53.57`	`bernardovieira`	`does anyone know what I have to put in the dial string for a zap channel to make asterisk pause for a couple of seconds?`
`02:54.02`	`Strom_C`	`w`
`02:54.04`	`file`	`watching... Dead Like Me and then a movie perhaps`
`02:54.07`	`hads`	`wwww`
`02:54.52`	`VeNoMouS_`	`bernardovieira: why would u need it to pause?`
`02:55.10`	`anonymouz666`	`bernardovieira: fala cara`
`02:55.25`	`bernardovieira`	`VeNoMouS_: legacy pbx behind a fxo interface...`
`02:55.49`	`bernardovieira`	`VeNoMouS_: the pbx takes some 2s to give an outside dialtone`
`02:55.49`	`anonymouz666`	`bernardovieira: vi sua mensagem na asterisk-brasil.... Voc� n�o precisa se matar pra portar o SIP-RTP novo pro Asterisk 1.2.9.1. D� uma olhada no asterisk-backports.org... J� est� feito.`
`02:56.09`	`VeNoMouS_`	`anonymouz666 parla english`
`02:56.24`	`VeNoMouS_`	`si?`
`02:56.33`	`bernardovieira`	`anonymouz666: thanks! I'll have a look....`
`02:56.34`	`anonymouz666`	`that's not spanish`
`02:56.48`	`Strom_C`	`portuguese!`
`02:56.53`	`anonymouz666`	`yeah`
`02:57.08`	`VeNoMouS_`	`CJ575058195 KBDownload - View - Edit04/07/2006 - 14:01:44`
`02:57.08`	`VeNoMouS_`	`<unavailable>190 KBDownload - View - Edit04/07/2006 - 14:01:41`
`02:57.08`	`VeNoMouS_`	`CJ574895111 KBDownload - View - Edit04/07/2006 - 14:01:29`
`02:57.08`	`VeNoMouS_`	`A522928523101 KBDownload - View - Edit04/07/2006 - 14:01:22`
`02:57.15`	`VeNoMouS_`	`fuck ignore that`
`02:57.18`	`VeNoMouS_`	`stupid mouse`
`02:57.44`	`bigmac4444`	`lol`
`02:58.12`	`[TK]D-Fender`	`JunK-Y : ! ! !`
`02:58.23`	`JunK-Y`	`tk!!!!!!`
`02:58.25`	`anonymouz666`	`Strom_C :)`
`02:59.18`	`Strom_C`	`of course, thanks to Terry Gilliam, Brazil to me is now "that place with all the paperwork"`
`03:02.27`	`bernardovieira`	`Strom_C: Terry Gilliam didn't know half of it....`
`03:04.05`	`*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)`
`03:04.26`	`rene-`	`this if off-topic`
`03:04.53`	`Strom_C`	`no more off topic than "aquarela do brasil"`
`03:04.55`	`rene-`	`how long a domain transfer takes in average?`
`03:05.16`	`rene-`	`tsup Strom_C`
`03:07.00`	`rene-`	`so does anybody knows the average time for a domain transfer (between registrars) to take place?`
`03:07.15`	`Strom_C`	`ive only done it once and i dont remember how long it took`
`03:07.19`	`rene-`	`i was like, yeah i will keep paying netsol more than twice what others charge`
`03:07.51`	`rene-`	`and netsol really hides the epp blocking for allowing transfers to take place`
`03:24.33`	`[TK]D-Fender`	`.`
`03:25.41`	`Strom_C`	`..`
`03:26.03`	`rene-`	`...`
`03:32.10`	`Juggie`	`..`
`03:32.13`	`[TK]D-Fender`	`I had a "point" the rest of you are jsut stuttering@`
`03:34.41`	`rene-`	`mine was a counterpoint`
`03:35.17`	`Strom_C`	`mine was an omgwtfbbq attack`
`03:35.25`	`*** join/#asterisk nvicf (n=v@201.250.166.197)`
`03:35.27`	`nvicf`	`hello`
`03:35.35`	`Strom_C`	`good afternoon`
`03:35.40`	`nvicf`	`how are you?`
`03:35.48`	`Strom_C`	`cheesy`
`03:36.26`	`nvicf`	`what do you mean?`
`03:37.14`	`Strom_C`	`well, as opposed to buttery`
`03:38.02`	`*** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)`
`03:41.18`	`Agrajag-`	`gday. i have an spa-3000 connected to an asterisk box. with some calls from soft phones (sip and iax) that go out or come in through the pstn line connected to the spa-3000, they can hear us fine but we can't hear them very well - it breaks up. using the line connected to the spa-3000 is always ok though (spa-3000 line1 is registered with asterisk too, not pstn direct)`
`03:41.47`	`Strom_C`	`what is between the spa and the asterisk box`
`03:42.35`	`Agrajag-`	`not sure i get what you're asking - a switch?`
`03:42.40`	`Agrajag-`	`they're on the same network`
`03:42.51`	`Strom_C`	`ok`
`03:43.05`	`Strom_C`	`they're both connected to the same switch?`
`03:43.11`	`Agrajag-`	`yep`
`03:43.41`	`Agrajag-`	`which is why i dont understand why the spa line1 always has fine quality but the other softphones (again on the same network) dont`
`03:44.20`	`Agrajag-`	`softphone to softphone works fine`
`03:48.04`	`nvicf`	`I have a little problem, my asterisk is giving me at 5038 no signal when user_suspended ocurrs, and the ip frame is giving me user_suspended when a user hangs, so this is giving me problems, any clues as to how can I fix this?`
`03:52.44`	`*** join/#asterisk bmg505 (n=leon@c1-230-3.rndf.isadsl.co.za)`
`03:52.52`	`Agrajag-`	`ok if that one's too tricky i have another one - if a phone doesn't support conference call, is there a way to add it as a feature in the same way you can add blind transfer?`
`03:54.15`	`*** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net)`
`03:55.10`	`[TK]D-Fender`	`Agrajag- : What kind of phone are you referring to?`
`03:55.33`	`*** part/#asterisk bernardovieira (n=bvieira@c911935d.static.bhz.virtua.com.br)`
`03:55.49`	`Agrajag-`	`[TK]D-Fender: iaxcomm for example?`
`03:56.25`	`[TK]D-Fender`	`Agrajag- : Doesn't support conference?`
`03:56.26`	`[TK]D-Fender`	`Eek`
`03:57.15`	`Agrajag-`	`i also have another phone in the office that's connected to the spa-3000 line1 that doesn't have a flash button`
`03:58.57`	`nvicf`	`hey, why not answering my question?`
`03:59.02`	`nvicf`	`:P`
`03:59.19`	`Strom_C`	`~ygwypf`
`03:59.31`	`Strom_C`	`damned bot`
`04:03.04`	`[TK]D-Fender`	`Agrajag- : Don't need a flash button... just hook-flash the old fashioned way`
`04:06.03`	`Agrajag-`	`[TK]D-Fender: how is that done?`
`04:07.58`	`*** join/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com)`
`04:08.14`	`seb-`	`is asterisk easy for a newbie to get some basic setup out of it?`
`04:08.23`	`seb-`	`i'd like to set up 3way calling w/ it`
`04:11.52`	`[TK]D-Fender`	`Agrajag- : hang up and pickup fast`
`04:12.05`	`[TK]D-Fender`	`Agrajag- : hence the term "hook flash"`
`04:13.28`	`russellb`	`asterisk sets itself up`
`04:13.31`	`russellb`	`it r0x0rz`
`04:16.47`	`Agrajag-`	`[TK]D-Fender: ahh.. never knew about that, ta`
`04:25.53`	`*** join/#asterisk lmcdowell88 (n=lmcdowel@prefect.oss.ntelos.net)`
`04:27.19`	`lmcdowell88`	`Busy?`
`04:27.55`	`nvicf`	`I have a little problem, my asterisk is giving me at 5038 no signal when user_suspended ocurrs, and the ip frame is giving me user_suspended when a user hangs, so this is giving me problems, any clues as to how can I fix this?`
`04:28.33`	`*** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au)`
`04:29.52`	`seb-`	`russellb: asterisk "just works" ?`
`04:30.04`	`russellb`	`yep.`
`04:30.24`	`seb-`	`russellb: but surely one must do work to understand voip and configure all the cool thingees you can do with it`
`04:30.42`	`russellb`	`yes, i'm just joking around`
`04:30.56`	`seb-`	`russellb: is there a nice url to learn it?`
`04:31.04`	`russellb`	`there is a book that you can get as a pdf`
`04:31.07`	`russellb`	`~book`
`04:31.16`	`jbot`	`methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`04:39.23`	`P-NuT`	`hi all..`
`04:44.07`	`*** join/#asterisk joe_acme (n=joe_acme@dyn-83-157-136-1.ppp.tiscali.fr)`
`04:46.18`	`joe_acme`	`ok, I think I got the configuration files OK this time, but... Asterisk doesn't seem able to bridge calls between two FXO cards. /var/log/asterisk/messages = http://codecomplete.free.fr/asterisk/asterisk_cli`
`04:48.07`	`Strom_C`	`joe_acme: are you in front of the computer this time`
`04:48.23`	`joe_acme`	`yup`
`04:48.57`	`Strom_C`	`OK`
`04:49.01`	`Strom_C`	`use pastebin.ca`
`04:49.08`	`joe_acme`	`ok`
`04:49.40`	`bkw_`	`you can't bridge two FXO's`
`04:49.56`	`joe_acme`	`bkw_ : positive?`
`04:50.06`	`bkw_`	`FXO <-> FXO`
`04:50.30`	`bkw_`	`is that what you're trying to do?`
`04:50.33`	`joe_acme`	`yes`
`04:50.41`	`bkw_`	`riddle me this.. who provides the dial tone then?`
`04:50.50`	`joe_acme`	`http://codecomplete.free.fr/asterisk/two_fxo.jpg`
`04:50.53`	`russellb`	`the <-> ?`
`04:51.04`	`P-NuT`	`guys.`
`04:51.06`	`joe_acme`	`the cares are connected to two different POTS lines`
`04:51.07`	`bkw_`	`oh taking a call in and back out`
`04:51.09`	`bkw_`	`IC`
`04:51.10`	`joe_acme`	`cards`
`04:51.12`	`joe_acme`	`yes`
`04:51.15`	`bkw_`	`thats simple`
`04:51.18`	`joe_acme`	`to call a remote phone`
`04:51.24`	`joe_acme`	`yes, that should be simple`
`04:51.30`	`bkw_`	`it is`
`04:51.33`	`joe_acme`	`ok`
`04:51.34`	`joe_acme`	`but...`
`04:51.35`	`joe_acme`	`doesn't work`
`04:51.51`	`joe_acme`	`asterisk says it has called out, but I get no calls on the remote number`
`04:52.40`	`Juggie`	`list of things to try.`
`04:52.57`	`Juggie`	`1) make a outbound call on port 2, or better yet receive an inbound call.`
`04:53.07`	`P-NuT`	`ok, IAX external connections that dial through the sip gateway, are getting congestion.`
`04:53.14`	`P-NuT`	`How can I fix this>`
`04:53.15`	`P-NuT`	`?`
`04:53.21`	`Juggie`	`2) find out why it thinks its answered, SOMETHING is on the other end, what is it... hook a phone up and find out.`
`04:53.22`	`Juggie`	`etc.`
`04:53.31`	`joe_acme`	`Juggie : ok`
`04:53.52`	`Strom_C`	`joe_acme: what happens when you hook a regular phone up to the fxo port?`
`04:53.52`	`Strom_C`	`er`
`04:53.53`	`Strom_C`	`to the line the fxo port is connected to`
`04:53.54`	`Strom_C`	`and try to dial out`
`04:54.06`	`joe_acme`	`both lines work`
`04:54.12`	`joe_acme`	`I get a dial tone and can use the POTS lines to call out`
`04:54.35`	`Strom_C`	`do you have a phone behind the asterisk box?`
`04:54.44`	`joe_acme`	`no`
`04:54.48`	`Strom_C`	`set one up`
`04:54.50`	`Juggie`	`can you dial the exact number your trying to dial from the port2`
`04:55.01`	`joe_acme`	`i'll do it know`
`04:55.03`	`Juggie`	`* thinks something answered`
`04:55.04`	`joe_acme`	`call into FXO 2`
`04:55.09`	`Strom_C`	`no no no`
`04:55.12`	`Strom_C`	`SET UP A PHONE`
`04:55.18`	`joe_acme`	`where?`
`04:55.18`	`Strom_C`	`dial out the fxo`
`04:55.20`	`Strom_C`	`see what happens`
`04:55.23`	`joe_acme`	`ok`
`04:55.24`	`Strom_C`	`softphone?`
`04:55.28`	`Strom_C`	`fxs port?`
`04:55.30`	`Strom_C`	`ata?`
`04:55.32`	`joe_acme`	`no`
`04:55.34`	`joe_acme`	`not yet`
`04:55.44`	`Strom_C`	`softphones are FREE`
`04:55.48`	`joe_acme`	`but I can set an SIP softphone on an other copputer`
`04:55.48`	`joe_acme`	`right`
`04:55.50`	`Juggie`	`joe_acme, what happens on the other end of the line when you dial into fxo1`
`04:55.57`	`Juggie`	`what do you hear?`
`04:56.13`	`joe_acme`	`Asterisk goes off hook, and I hear static`
`04:56.31`	`joe_acme`	`FWIW, two other people tried the same thing, and got the same result`
`04:56.45`	`joe_acme`	`asterisk sits there, silent`
`04:57.02`	`joe_acme`	`I can install a softphone on a computer if you want`
`04:57.02`	`Juggie`	`* thinks the other end answered.`
`04:57.05`	`joe_acme`	`yers`
`04:57.11`	`Juggie`	`what country?`
`04:57.13`	`joe_acme`	`but it doesn't actually dial out`
`04:57.14`	`joe_acme`	`FR`
`04:57.25`	`joe_acme`	`i set the ad hoc settings in zaptel.conf`
`04:57.26`	`Strom_C`	`joe_acme: install the bloody softphone already so we can test`
`04:57.30`	`joe_acme`	`ok`
`04:57.34`	`joe_acme`	`first time`
`04:58.51`	`joe_acme`	`anybody knows of a good tutorial on setting up SIP and a softphone?`
`04:58.58`	`joe_acme`	`so I don't waste time`
`04:59.02`	`joe_acme`	`figuring it out`
`04:59.07`	`joe_acme`	`and waste your time`
`04:59.30`	`Strom_C`	`sigh`
`04:59.38`	`joe_acme`	`no prob, I'll google`
`04:59.47`	`joe_acme`	`I didn't look into SIP yet because no need for it`
`05:04.17`	`joe_acme`	`fwiw, here's the output when I call into Zap/2 from a remote phone http://pastebin.ca/78516`
`05:04.39`	`joe_acme`	`i'll install a softphone`
`05:05.19`	`Strom_C`	`joe_acme: pastebin zaptel.conf and zapata.conf`
`05:05.36`	`Strom_C`	`also are you using regular asterisk or freepbx?`
`05:05.56`	`joe_acme`	`regular asterisk, compiled saturday from latest source`
`05:06.03`	`Strom_C`	`stable?`
`05:06.08`	`Strom_C`	`or svn trunk`
`05:06.08`	`joe_acme`	`yes`
`05:06.22`	`Strom_C`	`yes what`
`05:06.40`	`joe_acme`	`ye, stable version`
`05:06.47`	`joe_acme`	`downloaded zip`
`05:07.03`	`Strom_C`	`joe_acme: pastebin zaptel.conf and zapata.conf`
`05:07.09`	`joe_acme`	`2s`
`05:07.45`	`joe_acme`	`http://pastebin.ca/78518`
`05:08.25`	`Strom_C`	`try this`
`05:08.33`	`Strom_C`	`channel => 1-2`
`05:09.00`	`joe_acme`	`ok`
`05:09.20`	`joe_acme`	`must rmmod zaptel.conf?`
`05:09.27`	`Strom_C`	`what?`
`05:09.28`	`Strom_C`	`no`
`05:09.44`	`Strom_C`	`change zapata.conf, restart asterisk`
`05:09.58`	`*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)`
`05:10.27`	`joe_acme`	`same thing`
`05:11.00`	`joe_acme`	`http://pastebin.ca/78519`
`05:11.02`	`Strom_C`	`can I ssh into it from the outisde?`
`05:11.35`	`joe_acme`	`i'll have to open up the firewall and install/launch sshd`
`05:11.40`	`Strom_C`	`well, no, its not the same thing`
`05:11.41`	`Strom_C`	`look`
`05:11.49`	`Strom_C`	`the last time, the console returned congestion`
`05:11.56`	`Strom_C`	`this time, the console isn't.`
`05:12.02`	`*** join/#asterisk _omer (n=_omer@202.166.162.250)`
`05:12.12`	`joe_acme`	`ok`
`05:12.19`	`Strom_C`	`show me your extensions.conf`
`05:13.15`	`joe_acme`	`http://pastebin.ca/78520`
`05:13.31`	`*** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au)`
`05:14.35`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`05:15.01`	`Strom_C`	`joe_acme: and you're hearing NOTHING?`
`05:15.09`	`joe_acme`	`silence + static`
`05:15.15`	`joe_acme`	`like the other two people`
`05:15.22`	`joe_acme`	`who tried this`
`05:15.23`	`Strom_C`	`what happens when you dial the number from a telephone set`
`05:17.39`	`joe_acme`	`when I use line 2 to plug into a phone and dial out, regular dial tone and I can call out`
`05:17.46`	`joe_acme`	`both lines work`
`05:17.49`	`Strom_C`	`but what happens when you call that specific number`
`05:18.00`	`joe_acme`	`2s`
`05:18.55`	`joe_acme`	`when I call the number of line connected to FXO 2, Asterisk detects it but obviously can't work because the two lines use the same contexte`
`05:18.57`	`joe_acme`	`xt`
`05:19.02`	`Strom_C`	`NO NO NO`
`05:19.05`	`Strom_C`	`listen to me`
`05:19.19`	`Strom_C`	`Attach a telephone set to the telephone line that's currently plugged into Zap/2`
`05:19.29`	`Strom_C`	`then dial the number you're having asterisk dial`
`05:19.33`	`Strom_C`	`and then tell me what happens`
`05:20.27`	`file`	`Strom_C: my head exploded :(`
`05:20.31`	`Strom_C`	`oops`
`05:22.15`	`joe_acme`	`it rings OK`
`05:22.55`	`joe_acme`	`has _anyone_ done this before ?`
`05:23.07`	`joe_acme`	`maybe we're just trying to do sthing that * can't do at this poitn?`
`05:23.17`	`Strom_C`	`joe_acme: try prepending wwww to ${NUMBER}`
`05:24.09`	`joe_acme`	`NUMBER=wwwwwwwwww0145807013`
`05:24.24`	`Strom_C`	`no`
`05:24.25`	`joe_acme`	`2s`
`05:24.26`	`joe_acme`	`yes`
`05:24.27`	`Strom_C`	`no`
`05:24.28`	`joe_acme`	`I kjnow ;-)`
`05:24.30`	`Strom_C`	`NO`
`05:24.45`	`Strom_C`	`Dial(${TRUNK}/www${NUMBER})`
`05:24.47`	`file`	`you do not want Strom_C angry`
`05:24.55`	`joe_acme`	`exten => s,n,Dial(${TRUNK}/wwwwwwwwww${NUMBER})`
`05:25.09`	`Strom_C`	`three or four, not seven million`
`05:25.31`	`joe_acme`	`exten => s,n,Dial(${TRUNK}/wwww${NUMBER})`
`05:25.56`	`Strom_C`	`that's what I said, yes`
`05:26.26`	`trelane`	`all better`
`05:27.16`	`joe_acme`	`same thing`
`05:27.18`	`trelane`	`I'm concerned strom will start killing the users`
`05:27.21`	`trelane`	`that's my job`
`05:27.40`	`*** join/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer)`
`05:27.45`	`*** part/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer)`
`05:27.52`	`joe_acme`	`http://pastebin.ca/78525`
`05:28.33`	`Strom_C`	`try this`
`05:28.39`	`Strom_C`	`in your extensions.conf`
`05:28.40`	`Strom_C`	`do:`
`05:28.49`	`Strom_C`	`exten => s,n,Answer`
`05:28.56`	`Strom_C`	`exten => s,n,Wait(1)`
`05:29.09`	`Strom_C`	`exten => s,n,Dial(whatever)`
`05:31.15`	`*** join/#asterisk postel (n=jp@unaffiliated/postel)`
`05:35.09`	`joe_acme`	`same thing http://pastebin.ca/78527`
`05:36.04`	`Strom_C`	`switch the lines`
`05:36.07`	`joe_acme`	`ok`
`05:36.15`	`Strom_C`	`answer zap/2 and dial out over zap/2`
`05:36.16`	`Strom_C`	`er`
`05:36.20`	`Strom_C`	`dial out over sap/1`
`05:36.22`	`Strom_C`	`zap`
`05:36.24`	`Strom_C`	`dammit`
`05:39.35`	`*** join/#asterisk seb-- (n=cs@cpe-72-132-242-171.san.res.rr.com)`
`05:39.38`	`*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)`
`05:40.22`	`seb--`	`what hardware do i need to do conferencing? does everyone need a sip phone?`
`05:40.34`	`Strom_C`	`everyone needs some kind of phone`
`05:40.47`	`Strom_C`	`meetme needs a timing source - either a zaptel card or ztdummy`
`05:40.56`	`joe_acme`	`worked :-) http://pastebin.ca/78529`
`05:41.09`	`joe_acme`	`I hear a lot of static when calling into Zap/2`
`05:41.22`	`joe_acme`	`but it does go out through Zap/1 and dial the number`
`05:41.26`	`joe_acme`	`any idea why?`
`05:41.39`	`Strom_C`	`maybe your hardware is fucked`
`05:41.42`	`joe_acme`	`ok`
`05:41.49`	`Strom_C`	`get a TDM400`
`05:42.34`	`file`	`I like the TDM2400 better myself`
`05:42.50`	`Strom_C`	`oh, the TDM2400 is a fine piece of hardware`
`05:43.06`	`file`	`the hardware differences between the two... mucho`
`05:43.14`	`file`	`russellb: are you excited?!? CHEEBURGER!`
`05:43.26`	`Strom_C`	`CHEEBURGER CHEEBURGER!!!`
`05:43.27`	`*** join/#asterisk i2omani (n=i2omani@c-24-10-92-50.hsd1.ca.comcast.net)`
`05:43.36`	`Strom_C`	`I still have my cheeburger cheeburger receipt`
`05:43.43`	`i2omani`	`hello all`
`05:43.50`	`Strom_C`	`cheeburger`
`05:43.54`	`file`	`cheeburger`
`05:44.01`	`i2omani`	`anyone here used Linksys RT32P2`
`05:44.20`	`*** join/#asterisk geekster_ (n=steve@dns1.nyc.dns-roots.net)`
`05:44.24`	`joe_acme`	`ok, in any case, * doesn't close Zap/1 when I hang up on Zap/2, so I guess I'll look into either SIP or get some Digium hardware instead`
`05:44.31`	`russellb`	`file: meep`
`05:44.41`	`Strom_C`	`joe_acme: is your telco sending disconnect supervision?`
`05:44.47`	`file`	`hangup detection on analog is evil, unless you get it from the telco`
`05:44.51`	`file`	`russellb: you rock!`
`05:44.52`	`joe_acme`	`no idea`
`05:45.12`	`seb--`	`Strom_C: what if clients got pots (old) phones and asterisk server can't accept any new hardware? how call?`
`05:45.27`	`joe_acme`	`but I did read that analog lines weren't recommended`
`05:45.31`	`Strom_C`	`what do you mean "cant accept any new hardware"?`
`05:45.45`	`file`	`CONGA!`
`05:45.48`	`seb--`	`Strom_C: remote Xen server :)`
`05:45.50`	`joe_acme`	`because of this kind of issue or problems with caller ID (also impedence issues)`
`05:46.01`	`Strom_C`	`seb--: ztdummy and sip then`
`05:46.06`	`Strom_C`	`or even better`
`05:46.07`	`i2omani`	`anyone here used Linksys RT32P2 please help`
`05:46.09`	`Strom_C`	`ztdummy and iax`
`05:46.15`	`Strom_C`	`i2omani: just ask a question`
`05:46.53`	`seb--`	`Strom_C: i assume ztdummy is software? but pots is harware.....somewhere something has to convert pots to digital right?!`
`05:46.54`	`joe_acme`	`Strom_C : looks like problem solved :-) Thanks a bunch for putting up with me`
`05:47.25`	`Strom_C`	`joe_acme: any time`
`05:47.30`	`i2omani`	`Strom_C: i have one but it's locked for my older provider, and i was wondering if someone unlocked it`
`05:47.36`	`Strom_C`	`but next time I charge hourly :)`
`05:47.37`	`file`	`seb--: an ATA would provide you with an FXS port that you could plug a phone into, it would then go via VoIP to your Asterisk box`
`05:47.50`	`file`	`seb--: the timing source (ztdummy) is used in Asterisk to provide an audio mixing solution`
`05:47.55`	`joe_acme`	`Strom_C : sure :-)`
`05:48.02`	`file`	`keep it mad, keep it glad, keep it gay!`
`05:48.23`	`file`	`Strom_C: it's springtime for Hitler!`
`05:48.43`	`Strom_C`	`winter for poland and france`
`05:49.07`	`geekster_`	`hey all, looking for some skilled developers for a 3 month project.`
`05:49.38`	`russellb`	`file will do it for free`
`05:50.03`	`file`	`if by free you mean... not free`
`05:50.26`	`Strom_C`	`by free i think he means muffins`
`05:50.28`	`i2omani`	`anyone`
`05:50.58`	`Strom_C`	`i2omani: I've never unlocked one but I can try and crack it at my standard hourly rate`
`05:51.56`	`i2omani`	`Strom_C: lol, if i had the money i would go and buy me a tdm card man`
`05:52.17`	`seb--`	`Strom_C: thanks for help`
`05:53.42`	`*** part/#asterisk seb-- (n=cs@cpe-72-132-242-171.san.res.rr.com)`
`05:59.48`	`Corydon76-home`	`geekster_: to do what, exactly?`
`06:06.38`	`file`	`sleeeeepy`
`06:07.33`	`Strom_C`	`you people are no fun`
`06:24.40`	`*** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it)`
`06:25.09`	`*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`06:25.44`	`file`	`moo`
`06:26.28`	`Strom_C`	`floof`
`06:28.08`	`stephane_`	`jour`
`06:29.51`	`*** part/#asterisk joe_acme (n=joe_acme@dyn-83-157-136-1.ppp.tiscali.fr)`
`06:30.26`	`*** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz)`
`06:30.43`	`iceyp`	`hey guys... anyone know where i can get a list of landline prefixs in the UK?`
`06:31.20`	`iceyp`	`also looking for Australia... Want to allow mates to call countries free but not to mobile or that will cost me heaps`
`06:31.44`	`Strom_C`	`aren't UK mobile phones segregated off into their own area codes?`
`06:31.53`	`iceyp`	`thats what i mean`
`06:31.59`	`iceyp`	`Im looking for all UK area codes`
`06:32.59`	`Strom_C`	`holy wow! type "uk area codes" into google and you get http://www.ukphoneinfo.com/`
`06:33.15`	`iceyp`	`lol`
`06:33.20`	`iceyp`	`I was typing uk prefix codes`
`06:33.21`	`iceyp`	`duh`
`06:33.21`	`iceyp`	`:/`
`06:33.23`	`iceyp`	`thnx`
`06:33.30`	`Strom_C`	`AREA CODE FTW`
`06:35.21`	`Snake-Eyes`	`is there a trusted list in asterisk?`
`06:35.43`	`*** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net)`
`06:36.05`	`Strom_C`	`hmm`
`06:36.21`	`Strom_C`	`where can I get good new york style pizza at 11:36 PM in los angeles`
`06:36.50`	`CoffeeKid`	`I have a quick question. Does anyone know of a web based app (written in php) that will show the status on an inbound call queue, such as agents logged in, and people waiting in the queue?`
`06:37.52`	`CoffeeKid`	`Strom_C: good luck on that, its 4th of july, most places probably closed :(`
`06:38.11`	`CoffeeKid`	`at least they are here..`
`06:38.20`	`Strom_C`	`whre's "here"?`
`06:38.29`	`CoffeeKid`	`Salt Lake City, UT.`
`06:38.40`	`Strom_C`	`heh. This is Los Angeles!`
`06:38.47`	`CoffeeKid`	`heh, good point :)`
`06:42.32`	`Strom_C`	`there is a fairly good pizza joint in west hollywood`
`06:42.42`	`Strom_C`	`they're open till at least 2 AM`
`06:42.57`	`CoffeeKid`	`there you go..`
`06:43.07`	`CoffeeKid`	`LA is huge, how far away is that from you?`
`06:44.01`	`Strom_C`	`6.2 miles exactly`
`06:44.15`	`CoffeeKid`	`heh, not bad`
`06:44.23`	`*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)`
`06:44.24`	`CoffeeKid`	`let me guess.. google maps?`
`06:44.36`	`Strom_C`	`ESP`
`06:44.43`	`CoffeeKid`	`whats that?`
`06:45.01`	`Strom_C`	`er`
`06:45.02`	`Strom_C`	`http://en.wikipedia.org/wiki/Extra-sensory_perception`
`06:45.13`	`CoffeeKid`	`ah`
`06:45.18`	`CoffeeKid`	`or a good guess? :P`
`06:45.27`	`Strom_C`	`try a bad joke`
`06:45.34`	`Strom_C`	`of course it was google maps`
`06:45.36`	`CoffeeKid`	`heh`
`06:45.55`	`*** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net)`
`06:46.37`	`LoneShadow`	`Does video conferencing work in sync with asterisk voip connection ?`
`06:50.19`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`06:50.29`	`Strom_C`	`pho really`
`06:50.42`	`*** part/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com)`
`06:53.07`	`*** join/#asterisk MstlyHrmls (n=mh@66.195.193.151)`
`06:53.27`	`iceyp`	`looks like all landline numbers in uk are 0044 1`
`06:53.42`	`Strom_C`	`no`
`06:54.01`	`Strom_C`	`landline numbers in central london, for example, are +44 (0) 20 XXXX XXXX`
`06:54.28`	`iceyp`	`mmm`
`06:54.52`	`*** join/#asterisk acehunky (n=chat_jok@59.184.29.221)`
`06:55.10`	`iceyp`	`ok ill add that too`
`06:55.23`	`iceyp`	`i went by http://homepages.tcp.co.uk/~alounds/std-codes.html`
`06:55.44`	`Strom_C`	`note that the UK recently completely changed its numbering plan`
`06:55.55`	`iceyp`	`damn`
`06:56.14`	`iceyp`	`are mobiles locked into a specific range? i.e. 02X is cellphones here in NZ`
`06:56.22`	`Strom_C`	`I believe so`
`06:56.31`	`Strom_C`	`but as far as the specific code, I dont know offhand`
`06:56.40`	`iceyp`	`hehe`
`06:56.43`	`iceyp`	`thnx anyway`
`06:56.59`	`*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)`
`06:58.35`	`*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)`
`06:58.44`	`SplasPood`	`Hrm, I'm trying to setup a dynamic feature for my clients to dial when on a call that they wish to report audio trouble for... Can't figure out how to get the unique id of the call they're on... DumpChan shows it, but..`
`07:01.56`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`07:02.11`	`Juggie`	`you are over thinking this problem`
`07:02.20`	`Juggie`	`it already exists in ${UNIQUEID}`
`07:02.30`	`Strom_C`	`O say can you....uh....gentlemen, start your engines!`
`07:02.37`	`Juggie`	`automatically there for every call, enjoy :)`
`07:02.59`	`SplasPood`	`Juggie: yes, I realized that was a stupid question :)`
`07:03.13`	`SplasPood`	`Juggie: thank you tho... I was trying CDR(uniqueid) and stuff`
`07:04.39`	`SplasPood`	`hrm, be nice if I could do a SetCDRUserField from within a macro called via feature code`
`07:05.49`	`dlynes_home`	`iceyp: not here, they aren't...cell phone numbers could be all over the board here`
`07:06.12`	`dlynes_home`	`iceyp: here, being Canada`
`07:07.18`	`*** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za)`
`07:07.35`	`*** join/#asterisk Assid (i=assid@203.115.83.214)`
`07:07.42`	`Assid`	`heya`
`07:07.59`	`Nobbie`	`heya =)`
`07:11.48`	`*** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`07:13.59`	`SplasPood`	`Juggie: Hrm, do you know much about the uniqueid? Seems that what I get in my macro as called by feature code varies slightly from what is logged into my CDR`
`07:14.05`	`SplasPood`	`by like.. .03`
`07:14.27`	`SplasPood`	`1151997063.26 vs 1151997063.29`
`07:15.24`	`*** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au)`
`07:15.30`	`P-NuT`	`Hi all.`
`07:15.50`	`*** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au)`
`07:16.48`	`mbit`	`hey has anyone ever had an issue when receiving calls over an iax trunk and it hangs up after 3 rings?`
`07:18.01`	`Nobbie`	`mbit: have you checked debug info ? is the Dial() application not run with a 3 second timeout ?`
`07:18.23`	`P-NuT`	`speaking of IAX`
`07:18.33`	`mbit`	`how do i do that nobbie`
`07:19.16`	`*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)`
`07:19.39`	`Nobbie`	`mbit: run asterisk -r to attach the the CLI/API, then run: set verbose 3`
`07:19.47`	`Assid`	`sup poot`
`07:19.51`	`Nobbie`	`make your call and look for the Dial() application`
`07:19.53`	`Assid`	`pood`
`07:21.00`	`P-NuT`	`if you have an external IAX extension, apart from putting allow=guest in the top of sip.conf is there a way to let them dial out through your PSTN gateway?`
`07:21.07`	`P-NuT`	`a safer way>`
`07:21.18`	`P-NuT`	`can you do allow=2205`
`07:21.23`	`P-NuT`	`<-- the extension.`
`07:21.33`	`P-NuT`	`instead?`
`07:21.45`	`Assid`	`allow=2205 ???`
`07:21.55`	`P-NuT`	`well...`
`07:21.55`	`Assid`	`isnt allow for codecs?`
`07:22.03`	`P-NuT`	`well,`
`07:22.07`	`*** join/#asterisk boddy (n=e@212.58.24.138)`
`07:22.14`	`P-NuT`	`under general iff I put allow=guest`
`07:22.15`	`boddy`	`hii I all`
`07:22.16`	`P-NuT`	`it works.`
`07:22.27`	`Assid`	`allow= is for codecs that you wish to allow`
`07:22.41`	`P-NuT`	`Yeah I know.`
`07:22.53`	`P-NuT`	`but check under general in sip.conf.`
`07:23.03`	`P-NuT`	`it's also for allowing randoms from external.`
`07:23.15`	`Assid`	`thats lowbandwith etc.`
`07:23.15`	`P-NuT`	`my question is though..`
`07:23.20`	`P-NuT`	`yeah..`
`07:23.23`	`P-NuT`	`anyway,`
`07:23.39`	`Assid`	`since the codecs have been categorised for low /medium / high bandwith`
`07:23.52`	`P-NuT`	`instead of allowing everyone, how would I let this 1 external IAX extension dial out..`
`07:23.58`	`P-NuT`	`no.`
`07:24.10`	`P-NuT`	`if you have an IAX extension.`
`07:24.22`	`P-NuT`	`and you want to dial out,`
`07:24.27`	`Assid`	`that all depends upon the context`
`07:24.36`	`P-NuT`	`by default it gives you a congestion barage`
`07:24.39`	`Assid`	`where you can call and who can call you is done by contexts`
`07:24.49`	`P-NuT`	`yeeeeeaaaaaahhh....`
`07:24.53`	`P-NuT`	`I know that,`
`07:25.02`	`P-NuT`	`that's not what I'm getting at.`
`07:25.14`	`boddy`	`I am planing to buy 10 g729 codec but 20 sip client will register to server and almost 10 client will call in same time`
`07:25.17`	`P-NuT`	`I'm talking about IAX (from outside calls coming in.`
`07:25.27`	`Assid`	`if you want to block 1 person from dialling out.. just give him a different context to the locations he can dialout to`
`07:25.33`	`boddy`	`I have to buy 20 codec license`
`07:25.36`	`boddy`	`?`
`07:25.43`	`P-NuT`	`yeah.....`
`07:25.52`	`P-NuT`	`alright, I'll just wing it.`
`07:25.55`	`P-NuT`	`thanks.`
`07:25.56`	`Assid`	`iax is just a transport.. sip/iax .. nothing to do with calls coming in or out..`
`07:26.05`	`P-NuT`	`actally it is,`
`07:26.07`	`P-NuT`	`but thanks/`
`07:26.09`	`Assid`	`calls can be in either direction irrespective of what yuo chose`
`07:26.28`	`P-NuT`	`k`
`07:26.50`	`Assid`	`boddy: you need codecs for the number of concurrent calls`
`07:27.06`	`boddy`	`so 10 enough ?`
`07:27.11`	`Nobbie`	`anyone have an example of Pickup() application ?`
`07:27.32`	`Assid`	`so if you have only 10 calls which are being made simultanously.. i'd really just buy around 12-14 to be on the safe side incase more calls do expand`
`07:27.37`	`Assid`	`you can always buy more when you scale more`
`07:27.47`	`boddy`	`ok Assid thanks`
`07:27.50`	`boddy`	`for help`
`07:28.03`	`LoneShadow`	`anyone using video with thier voip ?`
`07:28.09`	`*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)`
`07:29.08`	`Nobbie`	`not yet`
`07:30.27`	`Assid`	`Nobbie: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup`
`07:32.46`	`*** join/#asterisk hads\|home (n=hads@mail.nice.net.nz)`
`07:33.21`	`*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)`
`07:33.27`	`*** join/#asterisk tparcina (n=tparcina@lns02-1292.dsl.iskon.hr)`
`07:33.33`	`tparcina`	`good morning`
`07:34.43`	`Nobbie`	`thanks =)`
`07:35.47`	`Nobbie`	`how can callgroup/pickupgroup be used without Pickup() application ?`
`07:36.42`	`tparcina`	`does anybody use unix-odbc (for storing cdr)?`
`07:37.40`	`Assid`	`Nobbie: thats for parked calls`
`07:37.49`	`Assid`	`pickup works on context`
`07:38.09`	`tparcina`	`i have problem with unix-odbc, it doesn't connect to mssql 2000 server and i don't know why. I can't find no logs... can someone help?`
`07:38.45`	`Assid`	`tparcina: you mnay want to check with #unixodbc ??`
`07:39.36`	`*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)`
`07:39.44`	`tparcina`	`Assid: there is no #unixodbc channel`
`07:40.40`	`tparcina`	`Assid: and i have checked on ##linux, but it doesn't seam that they know the answer`
`07:44.26`	`drray`	`are you sure that M$-SQL is set up for an ODBC connection?`
`07:44.57`	`tparcina`	`darry, no I'm not.`
`07:45.08`	`tparcina`	`darry, i'm not mssql admin, that is another person`
`07:45.08`	`Assid`	`so you need to verify that`
`07:45.17`	`drray`	`that is where I would start`
`07:45.22`	`tparcina`	`darry, what should i tell them to check?`
`07:45.55`	`Assid`	`a> odbc is working B> mssql allows odbc C> mssql logs for incoming connections d> verify user/pass`
`07:46.10`	`drray`	`"Is this server set up for an ODBC connection? Specifically, I'm having trouble connecting with unix-odbc"`
`07:46.12`	`Assid`	`e> correct unixodbc drivers are installed for mssql`
`07:46.53`	`drray`	`it would not shock me to hear that M$-SQL does not play nice with ODBC out of the box`
`07:47.04`	`tparcina`	`darry, should that be enabled by default, because i have allready used unix-odbc with mssql and i don't think that my coworker has changed anything particular - as far as i know he just created database and provide me username/pass`
`07:48.02`	`Sonderblade`	`anyone know of any softphones which has a configurable sip registration timeout setting?`
`07:48.10`	`drray`	`We are not the droids you are looking for`
`07:50.04`	`Assid`	`Sonderblade: eyebeam/xten`
`07:50.48`	`tparcina`	`assid, darry, thank you!`
`07:52.17`	`Sonderblade`	`Assid: i meant hardphones`
`07:53.13`	`Sonderblade`	`grandstream's phones doesn't seem to have such a setting`
`07:55.16`	`kmilitzer`	`Morning everyone ...`
`07:55.56`	`Assid`	`"<Sonderblade> anyone know of any softphones......... "`
`07:56.06`	`Assid`	`anyways.. polycoms do..`
`07:56.12`	`Assid`	`atleast while provisioning`
`07:56.38`	`kmilitzer`	`Does anyone have an idea why I have problems with the bufferings of packets with my TE205P? As it seems my zaptel input buffer runs full and then my output buffer is empty.`
`07:57.00`	`kmilitzer`	`... as I use chan_ss7 that results in my MTP-Layer flapping`
`07:57.10`	`kmilitzer`	`Any idea why something like this can happen?`
`07:57.15`	`tparcina`	`assid, darry, weary usefull comand is - isql -v MSSQL-asterisk username password : which connects you to mssql server or tell's you why it didn't connect`
`07:57.32`	`Sonderblade`	`Assid: i typoed :)`
`07:58.39`	`*** part/#asterisk tparcina (n=tparcina@lns02-1292.dsl.iskon.hr)`
`08:06.18`	*** join/#asterisk Lino` (n=Lino@i577BD86E.versanet.de)
`08:24.29`	`*** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net)`
`08:29.29`	`*** join/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`08:29.39`	`DHuang`	`hi!`
`08:29.51`	`Strom_C`	`good afternoon`
`08:30.11`	`*** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com)`
`08:30.35`	`DHuang`	`hi Strom_C`
`08:31.23`	`*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)`
`08:31.58`	`DHuang`	`anyone know how to setup or ideas how joining 2 SIP calls? like the dialer on http://www.jajah.com/ site?`
`08:32.41`	`Strom_C`	`what do you mean "joining"`
`08:33.42`	`DHuang`	`on that site it dials 2 numbers.. and how to bridge those two outgoing SIP calls?`
`08:34.07`	`Strom_C`	`it dials two numbers and conferences them together?`
`08:34.08`	`Strom_C`	`easy`
`08:34.11`	`Strom_C`	`.call file`
`08:34.57`	`DHuang`	`so you create conference room 1st and then make 2 SIP calls to join that conference room?`
`08:35.02`	`Strom_C`	`no`
`08:35.28`	`Strom_C`	`you place one call, and then once that call supervises, you have it connect the other call`
`08:35.56`	`DHuang`	`can u do that with SIP?`
`08:36.06`	`Strom_C`	`you can do that with any protocol`
`08:36.09`	`Strom_C`	`it doesn't matter`
`08:36.32`	`Strom_C`	`SIP, IAX, or even H.323 if you're feeling particularly masochistic`
`08:36.47`	`DHuang`	`Oh... :-) Great, where can I read or find sample on this?`
`08:37.18`	`Strom_C`	`http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1512.html`
`08:37.21`	`Strom_C`	`~docs`
`08:37.23`	`jbot`	`methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf 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`
`08:37.23`	`Strom_C`	`~book`
`08:37.24`	`jbot`	`well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`08:38.10`	`DHuang`	`Excellent.... I always those Channel: can only be ZAP`
`08:38.15`	`DHuang`	`thought`
`08:38.26`	`Strom_C`	`obviously you have read very little documentation :)`
`08:38.40`	`DHuang`	`:-)`
`08:39.17`	`DHuang`	`Extension is the 2nd number to dial... I've tried SIP but didn't work?`
`08:39.49`	`Strom_C`	`SIP is not an extension`
`08:40.07`	`Strom_C`	`do you understand the difference between an extension and the Dial application?`
`08:40.25`	`DHuang`	`Yes.`
`08:40.35`	`Strom_C`	`then look closely at that page`
`08:42.14`	`RoyK[at]`	`morning`
`08:42.17`	`DHuang`	`got ya.. call files specify an channel to call and an extension or application to connect with the called channel... so use the application to make the 2nd call..`
`08:42.25`	`Strom_C`	`bingo`
`08:42.52`	`DHuang`	`Thanks again.. :-) good to have someone hitting my head...`
`08:43.06`	`Strom_C`	`better to force you to figure it out than to hold your hand`
`08:43.16`	`Strom_C`	`makes you a better asterisk admin ;)`
`08:43.37`	`DHuang`	`<PROTECTED>`
`08:45.27`	`*** join/#asterisk canatella (n=dam@bigmonk.cosinux.org)`
`08:45.33`	`*** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr)`
`08:45.33`	`canatella`	`hello`
`08:45.49`	`Strom_C`	`good afternoon`
`08:47.38`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`08:47.54`	`canatella`	`I own a virtual server. I would like to have two sip address for my domain name (which is pointing to my server) and be able to register on that server from anywhere on the web to be able to call/receive sip calls. Can asterisk do something for me ?`
`08:48.18`	`Strom_C`	`ok, what?`
`08:49.33`	`DHuang`	`Asterisk can do that.`
`08:50.24`	`canatella`	`DHuang: for two address, do I need a powerfull server or my small xen virtual server will do ?`
`08:50.36`	`Strom_C`	`I'm still attempting to resolve just what the hell that means, although it's 2 in the morning here and I'm probably in no state of mind to start picking apart implied and express meaning`
`08:50.58`	`drray`	`If you mean two ip addresses?`
`08:51.01`	`DHuang`	`cannatella: depends how many simultaneous SIP connections you going to have?`
`08:51.02`	`*** join/#asterisk reza_ (i=reza@abort.boom.net)`
`08:51.04`	`drray`	`then asterisk does not care`
`08:51.08`	`canatella`	`DHuang: max 2`
`08:51.15`	`reza_`	`hey - anyone know if a cheap toll free did for the us?`
`08:51.20`	`DHuang`	`Sotrm: go to sleep...`
`08:51.32`	`canatella`	`drray: no two sip user address, for me and my wife ;)`
`08:52.56`	`DHuang`	`cannatella: can you explain more in detail what you are trying to achieve?`
`08:54.22`	`canatella`	`so I own the domain cosinux.org which points to a virtual server running debian. I would like to have two sip address for that domain say foo@cosinux.org and bar@cosinux.org`
`08:54.43`	`Strom_C`	`that's easy`
`08:54.48`	`canatella`	`I would like to be able to make and receive sip call from home or my work to these address`
`08:54.48`	`DHuang`	`;-)`
`08:55.03`	`Strom_C`	`canatella: that's ridiculously easy`
`08:55.11`	`Strom_C`	`in fact, it's easier than pie`
`08:55.12`	`DHuang`	`ya.. no need asterisk to do that`
`08:55.26`	`canatella`	`Strom_C: I've not said its not easy =P I just don't know where to start`
`08:55.32`	`Strom_C`	`~book`
`08:55.34`	`jbot`	`book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`08:55.34`	`Strom_C`	`~docs`
`08:55.35`	`jbot`	`i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf 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`
`08:55.41`	`DHuang`	`heehe..`
`08:55.45`	`Strom_C`	`does that work? :)`
`08:57.23`	`canatella`	`I've read it but not deep enough (I hope this is correct english) :P`
`08:57.40`	`Strom_C`	`well, read more deeply then :)`
`08:57.42`	`canatella`	`I'll reopen it then ;)`
`08:58.16`	`SheriF_WorK`	`Jul 4 12:06:18 WARNING[11064]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/130-6a31 compatible with SIP/20026-f1cf <-- any idea why ? both ends should use ULAW .. one is multitech device the other one is xlite softphone`
`08:58.45`	`Strom_C`	`are you sure they're both using ulaw?`
`08:59.19`	`DHuang`	`try "sip debug" and see the codec`
`08:59.47`	`SheriF_WorK`	`Strom_C: yes i only use ulaw in xlite and the device i use ulaw and also in sip.conf / disallow = all and allow = ulaw`
`09:00.19`	`Strom_C`	`pastebin your entire sip.conf`
`09:00.27`	`Strom_C`	`and also do a sip debug like DHuang said`
`09:00.36`	`DHuang`	`;-)`
`09:01.03`	`SheriF_WorK`	`ok 1 min`
`09:02.24`	`*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)`
`09:03.39`	`*** join/#asterisk Modcuts (n=bob@lan.proporta.com)`
`09:11.36`	`*** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net)`
`09:11.38`	`canatella`	`thx bye =P`
`09:11.41`	`*** part/#asterisk canatella (n=dam@bigmonk.cosinux.org)`
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`09:18.05`	`Strom_C`	`For the laaaaand of the cheeeeeeeeeeeeese in the caaaaaaan that you spraaaaaaaaaaay`
`09:20.14`	`DHuang`	`how to pass ARG in the .call files for the application? Application: appps ${Arg1} ${Arg2}`
`09:20.43`	`Strom_C`	`you dont`
`09:20.46`	`Strom_C`	`you use your custom script to generate the call file`
`09:21.00`	`DHuang`	`I see.. AGI file`
`09:21.08`	`Strom_C`	`no`
`09:21.57`	`mitcheloc`	`DHuang, the best way is to connect it to an extension in your extensions.conf file and have all your commands handled there`
`09:22.22`	`DHuang`	`mitcheloc: trying to connect to SIP connection using the .call file`
`09:23.26`	`mitcheloc`	`i don't use .call files, but ami instead...so i can't be of help regarding syntax, only concepts ;)`
`09:24.26`	`DHuang`	`mitcheloc: trying to do this http://www.jajah.com/`
`09:24.52`	`mitcheloc`	`what about it?`
`09:25.20`	`DHuang`	`You enter to numbers and it connects for you.`
`09:25.42`	`mitcheloc`	`do you mean you are trying to make your own?`
`09:25.50`	`DHuang`	`mit: yeah`
`09:26.13`	`mitcheloc`	`should be simple enough`
`09:26.47`	`DHuang`	`that's what I thought.. until I try... :-(`
`09:27.05`	`mitcheloc`	`are you a programmer?`
`09:27.12`	`DHuang`	`yes..`
`09:27.38`	`mitcheloc`	`whats the troubling part, i think you are looking at it the wrong way`
`09:28.02`	`DHuang`	`please shine some light on it..!!`
`09:28.13`	`mitcheloc`	`don't pass arg1/arg 2, there is no point`
`09:28.48`	`mitcheloc`	`in channel set it up like so: "Channel: ZAP/g1/extension" to call, and then "Extension: numbertoconnectto"`
`09:29.14`	`mitcheloc`	`and set up a context that matches any extension to connect ZAP/* to your outbound trunk`
`09:29.24`	`mitcheloc`	`(zap/* could be anything else, even another outbound trunk)`
`09:29.30`	`DHuang`	`:-) but Extensions can not make SIP out?`
`09:29.54`	`DHuang`	`Hmm... I see.. so realtime you mean..`
`09:30.07`	`mitcheloc`	`using the "Channel" and extensions.conf syntax you can do that, set up the extensions.conf to match and use the dial command to dial any channel`
`09:30.12`	`Strom_C`	`http://starspangledwtf.ytmnd.com/`
`09:30.47`	`mitcheloc`	`you are an odd one mister Strom_C`
`09:30.54`	`DHuang`	`Strom: I guess you are going to sleep`
`09:30.55`	`Strom_C`	`yes I am`
`09:33.11`	`DHuang`	`mit: Thanks.. I think I got it.. :-) Shall try it out l8r.... eg. extension 777xxxxxx = dial out xxxxxx and in the .call file you just set the number 777xxxxxx :-p`
`09:33.54`	`DHuang`	`Excellent.... can't wait to try it out later`
`09:33.56`	`mitcheloc`	`something like that ;)`
`09:34.40`	`DHuang`	`this should keep me busy tonight... :-) Okay Thanks again...`
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`09:50.14`	`stoffell`	`when dialing invalid (or busy) numbers on a zap channel, how can i make sure the SIP clients of the asterisk can hear the telco error? (and not play circuits are busy for example)`
`09:50.34`	`Strom_C`	`FXO, T1, or PRI?`
`09:51.29`	`stoffell`	`it's BRI, does this have to do with priindication ?`
`09:51.48`	`Strom_C`	`I have no experience with BRI`
`09:52.19`	`stoffell`	`guess it's the same as for PRI, at least the 'issue' is.. (I have 1 BRI and 1 PRI card in 2 different servers)`
`09:54.35`	`stoffell`	`you know how it's handled on an E1 (T1) ?`
`09:56.14`	`*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)`
`09:56.37`	`Strom_C`	`depends on whether you're talking channelized E1 or PRI`
`09:57.27`	`*** join/#asterisk thedotedge (n=thedoted@193-201-26-189.client.telesystems.ua)`
`09:57.55`	`thedotedge`	`help`
`09:58.18`	`Strom_C`	`well, that's a detailed description of the problem indeed`
`09:58.21`	`RoyK[at]`	`~docs`
`09:58.22`	`jbot`	`hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf 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`
`09:58.22`	`stoffell`	`I get 1 E1 cable coming in, wich connects directly to the digium te110p`
`09:58.26`	`thedotedge`	`sorry`
`09:58.35`	`Strom_C`	`stoffell: but is it PRI?`
`09:58.45`	`Strom_C`	`or is it boring old channelized E1?`
`09:58.47`	`stoffell`	`Strom_C, yes, a PRI, with 30 channels`
`09:58.51`	`stoffell`	`not boring old :)`
`09:58.51`	`thedotedge`	`I'm experiencing strange behavior while placing outgoing calls from SIP phone to PSTN via Zap channels (TE410P over E1).`
`09:59.20`	`RoyK[at]`	`:)`
`09:59.44`	`thedotedge`	`Incoming calls do work fine, and outgoing calls randomly fail like that: I pick up the hook on the PSTN phone and there is silence there while * says it's still ringing and SIP phone also continues to ring. But sometimes the call go through fine. I don't do any reloads etc, just place one call and then another.`
`09:59.44`	`thedotedge`	`pri debug span 1 for successful and unsuccesfull calls showed that sometimes * doesn't receive CONNECT (7) and CONNECT ACKNOWLEDGE (15) messages`
`10:01.45`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`10:03.03`	`thedotedge`	`please take a look at http://forums.digium.com/viewtopic.php?t=7807`
`10:03.46`	`RoyK[at]`	`Strom_C: PRI or channalised E1??`
`10:04.10`	`Strom_C`	`RoyK[at]: catsex on a stick`
`10:04.24`	`RoyK[at]`	`methinks Strom_C is evil`
`10:04.30`	`Strom_C`	`to the core`
`10:04.33`	`Strom_C`	`also cheese`
`10:13.47`	`*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)`
`10:13.50`	`Bert-`	`hello there :)`
`10:14.38`	`*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)`
`10:14.45`	`RoyK[at]`	`<PROTECTED>`
`10:15.15`	`Bert-`	`I have a little question about ring tone : When I dial a number (throught SIP Trunk), asterisk provide ringtone before receiving the SIP ringing msg. Is a way to wait the ringing msg before sending ringtone to the caller ?`
`10:15.51`	`RoyK[at]`	`don't use the r flag?`
`10:15.57`	`Bert-`	`nop`
`10:16.01`	`Bert-`	`Hi Roy ;)`
`10:16.04`	`Bert-`	`let me try`
`10:16.14`	`Bert-`	`I was wondering about Dial parameter`
`10:16.30`	`Bert-`	`hmm`
`10:16.32`	`Bert-`	`in fact`
`10:16.37`	`Bert-`	`I use it : exten => s,5,Dial(SIP/${ARG1}@Nextone_OUT,60,Tr)`
`10:16.50`	`RoyK[at]`	`s/Tr/T/`
`10:16.59`	`Bert-`	`ok`
`10:20.35`	`thedotedge`	`has anyone ever experienced random outgoing call connections not being detected on Zap channels?`
`10:22.10`	`drray`	`no, that never happens`
`10:22.34`	`drray`	`er, pardon me`
`10:23.57`	`*** join/#asterisk BertZ (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr)`
`10:29.55`	`stoffell`	`thedotedge, are you using bristuff?`
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`10:47.28`	`kay2`	`If my voicemail is saved in .gsm format, and I am using g729 with my softphone. Do I have to get a license for the g729 ?`
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`11:01.35`	`*** join/#asterisk markeyGB (n=mark@194.201.148.215)`
`11:08.30`	`thedotedge`	`stoffell no, it's just libpri 1.2.3 and zaptel 1.2.6`
`11:10.15`	`*** join/#asterisk _omer (n=_omer@202.166.162.250)`
`11:10.18`	`_omer`	`hi`
`11:10.52`	`_omer`	`anyone know where I can get DBD::ODBC for RH9 ??? I need to make ASTCC (AGI script using Perl) working with Microsoft SQL 2000....`
`11:15.22`	`markeyGB`	`Hi all is anyone using modems and or fax machines out of the back of asterisk? How are you doing it? I have read about people having a lot of problems. Some people say ATA some say TDM400 some say use a channel bank such as the rhino (though I think this look suspiciously like an asterisk box itself)`
`11:15.45`	`hwt`	`all my spawn extensions exit non-zero. is that normal?`
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`11:39.46`	`*** join/#asterisk Eimann_ (i=eimann@linoa.etherkiller.de)`
`11:39.50`	`Eimann_`	`hi`
`11:40.16`	`Eimann_`	`Hmm, how can i accept all calls with SER and send them to an asterisk, if the customers ATA is offline?`
`11:41.51`	`hwt`	`what's the technical difference between SER an OpenSER?`
`11:43.39`	`*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)`
`11:46.21`	`Eimann_`	`don't know.`
`11:47.10`	`znoG`	`hi, does anyone do QoS on Linux for VoIP traffic?`
`11:49.33`	`Mw3`	`markeyGB: We're using fax machines and data modemes behind asterisk with ATA's. sometimes there are problems (fax not getting through, low modem speed (4800 baud)), but most of the time it's working well.`
`11:52.53`	`stoffell`	`markeyGB, I'm using IAXMODEM since a few days, seems to work nice..`
`11:53.01`	`stoffell`	`(better then rxfax/txfax combo)`
`11:53.03`	`hwt`	`Mw3: have you looked at virtualizing the whole thing with spandsp/(rx\|tx)fax?`
`11:53.08`	`hwt`	`that worked great here.`
`11:53.24`	`hwt`	`given that you have a controlled network environment.`
`11:53.49`	`rob0`	`~stun`
`11:53.50`	`jbot`	`it has been said that stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs`
`11:53.51`	`hwt`	`stoffell: what's the difference between iaxmodem and spandsp?`
`11:54.23`	`rob0`	`Okay, but my SIP is not NATed.`
`11:54.52`	`hwt`	`oh, it uses spandsp.`
`11:55.11`	`stoffell`	`hwt, I've had better luck using iaxmodem and hylafax, but it also uses spandsp in a way.. (but without the txfax/rxfax apps)`
`11:56.05`	`DrkShdw`	`I've had the best luck with an analog line + a real fax machine ;)`
`11:56.12`	`markeyGB`	`thanks for the opinions guys... maybe i will just give them a whirl and if its slow and buggy get some analogue lines in.`
`11:57.26`	`hwt`	`stoffell: what problems have you had with rx/txfax? they've worked perfectly for me thus far.`
`11:57.53`	`Mw3`	`hwt: yes. but the girls here doesn't like that :) they love standard fax machines :D`
`11:58.41`	`stoffell`	`hwt, mainly txfax, rxfax sometimes missed some faxes (incorrect pages, etc..) but sending faxes worked almost never.. (only 5% worked when testing it out)`
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`11:59.33`	`*** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at)`
`12:00.10`	`rob0`	`Do I need udp dpt:3478 open for SIP with FWD?`
`12:00.34`	`hwt`	`stoffell: in what topology are you using it?`
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`12:02.00`	`stoffell`	`hwt, using bristuff on a few ISDN lines..`
`12:05.36`	`RoyK[at]`	`does bristuff work with 1.2?`
`12:06.22`	`stoffell`	`RoyK[at], yes, if you use the latest, and patch it a bit to make sure you don't suffer from the hangup bug:)`
`12:07.45`	`*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)`
`12:07.56`	`RoyK[at]`	`stoffell: if you have the patched-up version, please upload it to asterisk-backports.org :)`
`12:08.20`	`hwt`	`stoffell: ok, i'm using it in a pure voip-network.`
`12:08.25`	`hwt`	`stoffell: closed network, that is.`
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`12:13.09`	`*** join/#asterisk __undef (n=jochum@213.30.245.34)`
`12:13.12`	`__undef`	`hi`
`12:13.32`	`madikonda2`	`please point correct channell for astcc..`
`12:13.33`	`__undef`	`can anyone tell me how i can get rid of "zaphfc[0]: b channel buffer overflow: xxx, xxx"?`
`12:13.47`	`__undef`	`assigning irqs to the hfc cards didn't help`
`12:14.31`	`__undef`	`one card is on irq5, the other one on irq7`
`12:14.50`	`__undef`	`and zaphfc was compiled with florz' patches`
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`12:16.35`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`12:17.18`	`rob0`	`sigh ... Jul 4 07:15:10 miniluv postfix/smtpd[25477]: warning: Illegal address syntax from mail.pulver.com[192.246.69.184] in MAIL command: <forum-no-reply@freeworlddialup.com "fwd user forums">`
`12:18.15`	`madikonda2`	`astcc is not saving cdrs, what is the configuration to enable cdrs?`
`12:18.19`	`rob0`	`I can't register at the FWD forums. :(`
`12:18.42`	`kay2`	`is the video Grandstream phone a good phone ?`
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`12:25.44`	`Dr-Linux`	`hi`
`12:26.04`	`Dr-Linux`	`anybody knows about SCCP?`
`12:26.20`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`12:32.54`	`RoyK[at]`	`~sccp`
`12:33.03`	`jbot`	`sccp is, like, Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database.`
`12:33.38`	`RoyK[at]`	`Dr-Linux: doing ss7?`
`12:34.20`	`*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)`
`12:34.31`	`Dr-Linux`	`RoyK[at]: actually, i have a Cisco 7935 conference phone, tht only supports SCCP, so i just configured all on asterisk server.`
`12:34.43`	`Dr-Linux`	`everything works fine, but a few problems`
`12:35.46`	`potsboy`	`i hate to ask the obvious... err * what problems *`
`12:35.58`	`hwt`	`Dr-Linux: i think you can firmware-upgrade that one to support SIP.`
`12:36.44`	`Dr-Linux`	`hwt: as i said, this device only support SCCP`
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`12:36.59`	`Dr-Linux`	`hwt: but if you are clear with your statement, then please help me`
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`12:39.48`	`telenieko`	`Hi ppl. I have a queue with some people with some penalties (some of that people is with more than one penalty) it was working fine until I upgraded to 1.2.9.1, now everybody that was repeated on more than one penalty is only on the last one. any clue?`
`12:40.21`	`Dr-Linux`	`what's SCCP new version?`
`12:47.02`	`tehmaze`	`in my tcpdumps I see a lot of INVITE messages, but asterisk seems to just ignore them, I got a section [username] in my sip.conf and a context+extension for it, still it seems to ignore incoming calls .. any tips?`
`12:53.44`	`*** join/#asterisk acehunky (n=chat_jok@59.184.29.221)`
`12:54.39`	`RoyK[at]`	`~skinny`
`12:54.40`	`jbot`	`somebody said skinny was a common name for SCCP, the VoIP protocol used by many Cisco phones, or what people look like when they put computing above eating`
`12:55.59`	`znoG`	`if I have a Dial command with tTL(3600000), why would a call NOT get hanged up by Asterisk?`
`12:56.12`	`znoG`	`they normally get dropped by Asterisk, but this one got away somehow`
`12:56.14`	`znoG`	`18 hours and counting`
`12:56.50`	`znoG`	`should I bet setting an AbsoluteTimeout or something?`
`12:57.18`	`znoG`	`instead of limiting the call using L(X) ?`
`12:58.01`	`*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)`
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`13:06.51`	`*** part/#asterisk potsboy (n=chrisg@196.34.241.242)`
`13:09.39`	`yxa`	`how do I strip certain digits of a number {VARIABLE} ?`
`13:10.17`	`[TK]D-Fender`	`yxa : Can you give an example on exactly what kind of stipping you're looking to do.`
`13:11.17`	`yxa`	`ok. after stripping weird characters after using FILTER, I am left with the country and area code I want to strip, leaving only a local number`
`13:11.25`	`MooingLemur`	`${VARIABLE:3} = returns fourth position from the start, through to the end. ${VARIABLE:0:3} = returns from first position, with the length of 3.`
`13:12.21`	`yxa`	`cool. I thought I can only do that with {EXTEN}`
`13:12.31`	`[TK]D-Fender`	`yxa : Nope, any var`
`13:14.26`	`*** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net)`
`13:16.31`	`madikonda2`	`anyone know about astcc?`
`13:17.51`	`viperdude`	`hi all, bit off topic I know but does anyone know how to disable the settings button on a cisco 7940 with the sip image on it?`
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`13:22.04`	`yxa`	`can I specify a call file to not dial out immediately?`
`13:22.26`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`13:22.51`	`madikonda2`	`~astcc`
`13:22.57`	`jbot`	`from memory, astcc is the asterisk calling card platform. There have been patches so that now you can use it in either a pre-pay or post-pay model. You can find more information about it on the wiki (www.voip-info.org)`
`13:23.39`	`thedotedge`	`yxa just touch it to future date`
`13:24.33`	`MRH2`	`yxa then move it to the sppol directory`
`13:24.41`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`13:25.08`	`MRH2`	`*spool`
`13:25.59`	`*** join/#asterisk postel (n=jp@unaffiliated/postel)`
`13:26.57`	`*** join/#asterisk trym (n=trym@062249179047.customer.alfanett.no)`
`13:27.15`	`[TK]D-Fender`	`yxa : You can either cron the move of the call file into the folder or you can set the file date to the time/date you want it to execute.`
`13:27.27`	`trym`	`i want to execute a shell command of some sort that makes asterisk call a number and put the call in a certain extension`
`13:27.33`	`trym`	`how can I achieve this?`
`13:28.18`	`yxa`	`trym we are talking abt it. call files.`
`13:28.25`	`trym`	`nice`
`13:28.31`	`yxa`	`http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out`
`13:29.14`	`yxa`	`[TK]D-Fender what's the syntax of touch that does currenttime + x secs?`
`13:30.38`	`Dr-Linux`	`anybody knows about SCCP?`
`13:30.49`	`*** part/#asterisk madikonda2 (n=madikond@60-240-21-153-nsw-pppoe.tpgi.com.au)`
`13:30.55`	`yxa`	`Dr-Linux www.chan-sccp.org`
`13:31.42`	`Dr-Linux`	`yxa: i've configured everything on my asterisk box for one of my Cisco 7940, but i'm having some problems so need help.`
`13:32.20`	`yxa`	`Dr-Linux no in-depth knowledge. sorry`
`13:33.25`	`Dr-Linux`	`yxa: when i call outside, i can't hear other end, and i hear very strange ring ... that's going to other party`
`13:33.55`	`*** join/#asterisk Egonis (n=Egonis@207.245.14.10)`
`13:34.17`	`*** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy)`
`13:34.22`	`Egonis`	`i just made my own music on hold mp3, which bitrate, etc should it be in? it just crashed mpg123 and asterisk subsequently when I tried to play it back`
`13:34.56`	`znoG`	`[TK]D-Fender: for setting the maximum time of a call, should I be using L(x) in the Dial command or the AbsoluteTimeout setting?`
`13:35.06`	`[TK]D-Fender`	`yxa : no clue, I suck at linux.`
`13:35.07`	`l-fy`	`hello guys`
`13:35.25`	`[TK]D-Fender`	`znoG : no idea`
`13:35.42`	`[TK]D-Fender`	`Dr-Linux : Wait till Qwell shows up.`
`13:35.50`	`l-fy`	`my asterisk can carry only 250 calls with iax without trunking`
`13:35.58`	`l-fy`	`how can i increase the number of calls?`
`13:36.15`	`[TK]D-Fender`	`l-fy : Well... try trunking :)`
`13:36.22`	`Dr-Linux`	`[TK]D-Fender: yeah, i can't find help even on google :S`
`13:36.29`	`trym`	`call files worked like a charm`
`13:36.37`	`l-fy`	`[TK]D-Fender > in my setup trunking is useless`
`13:36.43`	`yxa`	`znoG L(x) should do the job`
`13:37.00`	`l-fy`	`so any idea how can i increase the number of iax calls`
`13:37.06`	`[TK]D-Fender`	`Egonis : I'd suggest 128kbit and maybe use Native MoH and not MPG123`
`13:37.09`	`yxa`	`znoG but becareful it is in milliseconds`
`13:37.30`	`[TK]D-Fender`	`l-fy : Whats the actualy proble with it? Connection limit or CPU?`
`13:37.51`	`l-fy`	`[TK]D-Fender > well, it starts to lose frames, i guess`
`13:37.52`	`l-fy`	`so......`
`13:38.01`	`l-fy`	`the connection is enough`
`13:38.19`	`yxa`	`l-fy there's only so many transcoding a cpu can do...`
`13:38.34`	`l-fy`	`yxa > i don't do transcoding`
`13:38.44`	`l-fy`	`everything else there except iax is irevelant`
`13:39.52`	`[TK]D-Fender`	`l-fy : Maybe a timing issue.... can't advise any more on this unfortunately.`
`13:40.57`	`yxa`	`l-fy what codec?`
`13:41.15`	`l-fy`	`yxa > g711`
`13:41.25`	`l-fy`	`[TK]D-Fender > it dosen't seems to`
`13:44.31`	`*** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net)`
`13:45.51`	`*** join/#asterisk bernardovieira (n=bvieira@c911935d.static.bhz.virtua.com.br)`
`13:56.02`	`*** part/#asterisk Egonis (n=Egonis@207.245.14.10)`
`13:58.41`	`*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`14:00.32`	`Skarmeth`	`hi all`
`14:01.28`	`Skarmeth`	`it's a normal behavior when your run pri debug span 1 and pri intense debug span 1 at the moment your receive a call, the asterisk server goes down?`
`14:02.06`	`tzanger`	`Skarmeth: nope`
`14:02.15`	`tzanger`	`you typically don't need intense debugging`
`14:02.23`	`tzanger`	`but it should not crash`
`14:08.16`	`*** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR)`
`14:08.36`	`nXOR`	`hello ppl, i have a question about visdn, is there a dedicated channel for it or can i ask here ?`
`14:08.58`	`*** join/#asterisk doolph (n=doolph@200.75.196.182)`
`14:09.36`	`doolph`	`anyone know why my X100P don't hungup inmediatly when the party hungup?`
`14:09.44`	`nXOR`	`channel full of sleeping people .......`
`14:14.13`	`*** join/#asterisk ionix (n=ionix@p1200-ipbfp05miyazaki.miyazaki.ocn.ne.jp)`
`14:14.21`	`tzanger`	`doolph: because your telco isn't supplying CPD, or the card does not recognize your telco's brand of CPD`
`14:14.40`	`doolph`	`CPD?`
`14:15.07`	`tzanger`	`called/calling party disconnect`
`14:16.02`	`tzanger`	`basically some way for the telco to tell your phone that the other side hung up on you. not all telcos support it. generally it's a momentary battery drop or polarity reversal before you hear dialtone again`
`14:16.09`	`tzanger`	`~cpd`
`14:20.57`	`nXOR`	`who can answer me some asterisk/visdn related questions please`
`14:21.06`	`nXOR`	`or point me to a channel that can`
`14:22.54`	`*** join/#asterisk LeXo (n=lexo@dsl-200-95-119-243.prod-infinitum.com.mx)`
`14:28.21`	`BertZ`	`hmm`
`14:28.34`	`BertZ`	`what is the best digital card to buy for 4 isdn lines please ?`
`14:28.42`	`BertZ`	`does I need a DSP module ?`
`14:29.11`	`BertZ`	`In France, we call this kind of lines T0 (a box wich provide two phone lines)`
`14:29.22`	`BertZ`	`but I don't knwo the english equivalent :(`
`14:32.21`	`BertZ`	`seems that digium card are standard euro compliant ;=)`
`14:32.46`	`*** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net)`
`14:33.52`	`RoyK[at]`	`BertZ: they work well in europe, yse`
`14:33.53`	`RoyK[at]`	`yes`
`14:34.05`	`BertZ`	`okay`
`14:34.06`	`RoyK[at]`	`perhaps even in france :)`
`14:34.10`	`BertZ`	`:p`
`14:34.15`	`BertZ`	`it must`
`14:34.16`	`BertZ`	`anyway`
`14:34.24`	`RoyK[at]`	`we use digium cards in .no`
`14:34.26`	`RoyK[at]`	`that is`
`14:34.32`	`RoyK[at]`	`right now we only use sangomas`
`14:34.44`	`BertZ`	`As I can see (I'm really not an expert), cards are capale to handle hundreds channels`
`14:34.56`	`gaupe`	`RoyK[at]: not BRI I guess :)`
`14:35.03`	`BertZ`	`what about one for just 4 digital lines ?`
`14:35.04`	`RoyK[at]`	`a four-port card can obviously handle 120 channels`
`14:35.12`	`BertZ`	`yep`
`14:35.20`	`BertZ`	`I will have only 4 channels`
`14:35.25`	`RoyK[at]`	`or 124 if you do ss7`
`14:35.30`	`_4d4m_`	`I've been playing with variations of what is stated on the wiki regarding the i extension (and more specifically how to set-up a catchall for all unrecognised numbers dialled by my users), but I'm not having any success. Anyone know of a place that will show/explain how to create such a catch-all?`
`14:35.30`	`BertZ`	`so I mean 4 channels, not 4 ports sorry :)`
`14:35.38`	`RoyK[at]`	`4 PRIs?`
`14:35.42`	`BertZ`	`yep`
`14:35.45`	`RoyK[at]`	`get a good box anyway`
`14:35.51`	`RoyK[at]`	`or boards with echocancel`
`14:35.57`	`RoyK[at]`	`the latter is best, of course`
`14:36.16`	`BertZ`	`Pentium D 3GHz 512 DDRAM 80GO SATA. Is that ok ?`
`14:36.19`	`RoyK[at]`	`more expensive, but then you can prolly run eight or sixteen PRIs in a box`
`14:36.43`	`RoyK[at]`	`the CPU might not handle 120 echocancel`
`14:36.46`	`RoyK[at]`	`dunno`
`14:36.47`	`RoyK[at]`	`test it`
`14:36.51`	`BertZ`	`actually`
`14:37.03`	`BertZ`	`I use asterisk with 4 concurrent calls, with transcoding`
`14:37.07`	`RoyK[at]`	`two crossed E1s`
`14:37.11`	`RoyK[at]`	`four calls :)`
`14:37.13`	`BertZ`	`on a P III 500 MHZ and 192 SDRAM`
`14:37.13`	`RoyK[at]`	`YES`
`14:37.19`	`RoyK[at]`	`but that is four calls`
`14:37.20`	`[TK]D-Fender`	`RoyK[at] : That's why you buy the 8-Port EC card from Sangoma :)`
`14:37.22`	`BertZ`	`It work`
`14:37.24`	`BertZ`	`fine`
`14:37.31`	`BertZ`	`I don''t need more`
`14:37.32`	`rob0`	`Hahaha ... messing around with FWD, I can get it working with IAX2 but not SIP. And I called their time extension, 612 ... the clock is off by 6 minutes!`
`14:37.41`	`websae`	`TK: i have that card!`
`14:37.53`	`rob0`	`Would that perhaps cause call failures?`
`14:37.54`	`RoyK[at]`	`[TK]D-Fender: I know, but it doesn't work yet with amd64 :(`
`14:38.02`	`RoyK[at]`	`[TK]D-Fender: the sangoma guys are really slow there`
`14:38.44`	`BertZ`	`Well clearly, we will have 5 outbound concurrent calls MAX, all through SIP account and 4 concurrent inbound, all from pstn.`
`14:38.47`	`[TK]D-Fender`	`RoyK[at] : Only in 64 bit mode maybe... then again at higher densities like this most people looks for a gateway device like AudioCodes anyways`
`14:38.56`	`*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)`
`14:39.04`	`BertZ`	`We haeonly 4 isdn lines`
`14:39.19`	`BertZ`	`and I don't know which PRI card is the best for us`
`14:39.31`	`*** join/#asterisk postel (n=jp@unaffiliated/postel)`
`14:39.53`	`*** join/#asterisk clive- (n=pirch@dsl-165-169-163.telkomadsl.co.za)`
`14:40.39`	`RoyK[at]`	`[TK]D-Fender: still it is _really_ annoying they haven't fixed it.`
`14:41.19`	`[TK]D-Fender`	`RoyK[at] : There are worse things....`
`14:41.43`	`RoyK[at]`	`well, i haven't heard from them in a month`
`14:41.48`	`RoyK[at]`	`and i think that is quite bad`
`14:42.36`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`14:45.36`	`xheliox`	`Anyone have any experience with the Sangoma A200?`
`14:46.52`	`*** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)`
`14:47.05`	`_4d4m_`	`i'm trying to create a catchall for any extension that a user dials that is not matched in the dialplan for that context. I've followed suggestions on the wiki, but it does not work for me --> http://pastebin.ca/78743`
`14:47.12`	`[TK]D-Fender`	`xheliox : I have. What's up?`
`14:47.17`	`_4d4m_`	`the catch-all catches everything (as i thought it would to be honest)`
`14:47.22`	`xheliox`	`[TK]D-Fender: My hero, I was hoping you were around. :)`
`14:47.48`	`xheliox`	`[TK]D-Fender: I bought one, just to play, and it won't recognize my FXS port. The driver loads, sees the card, but fails.`
`14:48.05`	`xheliox`	`Let me pastebin the error.`
`14:48.20`	`_4d4m_`	`anyone any hints or tips?`
`14:49.13`	`[TK]D-Fender`	`_4d4m_ : Your Goto is wrong. read up on its syntax`
`14:49.49`	`xheliox`	`[TK]D-Fender: http://pastebin.ca/78746`
`14:49.51`	`[TK]D-Fender`	`_4d4m_ : And for the way you set it up, you're better off just INCLUDE-ing the context`
`14:50.31`	`[TK]D-Fender`	`xheliox : Sure its all assembled properly?`
`14:50.37`	`*** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do)`
`14:50.45`	`xheliox`	`As sure as I can be.`
`14:50.47`	`*** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do)`
`14:51.28`	`*** join/#asterisk djulius (i=danj@bzq-88-155-230-183.red.bezeqint.net)`
`14:51.42`	`[TK]D-Fender`	`xheliox : pastebin your zaptel / zapata / wanpipe1`
`14:52.03`	`*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)`
`14:52.08`	`xheliox`	`Stand by.`
`14:52.57`	`_4d4m_`	`[TK]D-Fender: the goto got screwed up in pasting.. http://pastebin.ca/78748 is what i've tried and is not working`
`14:53.07`	`_4d4m_`	`is that still wrong context?`
`14:53.17`	`_4d4m_`	`sip debug shows it is getting passed to the right context and file is played`
`14:53.23`	`_4d4m_`	`but played no matter what i enter`
`14:53.33`	`[TK]D-Fender`	`_4d4m_ : Your syntax is STILL wrong. Go read the INSTRUCTIONS -> "show application goto"`
`14:54.55`	`xheliox`	`[TK]D-Fender: http://pastebin.ca/78750`
`14:56.48`	`_4d4m_`	`[TK]D-Fender: d'oh.. thanks`
`14:58.03`	`[TK]D-Fender`	`xheliox : you are defining 2 channels in zapata, but only 1 in zaptel. What does "wanrouter status" say?`
`14:58.27`	`*** join/#asterisk magic_1 (n=samuraiq@wbs-196-2-99-243.wbs.co.za)`
`14:58.28`	`xheliox`	`Yeah, my mistake, I changed that earlier, just to test to see if maybe there was a problem with the second module...`
`14:58.57`	`xheliox`	`Find if I c/p it to you in /msg?`
`14:59.16`	`BertZ`	`so nobody use Asterisk for only some calls, as an IVR, for a little company ?`
`14:59.30`	`xheliox`	`Mind*`
`14:59.30`	`BertZ`	`you guys work all in big companies which need 120 channels ?`
`14:59.39`	`BertZ`	`difficult to believe ;)`
`14:59.54`	`xheliox`	`[TK]D-Fender: http://pastebin.ca/78752`
`15:01.36`	`*** join/#asterisk iq\|mobile (n=iq@unaffiliated/iq)`
`15:02.09`	`[TK]D-Fender`	`xheliox : Doesn't look up. do wanrouter start`
`15:02.56`	`xheliox`	`http://pastebin.ca/78755`
`15:02.57`	`BertZ`	`TE110P should be ok`
`15:03.05`	`xheliox`	`And in /var/log/messages is the same error I pasted you to begin with..`
`15:03.17`	`*** join/#asterisk Egonis (n=Egonis@207.245.14.10)`
`15:03.53`	`Egonis`	`I am getting no outbound audio on an outside phone, which registers to an asterisk server on a direct net ip -- any obvious things I am forgetting?`
`15:04.25`	`*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)`
`15:05.43`	`_4d4m_`	`[TK]D-Fender: Goto sorted. catch-all catches all, but I want a 'catch-all-but-the-valid-extensions-i've-defined' - http://pastebin.ca/78753. any hints?`
`15:06.53`	`[TK]D-Fender`	`_4d4m_ : stop using that goto at all and just INCLUDE it.`
`15:10.55`	`[TK]D-Fender`	`xheliox : Hmmm... try the wanrouter card autodetect (don't recall the parm for it)`
`15:11.11`	`*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`15:11.31`	`[TK]D-Fender`	`xheliox : Though something seems off in your wanpipe config if you're getting : 22 - Invalid argument`
`15:11.43`	`*** join/#asterisk salviadud (n=ralfalfa@201.135.2.210)`
`15:12.04`	`xheliox`	`I created it with "wancfg zaptel"`
`15:12.21`	`[TK]D-Fender`	`xheliox : mind you there is a hardware concern getting this error earlier : Jul 3 21:02:23 quagmire kernel: wanpipe1: No FXO/FXS modules are found!`
`15:12.36`	`[TK]D-Fender`	`xheliox : Never used that method before..`
`15:13.05`	`xheliox`	`Do you have a sample I could just test with?`
`15:14.59`	`__undef`	`does anyone have asterisk running with 3 hfc cards?`
`15:15.18`	`__undef`	`one works fine, but with three i get buffer overruns and underruns after a few seconds`
`15:15.56`	`[TK]D-Fender`	`xheliox : Not offhand, sorry...`
`15:16.02`	`xheliox`	`Okie dokie.`
`15:16.22`	`[TK]D-Fender`	`xheliox : Try rebuilding your wancfg setup from scratch.`
`15:17.20`	`eKo1`	`Man, liquidating CDR information is the suX0rz`
`15:20.02`	`*** part/#asterisk Egonis (n=Egonis@207.245.14.10)`
`15:20.33`	`xheliox`	`[TK]D-Fender: Same. :(`
`15:20.39`	`*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)`
`15:20.50`	`xheliox`	`is there an easy way to verify for sure this is a pci 2.2 system?`
`15:20.58`	`[TK]D-Fender`	`xheliox : no ida.`
`15:21.00`	`[TK]D-Fender`	`idea*`
`15:21.20`	`xheliox`	`I'm 99% sure it is`
`15:23.03`	`*** part/#asterisk clive- (n=pirch@dsl-165-169-163.telkomadsl.co.za)`
`15:24.56`	`*** join/#asterisk boch (n=root@201.216.241.97)`
`15:24.59`	`boch`	`hello`
`15:26.46`	`Sonderblade`	`on http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone there is a tip for how to reboot a phone using curl, have anyone used that tip and/or know how to get it to work?`
`15:27.36`	`rob0`	`What ports do I need open for SIP? 5060/udp, or others too? (No NAT involved, FWIW.)`
`15:28.29`	`eKo1`	`That is all you need for SIP.`
`15:28.34`	`*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)`
`15:28.45`	`rob0`	`What are the 10000:20000 ports I see mentioned?`
`15:28.50`	`eKo1`	`RTP`
`15:28.55`	`[TK]D-Fender`	`rob0 : You need 5060 for SIP, and typically 10000-20000 for RTP`
`15:29.16`	`rob0`	`ah! So that could be by SIP trouble!`
`15:29.43`	`xheliox`	`[TK]D-Fender: I'm thinking I have a bum card, I just threw in another system.`
`15:29.49`	`[TK]D-Fender`	`rob0 : You'll need to set either EXTERNIP or EXTERNHOST + EXTERNREFRESH, nat=yes, and LOCALNET in [general] in sip.conf`
`15:30.08`	`[TK]D-Fender`	`xheliox : Try swapping the modules around.`
`15:30.23`	`xheliox`	`[TK]D-Fender: Did that last night.`
`15:30.49`	`rob0`	`I'm not NATing ... the * host is directly connected to the 'Net.`
`15:31.55`	`[TK]D-Fender`	`xheliox : wELL IT COULD EB A BUM CARD.. TRIED CALLING THEM UP?`
`15:32.19`	`[TK]D-Fender`	`rob0 : So * is PUBLIC and you have a NAT'd CLIENT?`
`15:32.52`	`xheliox`	`[TK]D-Fender: Yeah, they're on holiday today, I'm going to try tomorrow.`
`15:33.22`	`rob0`	`* is public, there's no SIP client. I was trying to set up inbound calls from a remote DID (ipkall.com this time.)`
`15:33.34`	`*** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net)`
`15:35.12`	`[TK]D-Fender`	`rob0 : then where is NAT involved?`
`15:37.49`	`rob0`	`um, I said none ... "(No NAT involved, FWIW.)" It's just that most of the instructions at the Wiki seem to assume NAT.`
`15:38.22`	`[TK]D-Fender`	`rob0 : Ok, so you should have no issues then... ignore everything related to it.`
`15:41.14`	`rob0`	`thanks`
`15:42.57`	`*** join/#asterisk ctaloi (n=Chris@nat-66-218-1-182.usadatanet.com)`
`15:44.04`	`boch`	`gentlemen, could you helpme understanding what is happening in my asterisk when i try to make a sip call, i can paste you the signaling`
`15:44.18`	`ctaloi`	`boch - i can try`
`15:44.22`	`ctaloi`	`what've you go`
`15:44.24`	`ctaloi`	`got`
`15:45.31`	`*** join/#asterisk DarKnesS_WolF (n=wolf@82.201.197.130)`
`15:45.45`	`*** join/#asterisk s0lid (n=s0lid@203.177.12.98)`
`15:47.01`	`ctaloi`	`anyone have any suggestions on a voicemail user interface that will allow moves adds and changes?`
`15:47.20`	`boch`	`ctaloi: this is the signaling since i pick up the phone and dial until hangup http://pastebin.ca/78794`
`15:47.31`	`*** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr)`
`15:48.18`	`ctaloi`	`boch - can you explain to me what happens when you try to make a call? fast busy, op message ?`
`15:48.22`	`*** join/#asterisk retentiveboy (n=retentiv@h189.81.40.69.ip.alltel.net)`
`15:49.03`	`*** part/#asterisk retentiveboy (n=retentiv@h189.81.40.69.ip.alltel.net)`
`15:49.10`	`boch`	`ctaloi: fast busy after almost 10 secs`
`15:49.57`	`ctaloi`	`extensions '12' and '123' are internal SIP phones on your LAN?`
`15:51.37`	`boch`	`12 is the peer, an ata186, and in its context there is the extension: exten => _X.,1,NoOp(ok!!)`
`15:52.56`	`boch`	`the point is, NoOp is not being exec`
`15:53.14`	`ctaloi`	`you might want to try "set verbose 10" at the Asterisk CLI - I see you've got SIP debugging going, but the Asterisk verbose or Debug will give you a beter idea of what Asterisk is up to`
`15:53.51`	`ctaloi`	`try to following at the Ast CLI: set verbose 10 \| set debug 10`
`15:54.01`	`ctaloi`	`then tail /var/log/astersik/full`
`15:54.45`	`boch`	`right, didnt know that log, thanks`
`15:55.19`	`ctaloi`	`sure`
`15:56.07`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`16:07.24`	`*** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net)`
`16:08.58`	`*** join/#asterisk tlow (n=tlowe@bgp.terrorist.net)`
`16:10.29`	`*** part/#asterisk fenlander (n=fenlande@82.152.81.57)`
`16:18.45`	`*** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`16:31.43`	`*** join/#asterisk linlin (n=linlin@c-67-184-152-231.hsd1.il.comcast.net)`
`16:32.56`	`*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)`
`16:34.14`	`linlin`	`im attempting to install freepbx/asterisk on my deian machine using this guide: http://powerontech.com/freepbx-on-debian.htm`
`16:34.26`	`linlin`	`when i get to the part about building the asterisk-addons package, make stops with an error`
`16:34.44`	`linlin`	`heres the output of make http://pastebin.ca/78826`
`16:34.53`	`*** join/#asterisk St1ckm4n (n=shortes9@c-71-193-166-111.hsd1.or.comcast.net)`
`16:35.33`	`St1ckm4n`	`Hello`
`16:37.10`	`St1ckm4n`	`I think one of my cronjobs is making my asterisk crash but it's happening intermittently`
`16:37.12`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`16:38.07`	`St1ckm4n`	`not every day but when it crashes it's right on the hour`
`16:38.33`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`16:39.18`	`St1ckm4n`	`I figure the culprit is either 00-makewhatiscron or loadQueueLogs but haven't been able to find any output from either that looks like there was an error`
`16:39.46`	`St1ckm4n`	`Doesn't seem like anyone is really out there today anyways so I guess I'm just venting`
`16:40.33`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`16:45.40`	`dlynes_office`	`St1ckm4n: Are you on freebsd?`
`16:46.46`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`16:47.19`	`St1ckm4n`	`no linux, We had a company build the box for us I believe it's using slackware`
`16:51.49`	`salviadud`	`i use slackware`
`16:52.12`	`salviadud`	`and like it`
`16:52.27`	`gaupe`	`I like to shoot my self in the head too`
`16:52.40`	`gaupe`	`but I'm not paying for it ;)`
`16:53.25`	`salviadud`	`are u saying slackware is like shooting yourself in the head?`
`16:53.41`	`gaupe`	`I would prefer shooting myself in the head really`
`16:53.42`	`Qwell`	`salviadud: no, shooting yourself in the head is a lot like slackware`
`16:54.13`	`salviadud`	`come on, it's not THAT hard...`
`16:54.27`	`gaupe`	`it's very hard to maintain`
`16:54.52`	`salviadud`	`depends on how you handle the packaging system`
`16:54.59`	`salviadud`	`i use checkinstall`
`16:55.05`	`dlynes_office`	`St1ckm4n: dood....slackware kicks ass`
`16:55.21`	`dlynes_office`	`Qwell: You're just jealous cause some goon is forcing you to use fedora :)`
`16:55.31`	`salviadud`	`dlynes_office, agreed`
`16:55.34`	`Qwell`	`pfft, gentoo`
`16:55.36`	`St1ckm4n`	`I really haven't developed an opinion on it yet but was listening to Qwell and gaupes discussion about it`
`16:55.55`	`dlynes_office`	`St1ckm4n: listen to salviadud and I :)`
`16:56.00`	`dlynes_office`	`St1ckm4n: but seriously`
`16:56.14`	`dlynes_office`	`St1ckm4n: I'd say there's probably at least 10% of the people in this channel using Slackware`
`16:56.18`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`16:56.32`	`dlynes_office`	`It's extremely stable...that's its main claim to fame`
`16:56.45`	`dlynes_office`	`It's other claim to fame is that it doesn't run all the latest and the greatest`
`16:56.46`	`gaupe`	`there's no fame, rather shame`
`16:56.51`	`dlynes_office`	`which is probably why it's stable`
`16:57.05`	`salviadud`	`hey, i can make slackware unstable`
`16:57.10`	`salviadud`	`just compile from source`
`16:57.12`	`dlynes_office`	`salviadud: yeah...just add rpms`
`16:57.14`	`salviadud`	`hehe`
`16:57.32`	`Strom_C`	`FWIW, I'm a big fan of debian stable - the default base installation is fairly small and apt is rather elegant`
`16:57.44`	`Strom_C`	`makes setting servers up a snap`
`16:57.46`	`St1ckm4n`	`I haven't been having very much luck with my asterisk since we purchased it, imo I think the company we bought it from did a poor job configuring it and I'm trying to learn the ropes to fix some of their mess`
`16:57.50`	`Strom_C`	`hey, alliteration`
`16:58.10`	`Strom_C`	`St1ckm4n: where are you located?`
`16:58.16`	`St1ckm4n`	`Portland, OR`
`16:58.17`	`Qwell`	`St1ckm4n: What company?`
`16:58.24`	`*** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net)`
`16:58.49`	`dlynes_office`	`St1ckm4n: try looking in your /etc/cron.hourly, /etc/cron.daily, /etc/cron.weekly and /etc/cron.monthly directories`
`16:59.07`	`St1ckm4n`	`Qwell: I don't want to throw out names, can I pm you their name?`
`16:59.09`	`dlynes_office`	`St1ckm4n: do a grep for the keyword 'asterisk' in there...see if there's some kind cron job for asterisk`
`16:59.14`	`Qwell`	`St1ckm4n: sure`
`16:59.24`	`Strom_C`	`St1ckm4n: also, just for our amusement, pastebin your extensions.conf file`
`16:59.37`	`dlynes_office`	`Strom_C: his issue he thinks is a cron job`
`16:59.42`	`Strom_C`	`ah ok`
`16:59.43`	`Qwell`	`St1ckm4n: never heard of em`
`17:00.01`	`St1ckm4n`	`I think we might of been one of their first clients`
`17:00.14`	`dlynes_office`	`St1ckm4n: they don't do any after market support?`
`17:00.15`	`St1ckm4n`	`our telecom provider recomended them to us`
`17:00.40`	`St1ckm4n`	`they do but they're expensive and after learning some of it myself I'm not that confident with them`
`17:01.09`	`St1ckm4n`	`when my asterisk crashes all calls get dropped and it seems to be right on the hour`
`17:01.11`	`Strom_C`	`how much are they charging you?`
`17:01.19`	`St1ckm4n`	`120/hr`
`17:01.49`	`Strom_C`	`thats about what I charge for consulting, usually`
`17:02.26`	`St1ckm4n`	`I know that's about standard, it just seems to take them a long time to fix something that seems trivial`
`17:02.54`	`St1ckm4n`	`plus I'm trying to learn this a little more so we can be more self sufficient, we're a small company :)`
`17:02.55`	`nassy`	`i am new to asterisk and am trying to get external calls to connect to my software sip phone on my computer. asteris 1.2.9.1 is behind a linksys router with NAT enabled. sip ports 5060-6082 and rdp ports 10000 - 20000 are forwarded to the asterisk server. i enabled sip debug on the sip channel and it looks as though i get some info from teliasip (my VoIP ISP). i can make outgoing calls ok. where can i find info on wha`
`17:03.04`	`*** join/#asterisk TeePOG (n=1234@dsl-145-143-190.telkomadsl.co.za)`
`17:03.13`	`TeePOG`	`evening`
`17:03.18`	`Strom_C`	`just out of curiosity, what are the specs on the entire system and how much did it end up costing you?`
`17:03.25`	`Dr-Linux`	`Qwell: hi,`
`17:03.25`	`[TK]D-Fender`	`nassy : pastebin your [general] section of sip.conf`
`17:03.27`	`[TK]D-Fender`	`~pb`
`17:03.30`	`jbot`	`from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/`
`17:03.42`	`Dr-Linux`	`Qwell: i was waiting for you for SCCP hints.`
`17:03.44`	`nassy`	`[TK]D-Fender: thanks. one sec`
`17:04.04`	`St1ckm4n`	`brb, I'm gonna go grab the specs`
`17:04.09`	`Strom_C`	`ok`
`17:04.47`	`Strom_C`	`yeah, looks like they're an IT firm that went "lol voip" one afternoon`
`17:06.44`	`St1ckm4n`	`Strom_C: HP Proliant DL360-G4 Server, w/Dual T1 card was $3655`
`17:07.14`	`Strom_C`	`how many phones, what kind of phones, and what did it end up costing you in consulting fees?`
`17:07.18`	`St1ckm4n`	`and for configuring it was around $6,000`
`17:07.42`	`St1ckm4n`	`we got 30 phones polycome 301 w/some POE hardware`
`17:07.48`	`St1ckm4n`	`*polycom`
`17:08.01`	`Qwell`	`so, about $15k?`
`17:08.16`	`Strom_C`	`$6000?! yow. does the system have to do anything particularly crazy?`
`17:08.17`	`St1ckm4n`	`yeah $17 was the total`
`17:08.27`	`St1ckm4n`	`we run a call center`
`17:08.46`	`St1ckm4n`	`about 30 agents taking roughtly 400 calls a day`
`17:08.49`	`Strom_C`	`so standard queues / agents stuff? or is it outbound`
`17:08.51`	`Qwell`	`yikes`
`17:08.59`	`[TK]D-Fender`	`St1ckm4n : thats really high...`
`17:09.01`	`*** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr)`
`17:09.27`	`nassy`	`aout hw many simultaneous cal St1ckm4n`
`17:09.29`	`St1ckm4n`	`I'm starting to realize that now`
`17:09.33`	`nassy`	`about how`
`17:09.49`	`St1ckm4n`	`we are only using one pri so at the most I've seen we've used 14-18 channels`
`17:09.50`	`nassy`	`pastebin seems to be responding slowly`
`17:10.00`	`Strom_C`	`try pastebin.ca`
`17:10.07`	`nassy`	`thanks`
`17:10.15`	`St1ckm4n`	`$17 seemed cheap after we got out of that Nortel BCM :)`
`17:10.18`	`Qwell`	`St1ckm4n: What problems are you actually having?`
`17:10.20`	`Strom_C`	`heh`
`17:10.29`	`jhiver`	`hi list`
`17:10.35`	`jhiver`	`I have this strange message on the CLI`
`17:10.43`	`jhiver`	`Jul 4 19:09:56 WARNING[99244]: channel.c:2492 ast_request: Channel H323 does not support requested formats (g729)`
`17:10.43`	`jhiver`	`Jul 4 19:09:56 NOTICE[99244]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown)`
`17:10.43`	`Qwell`	`$20 it's a warning`
`17:10.46`	`Qwell`	`ha`
`17:10.49`	`St1ckm4n`	`Qwell: asterisk consistently hangs, everyones calls will drop but everything looks normal`
`17:11.00`	`Qwell`	`St1ckm4n: oh, right, the hourly thing`
`17:11.03`	`jhiver`	`seems I can't do outbound H323 calling, don't know why`
`17:11.08`	`jhiver`	`any ideas?`
`17:11.12`	`St1ckm4n`	`logs show nothing but it always seems to happen right on the hour and the last thing I'll see in my asterisk -r window is remote unix connection`
`17:11.17`	`Qwell`	`jhiver: maybe because...it doesn't...I don't know...support g729?!`
`17:11.35`	`Qwell`	`St1ckm4n: pastebin that cdr cron`
`17:11.39`	`*** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net)`
`17:11.39`	`jhiver`	`Qwell: H323 = signaling, plus it works inbound`
`17:11.42`	`jhiver`	`so... ???????`
`17:11.53`	`jhiver`	`I'm not asking it to transcode or anything`
`17:11.53`	`Qwell`	`jhiver: it says it doesn't support it`
`17:12.07`	`nassy`	`[TK]D-Fender: http://pastebin.ca/78862`
`17:12.07`	`St1ckm4n`	`Qwell: which cron did you want me to paste? loadQueueLogs?`
`17:12.13`	`Qwell`	`St1ckm4n: yeah, that one`
`17:12.16`	`[TK]D-Fender`	`St1ckm4n : You sound like you were in the same setup I was. We're an all-Polycom + * setup now.`
`17:12.17`	`*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)`
`17:12.22`	`jhiver`	`so it doesn't support it outbound but works inbound? what kind of garbage is that?`
`17:12.40`	`Strom_C`	`jhiver: do you have g729 license(s)?`
`17:12.49`	`jhiver`	`I just do pass-through`
`17:12.53`	`jhiver`	`so I shouldn't need it`
`17:13.18`	`St1ckm4n`	`Qwell: http://pastebin.ca/78866`
`17:13.22`	`Strom_C`	`is this the same setup and call route in both directions?`
`17:13.30`	`[TK]D-Fender`	`nassy : Well first of all I should stop at the fact you're running AMP (or leftovers at the very least). But you are missing all the normally required setting for NAT to work on *. read up on EXTERNIP / EXTERNHOST, LOCALNET, and set nat=yes in there as well.`
`17:13.48`	`jhiver`	`it's not the same route, I have an inbound route and an outbound route`
`17:13.58`	`jhiver`	`I mean, I'll try ulaw`
`17:14.05`	`jhiver`	`and see if that changes anything`
`17:14.11`	`Strom_C`	`please do`
`17:14.19`	`Strom_C`	`and then explain your different routes`
`17:14.40`	`jhiver`	`well I have one incoming (H323) and a new one outgoing (H323 too)`
`17:15.06`	`Qwell`	`St1ckm4n: try disabling that cron for a while.`
`17:15.10`	`jhiver`	`the incoming one works because I've made a test connection with my SIP phone (H323 <-> SIP translation working, hurray!)`
`17:15.17`	`Qwell`	`St1ckm4n: it seems a bit useless - especially if you have much disk space`
`17:15.25`	`Qwell`	`low volume won't fill up your logs that fast at all`
`17:15.30`	`St1ckm4n`	`I need that cron, for my queue metrics reporting`
`17:15.41`	`jhiver`	`now the outbound route was /supposed/ to be SIP but then we had a "no ring tone" issue`
`17:15.52`	`nassy`	`[TK]D-Fender: thanks for helping. i didnt think that i had to add nat=1 if both asterisk and the software sip phone were behind the router.`
`17:16.01`	`Strom_C`	`jhiver: I asked you to try ulaw /first/`
`17:16.02`	`Qwell`	`St1ckm4n: I don't like that way it works at all`
`17:16.02`	`St1ckm4n`	`we pull the reporting throughout the day to see call stats`
`17:16.04`	`jhiver`	`so I've asked my provider to switch from SIP to H323 to see if it fixes things, and now it doesn't work at all anymore :)`
`17:16.11`	`jhiver`	`Strom_C, I'll do that now`
`17:16.33`	`St1ckm4n`	`seems kind of redundant the way it copies and moves the files`
`17:16.45`	`Strom_C`	`surely they can just use mysql?`
`17:16.58`	`Qwell`	`St1ckm4n: exactly`
`17:17.09`	`Strom_C`	`if you need to run metrics and whatnot, that would be the easier way to do it, wouldnt it?`
`17:17.23`	`Qwell`	`Strom_C: yeah, I would agree with that`
`17:17.24`	`St1ckm4n`	`my ideal would be real time reporting but I heard that this was buggy`
`17:17.29`	`Qwell`	`especially with such a small setup`
`17:17.38`	`St1ckm4n`	`so the cron runs hourly to dump the queue logs into psql for reporting`
`17:18.37`	`file`	`so, I just bought a new hard drive to replace my one that broke an hour or two ago...`
`17:18.42`	`St1ckm4n`	`I'm glad you guys are seeing my frustration`
`17:18.44`	`file`	`and it cost $5 more to send it via ground, then overnight`
`17:19.11`	`[TK]D-Fender`	`St1ckm4n : I'm surprised that you aren't using PSQL directly for storing your queue logs...`
`17:19.26`	`Qwell`	`can queuelogs be put directly into psql?`
`17:19.31`	`Strom_C`	`St1ckm4n: just for shits and giggles, what happens when you log into the asterisk console and type "logger rotate"?`
`17:19.36`	`[TK]D-Fender`	`Qwell : Sure. ODBC`
`17:19.40`	`Qwell`	`can queuelogs be put directly into odbc?`
`17:20.16`	`[TK]D-Fender`	`BIG*`
`17:20.16`	`Qwell`	`no..really`
`17:20.20`	`Qwell`	`it's not like cdr or anything`
`17:20.36`	`Qwell`	`you don't see res_queuelogs_odbc.so`
`17:21.34`	`St1ckm4n`	`outputs:Asterisk Event Logger restarted`
`17:21.34`	`St1ckm4n`	`Asterisk Queue Logger restarted`
`17:21.42`	`Qwell`	`St1ckm4n: do it from console`
`17:21.47`	`Qwell`	`asterisk -rx "logger rotate"`
`17:21.58`	`*** join/#asterisk MatsK (i=MatsK@83.233.97.229)`
`17:22.02`	`*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)`
`17:22.07`	`St1ckm4n`	`that's what I did`
`17:22.10`	`Strom_C`	`oh wait, must -rx be followed by a command in double-quotes?`
`17:22.11`	`Qwell`	`oh`
`17:22.21`	`St1ckm4n`	`yes`
`17:22.22`	`Strom_C`	`because the script has them in singles`
`17:22.38`	`Strom_C`	`/usr/sbin/asterisk -rx 'logger rotate'`
`17:22.55`	`St1ckm4n`	`works with single quotes too`
`17:22.55`	`Qwell`	`Strom_C: the quotes are for the console...* never sees them`
`17:23.00`	`Strom_C`	`hm ok`
`17:23.53`	`Strom_C`	`is there anything else that cron is doing hourly to asterisk?`
`17:23.55`	`St1ckm4n`	`there's the 00-makewhatiscron`
`17:24.09`	`St1ckm4n`	`but those are the only two running at the time of the crash`
`17:24.26`	`Strom_C`	`what is cron / baby dont run me / dont run me / no more`
`17:24.36`	`Strom_C`	`head bobbing`
`17:24.36`	`Qwell`	`Strom_C: ...`
`17:24.50`	`St1ckm4n`	`I'm really tempted to update my asterisk but scared that I'll break something`
`17:25.00`	`Qwell`	`St1ckm4n: what version?`
`17:25.23`	`St1ckm4n`	`1.2.5`
`17:25.24`	`Strom_C`	`Qwell: http://ckjcwf.ytmnd.com/`
`17:25.38`	`Strom_C`	`1.2.5 is kind of old...`
`17:26.13`	`file`	`if it works though...`
`17:26.27`	`St1ckm4n`	`what's the best way of going about preparing for an update, I run a backup nightly but I'm not 100% sure it's grabbing everything I might need in a worst case scenario`
`17:26.41`	`Strom_C`	`best? clone the drive :)`
`17:26.48`	`evisu`	`shot in the dark here, but would anyone happen to be familiar with the voip laws in israel?`
`17:29.43`	`*** join/#asterisk klictel (n=klictel@207.107.208.137)`
`17:30.14`	`*** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net)`
`17:30.34`	`evisu`	`got disconnected. did anyone happen to answer?`
`17:30.40`	`Strom_C`	`no`
`17:30.43`	`St1ckm4n`	`nope, it was pretty silent`
`17:30.44`	`evisu`	`thanks`
`17:31.20`	`St1ckm4n`	`is updating asterisk that difficult or does it tend to go pretty smoothly?`
`17:31.39`	`Strom_C`	`St1ckm4n: I've never had a problem upgrading within the same major release`
`17:32.35`	`St1ckm4n`	`any hints on what I should backup first in case I need to try and roll it back?`
`17:33.32`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`17:34.05`	`*** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66)`
`17:36.03`	`jhiver`	`ok, no luck with alaw either`
`17:37.21`	`*** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)`
`17:39.25`	`Strom_C`	`i guess either none of us know the answer, or we're all so confident that the upgrade will work that there's no need to worry about it`
`17:39.55`	`St1ckm4n`	`k, I know where to go when it's 3am and I don't have a working phone switch j/k`
`17:40.42`	`Strom_C`	`heheh`
`17:40.54`	`Strom_C`	`St1ckm4n: for kicks, i'd be curious to see your extensions.conf`
`17:41.16`	`Strom_C`	`St1ckm4n: they dont have you running freepbx or anything, do they?`
`17:41.26`	`St1ckm4n`	`whats freepbx?`
`17:41.36`	`salviadud`	`good answer`
`17:41.42`	`Strom_C`	`perfect answer`
`17:41.50`	`salviadud`	`lol`
`17:42.03`	`St1ckm4n`	`no the switch has amp on it but they said they didn't recommend using it so I've never even bothered opening it`
`17:42.13`	`Strom_C`	`St1ckm4n: wait wait wait`
`17:42.23`	`Strom_C`	`St1ckm4n: the switch has AMP but they said not to use it??`
`17:42.37`	`salviadud`	`well, that's actually a good move`
`17:42.45`	`Strom_C`	`well it`
`17:42.50`	`Strom_C`	`it's half a good move`
`17:42.59`	`St1ckm4n`	`yeah they showed it to me on install saying that it was a front end for configuring it but that it could be buggy`
`17:43.03`	`Strom_C`	`why install AMP in the first plave then`
`17:43.07`	`*** join/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net)`
`17:43.12`	`Strom_C`	`er`
`17:43.13`	`smackus2`	`happy 4th`
`17:43.14`	`Strom_C`	`place`
`17:43.33`	`Strom_C`	`smackus2: http://starspangledwtf.ytmnd.com/`
`17:43.39`	`*** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net)`
`17:43.39`	`Dr-Linux`	`what should i use for offhook/onhook`
`17:43.49`	`file`	`Strom_C: A PAYPHONE! IN THE CORNER!`
`17:43.57`	`Strom_C`	`OOOOOOOOOOOOOOOHHHHH`
`17:44.08`	`smackus2`	`nascar fan?`
`17:44.17`	`Strom_C`	`no`
`17:44.19`	`St1ckm4n`	`funny that AMP got brought up, I see it on the top of my extensions.conf`
`17:44.48`	`smackus2`	`looking for some assistance with the error message: Unable to request echo training on channel 50. Not finding much in google`
`17:45.07`	`Strom_C`	`St1ckm4n: I'm not saying you made a bad decision, but I could have probably done the job at 1/3 the cost, hung over, and in a bathrobe`
`17:45.12`	`smackus2`	`I have a E&M t1`
`17:45.18`	`smackus2`	`winkstart and such`
`17:45.24`	`file`	`Strom_C: and naked?`
`17:45.27`	`Strom_C`	`smackus2: channel 50?`
`17:45.36`	`Strom_C`	`smackus2: how many T1s do you have>`
`17:45.39`	`smackus2`	`49 and 50 so far`
`17:45.41`	`smackus2`	`4`
`17:45.45`	`St1ckm4n`	`lol, where you located at Strom_C`
`17:45.51`	`Strom_C`	`St1ckm4n: Los Angeles`
`17:45.57`	`smackus2`	`first two are pri second two are e&m wink`
`17:46.20`	`smackus2`	`we took 7000 calls on asterisk yesterday.`
`17:46.23`	`smackus2`	`aprox`
`17:46.28`	`smackus2`	`:-D`
`17:46.34`	`smackus2`	`only a couple of hiccups`
`17:46.38`	`Strom_C`	`smackus2: does your T1 card have a hardware echo can?`
`17:46.39`	`St1ckm4n`	`I know I'm not impressed with their abilities either, Asterisk was/is spawning extra processes intermittently and their solution was a cron that kills them`
`17:46.49`	`St1ckm4n`	`instead of figuring out why they are spawning`
`17:47.22`	`smackus2`	`it is the TE411P (so with echo cancel)`
`17:47.30`	`St1ckm4n`	`and they disabled safe asterisk, which after I looked at the script there were several errors because he used the wrong char to comment out lines`
`17:47.31`	`file`	`I wish I had mailing list access`
`17:48.15`	`St1ckm4n`	`I reenabled safe_asterisk because the processes still spawned even though it wasn't, and I'ld rather have it running as a safety net than not running at all`
`17:48.40`	`Dr-Linux`	`anybody knows about SCCP?`
`17:48.47`	`smackus2`	`The other one I am starting to see is: app_voicemail.c: No more messages possible`
`17:48.50`	`dlynes_office`	`St1ckm4n: you mean intermittent mpg123 processes?`
`17:49.08`	`St1ckm4n`	`I don't think they were mpg123 processes`
`17:49.23`	`St1ckm4n`	`whenever I did ps there would be several asterisk processes sitting dormant`
`17:49.28`	`dlynes_office`	`smackus2: you might be out of drive space, or you might be at the maximum number of messages for that user`
`17:49.45`	`St1ckm4n`	`and if I didn't kill -9 them more would multiply and the phone switch would hang and have to be rebooted`
`17:49.46`	`file`	`processes or threads?`
`17:49.48`	`dlynes_office`	`St1ckm4n: yeah...probably old call legs that haven't been killed off yet`
`17:49.57`	`dlynes_office`	`St1ckm4n: it was probably threads, as file has enquired`
`17:49.59`	`smackus2`	`ah, ok.`
`17:50.16`	`dlynes_office`	`smackus2: your max messages defaults to 99 if I remember correctly`
`17:50.19`	`file`	`I haven't seen a box that represented threads as processes for a long time`
`17:50.27`	`dlynes_office`	`smackus2: you can change that in your voicemail.conf file`
`17:50.35`	`dlynes_office`	`file: i've got several boxes like that`
`17:50.40`	`dlynes_office`	`file: it's how ps groups them`
`17:51.02`	`dlynes_office`	`file: basically if the person's running a 2.4 kernel, you'll see it`
`17:51.07`	`dlynes_office`	`file: with a 2.6 you won't`
`17:51.12`	`file`	`2.4, those were the days`
`17:51.12`	`[TK]D-Fender`	`dlynes_office : Slackware for the win!`
`17:51.23`	`[TK]D-Fender`	`file : How soon is now? ;)`
`17:51.25`	`dlynes_office`	`[TK]D-Fender: I dont' run 2.4 on slackware`
`17:51.41`	`[TK]D-Fender`	`dlynes_home : Yeah, and you have problems :)`
`17:51.46`	`dlynes_office`	`I do?`
`17:51.48`	`smackus2`	`is there a way to search within the cli output?`
`17:51.51`	`St1ckm4n`	`is anyone else here running a small-medium size inbound call center off of asterisk, and if so what are you using to see live time agent status and queue status?`
`17:51.58`	`smackus2`	`besides scrolling and looking`
`17:52.05`	`[TK]D-Fender`	`dlynes_office : Mistar smart-e pants and his fancy kernels! ;)`
`17:52.25`	`*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`17:52.38`	`smackus2`	`St1ckm4n: we bult an app yesterday that parses show agents and show queues`
`17:52.48`	`smackus2`	`not real time, but gives the info to the users`
`17:52.56`	`smackus2`	`astguiclient looks cool`
`17:52.58`	`[TK]D-Fender`	`St1ckm4n : I have a live queue stats display on the idle page of my Polycom IP 600 MicroBrowser and have done some web apps as well.`
`17:53.00`	`smackus2`	`have not used it yet though`
`17:53.12`	`dlynes_office`	`fscking hell`
`17:53.24`	`dlynes_office`	`2.6.17.3 is annoying`
`17:53.26`	`[TK]D-Fender`	`St1ckm4n : Also am preparing for ACD integration with them.`
`17:53.35`	`dlynes_office`	`trying to get it up and running on my laptop :)`
`17:53.37`	`[TK]D-Fender`	`dlynes_office : Told you so? :\|`
`17:53.40`	`salviadud`	`i'm on 2.6.16.19`
`17:53.47`	`St1ckm4n`	`I wrote a web app that I put up on our big screens but it has to poll every five seconds and only shows agents on a queue call, no outbound or personal inbound`
`17:53.52`	`[TK]D-Fender`	`8.6.7.5.3.0.9.?`
`17:53.57`	`dlynes_office`	`[TK]D-Fender: nah...bought a new dual core laptop`
`17:54.01`	`salviadud`	`they all look the same to me :(`
`17:54.06`	`St1ckm4n`	`I'm worried that this may be causing problems since it has to poll off the server so often`
`17:54.09`	`dlynes_office`	`[TK]D-Fender: trying to get a kernel up and running on it with dual core optimizations`
`17:54.32`	`paolob`	`Guys, I access my pstn line throug a sipura spa3000. The telcom gave me a code in order to restrict access to long distance call, so that in order to make a long distance call I must dial 12345 (a dialtone is given almost instantly) 1xxx-xxxx. How do I program asterisk so that it gives the 12345 code before the long distance number? thank you!`
`17:54.47`	`smackus2`	`is there a command in linux to count files in a directory?`
`17:54.48`	`[TK]D-Fender`	`dlynes_home : That kind of effort is validated when running a wimp like those C3's.... but FFS do yourself a favour and just celebrate getting your hands on anything better :)`
`17:55.08`	`Strom_C`	`paolob: easy`
`17:55.11`	`dlynes_office`	`[TK]D-Fender: i bought the laptop with my own money, not company money :)`
`17:55.26`	`[TK]D-Fender`	`dlynes_home : And spoiled already, I'm impressed...`
`17:55.29`	`Strom_C`	`paolob: prefix the dialed number in the Dial application with *12345`
`17:55.30`	`dlynes_office`	`[TK]D-Fender: heh`
`17:55.32`	`St1ckm4n`	`[TK]D-Fender & smackus2: did you guys use I forget the term but you telnet in and Action: show agents`
`17:55.32`	`Nugget`	`telnet is eeeeeeevil!`
`17:55.58`	`smackus2`	`i use ssh`
`17:56.02`	`dlynes_office`	`[TK]D-Fender: but after this last spate of c3's is exhausted`
`17:56.07`	`dlynes_office`	`[TK]D-Fender: we'll be using semprons`
`17:56.09`	`paolob`	`Strom_C, the fact is that I must wait one second before dialling the 1xxx-xxxx`
`17:56.16`	`Strom_C`	`then add ww`
`17:56.18`	`[TK]D-Fender`	`St1ckm4n : "show queues". I parse it, as well as dumping VM so I can compile in the VM msg count for the queue MB's`
`17:56.31`	`[TK]D-Fender`	`dlynes_office : Sempron is very livable.`
`17:56.37`	`dlynes_office`	`[TK]D-Fender: exactly`
`17:56.48`	`dlynes_office`	`[TK]D-Fender: the via's are huge pieces of crap`
`17:56.58`	`dlynes_office`	`[TK]D-Fender: i don't know how any deals with those on asterisk`
`17:57.02`	`smackus2`	`St1ckm4n: you are just trying to show live stats right?`
`17:57.04`	`dlynes_office`	`s/any/anyone/`
`17:57.09`	`smackus2`	`how often do you need to view them?`
`17:57.10`	`St1ckm4n`	`yes`
`17:57.19`	`smackus2`	`all the time?`
`17:57.23`	`[TK]D-Fender`	`St1ckm4n : 5s isn't bad...`
`17:57.24`	`St1ckm4n`	`yep`
`17:57.26`	`smackus2`	`do you have someone sitting and monitroing?`
`17:57.39`	`St1ckm4n`	`we have two big screens showing the stats to the call center`
`17:57.45`	`paolob`	`Strom_C, the *12345 gets the answer: "chan_sip.c:9559 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Oficina Mision" <sip:asterisk@10.152.58.1>;tag=as15018f5a'"`
`17:57.46`	`smackus2`	`do you have or are you your self a developer?`
`17:58.00`	`smackus2`	`can you do things with php and mysql`
`17:58.01`	`Strom_C`	`paolob: paste your dial statement`
`17:58.13`	`St1ckm4n`	`all I did was make some quick php pages that parse out show agents like you said and color them depending on their status and what queue they're on`
`17:58.21`	`rpm`	`should i get bri's or should i get pri's, a bri is a b-channel which is not always connected right?`
`17:58.28`	`Strom_C`	`rpm: no`
`17:58.36`	`Strom_C`	`rpm: BRI is two b-channels and a d-channel`
`17:58.40`	`smackus2`	`that will not be real time though... you have to have the page refresh every few seconds.`
`17:58.42`	`St1ckm4n`	`it polls every 5 seconds, and then I show the FOP queue screen,`
`17:58.45`	`Strom_C`	`rpm: PRI is 23 b-channels and a d-channel`
`17:58.48`	`smackus2`	`ok`
`17:58.49`	`smackus2`	`that works`
`17:59.03`	`smackus2`	`does that not work for you?`
`17:59.04`	`paolob`	`Strom_C, exten => _918[02]9NXXXXXX,1,Dial(SIP/*23153${EXTEN:1}@${TRANSPORTEBETANIA},60,Tt) , where TRANSPORTEMISION=pstn-spa3000-mision`
`17:59.08`	`St1ckm4n`	`it works but seems hokey`
`17:59.15`	`dlynes_office`	`rpm: pri is a 23 b-channels and a d-channel if it's a T1, and 29 b-channels and one d-channel if it's an E1`
`17:59.24`	`St1ckm4n`	`I can only see queue calls, I cannot tell if someone is on a personal call or an outbound`
`17:59.26`	`smackus2`	`well... hokey is a relative term`
`17:59.33`	`dlynes_office`	`rpm: pri's and bri's are both always connected`
`17:59.46`	`smackus2`	`unless you want to write more apps, take a look at astguiclient`
`17:59.58`	`smackus2`	`it sounds like it already has a lot of what you are looking for`
`18:00.13`	`St1ckm4n`	`and my next problem comes from mapping the DND button on the polycom to the not busy feature on asterisk so I can tell if someone is putting themselves on DND`
`18:00.28`	`Dr-Linux`	`anybody know what's this? >> rtptos = 184 ; sets the default rtp packets TOS`
`18:00.30`	`smackus2`	`astguiclient.sourceforge.net/`
`18:00.33`	`St1ckm4n`	`I did look at astguiclient, it looks like what we need`
`18:00.38`	`*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)`
`18:01.13`	`St1ckm4n`	`[TK]D-fender, I would be curious to see your web page you show on your 600, the managers are using the 600's right now and I could see them liking that ability`
`18:01.20`	`docelmo`	`Dr-Linux, did you bother consulting the Wiki or Google before dumping that question in here?`
`18:03.50`	`paolob`	`Strom_C, did you see the statement?`
`18:03.55`	`St1ckm4n`	`astguiclient I think will work for managers, but doesn't look like it would work on the big screens I need to be able to customize it and expand it so it's easily visible`
`18:03.57`	`Dr-Linux`	`docelmo: do you know something about SCCP?`
`18:04.43`	`Strom_C`	`paolob: yeah - im not quite sure, but i think it might be a config issue with your spa3000`
`18:05.02`	`St1ckm4n`	`FOP would work but I notice it has some ghost issues where calls show in queue that really aren't or it shows someone on the phone after they have left`
`18:05.14`	`[TK]D-Fender`	`St1ckm4n : Just disable DND. Thats a bad thing with queues`
`18:05.32`	`paolob`	`Strom_C, but is there a way to tell the dial application to wait a second before keeping on with the next numbers?`
`18:05.34`	`*** join/#asterisk HuSoft (n=apo@194stb46.codetel.net.do)`
`18:05.43`	`St1ckm4n`	`I wanted to disable it but people were complaining because they didn't have enough time to finish up their previous call`
`18:06.03`	`St1ckm4n`	`and needed a busy feature instead of having to log out to avoid the next call`
`18:06.46`	`Strom_C`	`paolob: if you are using a zaptel card, sure. with an spa3000 i dont know`
`18:06.52`	`[TK]D-Fender`	`St1ckm4n : Thats what the "wrapuptime" is for....`
`18:07.34`	`Qwell`	`Dr-Linux: ?`
`18:07.38`	`St1ckm4n`	`we have that set at 10 seconds right now, the problem is with the variations we have in calls, our talk time can be from 2 minutes to over an hour and we let the agents determine their wrapup on each call`
`18:07.41`	`docelmo`	`Dr-Linux, yes.. why? Its the default protocol used by cisco`
`18:08.00`	`Dr-Linux`	`docelmo: that's what i'm asking since few days`
`18:08.10`	`paolob`	`Anyone knows of a irc channel for sipura products?`
`18:08.24`	`docelmo`	`Well I havent been here what do you wanna know and why the HELL do you wanna use it?`
`18:08.37`	`Dr-Linux`	`docelmo: i have configured SCCP on asterisk for one of my Cisco 7935 conference phone. but i have a few problems, so i'm asking here, maybe someone knows already`
`18:08.51`	`docelmo`	`flash to sip!`
`18:09.11`	`Dr-Linux`	`docelmo: Cisco 7935 doesn't support SIP :(`
`18:09.27`	`docelmo`	`If you flash it with a new IOS Im sure it wil`
`18:09.29`	`docelmo`	`will`
`18:09.54`	`Strom_C`	`docelmo: uhm`
`18:10.03`	`Strom_C`	`7935 doesnt run IOS IIRC`
`18:10.11`	`[TK]D-Fender`	`St1ckm4n : have them use pause/unpause rather than DND. DND really screws with stats`
`18:10.18`	`docelmo`	`Well not IOS but whatever..`
`18:10.55`	`St1ckm4n`	`can I remap DND to the pause/unpause feature, so they don't have to dial that feature`
`18:11.12`	`Dr-Linux`	`docelmo: i'm already using a couple of cisco 7960/40 with SIP firmware, but this device doesn't support SIP, i think and i was told by few folks here`
`18:11.21`	`*** join/#asterisk qdk (n=qdk@x1-6-00-0f-66-90-6b-48.k441.webspeed.dk)`
`18:11.27`	`Dr-Linux`	`docelmo: if you know it does, then please help me how can i do tht?`
`18:11.48`	`HuSoft`	`can someone please help me? http://pastebin.ca/78915`
`18:12.50`	`Strom_C`	`HuSoft: run "sip show peers'`
`18:12.52`	`Strom_C`	`e`
`18:12.53`	`Strom_C`	`er`
`18:12.57`	`Strom_C`	`"sip show peers"`
`18:13.13`	`paolob`	`Strom_C, excuse me, when I dial 18097630026 with that dialplan statemente I get: Executing Dial("SIP/oficinamision-8255", "SIP/*2315318097630026@pstn-spa3000-betania\|60\|Tt") in new stack`
`18:13.14`	`paolob`	`<PROTECTED>`
`18:13.14`	`paolob`	`Jul 4 13:56:52 WARNING[4160]: chan_sip.c:9559 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Oficina Mision" <sip:asterisk@10.152.58.1>;tag=as15018f5a'`
`18:13.14`	`paolob`	`<PROTECTED>`
`18:13.24`	`HuSoft`	`zoel/zoel 10.0.0.1 D N 5060 Unmonitored`
`18:13.31`	`HuSoft`	`husoft/husoft 10.0.0.3 D N 5062 Unmonitored`
`18:13.40`	`HuSoft`	`2 sip peers [2 online , 0 offline]`
`18:13.56`	`paolob`	`Strom_C, who is giving asterisk the response "Forbidden etc."?`
`18:14.02`	`docelmo`	`Well have fun with that one then. I know what sccp is but never had to use it thank god`
`18:14.06`	`Strom_C`	`HuSoft: then you have to dial SIP/zoel`
`18:14.12`	`Strom_C`	`or dial SIP/husoft`
`18:14.20`	`*** join/#asterisk marv0997 (i=marv0997@190.4.2.86)`
`18:14.37`	`Strom_C`	`paolob: I already told you`
`18:14.44`	`Strom_C`	`paolob: it's the sipura rejecting the call`
`18:14.46`	`HuSoft`	`Strom_C, let me try that`
`18:15.02`	`reza_`	`hey, does anyone know of a cheap 1800 did provider?`
`18:15.11`	`Strom_C`	`reza_: define cheap`
`18:15.22`	`paolob`	`Strom_C, apparently it consider it a INVITE statement... what is that?`
`18:15.30`	`HuSoft`	`YES!!! IT WORKED`
`18:15.32`	`reza_`	`like under 3c/min ish`
`18:15.39`	`dlynes_office`	`Dr-Linux: have you talked to qwell at all?`
`18:16.02`	`HuSoft`	`thanks a LOT Storm_C!`
`18:16.17`	`reza_`	`i actually want a vanit did`
`18:16.18`	`Strom_C`	`HuSoft: I should really start charging for advice ;)`
`18:16.18`	`Dr-Linux`	`dlynes_home: he was here, but he didn't answer me ...`
`18:16.24`	`dlynes_office`	`Dr-Linux: ah`
`18:16.28`	`HuSoft`	`hehehe`
`18:16.35`	`HuSoft`	`:]`
`18:16.45`	`reza_`	`storm - do you know of any?`
`18:16.47`	`dlynes_office`	`Dr-Linux: anyways...if your boss greases his palms with a little cash, it might help, too...he might be more inclined to get something working`
`18:16.56`	`Dr-Linux`	`dlynes_home: i wanna share with you guys my sccp.conf , there is good comments for everything, maybe your veiw can help`
`18:17.06`	`Strom_C`	`reza_: there are dozens`
`18:17.21`	`Qwell`	`?`
`18:17.26`	`Strom_C`	`3c per minute is considered to be on the pricey end of things`
`18:17.34`	`reza_`	`so then that's fine`
`18:17.36`	`Dr-Linux`	`Qwell: hi`
`18:17.42`	`reza_`	`i just want a vanity one`
`18:17.50`	`reza_`	`800-the-reza or something like that`
`18:18.02`	`dlynes_office`	`Dr-Linux: see what i mean?`
`18:18.08`	`*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)`
`18:18.10`	`dlynes_office`	`Dr-Linux: his ears perk up more when cash is mentioned :)`
`18:18.27`	`Dovid`	`lol`
`18:18.35`	`HuSoft`	`Strom_C, is there a way to have these extensions in a database? or asterisk only reads it from extensions.conf?`
`18:18.54`	`Strom_C`	`HuSoft: what do you mean`
`18:18.58`	`[TK]D-Fender`	`HuSoft : look up Asterisk Realtime. But its more trouble than its worth most of the time.`
`18:19.07`	`Dr-Linux`	`Qwell: as you advised, i have compiled SCCP and configured it, everytihng works fine, voice quality is quite fine. but i have 2 problems`
`18:19.15`	`Strom_C`	`HuSoft: make life easier on yourself and just use the extension numbers as the names of the sip.conf entries`
`18:19.16`	`[TK]D-Fender`	`HuSoft : how big a setup ae you looking at having?`
`18:19.44`	`Dr-Linux`	`Qwell: 1, i can't hangup the call being on the phone`
`18:19.47`	`reza_`	`what time does the worldcup game start?`
`18:19.54`	`HuSoft`	`[TK]D-Fender, a big call center`
`18:20.16`	`Strom_C`	`HuSoft: where are you located?`
`18:20.22`	`dlynes_office`	`HuSoft: .do == dominican republic?`
`18:20.23`	`Dr-Linux`	`Qwell: 2 my voice quality goes very bad, if i make calls via asterisk trunks.`
`18:20.31`	`HuSoft`	`yeah`
`18:20.34`	`dlynes_office`	`ah`
`18:20.49`	`Qwell`	`Dr-Linux: send a message to the mailing list`
`18:21.16`	`dlynes_office`	`Dr-Linux: it sounds like it's probably not an sccp issue`
`18:21.19`	`St1ckm4n`	`[TK]D-Fender: when you say that DND screws up the stats, does this affect the call routing strategy, we have approx 20 queues right now and people complain about getting calls back to back while someone else is available, I've tried all available strategys with no luck`
`18:21.24`	`Dr-Linux`	`Qwell: what mailing list?`
`18:21.31`	`Qwell`	`Dr-Linux: chan-sccp-users`
`18:21.37`	`Qwell`	`chan-sccp.berlios.de`
`18:21.55`	`dlynes_office`	`Dr-Linux: you're using chan_sccp, not chan_skinny?`
`18:22.19`	`rob0`	`Host howdy.do not found: 3(NXDOMAIN)`
`18:22.23`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.162)`
`18:22.38`	`paolob`	`Strom_C, I found how to bypass the sipura barrier. Now I must find the way to give the code, wait for the dialtone (or wait a second) and send the long distance number`
`18:22.48`	`Qwell`	`dlynes_home: 793x hasn't been tested with skinny`
`18:22.52`	`Qwell`	`dlynes_home: feel free to send me one`
`18:22.55`	`dlynes_office`	`Qwell: ah`
`18:23.02`	`dlynes_office`	`Qwell: not much use to me`
`18:23.03`	`Dr-Linux`	`dlynes_office: i'm using chan_sccp bcoz Qwell told me to use it.`
`18:23.06`	`dlynes_office`	`Qwell: i'm not using cisco`
`18:23.07`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`18:23.16`	`dlynes_office`	`Qwell: if i was, i probably wouldn't have a problem sending you one`
`18:23.41`	`Dr-Linux`	`dlynes_office: yes, maybe you are right, it's not chan_sccp issue :S`
`18:23.50`	`reza_`	`so anyone here want to recommend a toll-free DID provider?`
`18:24.12`	`Dr-Linux`	`Qwell: please have a look > http://pastebin.ca/78921`
`18:24.12`	`dasenjo`	`Hi, I'm behind NAT and trying to register an iax extension, I got no error registration messages, but when I type "iax2 debug" on CLI I got messages, can you help me?`
`18:24.23`	`Dr-Linux`	`dlynes_office: here is my sccp.conf`
`18:24.27`	`Dr-Linux`	`http://pastebin.ca/78921`
`18:24.30`	`dlynes_office`	`Dr-Linux: and?`
`18:24.52`	`dlynes_office`	`Dr-Linux: i know absolutely nothing about chan_skinny, chan_sccp, or cisco phones, in general`
`18:25.05`	`Dr-Linux`	`dlynes_office: if that's not SCCP issue, then lemme show you my two lines for this device in extensions.conf`
`18:25.12`	`paolob`	`Guys, what does it happen if I put a dial(*12345,1) followed by a dial(some telephone number) ? after the 1st dial command has timed out, does it send the second?`
`18:25.20`	`Dr-Linux`	`dlynes_office: no problem`
`18:25.29`	`Strom_C`	`paolob: why not just try it`
`18:25.39`	`paolob`	`Strom_C, let me see`
`18:26.02`	`dlynes_office`	`Dr-Linux: are you running anything besides chan_sccp/cisco phones on that system?`
`18:26.23`	`reza_`	`so voxbone (my current did provider) added aix support, i'm already configured for sip with them -- any benefit in switching over?`
`18:26.36`	`paolob`	`Strom_C, it get an answer, and therefore it doesn't proceed until the central or I disconnect :-(`
`18:26.38`	`Strom_C`	`AIX?`
`18:26.55`	`reza_`	`axi?`
`18:27.03`	`Strom_C`	`IXA?`
`18:27.05`	`reza_`	`astrisk's own protocol`
`18:27.09`	`Strom_C`	`oh. IAX`
`18:27.25`	`reza_`	`hehe, yeah, one of those ;)`
`18:27.33`	`Dr-Linux`	`dlynes_office: yes, zap hardwares and SIP etc all kind of stuff`
`18:27.35`	`Strom_C`	`pronounced "eeks"`
`18:27.47`	`Dr-Linux`	`exten => 2123,1,Dial(SCCP/2123)`
`18:27.47`	`Dr-Linux`	`exten => h,2,Hangup`
`18:27.53`	`reza_`	`no "eaks"?`
`18:27.57`	`rob0`	`Eeks!`
`18:28.03`	`Qwell`	`~eeks`
`18:28.07`	`jbot`	`hmm... eeks is the Eeks eeks run for the hills IAX2 is here to stay`
`18:28.07`	`Dr-Linux`	`these are my extensions.conf entry for SCCP phone`
`18:28.32`	`reza_`	`so any benefit in using iax over sip with a did provider?`
`18:28.34`	`Strom_C`	`next up: chan_omgwtfbbq`
`18:28.44`	`rob0`	`Looks like "yacks" to me, sorry.`
`18:30.30`	`Qb3rt`	`is there a way to do load balancing and/or redundency with asterisk servers???`
`18:31.44`	`dasenjo`	`please help me. I'm using tp-link asdl modem, I dont know if Im using consistent nat .. or what can I do ..`
`18:31.45`	`dlynes_office`	`Dr-Linux: and it's only the sccp channels you're having issues with?`
`18:31.46`	`Dr-Linux`	`Qwell: where is sccp-mailing-list?`
`18:31.52`	`dasenjo`	`I really need to register this IAX phone ..`
`18:31.54`	`dlynes_office`	`Dr-Linux: he already told yo`
`18:31.55`	`Qwell`	`Dr-Linux: chan-sccp.berlios.de has a link`
`18:32.10`	`Dr-Linux`	`dlynes_office: yes, only with this 7935 phone`
`18:32.11`	`[TK]D-Fender`	`St1ckm4n : try "rrmemory". Works the best in my experience.`
`18:32.47`	`[TK]D-Fender`	`St1ckm4n : DCC'ing you a pic of my MB screen`
`18:32.58`	`Qb3rt`	`is there a way to do load balancing and/or redundency with asterisk servers???`
`18:32.59`	`Dr-Linux`	`"error in upgrading file format"`
`18:33.15`	`Strom_C`	`Qb3rt: dont ask the same question more than once every fifteen minutes please`
`18:33.45`	`Dr-Linux`	`Qwell: when i reboot the phone i see this on the phone screen "error in upgrading file format" but it boots sucessfully and works, not sure what this ERROR means.`
`18:33.57`	`Qwell`	`Dr-Linux: bad firmware in your tftp?`
`18:34.52`	`*** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net)`
`18:35.09`	`Dr-Linux`	`Qwell: i have 2 files in the TFTP, the phone grabs one and do not grab other one. not sure why. also not sure why i need other one.`
`18:40.18`	`*** join/#asterisk jetway2006 (n=asd@218.111.159.204)`
`18:40.26`	`jetway2006`	`hi`
`18:40.29`	`dlynes_office`	`Qb3rt: check the 'large asterisk systems' section of the wiki`
`18:40.33`	`dlynes_office`	`~wikis`
`18:40.34`	`jbot`	`i guess wikis is http://www.voip-info.org`
`18:40.39`	`dlynes_office`	`erm`
`18:40.42`	`dlynes_office`	`~asteriskwiki`
`18:40.44`	`jbot`	`somebody said asteriskwiki was at http://www.voip-info.org/wiki-Asterisk"`
`18:41.24`	`jetway2006`	`could someone pls tell me how do calling card system have one access number but are able to support many stimulaneous users calling`
`18:41.54`	`Dovid`	`jetway2006: you need to get a 'line' that can handle multiple calls`
`18:41.55`	`dlynes_office`	`jetway2006: pri's`
`18:42.07`	`Dovid`	`I have one with pure VOIP`
`18:42.20`	`jetway2006`	`ok..is it from the telco`
`18:42.37`	`dlynes_office`	`jetway2006: a pri is, yes`
`18:42.47`	`dlynes_office`	`i'm guessing you want it for local callers right?`
`18:43.22`	`jetway2006`	`i am just curious as how does a single number could support so many users calling at the same time`
`18:43.26`	`*** join/#asterisk s0lid (n=s0lid@61.28.161.132)`
`18:43.36`	`jetway2006`	`how many user could pri support`
`18:43.46`	`dlynes_office`	`you mean simultaneous callers?`
`18:43.51`	`dlynes_office`	`as many as you want`
`18:43.55`	`dlynes_office`	`23 per pri though`
`18:43.59`	`paolob`	`Guys, trying to use Dial(resource,timeout,D(DTMF)), but I must send the DTMF after 3 seconds. How do I do it? I'm exiting to pstn via a sipura spa3000`
`18:43.59`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-125-116.red.bezeqint.net)`
`18:44.14`	`dlynes_office`	`but you can have multiple pris per machine`
`18:44.36`	`dlynes_office`	`and you can have that number dropped on to anyone of them, or all of them, or a combination thereof`
`18:44.38`	`jetway2006`	`ok...thanks for the info ..is it same as buying did numbers`
`18:45.10`	`dlynes_office`	`sorta yeah...you need to go through your telco's business services section to order a pri`
`18:45.18`	`tzafrir_laptop`	`hmmm that scott withh that well-configured auto-responder... just out of curiosity: any idea what auto-responder he uses? Merak mail server?`
`18:45.26`	`dlynes_office`	`but it's a lot more expensive than simple dids`
`18:45.39`	`dlynes_office`	`did is a virtual concept...it's nothing tangible`
`18:45.44`	`jetway2006`	`what is pri actually....pls explain a bit further`
`18:45.48`	`dlynes_office`	`a pri is a physical link`
`18:45.58`	`Strom_C`	`ISDN Primary Rate Interface`
`18:46.05`	`dlynes_office`	`primary rate interface; 23 b-channels and and one d-channel`
`18:46.09`	`Strom_C`	`a digital trunk with 23 voice channels and a signaling channel`
`18:46.10`	`jetway2006`	`ok...is it like the copper wire phone line`
`18:46.21`	`dlynes_office`	`the d-channel is your control channel; each b channel is a 56Kbps voice channel`
`18:46.26`	`Strom_C`	`no no`
`18:46.31`	`dlynes_office`	`jetway2006: it's CAT5E UTP`
`18:46.32`	`Strom_C`	`b-channels are 64kbps, silly`
`18:46.39`	`dlynes_office`	`erm 64Kbps i mean yeah`
`18:46.41`	`jetway2006`	`ok`
`18:46.46`	`dlynes_office`	`i'm out to lunch`
`18:46.48`	`Strom_C`	`PRI should not be doing bit-robbing`
`18:46.54`	`jetway2006`	`thanks dlynes`
`18:47.04`	`jetway2006`	`go ahead strom`
`18:47.17`	`Strom_C`	`jetway2006: do you know what a T1 is?`
`18:47.18`	`jetway2006`	`do calling cards companies buy pri to support many users`
`18:47.23`	`dlynes_office`	`jetway2006: but pri is not internet; it's a different signalling`
`18:47.27`	`jetway2006`	`yes i know abt T!`
`18:47.30`	`jetway2006`	`t1`
`18:47.34`	`dlynes_office`	`jetway2006: you can get internet on pri`
`18:47.37`	`Strom_C`	`jetway2006: a PRI is delivered over a T1`
`18:47.40`	`dlynes_office`	`jetway2006: but it's used for much more than that`
`18:48.09`	`jetway2006`	`when a user calls the access number...the connection goes to the pri`
`18:48.40`	`Strom_C`	`jetway2006: it works like this`
`18:48.47`	`Strom_C`	`say I call 311-555-2368`
`18:48.47`	`jetway2006`	`pls do tell me`
`18:48.51`	`jetway2006`	`ok`
`18:49.26`	`Strom_C`	`the telco sends a setup message down the PRI saying "call on channel 14 from 323-212-3001 to 311-555-2368"`
`18:49.46`	`jetway2006`	`ok`
`18:49.48`	`Strom_C`	`the pbx on the other end of the line takes the call and does whatever the user wants with it`
`18:50.55`	`jetway2006`	`oh..so how many pri usually a calling card company have`
`18:51.06`	`Strom_C`	`however many they need`
`18:51.12`	`Strom_C`	`thats the science of traffic engineering`
`18:51.42`	`jetway2006`	`one pri could support 32 channels rite`
`18:51.49`	`Strom_C`	`are you in europe?`
`18:51.56`	`jetway2006`	`yes`
`18:52.17`	`Strom_C`	`30 or 29 b-channels`
`18:52.25`	`Strom_C`	`i dont remember exactly`
`18:52.34`	`dlynes_office`	`29 b channels, 1 d channel`
`18:52.36`	`drray`	`I've wondered about that, why not use a E1 channel bank in america to have more than 24?`
`18:52.46`	`jetway2006`	`how does pri relate to voice codec used....i heard some codec uses less banwitdh..so more stimulateous calls`
`18:52.48`	`Strom_C`	`thanks dlynes_office`
`18:52.58`	`Strom_C`	`jetway2006: pri will always be alaw/ulaw`
`18:53.01`	`dlynes_office`	`1-15, 17-30`
`18:53.09`	`dlynes_office`	`d-channel is channel 16`
`18:53.13`	`Igbothom_III`	`woohoo - shuttle is up, all looks good`
`18:53.32`	`dlynes_office`	`alaw for e1, ulaw for t1`
`18:53.57`	`jetway2006`	`how abt different codec like gsm ...and others...`
`18:54.08`	`Strom_C`	`jetway2006: not over PRI`
`18:54.12`	`dlynes_office`	`jetway2006: you need to transcode then`
`18:55.04`	`jetway2006`	`ok...another question abt asterisk....abt stimulatenous calls...calls between extension depends on the processor speed`
`18:55.26`	`jetway2006`	`and calls to outside * depends on bandwith`
`18:55.30`	`jetway2006`	`is it rite`
`18:55.40`	`Strom_C`	`in a very general sense, yes`
`18:55.47`	`dlynes_office`	`jetway2006: show translation to see what your transcoding times will be`
`18:56.12`	`jetway2006`	`do all of u have * set up in our place`
`18:56.20`	`jetway2006`	`pls share ur experience`
`18:56.20`	`Qb3rt`	`i am calling in 1-800 number with automated message... it answering and working good.. i am even sending digits to browse the menu and its working perfectly.. but after exactly 60 seconds it says number is not in service... in my CLI i see that ---> Nobody picked up in 60000 ms`
`18:56.26`	`dlynes_office`	`Nope...I don't have it set up your place...only my place`
`18:56.46`	`jetway2006`	`heheh ...i mean ur place`
`18:56.49`	`jetway2006`	`sorry...`
`18:56.52`	`dlynes_office`	`heh`
`18:56.55`	`Strom_C`	`jetway2006: please speak in English`
`18:57.00`	`Strom_C`	`jetway2006: "ur" is not English`
`18:57.05`	`jetway2006`	`ok..`
`18:57.08`	`dlynes_office`	`Strom_C: pls is :)`
`18:57.26`	`jetway2006`	`how large is your asterisk set up`
`18:57.33`	`dlynes_office`	`whatever the hell a weco 500 is :)`
`18:57.36`	`Strom_C`	`Qb3rt: are you sure the remote end is supervising?`
`18:57.44`	`Strom_C`	`dlynes_office: old rotary desk phone`
`18:57.51`	`dlynes_office`	`ah`
`18:58.11`	`dlynes_office`	`jetway2006: i've got several asterisk setups`
`18:58.16`	`jetway2006`	`do calling card companies use did numbers....what are the purpose for it`
`18:58.24`	`dlynes_office`	`jetway2006: everything from 6 extensions to about 40 extensions`
`18:58.25`	`Qb3rt`	`Strom_C: spervising?? what do you mean? i am browsing the menu with digits and its working good...`
`18:58.41`	`Strom_C`	`Qb3rt: is it actually sending an ANSWER message back to you?`
`18:59.05`	`dlynes_office`	`jetway2006: and from 1 phone line to a single pri`
`18:59.24`	`dlynes_office`	`Qb3rt: disconnect supervision`
`18:59.32`	`Strom_C`	`no`
`18:59.35`	`Strom_C`	`answer supervision`
`18:59.39`	`Strom_C`	`not disconnect supervision`
`18:59.42`	`dlynes_office`	`ah`
`18:59.47`	`dlynes_office`	`what's answer supervision?`
`18:59.50`	`dlynes_office`	`never heard of it`
`19:00.14`	`Strom_C`	`dlynes_office: the transmission of onhook / offhook state across the network`
`19:00.18`	`Qb3rt`	`Strom_C: in the CLI it is only saying -- Called g1/18006632275 and nothing else`
`19:00.24`	`evisu`	`how do calling cards survive with 800 numbers costing $.49 to connect from public payphones ?`
`19:00.26`	`dlynes_office`	`Strom_C: don't all phone lines do that, though?`
`19:00.32`	`jetway2006`	`i mean since pri support many channels ...why do calling card company need did numbers...`
`19:00.33`	`Strom_C`	`dlynes_office: not necessarily`
`19:00.57`	`Strom_C`	`dlynes_office: with an analog line, for example, you cant tell when the other party answers`
`19:01.15`	`dlynes_office`	`jetway2006: you need a phone number to call, or the calling card company is going to be out of business if nobody can call their number`
`19:01.30`	`*** join/#asterisk anonymouz666 (n=anonymou@201.29.65.18)`
`19:01.37`	`dlynes_office`	`Strom_C: then how do you charge the customer?`
`19:01.43`	`anonymouz666`	`Strom_C!`
`19:01.54`	`jetway2006`	`access number = did numbers?`
`19:02.01`	`dlynes_office`	`jetway2006: correct`
`19:02.02`	`Strom_C`	`dlynes_office: if you're smart, you're not using analog lines for trunking`
`19:02.04`	`anonymouz666`	`the one who likes aquarela do brasil :D`
`19:02.12`	`dlynes_office`	`Strom_C: i'm not, but still... :)`
`19:02.28`	`Strom_C`	`Qb3rt: what are you ising for trunking?`
`19:02.30`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.162)`
`19:02.52`	`Qb3rt`	`Strom_C: iax`
`19:03.03`	`jetway2006`	`but i heard did number voice data are automatically routed to sip address`
`19:03.05`	`Strom_C`	`Qb3rt: then the called number is not supervising`
`19:03.25`	`Strom_C`	`if you're not seeing an ANSWER message`
`19:03.28`	`Qb3rt`	`Strom_C: so there is no way to fix that??`
`19:03.40`	`Strom_C`	`not unless you have control of the called number :)`
`19:04.28`	`Qb3rt`	`can somebody try this number with asterisk server to see if its really not supervising? 1-800-663-2275 = futurshop canada`
`19:04.40`	`Qb3rt`	`:)`
`19:04.52`	`anonymouz666`	`Strom_C: callerid doesn't work unfortunely.`
`19:05.01`	`anonymouz666`	`dtmf mode with cidstart ring or polarity`
`19:05.08`	`dlynes_office`	`Qb3rt: you're in edmonton?`
`19:05.10`	`Strom_C`	`Qb3rt: one sec`
`19:05.24`	`dlynes_office`	`Qb3rt: what number shoudl i hit on the autoattendant?`
`19:05.48`	`Qb3rt`	`dlynes_office: whatgever number... it needs to play at least 60 seconds`
`19:06.10`	`*** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`19:06.11`	`Qb3rt`	`so just browse and make her repeat the menu with #`
`19:06.20`	`dlynes_office`	`yeah...that's what i'm doing`
`19:06.23`	`Strom_C`	`Qb3rt: yep, the'yre sending PROGRESS message, but not an ANSWER message`
`19:06.39`	`dlynes_office`	`ah...thought you were mon itoring it on your end :)`
`19:07.21`	`Qb3rt`	`Strom_C> :(`
`19:07.24`	`jetway2006`	`does anyone here has set up callback system in asterisk`
`19:07.33`	`*** part/#asterisk dasenjo (n=dasenjo@208.195.215.162)`
`19:07.48`	`Strom_C`	`it does, howeverm supervise once you get through the IVR menu (I pressed 1 repeatedly)`
`19:08.19`	`dlynes_office`	`Strom_C: maybe he's not using Answer()`
`19:08.28`	`*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)`
`19:08.30`	`Qb3rt`	`Strom_C> Can i setup my server to take the PROGRESS as ANSWER?`
`19:08.36`	`Strom_C`	`Qb3rt: uh, no`
`19:08.45`	`Qb3rt`	`i think it is not a good idea... anyway`
`19:08.55`	`jetway2006`	`is rhino channel bank better than digium cards`
`19:08.56`	`Dr-Linux`	`dlynes_office: http://pastebin.ca/78962`
`19:08.57`	`Strom_C`	`dlynes_office: it's not his PBX`
`19:08.58`	`jetway2006`	`??`
`19:09.04`	`dlynes_office`	`oh`
`19:09.07`	`Strom_C`	`dlynes_office: the far end has to send answer supervision`
`19:09.08`	`dlynes_office`	`Dr-Linux: ?`
`19:09.20`	`Strom_C`	`jetway2006: for what purpose`
`19:09.21`	`Dr-Linux`	`Qwell: please have a look, i phone doesn't hange up .. http://pastebin.ca/78962`
`19:10.37`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`19:10.37`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released with a critical security fix for chan_iax2, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx`
`19:10.37`	`Strom_C`	`jetway2006: how many analog phones`
`19:10.41`	`Dr-Linux`	`Qwell: i have only way to terminate the call, reboot the phone or restart the asterisk`
`19:11.08`	`Dr-Linux`	`even can't hangup the channel :(`
`19:11.09`	`Dr-Linux`	`*CLI> soft hangup SCCP/2123-0000000d`
`19:11.09`	`Dr-Linux`	`Requested Hangup on channel 'SCCP/2123-0000000d'`
`19:11.09`	`Dr-Linux`	`*CLI>`
`19:11.23`	`dlynes_office`	`Qb3rt: so you're in edmonton?`
`19:11.36`	`jetway2006`	`24`
`19:11.55`	`Strom_C`	`jetway2006: that's doable, though I'd recommend an Adtran channel bank over a Rhino one`
`19:12.04`	`anonymouz666`	`Strom_C: I will test right now with TDM`
`19:12.14`	`jetway2006`	`please tell me the advantage`
`19:12.25`	`Strom_C`	`jetway2006: Adtran is quality stuff`
`19:12.43`	`jetway2006`	`how abt digium cards`
`19:13.02`	`Strom_C`	`jetway2006: youd need a digium T1 card anyway for the channel bank`
`19:13.26`	`Strom_C`	`jetway2006: or you can use the TDM2400`
`19:13.46`	`jetway2006`	`i mean digium 24 analog fXs card..`
`19:13.53`	`jetway2006`	`is it better`
`19:13.54`	`Strom_C`	`jetway2006: yes, TDM2400`
`19:14.01`	`Strom_C`	`jetway2006: depends on the size of your install`
`19:14.07`	`Qb3rt`	`dlynes_office: no...`
`19:14.20`	`jetway2006`	`is adtrans compatible with *`
`19:14.32`	`Strom_C`	`yes`
`19:14.32`	`jetway2006`	`rhino channel banks states it is`
`19:14.42`	`jetway2006`	`how much would it cost?`
`19:14.46`	`Qb3rt`	`dlynes_office: quebec...`
`19:14.52`	`Strom_C`	`jetway2006: adtran channel bank is like the gold standard of channel banks`
`19:15.00`	`dlynes_office`	`Qb3rt: ah...so you're subcontracted out to future shop then?`
`19:15.25`	`Strom_C`	`jetway2006: channel bank -> T1 -> T1 card`
`19:15.57`	`jetway2006`	`how much is it...`
`19:16.08`	`Strom_C`	`jetway2006: $1500? $2500?`
`19:16.14`	`Strom_C`	`jetway2006: depends where you get it`
`19:16.29`	`Qb3rt`	`dlynes_office: no.. one of our customer is trying to call them and he is not able....`
`19:16.42`	`Strom_C`	`jetway2006: if you're doing max 24 phones, a TDM2400 is probably right for you`
`19:16.46`	`dlynes_office`	`Qb3rt: ah`
`19:17.08`	`Strom_C`	`jetway2006: if you're doing 48-96 phones, then do a quad-span T1 card and Adtran channel banks`
`19:17.14`	`Qb3rt`	`dlynes_office: why? you know them?`
`19:17.19`	`dlynes_office`	`Qb3rt: of course`
`19:17.31`	`dlynes_office`	`Qb3rt: That's where I just bought my laptop on Friday :p`
`19:17.32`	`jetway2006`	`oh okk..thx....any of u have experience with gsm terminal`
`19:17.41`	`*** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com)`
`19:17.47`	`Qb3rt`	`dlynes_office: hehe good stuff!`
`19:17.51`	`dlynes_office`	`good morning, Mattwj2005`
`19:18.06`	`dlynes_office`	`Qb3rt: salut, m'sieu`
`19:18.11`	`Mattwj2005`	`hi dlynes_office :)`
`19:18.24`	`Strom_C`	`Qb3rt: the essence of the problem is that futureshop is being a load of cheap bastards`
`19:18.31`	`Mattwj2005`	`actually it is 2:15 in the afternoon....I work night shifts`
`19:18.32`	`Strom_C`	`Qb3rt: if they dont answer the call, they dont have to pay for it`
`19:18.39`	`Qb3rt`	`dlynes_office: lol you speak french?`
`19:18.51`	`dlynes_office`	`Qb3rt: a little bit...I speak more Mandarin now though`
`19:18.58`	`dlynes_office`	`Qb3rt: I live in BC...not many French people here`
`19:19.01`	`Qb3rt`	`dlynes_office: yeah but they dont care about the futurshop name... they care about bestbuy now`
`19:19.02`	`dlynes_office`	`lots of Chinese, though`
`19:19.14`	`jetway2006`	`hi ppl....know abt gsm terminal`
`19:19.24`	`dlynes_office`	`yeah...bestbuy sucks though`
`19:19.31`	`dlynes_office`	`futureshop's prices are much better`
`19:19.32`	`Strom_C`	`jetway2006: dont ask the same question over and over again please`
`19:19.45`	`jetway2006`	`ok...`
`19:19.45`	`Qb3rt`	`dlynes_office: no! i like bestbuy they have good prices...`
`19:20.03`	`dlynes_office`	`Qb3rt: I got a much better deal on my laptop at futureshop`
`19:20.09`	`Qb3rt`	`dlynes_office: smaller store than bestbuy`
`19:20.11`	`jetway2006`	`http://www.buymin.com/`
`19:20.22`	`dlynes_office`	`Qb3rt: and I didn't have to resort to buying a crappy hp`
`19:20.28`	`jetway2006`	`any experience with this provider`
`19:20.38`	`dlynes_office`	`Qb3rt: bestbuy promotes the hell out of those crappy hp laptops`
`19:20.45`	`jetway2006`	`where could i find a good voip termination ...`
`19:20.58`	`dlynes_office`	`jetway2006: www.calltermination.com`
`19:21.03`	`*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)`
`19:21.04`	`Qb3rt`	`dlynes_office: hehehe yeah`
`19:21.06`	`jetway2006`	`are voip provider and voip termination the same`
`19:21.11`	`dlynes_office`	`jetway2006: pick your poison...there's like five hundred voip terminators on there`
`19:21.20`	`dlynes_office`	`jetway2006: yes`
`19:21.52`	`jetway2006`	`oh...i get confused...with them...could u recommend any company...from your experience`
`19:22.01`	`jetway2006`	`dont know whom to choose`
`19:22.06`	`dlynes_office`	`jetway2006: for residential market they're call voip providers; for wholesale market, they're called voip terminators`
`19:22.17`	`dlynes_office`	`same crap, different pile`
`19:22.45`	`dlynes_office`	`jetway2006: for retail, or wholesale?`
`19:22.50`	`jetway2006`	`hahah...yeah`
`19:22.56`	`jetway2006`	`wholesale`
`19:23.13`	`dlynes_office`	`jetway2006: try Five 9s Network...I've found them to be relatively reliable`
`19:23.14`	`jetway2006`	`it will be cheaper than residential`
`19:23.18`	`dlynes_office`	`Quite cheap, too`
`19:23.27`	`jetway2006`	`website?`
`19:23.39`	`dlynes_office`	`jetway2006: yeah, but wholesalers expect you to have certain minimum minutes every month`
`19:23.53`	`dlynes_office`	`www.five9snetwork.com`
`19:24.07`	`jetway2006`	`oh...i get it...minimum requirement`
`19:24.07`	`jetway2006`	`ok`
`19:24.09`	`jetway2006`	`thx`
`19:24.13`	`dlynes_office`	`and the higher your minimum is`
`19:24.19`	`dlynes_office`	`the better your pricing structure`
`19:24.27`	`Qb3rt`	`<dlynes_office> www.five9snetwork.com lol nice template!!`
`19:24.45`	`dlynes_office`	`generally the best pricing structure is when you have 1M minutes per month, or more`
`19:25.00`	`dlynes_office`	`Qb3rt: heh...I like ours better`
`19:25.09`	`dlynes_office`	`Qb3rt: we've got 247communications.com and a2zcommunications.com :)`
`19:25.40`	`jetway2006`	`another question...it is beter to find a provider nearner to my location or a cheaper provider but located far`
`19:25.54`	`dlynes_office`	`jetway2006: check the ping times on their sip server`
`19:26.09`	`dlynes_office`	`jetway2006: if the ping times are too high, find someone else`
`19:26.30`	`dlynes_office`	`jetway2006: you can get low ping times from providers far away, but on the same continent, too`
`19:27.06`	`dlynes_office`	`jetway2006: generally you want under 100ms ping times`
`19:27.24`	`jetway2006`	`hmmm..interesting...how abt stimultaneous calls....how many calls do they allow..`
`19:27.37`	`dlynes_office`	`jetway2006: depends on how many ports you ask them to open for you`
`19:27.40`	`jetway2006`	`is there any standard among provider`
`19:28.15`	`dlynes_office`	`we've got ten ports open right now, and I think we only use 3 or 4 of them`
`19:28.59`	`jetway2006`	`do u pay extra for number of ports`
`19:29.23`	`dlynes_office`	`I don't, but then again, they get their main wholesale phone line through us`
`19:29.35`	`dlynes_office`	`so we get special treatment`
`19:29.52`	`*** join/#asterisk nomego (i=boink@1-1-13-33a.sh.sth.bostream.se)`
`19:29.59`	`jetway2006`	`main wholesale phone line??`
`19:29.59`	`nomego`	`I want to replace my regular phone with some kind of voip-solution to be able to get regular phone-calls into mythphone and some wireless sip-phone.. is an usb-modem and asterisk the right way to go?`
`19:30.07`	`dlynes_office`	`they're based out of vancouver, but their main server is in toronto`
`19:30.14`	`Strom_C`	`nomego: "usb-modem"?`
`19:30.25`	`dlynes_office`	`Yeah...their 604-628-0029 number is coming in on our pri`
`19:30.33`	`Strom_C`	`you want a digium TDM card, my friend`
`19:30.37`	`dlynes_office`	`we just forward it off to their talkswitch pbx`
`19:30.46`	`nomego`	`Strom_C: yeah.. it's just a single analog phone line to my home`
`19:31.08`	`jetway2006`	`oh...ok...got it...how about usually providers...do need to pay extra for number of ports`
`19:31.08`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`19:31.09`	`nomego`	`shouldn't just a voice usb modem suffice?`
`19:31.11`	`Strom_C`	`nomego: digium tdm400`
`19:31.17`	`Strom_C`	`nomego: you cant use a modem`
`19:31.41`	`dlynes_office`	`jetway2006: i don't know...I would imagine that they'll only let you have a certain number of ports open, all dependent on how much traffic you generate`
`19:31.42`	`nomego`	`why not?`
`19:32.07`	`dlynes_office`	`i.e. they won't charge you so much per port, but will only open so many ports, depending on how many minutes you expect to generate per month`
`19:32.24`	`Strom_C`	`nomego: no drivers, shitty audio quality, etc etc etc`
`19:32.51`	`dlynes_office`	`Strom_C: whatchu talking about mang?`
`19:32.54`	`jetway2006`	`in the 5 9 rates....there is direct and white route...what is it....the rates..are very much cheaper than other providers`
`19:32.56`	`dlynes_office`	`Strom_C: they've got drivers`
`19:33.02`	`Strom_C`	`dlynes_office: shhhh`
`19:33.02`	`dlynes_office`	`Strom_C: nomego's going to write one`
`19:33.05`	`Strom_C`	`:)`
`19:33.07`	`nomego`	`Strom_C: in any way, I need something external (preferably USB), small (to hide behind my tv bench), cheap, and supported by asterisk and linux`
`19:33.22`	`Strom_C`	`nomego: for the analog line?`
`19:33.25`	`Strom_C`	`or for an analog phone`
`19:33.36`	`dlynes_office`	`nomego: just get a sipura 3000 and be done with it`
`19:33.40`	`Strom_C`	`for a phone, use a sipura or something`
`19:33.56`	`nomego`	`Strom_C: I have an analog phone, but I want to get rid of it and use SIP-phones or MythPhone instead`
`19:34.11`	`Strom_C`	`nomego: so then get a sipura spa-3000 or a digium tdm400`
`19:34.14`	`dlynes_office`	`nomego: at retail rates, a sipura 3000 is a hell of a lot cheaper than going with a tdm card`
`19:34.15`	`Strom_C`	`problem solved`
`19:34.31`	`dlynes_office`	`nomego: and probably more reliable than a tdm400p`
`19:34.43`	`*** join/#asterisk j3g (n=rafael@200.130.8.1)`
`19:34.55`	`nomego`	`alright, thanks alot`
`19:35.02`	`j3g`	`what is the best voip gateway to talk to an asterisk box? pap2? something from sipura?`
`19:35.06`	`jetway2006`	`Mr. dlynes..`
`19:35.23`	`anonymouz666`	`Strom_C: should I use callerid=asreceived ?`
`19:35.40`	`jetway2006`	`how abt me....pls explain the direct and white routes...i dont understand..`
`19:36.46`	`Strom_C`	`j3g: gateway for an analog telephone set?`
`19:37.00`	`Strom_C`	`anonymouz666: do you have the tdm400 in now?`
`19:37.05`	`anonymouz666`	`yes`
`19:37.18`	`Strom_C`	`is it working with the settings we originally tried?`
`19:37.54`	`anonymouz666`	`its with dtmf and ring.`
`19:38.06`	`anonymouz666`	`i tested with no callerid in the group config.`
`19:38.17`	`anonymouz666`	`should I use callerid=asreceived?`
`19:38.24`	`Strom_C`	`anonymouz666: yes`
`19:39.13`	`anonymouz666`	`testing`
`19:39.35`	`nomego`	`so how would I handle a linksys/sipura spa-3000? I connect one end to the pstn and then a network cable to my network and then configure it with an ip-address that I add as a channel in asterisk?`
`19:39.54`	`Strom_C`	`essentially, yes`
`19:39.56`	`j3g`	`Strom_C: yes`
`19:40.19`	`Strom_C`	`j3g: linksys pap2, sipura spa-whatever, digium iaxy`
`19:40.21`	`Strom_C`	`they're all good`
`19:40.22`	`nomego`	`nice, I'll have to look into that`
`19:40.29`	`anonymouz666`	`nada`
`19:40.32`	`anonymouz666`	`Jul 4 16:42:32 NOTICE[30150]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)...`
`19:42.02`	`nomego`	`would anyone know if the spa-3000 works in sweden?`
`19:42.35`	`anonymouz666`	`Strom_C dtmf, ring and callerid=asreceived`
`19:42.44`	`tzafrir_laptop`	`nomego, I figure that our fxs 8 ports usb box is not for you and you're basically looking for something like the discontinued digium s100u, right?`
`19:43.09`	`Strom_C`	`tzafrir_laptop: he wants FXO, not FXS`
`19:44.10`	`*** join/#asterisk BugKham (i=CKGLOB@125.24.1.76)`
`19:44.14`	`tzafrir_laptop`	`I heard someone here designing a cheaper (price in the order of magnitude of 100$) box of 4 FXS ports) but I didn't see anything with it`
`19:44.28`	`BugKham`	`where can I download mpg123 for x86_64?`
`19:45.26`	`dlynes_office`	`direct is routes they own; white routes are legal routes into countries where they often have routes with questionable legal implications`
`19:45.37`	`dlynes_office`	`jetway2006: direct is routes they own; white routes are legal routes into countries where they often have routes with questionable legal implications`
`19:45.44`	`*** join/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com)`
`19:46.24`	`tzafrir_laptop`	`one fxo port through USB... I saw someone here who was working on zaptel drivers for a certain USB modem. I also believe that coppice was working on chan_unicall drivers for something similar`
`19:46.25`	`dlynes_office`	`nomego: yes, it should`
`19:46.26`	`BugKham`	`just tried to compile * on my new 64-bit machine and had errors`
`19:46.28`	`anonymouz666`	`Strom_C: it will work if TELCO sends the callerid by FSK?`
`19:46.29`	`seb-`	`possible to 1. call someone 2. put them on hold 3. call someone else 4. talk to both at same time?`
`19:46.37`	`tzafrir_laptop`	`BugKham, why do you need mpg123?`
`19:46.46`	`Strom_C`	`seb-: yes`
`19:46.47`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.74)`
`19:46.49`	`nomego`	`dlynes_office: thanks`
`19:46.53`	`Strom_C`	`anonymouz666: oh god i dont know anymore`
`19:47.12`	`Strom_C`	`anonymouz666: pay me consulting fees and ill work on it for you`
`19:47.15`	`seb-`	`Strom_C: where is a HOWTO for this most simplest of 3way calling?`
`19:47.16`	`BugKham`	`tzafrir_laptop: hmm, isn't it required for playing mp3?`
`19:47.21`	`Strom_C`	`seb-: um`
`19:47.25`	`seb-`	`Strom_C: or whatever you call it`
`19:47.30`	`Strom_C`	`seb-: are you using a tdm card?`
`19:47.37`	`BugKham`	`tzafrir_laptop: I generally use mp3 for music on hold and stuff`
`19:47.38`	`seb-`	`Strom_C: what 's that?`
`19:47.56`	`seb-`	`Strom_C: i'm a voip newb`
`19:47.57`	`Strom_C`	`seb-: what kind of phone are you using`
`19:48.12`	`seb-`	`Strom_C: all i got is a gizmo to convert pots phone to digital`
`19:48.22`	`Strom_C`	`WHAT THE HELL IS IT CALLED`
`19:48.31`	`Strom_C`	`model number`
`19:48.32`	`seb-`	`Strom_C: Grandstream Handyphone 286`
`19:48.33`	`anonymouz666`	`Strom_C: digium already do that :D`
`19:48.36`	`Strom_C`	`manufacturer`
`19:48.50`	`seb-`	`Strom_C: Handytone*`
`19:48.53`	`Strom_C`	`seb-: it works just like three way calling on your home phone then`
`19:48.56`	`Strom_C`	`dial`
`19:48.59`	`Strom_C`	`wait for supervision`
`19:49.02`	`jetway2006`	`hi...dlynes...do you have the server ip..so i could test the ping time`
`19:49.02`	`Strom_C`	`flash`
`19:49.04`	`Strom_C`	`dial again`
`19:49.07`	`Strom_C`	`wait for supervision`
`19:49.08`	`Strom_C`	`flash`
`19:49.09`	`Strom_C`	`DONE`
`19:49.17`	`tzafrir_laptop`	`BugKham, do you need to stream mp3 music? If not: convert the sound fies to wav/gsm/speex and use native moh`
`19:49.47`	`seb-`	`Strom_C: what is "flash" ?`
`19:49.54`	`Strom_C`	`hookflash`
`19:50.04`	`Strom_C`	`momentarily depress the hookswitch`
`19:50.15`	`Strom_C`	`or use the button labeled FLASH`
`19:50.18`	`seb-`	`Strom_C: are you saying my Handytone has 3 way calling built in?`
`19:50.35`	`Strom_C`	`seb-: why not just TRY IT`
`19:50.38`	`Strom_C`	`AAAHHHHHHH`
`19:50.49`	`seb-`	`Strom_C: aren't newbs fun? :)`
`19:51.13`	`BugKham`	`tzafrir_laptop: hmm, so mpg123 doesn't support 64-bit CPU?`
`19:51.20`	`tzafrir_laptop`	`hmmm not speex`
`19:51.24`	`seb-`	`Strom_C: i don't have asterisk yet...i'm using a voip provider`
`19:51.30`	`seb-`	`Strom_C: will it still work?`
`19:51.40`	`Strom_C`	`seb-: just fucking try it already`
`19:51.43`	`nomego`	`how about the linksys spa3102-eu, would that work as well as sipura spa-3000 with asterisk?`
`19:51.44`	`seb-`	`ok`
`19:53.14`	`BugKham`	`tzafrir_laptop: it's just easier playing with mp3 that's all`
`19:56.19`	`rob0`	`BugKham: * is wonderful on my x86_64 machine ... aaahhhhhhhh ... no more IRQ problems :)`
`19:58.45`	`tzafrir_laptop`	`http://packages.debian.org/unstable/sound/mpg123 . amd64 is not there. It's an unmaintained package`
`19:59.37`	`tzafrir_laptop`	`(labled as "nonfree" because the author won't do the little required work to change the license officially)`
`20:01.16`	`tzafrir_laptop`	`rob0, an amd64 system is not that different from an i386 system. Same BIOS. Same baic interrupts handling, right?`
`20:03.05`	`*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)`
`20:04.55`	`*** join/#asterisk dlynes_office (n=dlynes@216.251.149.66)`
`20:13.47`	`*** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net)`
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`20:16.32`	`*** join/#asterisk jetaway2007 (n=asd@60.48.48.183)`
`20:19.04`	`rob0`	`tzafrir_laptop: http://pastebin.ca/79020`
`20:20.03`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`20:21.55`	`tzafrir_laptop`	`rob0, lspci --help; lspci -vb`
`20:22.15`	`tzafrir_laptop`	`and look for IRQ in the otput of te latter`
`20:24.59`	`seb-`	`Strom_C: thanks for help...i think i got it`
`20:25.03`	`rob0`	`The FXO says IRQ 9, the TDM says IRQ 5.`
`20:25.05`	`Strom_C`	`yay`
`20:32.05`	`anonymouz666`	`Strom_C: doesn't work callerid here. there is nothing else I can do.`
`20:32.17`	`anonymouz666`	`I tried almost everything`
`20:34.01`	`Strom_C`	`anonymouz666: I wish I knew how caller ID worked in brazil`
`20:34.56`	`anonymouz666`	`analog line sucks`
`20:35.30`	`*** join/#asterisk bnafh (n=derek@c-67-185-114-199.hsd1.wa.comcast.net)`
`20:36.19`	`*** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl)`
`20:36.20`	`kFuQ-lap`	`question: what exactly in the hell does this mean ?`
`20:36.23`	`kFuQ-lap`	`Jul 4 13:35:48 WARNING[29148]: app_voicemail.c:2105 messagecount: Failed to obtain database object for 'asterisk'!`
`20:36.35`	`kFuQ-lap`	`besides the obvious :-D`
`20:38.40`	`*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)`
`20:39.19`	`kFuQ-lap`	`grrrrrrrrrr....`
`20:39.23`	`kFuQ-lap`	`dam realtime voicemail`
`20:40.10`	`Dovid`	`:)`
`20:40.37`	`kFuQ-lap`	`google hasn't been any help with that error either`
`20:40.39`	`kFuQ-lap`	`:\`
`20:42.30`	`Dovid`	`Whats the error ?`
`20:42.52`	`dlynes_office`	`<kFuQ-lap> Jul 4 13:35:48 WARNING[29148]: app_voicemail.c:2105 messagecount: Failed to obtain database object for 'asterisk'!`
`20:43.24`	`Dovid`	`I know this is obvoius but it isnt connecting to ur db`
`20:43.27`	`Dovid`	`What r u using ?`
`20:47.27`	`nomego`	`the linksys spa3102-eu, would that work as well as sipura spa-3000 with asterisk?`
`20:47.59`	`nomego`	`it feels like it should be essentially the same piece of equipment..`
`20:52.13`	`nomego`	`and how about the D-Link DVG-2001S/E, is that the same type of equipment? does it work with asterisk? is it stable?`
`21:04.14`	`*** join/#asterisk ozverenm (n=ozverenm@125.27.103-84.rev.gaoland.net)`
`21:04.22`	`ozverenm`	`hello all`
`21:04.34`	`*** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com)`
`21:04.45`	`Strom_C`	`hello`
`21:04.49`	`ozverenm`	`I have a samgoma A102 pri card and have some problems`
`21:05.38`	`ozverenm`	`when I activate zaptel DACS functionality, sometimles CPU goes up to 60% on a P4 3ghz hardware`
`21:05.55`	`ozverenm`	`did'nt understant why`
`21:06.12`	`ozverenm`	`When I done tests on my telco access, no problem`
`21:06.24`	`ozverenm`	`but on some sites, CPU load is very high`
`21:06.47`	`ozverenm`	`what about echo cancelling in zaptel module ?`
`21:06.56`	`mpruett`	`Does anyone know what the "setvar" field is for on the realtime sipusers table is for?`
`21:17.33`	`*** join/#asterisk blebleble (i=godie@caesar.godie.net)`
`21:18.26`	`*** join/#asterisk Icheb (n=icheb@sebsoft.xs4all.nl)`
`21:19.15`	`Icheb`	`Anyone here has an idea on how to use the manager api to issue '!<shellcommand>', it doesn't seem to be working for some reason`
`21:20.48`	`Skarmeth`	`hi all`
`21:21.21`	`anonymouz666`	`tzafrir_laptop?`
`21:21.36`	`Strom_C`	`pls 2 help me asterisk my internets together`
`21:21.37`	`Skarmeth`	`I am fighting with my telco and my E1, and after been able to call non-local numbers, now I just get this ' Channel 0/1, span 1 received AOC-E charging 0 units'`
`21:21.44`	`Skarmeth`	`when I try to make a call`
`21:21.51`	`Skarmeth`	`what it means?`
`21:22.07`	`anonymouz666`	`Strom_C needs a TODO list :PPP`
`21:22.14`	`Strom_C`	`haha`
`21:22.21`	`Strom_C`	`im just being silly`
`21:22.23`	`tzafrir_laptop`	`anonymouz666, here as well`
`21:23.09`	`anonymouz666`	`tzafrir: I can't put CallerID (DTMF) to work here in Brazil on a TDM card.`
`21:23.21`	`anonymouz666`	`I just tried almost everything`
`21:23.34`	`anonymouz666`	`I am thinking that I need something to convert from DTMF->FSK`
`21:24.54`	`Skarmeth`	`anonymouz666, you will, look at asteriskbrasil.org mailing list`
`21:25.20`	`anonymouz666`	`everyone there uses something to convert from DTMF to FSK`
`21:26.06`	`*** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com)`
`21:29.26`	`*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)`
`21:29.27`	`tzafrir_laptop`	`anonymouz666, is the problem sening caller ID or recieving caller ID?`
`21:29.38`	`anonymouz666`	`receiving`
`21:29.41`	`anonymouz666`	`from telco`
`21:30.07`	`tzafrir_laptop`	`And what's the problem?`
`21:30.09`	`*** part/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net)`
`21:30.40`	`anonymouz666`	`well, I can't see in Asterisk what caller id is`
`21:30.53`	`anonymouz666`	`so i'dont know who is calling me.`
`21:33.35`	`*** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR)`
`21:33.56`	`anonymouz666`	`Strom_C`
`21:34.09`	`nXOR`	`hello people, will sangoma a200 card allow me to connect my internal network to PSTN line outside, i have an nt switch which has 2 nalog and 2 digital ports`
`21:34.21`	`nXOR`	`im thinking of buying one`
`21:34.28`	`anonymouz666`	`I pay you 300 dollars. but If you can't get this working, you pay me 600. What about?`
`21:34.45`	`nXOR`	`anyone have any experience`
`21:35.36`	`nXOR`	`ok off to watch some tv`
`21:35.39`	`*** part/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR)`
`21:39.14`	`tzafrir_laptop`	`anonymouz666, just to avoid the obvious: you did set 'callerid=asrecieved' in zapata.conf, right?`
`21:39.20`	`anonymouz666`	`sure`
`21:39.44`	`*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)`
`21:40.05`	`tzafrir_laptop`	`Any chance the telco does not send callerid?`
`21:40.10`	`anonymouz666`	`no chance.`
`21:40.25`	`anonymouz666`	`I have another telephone in here that shows the number when I connect that line.`
`21:40.27`	`tzafrir_laptop`	`you can record the line with ztmonitor`
`21:40.58`	`tzafrir_laptop`	`you'll hear the caller ID as a short fax-like voice`
`21:43.59`	`anonymouz666`	`you mean -a option?`
`21:47.26`	`*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)`
`21:50.32`	`*** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com)`
`21:52.10`	`anonymouz666`	`tzafrir I don't think that this can have any utility`
`21:52.16`	`anonymouz666`	`I know it's coming`
`21:53.06`	`tzafrir_laptop`	`"utility"?`
`21:53.27`	`tzafrir_laptop`	`ah, ok`
`21:54.31`	`tzafrir_laptop`	`The funny thing is that the chip that the tdm card uses should be able to detect caller id on its own`
`21:55.13`	`tzafrir_laptop`	`but that capability is not used`
`21:55.20`	`*** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)`
`21:56.34`	`tzafrir_laptop`	`Do you see anything interesting in the debug trace of asterisk when you enable debug logging?`
`21:56.54`	`Dovid`	`Tzafir: boker tov`
`21:58.18`	`l-fy`	`is there anyone here using sangoma A104d?`
`21:59.32`	`*** join/#asterisk adorah (n=Administ@87.69.72.228)`
`22:00.17`	`anonymouz666`	`tzafrir set debug 1`
`22:00.39`	`anonymouz666`	`and nothing special shows`
`22:00.58`	`tzafrir_laptop`	`set debug 5`
`22:01.07`	`anonymouz666`	`after that i set debug 255`
`22:01.12`	`anonymouz666`	`just to be sure`
`22:01.13`	`tzafrir_laptop`	`or even more. 1 misses some things`
`22:04.33`	`anonymouz666`	`set debug 255`
`22:04.39`	`anonymouz666`	`no hint`
`22:05.13`	`Dovid`	`Tzafir: is sangoma any better than the digium tdm400p ?`
`22:08.03`	`Dovid`	`Tzafir: do u know if I can set CID here in Israel ? I was allways told no and just got a call from the us and the CID was +718XXXXXXX`
`22:08.31`	`Strom_C`	`718 == brooklyn :)`
`22:08.48`	`Dovid`	`I know that`
`22:08.58`	`Dovid`	`I want to know what providers here let u set ur own CID`
`22:09.16`	`Dovid`	`Its something pretty new`
`22:10.43`	`Dovid`	`And I would luv to be able to set Cid here`
`22:10.44`	`Dovid`	`CID*`
`22:12.15`	`mishehu`	`Dovid: cid is new in .il ?`
`22:12.37`	`mishehu`	`since when? I had cid when I had isdn back in 1999...`
`22:12.54`	`mishehu`	`or did they finally add the name field?`
`22:12.57`	`*** join/#asterisk Dimitripietro (i=Wut@modemcable069.5-202-24.mc.videotron.ca)`
`22:13.01`	`Dovid`	`lol`
`22:13.16`	`Dovid`	`Nno. I never had a (VOIP) provider that let me set the CID`
`22:13.29`	`mishehu`	`Dovid: oh that, I thought you were talking about bezeq`
`22:13.42`	`*** part/#asterisk tlow (n=tlowe@bgp.terrorist.net)`
`22:13.53`	`Dovid`	`nah`
`22:13.59`	`Dimitripietro`	`<Dovid> Here in Canada we have Unlimitel that let us set the callerid on the fly`
`22:14.03`	`Dovid`	`So u can set the CID with a bezeq ISDN ?`
`22:14.24`	`Dovid`	`Yes. I can set CID to the US. Just cant here`
`22:15.09`	`mishehu`	`Dovid: you mean that when you call a phone in .il via a VoIP provider, the person you call does not see the CID that you set and instead a bunch of gibberish?`
`22:15.18`	`*** part/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com)`
`22:15.30`	`tzafrir_laptop`	`I figure Bezeq won't let you set CID on analog lines`
`22:15.30`	`Dovid`	`yup`
`22:15.43`	`Dovid`	`Lol. Not analog.`
`22:15.46`	`tzafrir_laptop`	`(technically: will ignore the value you et)`
`22:16.01`	`Dovid`	`But some one called me with a calling from the US and I got the US CID`
`22:16.31`	`mishehu`	`Dovid: I don't know what protocol bezeq uses for cid, but the only time I ever received accurate cid from the US was once when somebody called me from florida.`
`22:16.43`	`tzafrir_laptop`	`on PRI: you should be able to set the CID , at least within a limit. Not sure exactly what they do`
`22:16.46`	`mishehu`	`I think there's an incompatibility between bezeq and US cid`
`22:16.49`	`Dovid`	`hmm`
`22:17.08`	`Dovid`	`hmm`
`22:17.13`	`mishehu`	`tzafrir_laptop: have they started supporting cidname in .il ?`
`22:17.14`	`Dovid`	`Is it how asterisk sends it ?`
`22:17.33`	`mishehu`	`I was in a year ago and I didn't pay attention to the phones there`
`22:17.37`	`Dovid`	`Hehe. Gona be a while cause all the boxes are english based and they goto have em support ivrit`
`22:17.43`	`tzafrir_laptop`	`mishehu, no idea`
`22:18.03`	`Dovid`	`So if I get ISDN mishehu I can set the CID to what ever I want ?`
`22:18.14`	`Dovid`	`I want it when people call me from the US I can see who it is on my cell`
`22:18.28`	`tzafrir_laptop`	`Dovid, is UTF-8 supported in CID?`
`22:18.28`	`mishehu`	`Dovid: that was back in 1999. it may or many not allow you to override your outbound cid.`
`22:18.41`	`Dovid`	`Dont know what that is`
`22:18.43`	`mishehu`	`Dovid: I haven't done work with bezeq since a year after I got out of tsahal`
`22:18.44`	`tzafrir_laptop`	`Asterisk sends cidname just fine on analog lines`
`22:19.05`	`mishehu`	`utf8 is a codepage, for encoding characters for textual display`
`22:19.06`	`Dovid`	`?`
`22:19.31`	`Dovid`	`Tzafir: if I have a bezeq analog I can set the CID Name here now ?`
`22:20.17`	`mishehu`	`Dovid: it doesn't sound like the infrastructure supports cid name display there.`
`22:20.21`	`tzafrir_laptop`	`IIRC it won't be set`
`22:20.39`	`Dovid`	`ok`
`22:20.53`	`Dovid`	`So basicly what do I need to set my own CID number here ? ISDN ?`
`22:21.08`	`tzafrir_laptop`	`At least from what we tested with our boxesL they can set cidname/num on FXO, but it is not set on the FXO lines`
`22:21.14`	`Dovid`	`Whats the cheapest way and are there VOIP providers that let me set the CID here in.il`
`22:21.38`	`Dovid`	`Tazfir: set it meaning ? In hebrew or over bezeq ?`
`22:22.25`	`tzafrir_laptop`	`hebrew: I am yet to see hebrew caller id anywhere`
`22:22.44`	`tzafrir_laptop`	`over bezeq: again: doesn't work here`
`22:22.51`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`22:23.15`	`tzafrir_laptop`	`is CID limited to ascii or is UTF-8 used?`
`22:23.36`	`mishehu`	`we'd have to consult the spec`
`22:23.40`	`mishehu`	`and there might be more than one spec`
`22:24.00`	`mishehu`	`I'd imagine that if it didn't support it originally, there is a newer spec that allows for utf8`
`22:24.24`	`Dovid`	`mishehu: whats the cheapest way so I can set CID here ?`
`22:24.25`	`tzafrir_laptop`	`Are accented latin characters supported? cirrylic?`
`22:25.02`	`mishehu`	`gah, prepared statements are annoying me at the moment...`
`22:25.26`	`mishehu`	`Dovid: I've been too far removed from .il to be able to give good advice on that.`
`22:25.36`	`mishehu`	`been gone for about 5 years.`
`22:25.53`	`mishehu`	`shit, they built kvish 6 in my absence`
`22:25.55`	`Dovid`	`ok`
`22:25.58`	`tzafrir_laptop`	`well, off to bed...`
`22:25.59`	`mishehu`	`and a new natba"g`
`22:26.03`	`Dovid`	`Do u remmebr the rates for ISDN ?`
`22:26.07`	`Dovid`	`hehe`
`22:26.15`	`Dovid`	`Took em for ever to do that`
`22:26.22`	`Dovid`	`Layla tov tzafir`
`22:26.37`	`tzafrir_laptop`	`good night`
`22:26.44`	`mishehu`	`Dovid: I'm sure they're not relevant to today's pricing. I still have an old nt1 device and a fritzcard pci isdn card`
`22:34.05`	`_4d4m_`	`hi all. can anyone tell me why the t extension and ResponseTimout dont work in this model, whilst everything else does? - http://pastebin.ca/79106`
`22:34.05`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`22:34.05`	`Dovid`	`Model /`
`22:34.06`	`_4d4m_`	`nm.. found out`
`22:34.06`	`_4d4m_`	`autofallthrough=yes was my problem fwiw`
`22:34.06`	`*** join/#asterisk jetway2008 (n=asd@218.111.221.11)`
`22:35.03`	`jetway2008`	`hi`
`22:50.39`	`*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)`
`22:52.52`	`*** join/#asterisk techie (n=gus@voipops.net)`
`22:55.32`	`*** join/#asterisk AuPix (n=AuPix@adsl-04-85.abel.net.uk)`
`22:57.01`	`AuPix`	`Anyone know how to get trunk cdr_addon_mysql compiled when mysql hasn't been installed from source?`
`22:58.32`	`MikeJ__`	`AuPix, you should just need headers`
`22:58.52`	`MikeJ__`	`there is probably a -devel package that goes with it that includes what you need`
`22:59.23`	`AuPix`	`Hmmm... I can't get the configure script to find myql when its in /usr/lib/mysql and /usr/include/mysql`
`23:00.07`	`AuPix`	`There also seems to be an issue with the script trying to load -lmysqlclient when all I have is libmysqlclient`
`23:00.11`	`Agrajag-`	`does iax have presence capabilities?`
`23:00.56`	`AuPix`	`MikeJ it all worked fine before the updates to menuselect etc`
`23:01.16`	`MikeJ__`	`dunno.... what dir's is it looking in now for it?`
`23:02.01`	`MikeJ__`	`the -l looks fine to me`
`23:02.10`	`AuPix`	`Well I can do the configure --with-mysqlclient= something but the configure script seems to want to then look in that dir /lib and /include`
`23:02.43`	`AuPix`	`and my install has /usr/include/mysql and /usr/lib/mysql`
`23:05.08`	`AuPix`	`MikeJ do you have the trunk cdr_addon_mysql working?`
`23:05.33`	`MikeJ__`	`no.. havent tried`
`23:05.44`	`MikeJ__`	`but should be easy enough to add to the dirs it searches in...`
`23:05.56`	`MikeJ__`	`just patch configure.ac`
`23:06.08`	`russellb`	`./configure --with-mysqlclient=/usr/lib/mysql`
`23:06.10`	`russellb`	`that doesn't do it?`
`23:06.23`	`MikeJ__`	`or.. patch russellb`
`23:06.27`	`russellb`	`argh ... dinner time ...`
`23:06.36`	`AuPix`	`russellb no I tried /usr/lib/mysql`
`23:06.52`	`russellb`	`well why are they in /usr/lib/mysql?`
`23:06.54`	`russellb`	`that's sillyness`
`23:07.00`	`AuPix`	`:-)`
`23:07.15`	`AuPix`	`That's where my CentOS install put them :-(`
`23:07.19`	`russellb`	`ugh`
`23:07.23`	`russellb`	`that means i'm going to have to address it`
`23:07.27`	`MikeJ__`	`heh`
`23:07.30`	`russellb`	`stupid fedora/redhat/centos/crap`
`23:07.41`	`AuPix`	`:-)`
`23:07.45`	`MikeJ__`	`russellb broke it!`
`23:07.51`	`russellb`	`centos broke it`
`23:07.54`	`MikeJ__`	`easy fix tho`
`23:08.16`	`AuPix`	`I'll have a look at configure.ac... thanks all.`
`23:09.18`	`russellb`	`and note, i didn't break it`
`23:09.26`	`russellb`	`asterisk-addons in trunk was completely broken before i got to it`
`23:09.29`	`russellb`	`i made it working again :)`
`23:10.43`	`*** join/#asterisk Asterisk_Newbie (n=a_ti_tu_@bl7-133-238.dsl.telepac.pt)`
`23:10.54`	`Asterisk_Newbie`	`Hi all`
`23:12.01`	`AuPix`	`Russelb, can trunk be updated simply, or should I look at it myself in the meantime?`
`23:12.01`	`MikeJ__`	`yeah yeah yeah.. that's what they all say :P`
`23:12.13`	`Asterisk_Newbie`	`I have a cool idea for asterisk that i need someone to confirm me if it's possible`
`23:12.20`	`Asterisk_Newbie`	`can i put in on the channel?`
`23:13.19`	`AuPix`	`I guess you didn't mean that you'd fixed my issue already :-)`
`23:13.24`	`l-fy`	`Asterisk_Newbie > try`
`23:13.34`	`Asterisk_Newbie`	`I from Portugal and i think that i'm lagged`
`23:13.49`	`l-fy`	`Asterisk_Newbie > ok`
`23:13.52`	`Asterisk_Newbie`	`supose that you have a small company`
`23:13.56`	`jetway2008`	`does any here could pls recommend a good voip termination provider?`
`23:14.22`	`Asterisk_Newbie`	`with 40 people who works in a shop-floor`
`23:14.23`	`l-fy`	`jetway2008 > which country?`
`23:14.48`	`jetway2008`	`to malaysia`
`23:14.55`	`jetway2008`	`sri lanka ,india`
`23:14.59`	`Asterisk_Newbie`	`they need to fill a form with their number, the order number and the time they took to do it`
`23:15.31`	`Asterisk_Newbie`	`I'm thinking on using IVR from asterisk`
`23:15.43`	`Asterisk_Newbie`	`so they dial a special extension like 100`
`23:15.58`	`l-fy`	`Asterisk_Newbie > ok, what's the problem in doing that?`
`23:16.01`	`Asterisk_Newbie`	`and the asterisk server ask them the order number`
`23:16.07`	`Asterisk_Newbie`	`then their number`
`23:16.12`	`Asterisk_Newbie`	`then the time they took`
`23:16.28`	`Asterisk_Newbie`	`every value they dial will be saved on a database`
`23:16.54`	`Asterisk_Newbie`	`it's possible to store this values on a databse using a dialplan?`
`23:17.18`	`l-fy`	`Asterisk_Newbie > no`
`23:17.18`	`l-fy`	`people use agi for that`
`23:17.19`	`Asterisk_Newbie`	`hope i make myself clear`
`23:17.20`	`Asterisk_Newbie`	`agi...`
`23:17.24`	`Asterisk_Newbie`	`mmmm thanks`
`23:17.33`	`Asterisk_Newbie`	`i will try to read something about that`
`23:19.57`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
`23:21.48`	`Asterisk_Newbie`	`I want to thanks l-fy. I think agi will do the job`
`23:22.02`	`*** join/#asterisk Kis (i=vlad@p5080F520.dip.t-dialin.net)`
`23:22.44`	`Asterisk_Newbie`	`I have to learn how to program :P`
`23:23.51`	`jetway2008`	`does anyyoe know a good voip termination`
`23:24.17`	`l-fy`	`Asterisk_Newbie > use fastagi`
`23:26.52`	`*** join/#asterisk syle (n=blah@unaffiliated/syle)`
`23:26.54`	`*** join/#asterisk peanuter (n=peanuter@216.176.177.138)`
`23:27.40`	`Asterisk_Newbie`	`thanks again l-fy, i'm already googling fastagi`
`23:28.12`	`peanuter`	`does anyone know who runs nufone.net ?`
`23:29.49`	`*** join/#asterisk nagl (n=nagl@86.59.54.237)`
`23:33.14`	`*** join/#asterisk mog (i=ejabberd@68.62.237.103)`
`23:34.25`	`l-fy`	`peanuter > the moron named JerJer`
`23:34.36`	`l-fy`	`never ever asume that nufone works`
`23:34.37`	`peanuter`	`ya figured that out`
`23:34.38`	`peanuter`	`thank you though`
`23:34.40`	`l-fy`	`because it dosen't`
`23:34.44`	`russellb`	`l-fy: personal attacks are not acceptable`
`23:34.46`	`peanuter`	`thats fine :)`
`23:35.00`	`peanuter`	`russellb: truth hurts :)`
`23:35.06`	`l-fy`	`russellb > JerJer was started first`
`23:35.28`	`l-fy`	`and him also has that attitude of "you're not american so you don't deserve to live"`
`23:35.29`	`l-fy`	`sorry`
`23:35.41`	`l-fy`	`you're not white american you don't deserve to live`
`23:35.53`	`russellb`	`i'd say the same thing to him if i saw him here saying things about you`
`23:36.15`	`l-fy`	`russellb > than consider the fact that i have a ban on #asterisk-dev from him`
`23:44.38`	`kFuQ-lap`	`<PROTECTED>`
`23:47.47`	`*** join/#asterisk dongs (n=HPUX@h193071.ppp.asahi-net.or.jp)`
`23:50.51`	`dongs`	`whats everyones opinion on that utstar SIP wifi phone?`
`23:50.57`	`dongs`	`F1000 or whatever`
`23:51.05`	`dongs`	`does it suck?`

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